From ken at ukgb.net  Fri Jan  1 02:45:34 2010
From: ken at ukgb.net (Ken Gillett)
Date: Fri, 1 Jan 2010 10:45:34 +0000
Subject: [Freeswitch-users] video
Message-ID: <90AC63C4-48ED-43F7-A534-CB90C208604F@ukgb.net>

Sorry if this is a really basic question, but how do SIP video phones work and what capabilities does FreeSwitch have in this regard?

Does a PBX such as FS have to deal with the video stream, or does it just pass around a URL that the phone then uses as the source for the streaming video? Or is the video data all 'switched' along with the audio data? How would conference calls be handled?

Is this part of the SIP standard, or is it all proprietary and just down to the phone designer/manufacturer?


Ken G i l l e t t

_/_/_/_/_/_/_/_/





From tculjaga at gmail.com  Fri Jan  1 04:52:44 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 1 Jan 2010 13:52:44 +0100
Subject: [Freeswitch-users] freeswitch and H323
In-Reply-To: <87f2f3b90912310920q1821fe0eu5fd74e553e4cb12@mail.gmail.com>
References: <7aa8bd9d0912300514r68c90b12u7c631a649981cfa3@mail.gmail.com>
	<87f2f3b90912310920q1821fe0eu5fd74e553e4cb12@mail.gmail.com>
Message-ID: <65d96fc81001010452k37068e87sc0a229cdfe481c40@mail.gmail.com>

well, mod_h323 works for me... there are still some missing things and of
course bugs ... e.g. incorrect releaseCause mapping, no automatic codec
ptime sync... but it is usable ....


if you'd like to go mod_h323 way i can help you... it builds as a charm for
me...


T.




On Thu, Dec 31, 2009 at 6:20 PM, Michael Collins <msc at freeswitch.org> wrote:

> Are you trying to use mod_h323 or mod_opal? They are both works in
> progress, but the latter is farther along than the former. Use the latest
> FreeSWITCH trunk (or latest.freeswitch.org) and run the buildopal.sh
> script in the build directory. If you have any build issues then paste the
> log on pastebin.freeswitch.org and reply to this thread with the PB URL so
> that we can take a look.
> -MC
>
>
> On Wed, Dec 30, 2009 at 5:14 AM, Pete Kay <petedao at gmail.com> wrote:
>
>> Hi,
>>
>> has anyone been able to get H323 to work?
>>
>> I have problem trying to get it compiled with either 1.0.4 or 1.0.5.
>>
>> Thanks,
>> pete
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ken at ukgb.net  Fri Jan  1 05:03:31 2010
From: ken at ukgb.net (Ken Gillett)
Date: Fri, 1 Jan 2010 13:03:31 +0000
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <207e7a5e0912301103h17133fdfk22430bbce5a1718b@mail.gmail.com>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com>
	<B6763940-A3CB-4006-8DF1-ADEDD9C68294@ukgb.net>
	<0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org>
	<CA53F057-4A51-457C-A673-B0B6E4DF482A@ukgb.net>
	<207e7a5e0912301103h17133fdfk22430bbce5a1718b@mail.gmail.com>
Message-ID: <090E289D-76EE-4F4A-82DF-513626A915E6@ukgb.net>

Brilliant. Thank you. That's exactly what I needed to know.


On 30 Dec 2009, at 19:03, jonathan augenstine wrote:

> Ken,
> 
> configure
> make
> make install
> 
> This sequence of steps builds and installs the default configuration but without the audio files.  If you want the sound files installed also then:
> 
> make install sounds-install moh-install
> 
> Now the default sound files for conferencing, voicemail and music on hold are installed.
> 
> If you want to modify the default install to customize the build you can add and remove modules in modules.conf.  Then you run make/make install again to build those modules that are now included in the edited modules.conf file.
> 
> Jonathan
> 
> On Wed, Dec 30, 2009 at 10:45 AM, Ken Gillett <ken at ukgb.net> wrote:
> This is beginning to confuse me. Some say just:
> 
> > - configure
> > - make
> > - make install
> 
> is required, but the docs say more is needed for modules.conf. I'm still not sure if this only applies when modules.conf has been edited. Anyone help there?
> 
> On 28 Dec 2009, at 14:37, Brian West wrote:
> 
> > "all" is no longer needed.
> >
> > /b
> >
> > On Dec 25, 2009, at 3:07 AM, Ken Gillett wrote:
> >
> >>      make all install sounds-install moh-install.
> 
> So
> 
>        make install sounds-install moh-install.
> 
> is required? Always? Why?
> 
> Also, to bring this topic back to my original question (not that the diversity hasn't been interesting:-)
> 
> How can I best compile FS on one Mac and install it onto a different Mac?
> 
> 
> 
> Ken G i l l e t t
> 
> _/_/_/_/_/_/_/_/
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



Ken G i l l e t t

_/_/_/_/_/_/_/_/





From mike at jerris.com  Fri Jan  1 07:25:02 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 1 Jan 2010 10:25:02 -0500
Subject: [Freeswitch-users] Self alarm
In-Reply-To: <1262326847726-4238924.post@n2.nabble.com>
References: <1262250725607-4235713.post@n2.nabble.com>
	<1262326847726-4238924.post@n2.nabble.com>
Message-ID: <2A1E0DAF-A680-47B6-AC49-6A80FEC312A2@jerris.com>

The same what?

On Jan 1, 2010, at 1:20 AM, Sharad <sharad at coraltele.com> wrote:

>
> Hi
>
> I am also intresting in the same.
>
> Is there any script for this functionality.
>
> Regards
> --  
> View this message in context: http://n2.nabble.com/Self-alarm-tp4235713p4238924.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From mike at jerris.com  Fri Jan  1 07:27:26 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 1 Jan 2010 10:27:26 -0500
Subject: [Freeswitch-users] video
In-Reply-To: <90AC63C4-48ED-43F7-A534-CB90C208604F@ukgb.net>
References: <90AC63C4-48ED-43F7-A534-CB90C208604F@ukgb.net>
Message-ID: <C0A6C497-2073-4D09-8847-C078C4C81577@jerris.com>

Just like the audio, but in a different stream.  We don't do video  
transcoding at this time so it is passthrough only.  We have basic  
support for video follow audio in conference bit it is still rough on  
transitions.

Mike

On Jan 1, 2010, at 5:45 AM, Ken Gillett <ken at ukgb.net> wrote:

> Sorry if this is a really basic question, but how do SIP video  
> phones work and what capabilities does FreeSwitch have in this regard?
>
> Does a PBX such as FS have to deal with the video stream, or does it  
> just pass around a URL that the phone then uses as the source for  
> the streaming video? Or is the video data all 'switched' along with  
> the audio data? How would conference calls be handled?
>
> Is this part of the SIP standard, or is it all proprietary and just  
> down to the phone designer/manufacturer?
>
>
> Ken G i l l e t t
>
> _/_/_/_/_/_/_/_/
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From aep.lists at it46.se  Fri Jan  1 10:03:28 2010
From: aep.lists at it46.se (Alberto Escudero)
Date: Fri, 1 Jan 2010 19:03:28 +0100
Subject: [Freeswitch-users] XML IVR and Javascript menu-exec-app
Message-ID: <7d4c9553f55e0af32200eebdd332b2f9.squirrel@correo.nodo50.org>

Hi,

I am writing several IVRs using Freeswitch XML
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr

One of the nodes of the IVR is a Javascript application that records a
message.
e.g.:
<entry action="menu-exec-app" digits="3" param="javascript foo.js"/>

The Javascript application starts by issuing a
session.answer()

[records the voice message]

exit();

Once the Javascript exits, the channel is dropped and hence the IVR
terminates.
Is it possible to write a Javascript application that once is completed,
the channel returns back to the top menu of the IVR? I want to emulate the
same behavior that "menu-play-sound", that once the file is played, the
IVR logic returns to the top menu.

-aep

-- 
Stopping junk mailers is good for the environment





From jcasale at activenetwerx.com  Fri Jan  1 10:06:30 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 1 Jan 2010 18:06:30 +0000
Subject: [Freeswitch-users] Zap dialplan
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9068B44@prato.activenetwerx.local>

All the examples show bridging calls to an fx(s|o) port, but none show how
to handle an incoming call from the pstn on an fxo port. How do you distinguish
this call in your dialplan and begin routing it?

Does fs just place it in the public context, and if so iirc pstn calls don't have
a "destination_number" to match on, but a caller id only?

Thanks,
jlc


From codecomplete at free.fr  Fri Jan  1 14:18:05 2010
From: codecomplete at free.fr (Fred-145)
Date: Fri, 1 Jan 2010 14:18:05 -0800 (PST)
Subject: [Freeswitch-users] [Windows] Stable enough for production use?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C55540C9EAA@cooper>
References: <26807322.post@talk.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C55540C9EAA@cooper>
Message-ID: <26988383.post@talk.nabble.com>


Thanks everyone for the feedback.
-- 
View this message in context: http://old.nabble.com/-Windows--Stable-enough-for-production-use--tp26807322p26988383.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From mrene_lists at avgs.ca  Fri Jan  1 18:07:49 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sat, 2 Jan 2010 03:07:49 +0100
Subject: [Freeswitch-users] Zap dialplan
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9068B44@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E9068B44@prato.activenetwerx.local>
Message-ID: <3C85D953-8862-4006-9DE4-FFB40AF4BD8A@avgs.ca>

You define the destination number and context in the port's config  
section, in openzap.conf.xml

Sent from my iPhone

On 2010-01-01, at 7:06 PM, "Joseph L. Casale"  
<jcasale at activenetwerx.com> wrote:

> All the examples show bridging calls to an fx(s|o) port, but none  
> show how
> to handle an incoming call from the pstn on an fxo port. How do you  
> distinguish
> this call in your dialplan and begin routing it?
>
> Does fs just place it in the public context, and if so iirc pstn  
> calls don't have
> a "destination_number" to match on, but a caller id only?
>
> Thanks,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From jcasale at activenetwerx.com  Fri Jan  1 20:28:36 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 2 Jan 2010 04:28:36 +0000
Subject: [Freeswitch-users] More Zap issues
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906996A@prato.activenetwerx.local>

Now that the incoming zap is routed (and not hitting enum to get where it
needs to be) I thought I would just send it to the example ivr (as I am remote)
so I could see what happens at the cli.

I can see it enter the public context then find the extension which sends it to
5000 XML default. I see all the prompts being played at the cli but there is nothing
heard at the remote end?

Any ideas what's wrong? During initial setup when it was hitting enum, people on the
local end heard the ring and where able to answer and communicate with their sip phones?

Thanks!
jlc


From sharad at coraltele.com  Fri Jan  1 20:35:14 2010
From: sharad at coraltele.com (Sharad)
Date: Fri, 1 Jan 2010 20:35:14 -0800 (PST)
Subject: [Freeswitch-users] [!! SPAM]  Re:  Self alarm
In-Reply-To: <2A1E0DAF-A680-47B6-AC49-6A80FEC312A2@jerris.com>
References: <1262250725607-4235713.post@n2.nabble.com>
	<1262326847726-4238924.post@n2.nabble.com>
	<2A1E0DAF-A680-47B6-AC49-6A80FEC312A2@jerris.com>
Message-ID: <002b01ca8b65$16762640$0c04a8c0@compaq77db609e>


Self Alarm..
  ----- Original Message ----- 
  From: Michael Jerris [via freeswitch-users] 
  To: Sharad 
  Sent: Friday, January 01, 2010 9:04 PM
  Subject: [!! SPAM] Re: [Freeswitch-users] Self alarm


  The same what? 

  On Jan 1, 2010, at 1:20 AM, Sharad <[hidden email]> wrote: 


  > 
  > Hi 
  > 
  > I am also intresting in the same. 
  > 
  > Is there any script for this functionality. 
  > 
  > Regards 
  > --   
  > View this message in context: http://n2.nabble.com/Self-alarm-tp4235713p4238924.html
  > Sent from the freeswitch-users mailing list archive at Nabble.com. 
  > 
  > _______________________________________________ 
  > FreeSWITCH-users mailing list 
  > [hidden email] 
  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-  
  > users 
  > http://www.freeswitch.org

  _______________________________________________ 
  FreeSWITCH-users mailing list 
  [hidden email] 
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org



------------------------------------------------------------------------------

  View message @ http://n2.nabble.com/Self-alarm-tp4235713p4239714.html 
  To unsubscribe from Re: Self alarm, click here. 


-- 
View this message in context: http://n2.nabble.com/Self-alarm-tp4235713p4241557.html
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From jaugenstine at gmail.com  Fri Jan  1 20:50:55 2010
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Fri, 1 Jan 2010 20:50:55 -0800
Subject: [Freeswitch-users] [!! SPAM] Re: Self alarm
In-Reply-To: <002b01ca8b65$16762640$0c04a8c0@compaq77db609e>
References: <1262250725607-4235713.post@n2.nabble.com>
	<1262326847726-4238924.post@n2.nabble.com>
	<2A1E0DAF-A680-47B6-AC49-6A80FEC312A2@jerris.com>
	<002b01ca8b65$16762640$0c04a8c0@compaq77db609e>
Message-ID: <207e7a5e1001012050g18e5563dpe99e55b0509c5625@mail.gmail.com>

I believe that the question is, what do you want to alarm?  Do you want to
setup basic monitoring of the system?  Are you trying to track T1 alarms?
 Your question is too vague to answer.

On Fri, Jan 1, 2010 at 8:35 PM, Sharad <sharad at coraltele.com> wrote:

> Self Alarm..
>
> ----- Original Message -----
> *From:* [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4241557&i=0>
> *To:* [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4241557&i=1>
> *Sent:* Friday, January 01, 2010 9:04 PM
> *Subject:* [!! SPAM] Re: [Freeswitch-users] Self alarm
>
> The same what?
>
> On Jan 1, 2010, at 1:20 AM, Sharad <[hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4239714&i=0>>
> wrote:
>
> >
> > Hi
> >
> > I am also intresting in the same.
> >
> > Is there any script for this functionality.
> >
> > Regards
> > --
> > View this message in context:
> http://n2.nabble.com/Self-alarm-tp4235713p4238924.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4239714&i=1>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4239714&i=2>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
> View this message in context: Re: [!! SPAM] Re: [Freeswitch-users] Self
> alarm <http://n2.nabble.com/Self-alarm-tp4235713p4241557.html>
>
> Sent from the freeswitch-users mailing list archive<http://n2.nabble.com/freeswitch-users-f2379917.html>at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
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From mcampbellsmith at gmail.com  Fri Jan  1 23:30:02 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sat, 2 Jan 2010 18:30:02 +1100
Subject: [Freeswitch-users] No audio after Remote SDP:
In-Reply-To: <7D75E6D6-8AB9-429E-A5A9-1639C0D5AD09@freeswitch.org>
References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com>
	<11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org>
	<33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com>
	<7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org>
	<33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com>
	<772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org>
	<33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com>
	<191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com>
	<33c87fa30912301213g3d24b773k86b95e7ce08f6761@mail.gmail.com>
	<7D75E6D6-8AB9-429E-A5A9-1639C0D5AD09@freeswitch.org>
Message-ID: <33c87fa31001012330g69fc8e88m61e648d893c4d8f7@mail.gmail.com>

Hi!

Both are auto-nat:
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>

FreeSWITCH Version 1.0.trunk (15490)

However, isn't it the IP address that is reported by the remote
SPA3102 that is incorrect?  Or?

On Thu, Dec 31, 2009 at 7:21 AM, Brian West <brian at freeswitch.org> wrote:
> show me the ext-rtp-ip and ext-sip-ip settings you're using along with SVN rev please.
>
> /b
>
> On Dec 30, 2009, at 2:13 PM, Mark Campbell-Smith wrote:
>
>> Hi Anthony,
>>
>> The only profiles I have defined are external and internal. ? These
>> should be using internal...
>>
>> 192.168.1.120 is the FS box, which is NAT'd. ?Never had any problems
>> with this being NAT'd though
>> 192.168.1.121 is a PAP2 ATA connected to FS
>>
>> I don't use proxy media.
>>
>> I am trying to call an SPA3102, which is on the internet and NAT'd
>> (external IP address 11.11.11.11 in the trace and internal/private ip
>> address of 192.168.1.3).
>>
>> Thanks!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From sharad at coraltele.com  Fri Jan  1 23:58:59 2010
From: sharad at coraltele.com (Sharad)
Date: Fri, 1 Jan 2010 23:58:59 -0800 (PST)
Subject: [Freeswitch-users]  User's Mailbox Password
Message-ID: <1262419139401-4241899.post@n2.nabble.com>


Hi

When a user changes his mailbox password from his phone using advance
options, the corresponding XML does not show the new password. 

Can someone tell me what is the use of vm-password parameter which is shown
in the XML of that user.

regards
Sharad
-- 
View this message in context: http://n2.nabble.com/User-s-Mailbox-Password-tp4241899p4241899.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ken at ukgb.net  Sat Jan  2 05:17:57 2010
From: ken at ukgb.net (Ken Gillett)
Date: Sat, 2 Jan 2010 13:17:57 +0000
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <090E289D-76EE-4F4A-82DF-513626A915E6@ukgb.net>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com>
	<B6763940-A3CB-4006-8DF1-ADEDD9C68294@ukgb.net>
	<0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org>
	<CA53F057-4A51-457C-A673-B0B6E4DF482A@ukgb.net>
	<207e7a5e0912301103h17133fdfk22430bbce5a1718b@mail.gmail.com>
	<090E289D-76EE-4F4A-82DF-513626A915E6@ukgb.net>
Message-ID: <79B790D4-DE06-47FF-B9FD-52D9E1EB8BCB@ukgb.net>

One question still outstanding:-

How can I compile FS on one Mac and install it onto a different Mac? This means compiling on a MacPro running Snow Leopard and then installing onto a Snow Leopard Server which doesn't have the developer tools installed (and I don't want it to).

Advice on this would be appreciated.



Ken G i l l e t t

_/_/_/_/_/_/_/_/





From brian at freeswitch.org  Sat Jan  2 08:03:59 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 2 Jan 2010 10:03:59 -0600
Subject: [Freeswitch-users] No audio after Remote SDP:
In-Reply-To: <33c87fa31001012330g69fc8e88m61e648d893c4d8f7@mail.gmail.com>
References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com>
	<11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org>
	<33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com>
	<7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org>
	<33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com>
	<772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org>
	<33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com>
	<191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com>
	<33c87fa30912301213g3d24b773k86b95e7ce08f6761@mail.gmail.com>
	<7D75E6D6-8AB9-429E-A5A9-1639C0D5AD09@freeswitch.org>
	<33c87fa31001012330g69fc8e88m61e648d893c4d8f7@mail.gmail.com>
Message-ID: <180A53B1-EFB6-4B17-AA7C-19B007ADE04C@freeswitch.org>

Are you behind a nat-pmp/upnp router?

/b

On Jan 2, 2010, at 1:30 AM, Mark Campbell-Smith wrote:

> Hi!
> 
> Both are auto-nat:
>    <param name="ext-rtp-ip" value="auto-nat"/>
>    <param name="ext-sip-ip" value="auto-nat"/>
> 
> FreeSWITCH Version 1.0.trunk (15490)
> 
> However, isn't it the IP address that is reported by the remote
> SPA3102 that is incorrect?  Or?
> 
> On Thu, Dec 31, 2009 at 7:21 AM, Brian West <brian at freeswitch.org> wrote:
>> show me the ext-rtp-ip and ext-sip-ip settings you're using along with SVN rev please.
>> 
>> /b

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From brian at freeswitch.org  Sat Jan  2 08:04:52 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 2 Jan 2010 10:04:52 -0600
Subject: [Freeswitch-users] User's Mailbox Password
In-Reply-To: <1262419139401-4241899.post@n2.nabble.com>
References: <1262419139401-4241899.post@n2.nabble.com>
Message-ID: <3B8B47CA-3751-4FAE-BB17-3E482832869F@freeswitch.org>

The one from the XML will never change its stored in the db table in voicemail if you change it... but if you're using XML curl we do a request to let you know its updated so you can do what ever to update the db.

/b

On Jan 2, 2010, at 1:58 AM, Sharad wrote:

> 
> Hi
> 
> When a user changes his mailbox password from his phone using advance
> options, the corresponding XML does not show the new password. 
> 
> Can someone tell me what is the use of vm-password parameter which is shown
> in the XML of that user.
> 
> regards
> Sharad



From jaugenstine at gmail.com  Sat Jan  2 08:40:15 2010
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Sat, 2 Jan 2010 08:40:15 -0800
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <79B790D4-DE06-47FF-B9FD-52D9E1EB8BCB@ukgb.net>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com>
	<B6763940-A3CB-4006-8DF1-ADEDD9C68294@ukgb.net>
	<0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org>
	<CA53F057-4A51-457C-A673-B0B6E4DF482A@ukgb.net>
	<207e7a5e0912301103h17133fdfk22430bbce5a1718b@mail.gmail.com>
	<090E289D-76EE-4F4A-82DF-513626A915E6@ukgb.net>
	<79B790D4-DE06-47FF-B9FD-52D9E1EB8BCB@ukgb.net>
Message-ID: <207e7a5e1001020840y1bd3bae2y2b683a19659c9d7d@mail.gmail.com>

A dmg install package would need to be created.  A default package does not
currently exist and it would need to be created.  You would need to do this,
particularly if you are going to customize the build/install.

On Sat, Jan 2, 2010 at 5:17 AM, Ken Gillett <ken at ukgb.net> wrote:

> One question still outstanding:-
>
> How can I compile FS on one Mac and install it onto a different Mac? This
> means compiling on a MacPro running Snow Leopard and then installing onto a
> Snow Leopard Server which doesn't have the developer tools installed (and I
> don't want it to).
>
> Advice on this would be appreciated.
>
>
>
> Ken G i l l e t t
>
> _/_/_/_/_/_/_/_/
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jcasale at activenetwerx.com  Sat Jan  2 09:34:21 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 2 Jan 2010 17:34:21 +0000
Subject: [Freeswitch-users] Mixing zap and sip users in the directory
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9069EE7@prato.activenetwerx.local>

While trying to uncover my issues with no sound on a zap channel, somewhere
in the archive I came across the mention of mixing analog and digital extensions
in the directory, is there anything special that needs to be done for this?

Also, to answer an incoming call from an fxo channel, can I simply transfer it:

<include>
	<extension name="Openzap_in">
		<condition field="source" expression="mod_openzap">
			<action application="set" data="domain_name=$${domain}"/>
			<action application="transfer" data="2000 XML default"/>
		</condition>
	</extension>
</include>

Or do I have to answer and do other things?

Thanks!
jlc 


From aep.lists at it46.se  Sat Jan  2 10:08:38 2010
From: aep.lists at it46.se (Alberto Escudero)
Date: Sat, 2 Jan 2010 19:08:38 +0100
Subject: [Freeswitch-users] PHP ESL Problem
In-Reply-To: <285BD733E19541989B31B95871BF5642@fromage>
References: <285BD733E19541989B31B95871BF5642@fromage>
Message-ID: <6204a6f4ea0160933ba6493a292c1b9b.squirrel@correo.nodo50.org>

I do not know if really helps you but we are facing the same problem in
one of our implementations using the ESL.so for PHP.

We have only see this problem when subscribing to the CHANNEL_STATE

getType() should always match EventName... but it does not
./aep

-- 
Stopping junk mailers is good for the environment

> Would someone please take a look at this simple PHP event socket script
> and
> tell me what I am doing wrong - or tell me that this could be a bug
> elsewhere?  Any help would be appreciated.
>
> When I run the script without the call to execute(), everything seems
> fine.
> When I include the call to execute(), the calls to getType() return CUSTOM
> for a while, then later start to return the correct name.
>
> #!/usr/bin/php
> <?php
> require_once 'ESL.php';
> $endPoint = 'sofia/internal/695%192.168.100.132';
>
> $eventSocket = New ESLconnection('192.168.100.132', '8021', 'ClueCon');
> $event = $eventSocket->events('plain', 'ALL');
>
> // call endpoint, get uuid
> $event = $eventSocket->api('originate', $endPoint . ' &park');
> $serializedEvent = explode("\n", $event->serialize());
> foreach ($serializedEvent as $eventLine) {
>     list($dummy, $uuid) = explode('+OK ', $eventLine);
>     if ($uuid) { break; }
> }
>
> // play announcement to endpoint
> $event = $eventSocket->execute('playback', '/opt/ann/user-busy.wav',
> $uuid);
>
> // monitor events
> while (TRUE) {
>     echo "getType: " . $event->getType() . "\n";
>     $serializedEvent = explode("\n", $event->serialize());
>     foreach ($serializedEvent as $eventLine) {
>         list($header, $value) = explode(': ', $eventLine);
>         if ($header == "Event-Name")   { printf($eventLine . "\n"); }
>         if ($header == "Content-Type") { printf($eventLine . "\n"); }
>     }
>
> 	printf("\n");
>     $event = $eventSocket->recvEvent();
> }?>
>
>
> Run without the call to execute():
> ==================================
> getType: CUSTOM
> Content-Type: api/response
>
> getType: CHANNEL_CREATE
> Event-Name: CHANNEL_CREATE
>
> getType: CHANNEL_OUTGOING
> Event-Name: CHANNEL_OUTGOING
>
> getType: CHANNEL_ORIGINATE
> Event-Name: CHANNEL_ORIGINATE
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
> getType: PRESENCE_IN
> Event-Name: PRESENCE_IN
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
> getType: PRESENCE_IN
> Event-Name: PRESENCE_IN
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
> getType: CALL_UPDATE
> Event-Name: CALL_UPDATE
>
> getType: CHANNEL_PROGRESS
> Event-Name: CHANNEL_PROGRESS
>
> getType: HEARTBEAT
> Event-Name: HEARTBEAT
>
> getType: HEARTBEAT
> Event-Name: RE_SCHEDULE
>
> getType: CALL_UPDATE
> Event-Name: CALL_UPDATE
>
> getType: CODEC
> Event-Name: CODEC
>
> getType: CODEC
> Event-Name: CODEC
>
> getType: CHANNEL_ANSWER
> Event-Name: CHANNEL_ANSWER
>
> getType: PRESENCE_IN
> Event-Name: PRESENCE_IN
>
> getType: API
> Event-Name: API
>
> getType: PRESENCE_IN
> Event-Name: PRESENCE_IN
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
> getType: CHANNEL_EXECUTE
> Event-Name: CHANNEL_EXECUTE
>
> getType: CHANNEL_PARK
> Event-Name: CHANNEL_PARK
>
> getType: CHANNEL_HANGUP
> Event-Name: CHANNEL_HANGUP
>
> getType: CHANNEL_UNPARK
> Event-Name: CHANNEL_UNPARK
>
> getType: CHANNEL_EXECUTE_COMPLETE
> Event-Name: CHANNEL_EXECUTE_COMPLETE
>
> getType: PRESENCE_IN
> Event-Name: PRESENCE_IN
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
> getType: PRESENCE_IN
> Event-Name: PRESENCE_IN
>
> getType: CHANNEL_HANGUP_COMPLETE
> Event-Name: CHANNEL_HANGUP_COMPLETE
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
> getType: CHANNEL_DESTROY
> Event-Name: CHANNEL_DESTROY
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
>
> Run with the call to execute():
> ===============================
> getType: CUSTOM
> Content-Type: command/reply
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_CREATE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_OUTGOING
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_ORIGINATE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_STATE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: PRESENCE_IN
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_STATE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: PRESENCE_IN
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_STATE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CALL_UPDATE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_PROGRESS
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CALL_UPDATE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CODEC
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CODEC
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_ANSWER
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: PRESENCE_IN
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: API
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: PRESENCE_IN
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_STATE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_EXECUTE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_PARK
>
> getType: CHANNEL_EXECUTE
> Event-Name: CHANNEL_EXECUTE
>
> getType: CHANNEL_HANGUP
> Event-Name: CHANNEL_HANGUP
>
> getType: CHANNEL_EXECUTE_COMPLETE
> Event-Name: CHANNEL_EXECUTE_COMPLETE
>
> getType: COMMAND
> Event-Name: COMMAND
>
> getType: CHANNEL_UNPARK
> Event-Name: CHANNEL_UNPARK
>
> getType: CHANNEL_EXECUTE_COMPLETE
> Event-Name: CHANNEL_EXECUTE_COMPLETE
>
> getType: PRESENCE_IN
> Event-Name: PRESENCE_IN
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
> getType: PRESENCE_IN
> Event-Name: PRESENCE_IN
>
> getType: CHANNEL_HANGUP_COMPLETE
> Event-Name: CHANNEL_HANGUP_COMPLETE
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
> getType: CHANNEL_DESTROY
> Event-Name: CHANNEL_DESTROY
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
>
> Thanks,
> Ron
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>




From jcasale at activenetwerx.com  Sat Jan  2 14:17:50 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 2 Jan 2010 22:17:50 +0000
Subject: [Freeswitch-users] system functions silently ignored
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906A04C@prato.activenetwerx.local>

What could cause this? I have a fax script which just stopped working.
I see the logs display the complete diaplan which involves the system
call, I can manually execute the very command from the cli, but during
the call, it just skips the system application and hangups without printing
anything in the logs?

I even tried something simple like:
<action application="system" data="touch /tmp/rxfax"/>

Thanks,
jlc

From Russell.Mosemann at cune.org  Sat Jan  2 14:32:40 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sat, 2 Jan 2010 16:32:40 -0600
Subject: [Freeswitch-users] system functions silently ignored
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E906A04C@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E906A04C@prato.activenetwerx.local>
Message-ID: <CFF81042AFE645CCA2D6AE3276D63283@cune.pri>

Joseph L. Casale asked:

> What could cause this

Have you updated to the latest release in SVN?

--
Russell Mosemann



From mcampbellsmith at gmail.com  Sat Jan  2 15:19:39 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 3 Jan 2010 10:19:39 +1100
Subject: [Freeswitch-users] No audio after Remote SDP:
In-Reply-To: <180A53B1-EFB6-4B17-AA7C-19B007ADE04C@freeswitch.org>
References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com>
	<7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org>
	<33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com>
	<772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org>
	<33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com>
	<191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com>
	<33c87fa30912301213g3d24b773k86b95e7ce08f6761@mail.gmail.com>
	<7D75E6D6-8AB9-429E-A5A9-1639C0D5AD09@freeswitch.org>
	<33c87fa31001012330g69fc8e88m61e648d893c4d8f7@mail.gmail.com>
	<180A53B1-EFB6-4B17-AA7C-19B007ADE04C@freeswitch.org>
Message-ID: <33c87fa31001021519j41ead649rb4a85e6be6236254@mail.gmail.com>

I have a Linksys SPA3102, NAT'd on the internet (remotely) and
connected to my FS on the otherside of the world, which is also
natted.  A PAP2T is connected on the same subnet as the FS.  The 3102
registers successfully and a call can be set up from the PAP2 to the
3102.

However, after FS receives the Remote SDP the audio stops (ring tone
stops in my case)

The FS router is upnp enabled.  The SPA3102 router is NOT upnp enabled
(SPA3102 does not support upnp anyway I think).



On Sun, Jan 3, 2010 at 3:03 AM, Brian West <brian at freeswitch.org> wrote:
> Are you behind a nat-pmp/upnp router?
> /b
> On Jan 2, 2010, at 1:30 AM, Mark Campbell-Smith wrote:
>
> Hi!
>
> Both are auto-nat:
> ???<param name="ext-rtp-ip" value="auto-nat"/>
> ???<param name="ext-sip-ip" value="auto-nat"/>
>
> FreeSWITCH Version 1.0.trunk (15490)
>
> However, isn't it the IP address that is reported by the remote
> SPA3102 that is incorrect? ?Or?
>
> On Thu, Dec 31, 2009 at 7:21 AM, Brian West <brian at freeswitch.org> wrote:
>
> show me the ext-rtp-ip and ext-sip-ip settings you're using along with SVN
> rev please.
>
> /b
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jcasale at activenetwerx.com  Sat Jan  2 15:48:09 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 2 Jan 2010 23:48:09 +0000
Subject: [Freeswitch-users] system functions silently ignored
In-Reply-To: <CFF81042AFE645CCA2D6AE3276D63283@cune.pri>
References: <CA5A491E9DEFBE4CB777DE97E21575E906A04C@prato.activenetwerx.local>,
	<CFF81042AFE645CCA2D6AE3276D63283@cune.pri>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906A13E@prato.activenetwerx.local>

>> What could cause this
>
>Have you updated to the latest release in SVN?

Yeah, sorry that was on latest, I just recompiled pre10 and same behaviour now?


From mcampbellsmith at gmail.com  Sat Jan  2 16:09:23 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 3 Jan 2010 11:09:23 +1100
Subject: [Freeswitch-users] Dropped calls
Message-ID: <33c87fa31001021609t60ed802eyd6d6d4db7d314e1d@mail.gmail.com>

Hi!

I just got a couple of dropped calls.  Unfortunately I lost the sip
traces, but I do have the debug logs... in both cases FS shows a
Duplicate SDP received.  I'm not sure if this is a cause - do these
show anything to anyone as to why the calls dropped?

FS version is FreeSWITCH Version 1.0.trunk (15490)

Drop 1:
2010-01-02 17:18:50.806686 [DEBUG] switch_core_io.c:234
sofia/internal/2001 at myddns.dydns.org:442 receive message
[TRANSCODING_NECESSARY]
2010-01-02 17:18:50.826355 [DEBUG] switch_rtp.c:1972 Correct ip/port confirmed.
2010-01-02 17:18:51.645569 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state [ready][200]
2010-01-02 17:18:51.666115 [DEBUG] switch_core_session.c:704 Send
signal sofia/internal/sip:1000 at 192.168.1.121:5060 [BREAK]
2010-01-02 17:18:51.666115 [DEBUG] switch_core_session.c:704 Send
signal sofia/internal/2001 at myddns.dydns.org:442 [BREAK]
2010-01-02 17:18:51.685576 [DEBUG] switch_ivr_bridge.c:122
sofia/internal/2001 at myddns.dydns.org:442 receive message [DISPLAY]
2010-01-02 17:18:51.685576 [DEBUG] switch_ivr_bridge.c:122
sofia/internal/sip:1000 at 192.168.1.121:5060 receive message [DISPLAY]
2010-01-02 17:19:40.169692 [DEBUG] switch_rtp.c:2344 RTP RECV DTMF 1:404
2010-01-02 17:19:40.189926 [DEBUG] switch_rtp.c:1641 Send start packet
for [1] ts=24065640 dur=160/160/404 seq=15337
2010-01-02 17:19:40.205843 [DEBUG] switch_rtp.c:1577 Send middle
packet for [1] ts=24065640 dur=320/320/404 seq=15338
2010-01-02 17:19:40.225669 [DEBUG] switch_rtp.c:1577 Send end packet
for [1] ts=24065640 dur=480/480/404 seq=15339
2010-01-02 17:19:40.225669 [DEBUG] switch_rtp.c:1577 Send end packet
for [1] ts=24065640 dur=480/480/404 seq=15340
2010-01-02 17:19:40.225669 [DEBUG] switch_rtp.c:1577 Send end packet
for [1] ts=24065640 dur=480/480/404 seq=15341
2010-01-02 17:19:48.942440 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state [calling][0]
2010-01-02 17:19:52.093594 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state [ready][200]
2010-01-02 17:19:52.093594 [DEBUG] sofia.c:3654 Duplicate SDP
v=0
o=- 238296 238296 IN IP4 192.168.1.3
s=-
c=IN IP4 192.168.1.3
t=0 0
m=audio 19428 RTP/AVP 2 0 8 4 96 97 98 100 101
a=rtpmap:2 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2010-01-02 17:19:59.713933 [DEBUG] switch_rtp.c:2344 RTP RECV DTMF 1:404
2010-01-02 17:19:59.725822 [DEBUG] switch_rtp.c:1641 Send start packet
for [1] ts=24225880 dur=160/160/404 seq=16315
2010-01-02 17:19:59.745728 [DEBUG] switch_rtp.c:1577 Send middle
packet for [1] ts=24225880 dur=320/320/404 seq=16316
2010-01-02 17:19:59.765676 [DEBUG] switch_rtp.c:1577 Send end packet
for [1] ts=24225880 dur=480/480/404 seq=16317
2010-01-02 17:19:59.765676 [DEBUG] switch_rtp.c:1577 Send end packet
for [1] ts=24225880 dur=480/480/404 seq=16318
2010-01-02 17:19:59.765676 [DEBUG] switch_rtp.c:1577 Send end packet
for [1] ts=24225880 dur=480/480/404 seq=16319
2010-01-02 17:20:49.969621 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state [calling][0]
2010-01-02 17:20:54.036198 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state
[terminating][503]
2010-01-02 17:20:54.036198 [NOTICE] sofia.c:4247 Hangup
sofia/internal/2001 at myddns.dydns.org:442 [CS_EXECUTE]
[NORMAL_TEMPORARY_FAILURE]
2010-01-02 17:20:54.040419 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/2001 at myddns.dydns.org:442 [KILL]
2010-01-02 17:20:54.040419 [DEBUG] switch_core_session.c:982 Send
signal sofia/internal/2001 at myddns.dydns.org:442 [BREAK]

Drop 2:
2010-01-02 17:10:42.177817 [DEBUG] switch_core_session.c:704 Send
signal sofia/internal/sip:1000 at 192.168.1.121:5060 [BREAK]
2010-01-02 17:10:42.177817 [DEBUG] sofia.c:411
sofia/internal/sip:1000 at 192.168.1.121:5060 receive message [DISPLAY]
2010-01-02 17:10:42.177817 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state
[completed][200]
2010-01-02 17:10:42.185735 [DEBUG] switch_core_session.c:643 Send
signal sofia/internal/2001 at myddns.dydns.org:442 [BREAK]
2010-01-02 17:10:42.189574 [NOTICE] switch_ivr_originate.c:2836
Channel [sofia/internal/2001 at myddns.dydns.org:442] has been answered
2010-01-02 17:10:42.193802 [DEBUG] switch_channel.c:182
sofia/internal/2001 at myddns.dydns.org:442 receive message [AUDIO_SYNC]
2010-01-02 17:10:42.193802 [DEBUG] switch_ivr_originate.c:2881
Originate Resulted in Success:
[sofia/internal/sip:1000 at 192.168.1.121:5060]
2010-01-02 17:10:42.197882 [DEBUG] switch_channel.c:182
sofia/internal/sip:1000 at 192.168.1.121:5060 receive message
[AUDIO_SYNC]
2010-01-02 17:10:42.201876 [DEBUG] switch_channel.c:182
sofia/internal/2001 at myddns.dydns.org:442 receive message [AUDIO_SYNC]
2010-01-02 17:10:42.217738 [DEBUG] switch_ivr_originate.c:2881
Originate Resulted in Success:
[sofia/internal/sip:1000 at 192.168.1.121:5060]
2010-01-02 17:10:42.221921 [DEBUG] switch_channel.c:182
sofia/internal/sip:1000 at 192.168.1.121:5060 receive message
[AUDIO_SYNC]
2010-01-02 17:10:42.221921 [DEBUG] switch_channel.c:182
sofia/internal/2001 at myddns.dydns.org:442 receive message [AUDIO_SYNC]
2010-01-02 17:10:42.237732 [DEBUG] switch_ivr_bridge.c:1004
sofia/internal/sip:1000 at 192.168.1.121:5060 receive message [BRIDGE]
2010-01-02 17:10:42.237732 [DEBUG] switch_core_session.c:643 Send
signal sofia/internal/sip:1000 at 192.168.1.121:5060 [BREAK]
2010-01-02 17:10:42.241599 [DEBUG] switch_ivr_bridge.c:1011
sofia/internal/2001 at myddns.dydns.org:442 receive message [BRIDGE]
2010-01-02 17:10:42.241599 [DEBUG] switch_core_session.c:643 Send
signal sofia/internal/2001 at myddns.dydns.org:442 [BREAK]
2010-01-02 17:10:42.241599 [DEBUG] switch_ivr_bridge.c:1055
(sofia/internal/sip:1000 at 192.168.1.121:5060) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2010-01-02 17:10:42.246345 [DEBUG] switch_core_session.c:982 Send
signal sofia/internal/sip:1000 at 192.168.1.121:5060 [BREAK]
2010-01-02 17:10:42.246345 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1000 at 192.168.1.121:5060) Running State Change
CS_EXCHANGE_MEDIA
2010-01-02 17:10:42.249596 [DEBUG] switch_core_state_machine.c:351
(sofia/internal/sip:1000 at 192.168.1.121:5060) State EXCHANGE_MEDIA
2010-01-02 17:10:42.249596 [DEBUG] mod_sofia.c:464 SOFIA LOOPBACK
2010-01-02 17:10:42.265702 [DEBUG] switch_core_io.c:234
sofia/internal/2001 at myddns.dydns.org:442 receive message
[TRANSCODING_NECESSARY]
2010-01-02 17:10:42.285925 [DEBUG] switch_rtp.c:1972 Correct ip/port confirmed.
2010-01-02 17:10:44.402422 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state [ready][200]
2010-01-02 17:10:44.413899 [DEBUG] switch_core_session.c:704 Send
signal sofia/internal/sip:1000 at 192.168.1.121:5060 [BREAK]
2010-01-02 17:10:44.419743 [DEBUG] switch_core_session.c:704 Send
signal sofia/internal/2001 at myddns.dydns.org:442 [BREAK]
2010-01-02 17:10:44.425721 [DEBUG] switch_ivr_bridge.c:122
sofia/internal/sip:1000 at 192.168.1.121:5060 receive message [DISPLAY]
2010-01-02 17:10:44.433714 [DEBUG] switch_ivr_bridge.c:122
sofia/internal/2001 at myddns.dydns.org:442 receive message [DISPLAY]
2010-01-02 17:11:44.822897 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state [calling][0]
2010-01-02 17:11:45.353266 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state [ready][200]
2010-01-02 17:11:45.353266 [DEBUG] sofia.c:3654 Duplicate SDP
v=0
o=- 188941 188941 IN IP4 192.168.1.3
s=-
c=IN IP4 192.168.1.3
t=0 0
m=audio 19424 RTP/AVP 2 0 8 4 96 97 98 100 101
a=rtpmap:2 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2010-01-02 17:12:44.834908 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state [calling][0]
2010-01-02 17:12:45.125887 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state
[terminating][503]
2010-01-02 17:12:45.125887 [NOTICE] sofia.c:4247 Hangup
sofia/internal/2001 at myddns.dydns.org:442 [CS_EXECUTE]
[NORMAL_TEMPORARY_FAILURE]
2010-01-02 17:12:45.125887 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/2001 at myddns.dydns.org:442 [KILL]
2010-01-02 17:12:45.125887 [DEBUG] switch_core_session.c:982 Send
signal sofia/internal/2001 at myddns.dydns.org:442 [BREAK]


From jcasale at activenetwerx.com  Sat Jan  2 21:21:42 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sun, 3 Jan 2010 05:21:42 +0000
Subject: [Freeswitch-users] Zap dialplan characteristics
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906A406@prato.activenetwerx.local>

It seems it was permissions problems which were causing the
audio issues for me, I was attempting to run a manually
compiled instance of freeswitch with the stock init scripts.

Apparently, setting the udev rules to freeswitch/daemon as
fs runs won't work. I got it running finally tonight as a
user and group 'freeswitch' but not till after I tried the
zaptel release from the wiki's reco which didn't work until
the perms issue was discovered. I am sure I can go back to
using Digiums Dahdi package for Centos.

So last question. W/ Asterisk, I had to answer the dahdi line
so the far end didn't activate the call forward on no answer.
Would it be safe in assuming this needs to be replicated here
as well. If so, do I understand this right if I do this:

<include>
  <extension name="openzap_in">
    <condition field="source" expression="mod_openzap">
      <action application="set" data="domain_name=$${domain}"/>
      <action application="ring_ready"/>
      <action application="transfer" data="2000 XML default"/>
    </condition>
  </extension>
</include>

So that I can ring the call group for its preset time which surely
exceeds that of the call-forward from the telco?

Thanks!
jlc




From vmknott at gmail.com  Sun Jan  3 06:06:16 2010
From: vmknott at gmail.com (VM Knott)
Date: Sun, 3 Jan 2010 09:06:16 -0500
Subject: [Freeswitch-users] Voicemail Question (using multiple servers)
Message-ID: <ed9046461001030606w679e2d9cx60b54fa8607b6a9f@mail.gmail.com>

Bill,

The original plan was to share the db across multiple switches.

However, if I have to implement functionality to track records that
identify a specific IP address for every switch in the cluster, I was
thinking that a temporary work-around for our application would be a
standard greeting for all mailboxes.  We do not want the default TTS
sounding greeting that identifies the mailbox number, so I was going
to assign a standard greeting for all new mailboxes, and then if the
owner submits a custom greeting to their mailbox, it will override the
default.

We can implement logic to synchronize the db, not a huge deal.  But
this just feels like I?m over-complicating the solution.

- VMK




---------- Forwarded message ----------
From: "Bill W." <freeswitch at aastral.net>
To: freeswitch-users at lists.freeswitch.org
Date: Fri, 01 Jan 2010 01:31:14 -0500
Subject: Re: [Freeswitch-users] Voicemail Question (using multiple servers)
So is the problem that you're having to replicate the voicemail database
 across switches in the cluster or is the problem the content of the
entries in voicemail database?

Because in your original post you're speaking of trying to share the
voicemail db over NFS.

Thanks,
Bill

VM Knott wrote:
> Thank you Bill for the input, but changing how I access the data
> source does not solve my problem.
> I was hoping to avoid the management of IP Addresses for every
> voicemail box on the system.
>
> Is there a way for me to set a default greeting to all voicemail boxes
> globally, without having to go to a repository (regardless of means of
> access) for each mailbox?
>
>
>


From freeswitch at aastral.net  Sun Jan  3 07:24:49 2010
From: freeswitch at aastral.net (Bill W.)
Date: Sun, 03 Jan 2010 10:24:49 -0500
Subject: [Freeswitch-users] Voicemail Question (using multiple servers)
In-Reply-To: <ed9046461001030606w679e2d9cx60b54fa8607b6a9f@mail.gmail.com>
References: <ed9046461001030606w679e2d9cx60b54fa8607b6a9f@mail.gmail.com>
Message-ID: <4B40B6C1.1020009@aastral.net>

Hey VM,

I'm interested in your issue because I will need to implement this
feature in probably 6 months or so.  But I'm not currently familiar with
the VM database records or how they're used.

The weird thing is I'm using a shared sofia database for registrations
across a cluster and it works fine.  I know the IP address of the switch
the UA registered to gets stored in the registration database, but any
switch can use that registration record.

Looking at the sql in the voicemail module, it shows a column for
'domain'.  Is this where the IP address is being stored?   If so, maybe
you can find a way to change that to a domain name as Tony suggested.
Is the domain name/ip address being used in the filesystem path?

In any case I'm willing to help you solve this because I need to solve
this issue as well.   If you don't want to clog up the list with all the
troubleshooting,  we can take this off-list and then post our results to
the thread when it's all done.

Thanks,
Bill




VM Knott wrote:
> Bill,
> 
> The original plan was to share the db across multiple switches.
> 
> However, if I have to implement functionality to track records that
> identify a specific IP address for every switch in the cluster, I was
> thinking that a temporary work-around for our application would be a
> standard greeting for all mailboxes.  We do not want the default TTS
> sounding greeting that identifies the mailbox number, so I was going
> to assign a standard greeting for all new mailboxes, and then if the
> owner submits a custom greeting to their mailbox, it will override the
> default.
> 
> We can implement logic to synchronize the db, not a huge deal.  But
> this just feels like I?m over-complicating the solution.
> 
> - VMK
> 
> 


From yehavi.bourvine at gmail.com  Sun Jan  3 07:57:20 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 3 Jan 2010 17:57:20 +0200
Subject: [Freeswitch-users] Polycom & TLS - help please...
Message-ID: <d361ca7b1001030757m2f55a400n3e2b3b74185c4202@mail.gmail.com>

Hello,

  I am writing again because I am quite desparate... I fail to enable TLS on
Polycom while on SNOM I can make it work with the same configuration. From
TCPDUMP I see the following difference between the two handshakes:


   - SNOM: The protocol suite used is TLS_RSA_WITH_RC4_128_SHA
   - Polyco: The protocol suite used is TLS_RSA_WITH_AES_256_CBC_SHA and
   sessionTicketTLS also appears there. After the key exchange the phone
   disconnects the connection.

Any idea how to debug it?

                               Thanks! __Yehavi:
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From kdjakovic at hotmail.com  Sun Jan  3 08:50:21 2010
From: kdjakovic at hotmail.com (katarina djakovic)
Date: Sun, 3 Jan 2010 17:50:21 +0100
Subject: [Freeswitch-users] How to suspend certain users from the system
	using flush_inbound_reg
Message-ID: <SNT137-w54C41FF82F0E62B26415E9B3750@phx.gbl>


Hi,

 

we are trying to figure out how to suspend certain subscribers from our system and we have some problems with removing thier 

registrations. The UAs are ATAs.

 

This is what we do:

 

1) We remove the subscriber extension from the conf\directory .xml files
2) We do reloadxml
3) We flush user's registration with flush_inbound_reg 

 

but, the users are still able to make calls as if they were still registered. To make it clearer, their registrations are removed from 

the registration list (checked with sofia status), but they system still accepts the calls from them. From this, it seems that if ATA is never rebooted - we are not able to ban these users from the system.

 

Only after the ATA is rebooted user is not able to make calls any more, as the ATA can not register any more - since they users are 

removed from the directory. But before we reboot ATA everything works as nothing had been done.

 

Does anyone have an idea what are we doing wrong? We expect that after the registration is removed from the FS the UA should not be 

able to make a call but this is not what happnes.

 

Can anybody help please?

Thanks,
Katarina
 		 	   		  
_________________________________________________________________
Keep your friends updated?even when you?re not signed in.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010
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From linux4michelle at tamay-dogan.net  Sun Jan  3 10:46:36 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Sun, 3 Jan 2010 19:46:36 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
Message-ID: <20100103184636.GW5547@tamay-dogan.net>

Hello *,

I am owner and developer of the  enterprise  "electronica at tdnet UG"  and
currently I design a  GSM  router  with  VoIP   gateway  using  a  Texas
Instruments Sitara or OMAP  with  an  attached  CologneChip  4port  ISDN
Controller and a Silicon Laboratories Quad ProSLIC.  Also it will have a
GSM/HSPA modem.

My questionis, does someone use FreeSwitch with a GSM/HSPA Modem and can
use internet connectivity, telephonie and SMS?

If yes, which GSM Modem do you use?

I have this question since in theorie my cell-phone "Nokia 6120 classic"
can do this, but I was not able to get the streams...

Otherwise I would develop a GSM/HSPA Modem which CAN DO THIS my own.

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From a.alalousi at gmail.com  Sun Jan  3 12:25:50 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Sun, 3 Jan 2010 20:25:50 +0000
Subject: [Freeswitch-users] How to suspend certain users from the system
	using flush_inbound_reg
In-Reply-To: <SNT137-w54C41FF82F0E62B26415E9B3750@phx.gbl>
References: <SNT137-w54C41FF82F0E62B26415E9B3750@phx.gbl>
Message-ID: <cd93b3701001031225p44cd0d0fk79ed7ae05639417c@mail.gmail.com>

Hi Katarina,

Sounds like you have enabled ipauth by having cidr attributes within the
extension file. E.g:

<user id="test" cidr="192.168.0.0/16">

If this is the case, then username/password tuplles will fail (because you
have disabled them) but ipauth will work, and FS will allow unregistered
calls.

Also, check you conf/autoload_configs/acl.conf.xml to see if your default
domains acl state is one of allow rather than deny. E.g:

<list name="domains" default="allow">
    <node type="allow" domain="$${domain}"/>
</list>

This type of list is bad news anyway. I've seen it allow unregistered calls
from anyone simply by them knowing your domain.

What we do in our set-up is remove the default ACLs altogether, and apply
our own custom ones + firewalling at the border.

Hope this helps you a little.




2010/1/3 katarina djakovic <kdjakovic at hotmail.com>

>  Hi,
>
> we are trying to figure out how to suspend certain subscribers from our
> system and we have some problems with removing thier
> registrations. The UAs are ATAs.
>
> This is what we do:
>
> 1) We remove the subscriber extension from the conf\directory .xml files
> 2) We do reloadxml
> 3) We flush user's registration with flush_inbound_reg
>
> but, the users are still able to make calls as if they were still
> registered. To make it clearer, their registrations are removed from
> the registration list (checked with sofia status), but they system still
> accepts the calls from them. From this, it seems that if ATA is never
> rebooted - we are not able to ban these users from the system.
>
> Only after the ATA is rebooted user is not able to make calls any more, as
> the ATA can not register any more - since they users are
> removed from the directory. But before we reboot ATA everything works as
> nothing had been done.
>
> Does anyone have an idea what are we doing wrong? We expect that after the
> registration is removed from the FS the UA should not be
> able to make a call but this is not what happnes.
>
> Can anybody help please?
> Thanks,
> Katarina
>
> ------------------------------
> Keep your friends updated? even when you?re not signed in.<http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From freeswitch at aastral.net  Sun Jan  3 15:27:04 2010
From: freeswitch at aastral.net (Bill W.)
Date: Sun, 03 Jan 2010 18:27:04 -0500
Subject: [Freeswitch-users] How to suspend certain users from the system
 using flush_inbound_reg
In-Reply-To: <SNT137-w54C41FF82F0E62B26415E9B3750@phx.gbl>
References: <SNT137-w54C41FF82F0E62B26415E9B3750@phx.gbl>
Message-ID: <4B4127C8.30404@aastral.net>

Hello Katarina,

You could do this several ways, but it has to be more than just removing
their extension.

* You could use mod_nibblebill to enforce  a zero balance so they can't
make calls.

* You could use dialplan logic and xml_curl where a variable set in the
database gets populated in the user's directory entry and enforced in
the dialplan condition.

* You could use ACL logic and xml_curl where a variable set in the
database gets populated in the user's directory entry and enforced via
ACLs.   (auth-calls in sofia combined with auth-acl in the directory)

Hope this helps,
-Bill



katarina djakovic wrote:
> Hi,
>  
> we are trying to figure out how to suspend certain subscribers from our
> system and we have some problems with removing thier
> registrations. The UAs are ATAs.
>  
> This is what we do:
>  
> 1) We remove the subscriber extension from the conf\directory .xml files
> 2) We do reloadxml
> 3) We flush user's registration with flush_inbound_reg
>  
> but, the users are still able to make calls as if they were still
> registered. To make it clearer, their registrations are removed from
> the registration list (checked with sofia status), but they system still
> accepts the calls from them. From this, it seems that if ATA is never
> rebooted - we are not able to ban these users from the system.
>  
> Only after the ATA is rebooted user is not able to make calls any more,
> as the ATA can not register any more - since they users are
> removed from the directory. But before we reboot ATA everything works as
> nothing had been done.
>  
> Does anyone have an idea what are we doing wrong? We expect that after
> the registration is removed from the FS the UA should not be
> able to make a call but this is not what happnes.
>  
> Can anybody help please?
> Thanks,
> Katarina
> 
> ------------------------------------------------------------------------
> Keep your friends updated? even when you?re not signed in.
> <http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010>
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From saeedahmad1981 at gmail.com  Sun Jan  3 16:36:34 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Mon, 4 Jan 2010 01:36:34 +0100
Subject: [Freeswitch-users] freeswitch and H323
In-Reply-To: <65d96fc81001010452k37068e87sc0a229cdfe481c40@mail.gmail.com>
References: <7aa8bd9d0912300514r68c90b12u7c631a649981cfa3@mail.gmail.com> 
	<87f2f3b90912310920q1821fe0eu5fd74e553e4cb12@mail.gmail.com> 
	<65d96fc81001010452k37068e87sc0a229cdfe481c40@mail.gmail.com>
Message-ID: <c96b95651001031636q1ecfec18x2bba1c22ff6ca9bb@mail.gmail.com>

HI,

It would be really nice if you can create a wiki page.

Thanks

On Fri, Jan 1, 2010 at 1:52 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> well, mod_h323 works for me... there are still some missing things and of
> course bugs ... e.g. incorrect releaseCause mapping, no automatic codec
> ptime sync... but it is usable ....
>
>
> if you'd like to go mod_h323 way i can help you... it builds as a charm for
> me...
>
>
> T.
>
>
>
>
>
> On Thu, Dec 31, 2009 at 6:20 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Are you trying to use mod_h323 or mod_opal? They are both works in
>> progress, but the latter is farther along than the former. Use the latest
>> FreeSWITCH trunk (or latest.freeswitch.org) and run the buildopal.sh
>> script in the build directory. If you have any build issues then paste the
>> log on pastebin.freeswitch.org and reply to this thread with the PB URL
>> so that we can take a look.
>>  -MC
>>
>>
>> On Wed, Dec 30, 2009 at 5:14 AM, Pete Kay <petedao at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> has anyone been able to get H323 to work?
>>>
>>> I have problem trying to get it compiled with either 1.0.4 or 1.0.5.
>>>
>>> Thanks,
>>> pete
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sun Jan  3 16:43:24 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 3 Jan 2010 19:43:24 -0500
Subject: [Freeswitch-users] Zap dialplan characteristics
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E906A406@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E906A406@prato.activenetwerx.local>
Message-ID: <4145E5D0-AF8C-4113-B9E6-D87B23E7CA97@jerris.com>

If you want the call to "ring" longer than the telco would allow you  
will need to answer the call first but I find this approach very  
suboptimal.  I would adjust the call forward no answer times on the  
carrier side so you can have consistant cdrs and normal flow of  
progress instead of hacks.

Mike

On Jan 3, 2010, at 12:21 AM, "Joseph L. Casale" <jcasale at activenetwerx.com 
 > wrote:

> It seems it was permissions problems which were causing the
> audio issues for me, I was attempting to run a manually
> compiled instance of freeswitch with the stock init scripts.
>
> Apparently, setting the udev rules to freeswitch/daemon as
> fs runs won't work. I got it running finally tonight as a
> user and group 'freeswitch' but not till after I tried the
> zaptel release from the wiki's reco which didn't work until
> the perms issue was discovered. I am sure I can go back to
> using Digiums Dahdi package for Centos.
>
> So last question. W/ Asterisk, I had to answer the dahdi line
> so the far end didn't activate the call forward on no answer.
> Would it be safe in assuming this needs to be replicated here
> as well. If so, do I understand this right if I do this:
>
> <include>
>  <extension name="openzap_in">
>    <condition field="source" expression="mod_openzap">
>      <action application="set" data="domain_name=$${domain}"/>
>      <action application="ring_ready"/>
>      <action application="transfer" data="2000 XML default"/>
>    </condition>
>  </extension>
> </include>
>
> So that I can ring the call group for its preset time which surely
> exceeds that of the call-forward from the telco?
>
> Thanks!
> jlc
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From tom at tomcarlson.com  Sun Jan  3 12:31:35 2010
From: tom at tomcarlson.com (Tom Carlson)
Date: Sun, 3 Jan 2010 12:31:35 -0800
Subject: [Freeswitch-users] jitter buffer effecting input callback in lua
Message-ID: <21e9d36c1001031231yed4a914u5c8e7e8568208c45@mail.gmail.com>

I have a very simple lua script (shown below my message)

This script plays a greeting, lets the caller record a message, detecting
when caller is done by sensing a keypress.  It then plays the message back
to the caller.

This works perfectly, except the audio quality of the recorded message is
less than I had hoped.  To try to fix this, I have added a line to activate
the jitter buffer.  This single line keeps the script from detecting the
dtmf tones that end the recording, so the script just stays locked in record
mode forever, until you hang up.

The log shows no problems.

How can I activate the jitter buffer, and still detect dtmf events?

Thanks for your help.

Tom

-- ---------------------------------------------------------------------
function key_press(session, input_type, data, args)
  if input_type == "dtmf" then
    freeswitch.consoleLog("info", "Key pressed: " .. data["digit"])
    return "break"
  end
end

session:setVariable("jitterbuffer_msec", "200");
if session:ready() then
  session:answer();
  while (session:ready() == true) do
    session:setAutoHangup(false);
    session:sleep(1000);


session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-record_message.wav");


    session:setInputCallback("key_press", "");
    session:recordFile("/tmp/blah.wav", 5000, 10, 10); -- pressing key ends
the recording
    session:streamFile("/tmp/blah.wav");
  end
end
--
--------------------------------------------------------------------------------------------------
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From mike at jerris.com  Sun Jan  3 17:24:15 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 3 Jan 2010 20:24:15 -0500
Subject: [Freeswitch-users] How to suspend certain users from the system
	using flush_inbound_reg
In-Reply-To: <SNT137-w54C41FF82F0E62B26415E9B3750@phx.gbl>
References: <SNT137-w54C41FF82F0E62B26415E9B3750@phx.gbl>
Message-ID: <59CAAB6B-5259-4B78-9C9C-676ACF04D6B1@jerris.com>

First a note.  Registration and authentication are completely  
different. Removing the registration has to do with the switch knowin  
where to send the calls and nothing to do with auth for receiving  
calls.  There is one caveat to this.  We do support nonce count, and  
it could be using the auth from the previous registration that is  
still valid.  Double check the nc from the registrations and the call  
and see if that rings true.  We may want to add something to  
explicitly expire the nonce when youflush reg but I need some  
confirmation on that first.  Otherwise the other responces seem to  
cover the possibilities.  Crank up the debug and check sip trace for  
more details on what is allowing the call through and report back.

Mike

On Jan 3, 2010, at 11:50 AM, katarina djakovic <kdjakovic at hotmail.com>  
wrote:

> Hi,
>
> we are trying to figure out how to suspend certain subscribers from  
> our system and we have some problems with removing thier
> registrations. The UAs are ATAs.
>
> This is what we do:
>
> 1) We remove the subscriber extension from the conf\directory .xml  
> files
> 2) We do reloadxml
> 3) We flush user's registration with flush_inbound_reg
>
> but, the users are still able to make calls as if they were still  
> registered. To make it clearer, their registrations are removed from
> the registration list (checked with sofia status), but they system  
> still accepts the calls from them. From this, it seems that if ATA  
> is never rebooted - we are not able to ban these users from the  
> system.
>
> Only after the ATA is rebooted user is not able to make calls any  
> more, as the ATA can not register any more - since they users are
> removed from the directory. But before we reboot ATA everything  
> works as nothing had been done.
>
> Does anyone have an idea what are we doing wrong? We expect that  
> after the registration is removed from the FS the UA should not be
> able to make a call but this is not what happnes.
>
> Can anybody help please?
> Thanks,
> Katarina
>
> Keep your friends updated? even when you?re not signed in.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From mike at jerris.com  Sun Jan  3 17:25:17 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 3 Jan 2010 20:25:17 -0500
Subject: [Freeswitch-users] Polycom & TLS - help please...
In-Reply-To: <d361ca7b1001030757m2f55a400n3e2b3b74185c4202@mail.gmail.com>
References: <d361ca7b1001030757m2f55a400n3e2b3b74185c4202@mail.gmail.com>
Message-ID: <BFB88CA0-8319-4D05-B3CD-39E3450DE157@jerris.com>

Perhapse cranking up the Sofia tport log to level 9 may help.

Mike

On Jan 3, 2010, at 10:57 AM, Yehavi Bourvine  
<yehavi.bourvine at gmail.com> wrote:

> Hello,
>
>   I am writing again because I am quite desparate... I fail to  
> enable TLS on Polycom while on SNOM I can make it work with the same  
> configuration. From TCPDUMP I see the following difference between  
> the two handshakes:
>
> SNOM: The protocol suite used is TLS_RSA_WITH_RC4_128_SHA
> Polyco: The protocol suite used is TLS_RSA_WITH_AES_256_CBC_SHA and  
> sessionTicketTLS also appears there. After the key exchange the  
> phone disconnects the connection.
> Any idea how to debug it?
>
>                                Thanks! __Yehavi:
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From mike at jerris.com  Sun Jan  3 17:29:40 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 3 Jan 2010 20:29:40 -0500
Subject: [Freeswitch-users] jitter buffer effecting input callback in lua
In-Reply-To: <21e9d36c1001031231yed4a914u5c8e7e8568208c45@mail.gmail.com>
References: <21e9d36c1001031231yed4a914u5c8e7e8568208c45@mail.gmail.com>
Message-ID: <F46682E5-43C4-4B1F-A318-ECAC7608353E@jerris.com>

There is no reason I can think of that would cause this.  Have you  
tried different phones to eliminate if it is an issue just with one  
type of phone?  Please open a bug on jira.freeswitch.org with a  
minimal script example to reproduce and details of the devices it has  
been reproduced with.

Mike

On Jan 3, 2010, at 3:31 PM, Tom Carlson <tom at tomcarlson.com> wrote:

> I have a very simple lua script (shown below my message)
>
> This script plays a greeting, lets the caller record a message,  
> detecting when caller is done by sensing a keypress.  It then plays  
> the message back to the caller.
>
> This works perfectly, except the audio quality of the recorded  
> message is less than I had hoped.  To try to fix this, I have added  
> a line to activate the jitter buffer.  This single line keeps the  
> script from detecting the dtmf tones that end the recording, so the  
> script just stays locked in record mode forever, until you hang up.
>
> The log shows no problems.
>
> How can I activate the jitter buffer, and still detect dtmf events?
>
> Thanks for your help.
>
> Tom
>
> --  
> ---------------------------------------------------------------------
> function key_press(session, input_type, data, args)
>   if input_type == "dtmf" then
>     freeswitch.consoleLog("info", "Key pressed: " .. data["digit"])
>     return "break"
>   end
> end
>
> session:setVariable("jitterbuffer_msec", "200");
> if session:ready() then
>   session:answer();
>   while (session:ready() == true) do
>     session:setAutoHangup(false);
>     session:sleep(1000);
>
>     session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/ 
> voicemail/8000/vm-record_message.wav");
>
>     session:setInputCallback("key_press", "");
>     session:recordFile("/tmp/blah.wav", 5000, 10, 10); -- pressing  
> key ends the recording
>     session:streamFile("/tmp/blah.wav");
>   end
> end
> --   
> --- 
> --- 
> --- 
> --- 
> --- 
> --- 
> --- 
> --- 
> --- 
> --- 
> --------------------------------------------------------------------
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From vmknott at gmail.com  Sun Jan  3 18:01:17 2010
From: vmknott at gmail.com (VM Knott)
Date: Sun, 3 Jan 2010 21:01:17 -0500
Subject: [Freeswitch-users] Voicemail Question (using multiple servers)
Message-ID: <ed9046461001031801p24542862p3e7b0409957c713a@mail.gmail.com>

Bill,

The "domain name" solution will not work for me.
My FreeSWITCH servers are spread across multiple domain names, so I
would prefer a solution that encompasses a more generic model.

I'm guessing that my situation is more uncommon than I originally thought.

I can setup something that is more specific to my architecture, and
create a service-component that manages the central database to
identify the greeting messages on all servers in the cluster.

A less desirable approach, but in consideration of all of the other
inherent features of FreeSWITCH, a small price to pay.

- VMK



---------- Forwarded message ----------
From: "Bill W." <freeswitch at aastral.net>
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 03 Jan 2010 10:24:49 -0500
Subject: Re: [Freeswitch-users] Voicemail Question (using multiple servers)
Hey VM,

I'm interested in your issue because I will need to implement this
feature in probably 6 months or so.  But I'm not currently familiar with
the VM database records or how they're used.

The weird thing is I'm using a shared sofia database for registrations
across a cluster and it works fine.  I know the IP address of the switch
the UA registered to gets stored in the registration database, but any
switch can use that registration record.

Looking at the sql in the voicemail module, it shows a column for
'domain'.  Is this where the IP address is being stored?   If so, maybe
you can find a way to change that to a domain name as Tony suggested.
Is the domain name/ip address being used in the filesystem path?

In any case I'm willing to help you solve this because I need to solve
this issue as well.   If you don't want to clog up the list with all the
troubleshooting,  we can take this off-list and then post our results to
the thread when it's all done.

Thanks,
Bill


From mike at jerris.com  Sun Jan  3 18:22:50 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 3 Jan 2010 21:22:50 -0500
Subject: [Freeswitch-users] Voicemail Question (using multiple servers)
In-Reply-To: <ed9046461001031801p24542862p3e7b0409957c713a@mail.gmail.com>
References: <ed9046461001031801p24542862p3e7b0409957c713a@mail.gmail.com>
Message-ID: <AC15E5EE-FDC1-434C-ADB0-FC0A576E78CD@jerris.com>

You misunderstand the domain name suggestion.  Look a bit closer at  
how the call to voicemail looks in the dialplan.  You can pass  
whatever you like.  Also, you can force domain for registration and  
the like as well and that may be the right solution for you as well  
but depends on full details of your setup.
Mike

On Jan 3, 2010, at 9:01 PM, VM Knott <vmknott at gmail.com> wrote:

> Bill,
>
> The "domain name" solution will not work for me.
> My FreeSWITCH servers are spread across multiple domain names, so I
> would prefer a solution that encompasses a more generic model.
>
> I'm guessing that my situation is more uncommon than I originally  
> thought.
>
> I can setup something that is more specific to my architecture, and
> create a service-component that manages the central database to
> identify the greeting messages on all servers in the cluster.
>
> A less desirable approach, but in consideration of all of the other
> inherent features of FreeSWITCH, a small price to pay.
>
> - VMK
>
>
>
> ---------- Forwarded message ----------
> From: "Bill W." <freeswitch at aastral.net>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sun, 03 Jan 2010 10:24:49 -0500
> Subject: Re: [Freeswitch-users] Voicemail Question (using multiple  
> servers)
> Hey VM,
>
> I'm interested in your issue because I will need to implement this
> feature in probably 6 months or so.  But I'm not currently familiar  
> with
> the VM database records or how they're used.
>
> The weird thing is I'm using a shared sofia database for registrations
> across a cluster and it works fine.  I know the IP address of the  
> switch
> the UA registered to gets stored in the registration database, but any
> switch can use that registration record.
>
> Looking at the sql in the voicemail module, it shows a column for
> 'domain'.  Is this where the IP address is being stored?   If so,  
> maybe
> you can find a way to change that to a domain name as Tony suggested.
> Is the domain name/ip address being used in the filesystem path?
>
> In any case I'm willing to help you solve this because I need to solve
> this issue as well.   If you don't want to clog up the list with all  
> the
> troubleshooting,  we can take this off-list and then post our  
> results to
> the thread when it's all done.
>
> Thanks,
> Bill
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From brian at freeswitch.org  Sun Jan  3 18:32:22 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 3 Jan 2010 20:32:22 -0600
Subject: [Freeswitch-users] Polycom & TLS - help please...
In-Reply-To: <BFB88CA0-8319-4D05-B3CD-39E3450DE157@jerris.com>
References: <d361ca7b1001030757m2f55a400n3e2b3b74185c4202@mail.gmail.com>
	<BFB88CA0-8319-4D05-B3CD-39E3450DE157@jerris.com>
Message-ID: <D950E561-1B94-48E0-9221-3739BAB8F91A@freeswitch.org>

If you didn't manually install the CA cert into the phone as per the wiki it won't ever work.

/b

On Jan 3, 2010, at 7:25 PM, Michael Jerris wrote:

> Perhapse cranking up the Sofia tport log to level 9 may help.
> 
> Mike
> 
> On Jan 3, 2010, at 10:57 AM, Yehavi Bourvine  
> <yehavi.bourvine at gmail.com> wrote:
> 
>> Hello,
>> 
>>  I am writing again because I am quite desparate... I fail to  
>> enable TLS on Polycom while on SNOM I can make it work with the same  
>> configuration. From TCPDUMP I see the following difference between  
>> the two handshakes:
>> 
>> SNOM: The protocol suite used is TLS_RSA_WITH_RC4_128_SHA
>> Polyco: The protocol suite used is TLS_RSA_WITH_AES_256_CBC_SHA and  
>> sessionTicketTLS also appears there. After the key exchange the  
>> phone disconnects the connection.
>> Any idea how to debug it?
>> 
>>                               Thanks! __Yehavi:

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From brian at freeswitch.org  Sun Jan  3 18:36:15 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 3 Jan 2010 20:36:15 -0600
Subject: [Freeswitch-users] system functions silently ignored
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E906A13E@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E906A04C@prato.activenetwerx.local>,
	<CFF81042AFE645CCA2D6AE3276D63283@cune.pri>
	<CA5A491E9DEFBE4CB777DE97E21575E906A13E@prato.activenetwerx.local>
Message-ID: <78ED9C83-CC46-41D5-AA2F-85291381B5C3@freeswitch.org>

Funny Pre's no longer exist use this:

http://latest.freeswitch.org/freeswitch-1.0.5-latest.tar.gz

/b

On Jan 2, 2010, at 5:48 PM, Joseph L. Casale wrote:

>>> What could cause this
>> 
>> Have you updated to the latest release in SVN?
> 
> Yeah, sorry that was on latest, I just recompiled pre10 and same behaviour now?



From brian at freeswitch.org  Sun Jan  3 18:36:45 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 3 Jan 2010 20:36:45 -0600
Subject: [Freeswitch-users] Dropped calls
In-Reply-To: <33c87fa31001021609t60ed802eyd6d6d4db7d314e1d@mail.gmail.com>
References: <33c87fa31001021609t60ed802eyd6d6d4db7d314e1d@mail.gmail.com>
Message-ID: <159CEFD5-4E6C-42EC-AB2B-29A9C6649A50@freeswitch.org>

Fairly old SVN rev please update, try again.. then post if it's not fixed.

http://latest.freeswitch.org/freeswitch-1.0.5-latest.tar.gz

/b

On Jan 2, 2010, at 6:09 PM, Mark Campbell-Smith wrote:

> FS version is FreeSWITCH Version 1.0.trunk (15490)



From brian at freeswitch.org  Sun Jan  3 18:40:52 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 3 Jan 2010 20:40:52 -0600
Subject: [Freeswitch-users] jitter buffer effecting input callback in lua
In-Reply-To: <F46682E5-43C4-4B1F-A318-ECAC7608353E@jerris.com>
References: <21e9d36c1001031231yed4a914u5c8e7e8568208c45@mail.gmail.com>
	<F46682E5-43C4-4B1F-A318-ECAC7608353E@jerris.com>
Message-ID: <7073E494-191F-4AAD-90EF-10CA6BEDA7C8@freeswitch.org>

Also please check collect an RTP packet cap if possible this will help too.

/b

On Jan 3, 2010, at 7:29 PM, Michael Jerris wrote:

> There is no reason I can think of that would cause this.  Have you  
> tried different phones to eliminate if it is an issue just with one  
> type of phone?  Please open a bug on jira.freeswitch.org with a  
> minimal script example to reproduce and details of the devices it has  
> been reproduced with.
> 
> Mike

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From jcasale at activenetwerx.com  Sun Jan  3 19:09:18 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Mon, 4 Jan 2010 03:09:18 +0000
Subject: [Freeswitch-users] system functions silently ignored
In-Reply-To: <78ED9C83-CC46-41D5-AA2F-85291381B5C3@freeswitch.org>
References: <CA5A491E9DEFBE4CB777DE97E21575E906A04C@prato.activenetwerx.local>,
	<CFF81042AFE645CCA2D6AE3276D63283@cune.pri>
	<CA5A491E9DEFBE4CB777DE97E21575E906A13E@prato.activenetwerx.local>
	<78ED9C83-CC46-41D5-AA2F-85291381B5C3@freeswitch.org>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906AC73@prato.activenetwerx.local>

>Funny Pre's no longer exist use this:
>
>http://latest.freeswitch.org/freeswitch-1.0.5-latest.tar.gz

That's what I was on, I just tried pre10 to see if it was a bug
in that "latest" build.

This is on hold for a day or two now, when this server shuts off
it panic's with a fault related to dahdi, so given that, I am not
going to trouble shoot this until obviously that's resolved.

I'll get back to this once we have some new hardware...


From yehavi.bourvine at gmail.com  Sun Jan  3 19:32:35 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 4 Jan 2010 05:32:35 +0200
Subject: [Freeswitch-users] Polycom & TLS - help please...
In-Reply-To: <D950E561-1B94-48E0-9221-3739BAB8F91A@freeswitch.org>
References: <d361ca7b1001030757m2f55a400n3e2b3b74185c4202@mail.gmail.com>
	<BFB88CA0-8319-4D05-B3CD-39E3450DE157@jerris.com>
	<D950E561-1B94-48E0-9221-3739BAB8F91A@freeswitch.org>
Message-ID: <d361ca7b1001031932m7695150fv967c0df683083af9@mail.gmail.com>

I've built the slef-signed root certificate and server;s certificate per the
TLS wiki, and installed the root certificate on the phone (both manually and
via the config files). I did not enter the "== untrusted ==" instead of the
cerificate (as the Polycom's wiki suggests) as the pone doesn't allow this.
It accepted the certificate.

I've tried this on 501 (running 3.1.3 which is the last supported version on
it), and 550 & 650 running 3.2.2.

                            Thans, __Yehavi:

 2010/1/4 Brian West <brian at freeswitch.org>

>  If you didn't manually install the CA cert into the phone as per the wiki
> it won't ever work.
>
> /b
>
>  On Jan 3, 2010, at 7:25 PM, Michael Jerris wrote:
>
> Perhapse cranking up the Sofia tport log to level 9 may help.
>
> Mike
>
> On Jan 3, 2010, at 10:57 AM, Yehavi Bourvine
> <yehavi.bourvine at gmail.com> wrote:
>
> Hello,
>
>
>  I am writing again because I am quite desparate... I fail to
>
> enable TLS on Polycom while on SNOM I can make it work with the same
>
> configuration. From TCPDUMP I see the following difference between
>
> the two handshakes:
>
>
> SNOM: The protocol suite used is TLS_RSA_WITH_RC4_128_SHA
>
> Polyco: The protocol suite used is TLS_RSA_WITH_AES_256_CBC_SHA and
>
> sessionTicketTLS also appears there. After the key exchange the
>
> phone disconnects the connection.
>
> Any idea how to debug it?
>
>
>                               Thanks! __Yehavi:
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nicolas at medularis.com  Sun Jan  3 22:45:20 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 4 Jan 2010 03:45:20 -0300
Subject: [Freeswitch-users] How to control call volume?
Message-ID: <1b46b4e81001032245k7d10e027p46b21e50bfb2264f@mail.gmail.com>

Hi, is there a way of controlling the volume of a call? I'm bridging
calls with a JS script. Sometimes the people getting the calls
complain the volume is too low. I've recorded a few of the calls and
most of the times, while playing the recorded wav files, the volume of
LegB (second leg of the bridge) is pretty hard to hear, even with the
computer and player volume to the max (ok, it's a laptop, but even
with headphones). I saw there are some volume control options for
conferences, but I couldn't find anything for regular "originate
calls" or bridges. I am doing transcoding, so that might help (?).

Thank you very much for your help.

Best,

Nicolas


From ken at ukgb.net  Mon Jan  4 01:14:17 2010
From: ken at ukgb.net (Ken Gillett)
Date: Mon, 4 Jan 2010 09:14:17 +0000
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <207e7a5e1001020840y1bd3bae2y2b683a19659c9d7d@mail.gmail.com>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com>
	<B6763940-A3CB-4006-8DF1-ADEDD9C68294@ukgb.net>
	<0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org>
	<CA53F057-4A51-457C-A673-B0B6E4DF482A@ukgb.net>
	<207e7a5e0912301103h17133fdfk22430bbce5a1718b@mail.gmail.com>
	<090E289D-76EE-4F4A-82DF-513626A915E6@ukgb.net>
	<79B790D4-DE06-47FF-B9FD-52D9E1EB8BCB@ukgb.net>
	<207e7a5e1001020840y1bd3bae2y2b683a19659c9d7d@mail.gmail.com>
Message-ID: <6E6E6D36-6E93-49BF-9516-FC028E73D279@ukgb.net>

I will look into this, but in the meantime, would it not be possible to simply copy the installed files? I realise that basically this must be possible, but practically it depends on what is installed where. I believe the default prefix is /usr/local/freeswitch, but is EVERYTHING in there or is other stuff scattered in other directories?

If it's all together I'm thinking I could simply tar the folder and copy the tarball. Wouldn't this be possible?


On 2 Jan 2010, at 16:40, jonathan augenstine wrote:

> A dmg install package would need to be created.  A default package does not currently exist and it would need to be created.  You would need to do this, particularly if you are going to customize the build/install.
> 
> On Sat, Jan 2, 2010 at 5:17 AM, Ken Gillett <ken at ukgb.net> wrote:
> One question still outstanding:-
> 
> How can I compile FS on one Mac and install it onto a different Mac? This means compiling on a MacPro running Snow Leopard and then installing onto a Snow Leopard Server which doesn't have the developer tools installed (and I don't want it to).
> 
> Advice on this would be appreciated.
> 



Ken G i l l e t t

_/_/_/_/_/_/_/_/





From sharad at coraltele.com  Mon Jan  4 01:54:51 2010
From: sharad at coraltele.com (Sharad)
Date: Mon, 4 Jan 2010 01:54:51 -0800 (PST)
Subject: [Freeswitch-users]  time stamp for sched_api
Message-ID: <1262598891498-4249284.post@n2.nabble.com>


I am trying the following API. I want this API to run on 3rd Jan 2010 at
15:13 hours.

sched_api 010315132010  none originate sofia/external/1006 at 192.168.4.106
&playback(ivr/alarm.wav)

But it is not getting activated on the maturity of this time.

Can someone let me know the error in the time format.

Regards 
Sharad


-- 
View this message in context: http://n2.nabble.com/time-stamp-for-sched-api-tp4249284p4249284.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From wasim at convergence.pk  Mon Jan  4 02:23:07 2010
From: wasim at convergence.pk (Wasim Baig)
Date: Mon, 4 Jan 2010 15:23:07 +0500
Subject: [Freeswitch-users] time stamp for sched_api
In-Reply-To: <1262598891498-4249284.post@n2.nabble.com>
References: <1262598891498-4249284.post@n2.nabble.com>
Message-ID: <b8ad2a5b1001040223yc7ab19dm57e0101ee335a2f1@mail.gmail.com>

On Mon, Jan 4, 2010 at 2:54 PM, Sharad <sharad at coraltele.com> wrote:

>
> I am trying the following API. I want this API to run on 3rd Jan 2010 at
> 15:13 hours.
>
> sched_api 010315132010  none originate sofia/external/1006 at 192.168.4.106
> &playback(ivr/alarm.wav)
>
> But it is not getting activated on the maturity of this time.
>
> Can someone let me know the error in the time format.
>

unixtimestamp ......seconds since Jan 01 1970.

-- 
wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
peace be upon you ...
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From sharad at coraltele.com  Mon Jan  4 02:51:39 2010
From: sharad at coraltele.com (Sharad)
Date: Mon, 4 Jan 2010 02:51:39 -0800 (PST)
Subject: [Freeswitch-users] time stamp for sched_api
In-Reply-To: <b8ad2a5b1001040223yc7ab19dm57e0101ee335a2f1@mail.gmail.com>
References: <1262598891498-4249284.post@n2.nabble.com>
	<b8ad2a5b1001040223yc7ab19dm57e0101ee335a2f1@mail.gmail.com>
Message-ID: <1262602299063-4249417.post@n2.nabble.com>


Thanks Mr. Baig..

So can you plz write the syntax for the same API if we want this to get
executed on 4th Jan 2010 at 1800 hours.

Regards



Wasim Baig wrote:
> 
> On Mon, Jan 4, 2010 at 2:54 PM, Sharad <sharad at coraltele.com> wrote:
> 
>>
>> I am trying the following API. I want this API to run on 3rd Jan 2010 at
>> 15:13 hours.
>>
>> sched_api 010315132010  none originate sofia/external/1006 at 192.168.4.106
>> &playback(ivr/alarm.wav)
>>
>> But it is not getting activated on the maturity of this time.
>>
>> Can someone let me know the error in the time format.
>>
> 
> unixtimestamp ......seconds since Jan 01 1970.
> 
> -- 
> wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
> peace be upon you ...
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/time-stamp-for-sched-api-tp4249284p4249417.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From nicolas at medularis.com  Mon Jan  4 07:10:42 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 4 Jan 2010 12:10:42 -0300
Subject: [Freeswitch-users] Hangup on silence?
Message-ID: <1b46b4e81001040710x4886d57ci8227c80c7fe3b796@mail.gmail.com>

Hi, is there a way to hangup after a certain amount of silence? my
problem is that with regular PSTN calls I don't get a BYE from my
provider until around 1 minute after the phone has been hanged up.
This is pretty standard, and the provider is not at fault, but I would
like to hangup the call after I detect a certain amount of silence
time. The only thing I could find was wait_for_silence on the wiki,
but that application only delays dialplan execution, and I would need
something like: hangup_on_silence=30, where 30 means to hangup if 30
seconds of silence have passed.

Thanks for your help!

Nicolas


From brian at freeswitch.org  Mon Jan  4 07:31:22 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 4 Jan 2010 09:31:22 -0600
Subject: [Freeswitch-users] Polycom & TLS - help please...
In-Reply-To: <d361ca7b1001031932m7695150fv967c0df683083af9@mail.gmail.com>
References: <d361ca7b1001030757m2f55a400n3e2b3b74185c4202@mail.gmail.com>
	<BFB88CA0-8319-4D05-B3CD-39E3450DE157@jerris.com>
	<D950E561-1B94-48E0-9221-3739BAB8F91A@freeswitch.org>
	<d361ca7b1001031932m7695150fv967c0df683083af9@mail.gmail.com>
Message-ID: <3968BF39-7A15-4F9F-8FA2-A78BC4F0F38E@freeswitch.org>

OK to properly use TLS you have to setup NAPTR and SRV records and the DNS domain has to match the cert.  Did you do that?

/b

On Jan 3, 2010, at 9:32 PM, Yehavi Bourvine wrote:

> I've built the slef-signed root certificate and server;s certificate per the TLS wiki, and installed the root certificate on the phone (both manually and via the config files). I did not enter the "== untrusted ==" instead of the cerificate (as the Polycom's wiki suggests) as the pone doesn't allow this. It accepted the certificate.
>  
> I've tried this on 501 (running 3.1.3 which is the last supported version on it), and 550 & 650 running 3.2.2.
>  
>                             Thans, __Yehavi:



From brian at freeswitch.org  Mon Jan  4 07:35:15 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 4 Jan 2010 09:35:15 -0600
Subject: [Freeswitch-users] time stamp for sched_api
In-Reply-To: <1262602299063-4249417.post@n2.nabble.com>
References: <1262598891498-4249284.post@n2.nabble.com>
	<b8ad2a5b1001040223yc7ab19dm57e0101ee335a2f1@mail.gmail.com>
	<1262602299063-4249417.post@n2.nabble.com>
Message-ID: <B3DC0945-8C81-4BD5-8821-F48A3B894E83@freeswitch.org>

You could open a bounty and pay someone to write it...  also you can prepend it with a +sign... IE +60 and make it execute 60 seconds from NOW.

/b

On Jan 4, 2010, at 4:51 AM, Sharad wrote:

> 
> Thanks Mr. Baig..
> 
> So can you plz write the syntax for the same API if we want this to get
> executed on 4th Jan 2010 at 1800 hours.
> 
> Regards



From vinuth.madinur at gmail.com  Mon Jan  4 07:41:30 2010
From: vinuth.madinur at gmail.com (Vinuth Madinur)
Date: Mon, 4 Jan 2010 21:11:30 +0530
Subject: [Freeswitch-users] Hangup on silence?
In-Reply-To: <1b46b4e81001040710x4886d57ci8227c80c7fe3b796@mail.gmail.com>
References: <1b46b4e81001040710x4886d57ci8227c80c7fe3b796@mail.gmail.com>
Message-ID: <910309031001040741x219d4085k5821045f5b5edce8@mail.gmail.com>

You can call wait_for_silence with timeout as 1 minute. It'll wait for max 1
minute, but it'll return immediately when there is silence of mentioned
length. You can next invoke hangup in the dialplan.

Or, you can use event socket library to call "wait_for_silence" and wait for
"CHANNEL_EXECUTE_COMPLETE" with
variable_current_application=wait_for_silence, upon which you can hangup.

Helps?


On Mon, Jan 4, 2010 at 8:40 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Hi, is there a way to hangup after a certain amount of silence? my
> problem is that with regular PSTN calls I don't get a BYE from my
> provider until around 1 minute after the phone has been hanged up.
> This is pretty standard, and the provider is not at fault, but I would
> like to hangup the call after I detect a certain amount of silence
> time. The only thing I could find was wait_for_silence on the wiki,
> but that application only delays dialplan execution, and I would need
> something like: hangup_on_silence=30, where 30 means to hangup if 30
> seconds of silence have passed.
>
> Thanks for your help!
>
> Nicolas
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nicolas at medularis.com  Mon Jan  4 07:59:29 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 4 Jan 2010 12:59:29 -0300
Subject: [Freeswitch-users] Hangup on silence?
In-Reply-To: <910309031001040741x219d4085k5821045f5b5edce8@mail.gmail.com>
References: <1b46b4e81001040710x4886d57ci8227c80c7fe3b796@mail.gmail.com>
	<910309031001040741x219d4085k5821045f5b5edce8@mail.gmail.com>
Message-ID: <1b46b4e81001040759n1cb84b18n1cab0138a6ed05cf@mail.gmail.com>

Vinuth, thank you very much for your response. I guess the event
socket solution is the best for me, since I'm not using the dialplan.
for these calls Do you think there's an alternative for Javascript
though? hehe.


On Mon, Jan 4, 2010 at 12:41 PM, Vinuth Madinur
<vinuth.madinur at gmail.com> wrote:
> You can call wait_for_silence with timeout as 1 minute. It'll wait for max 1
> minute, but it'll return immediately when there is silence of mentioned
> length. You can next invoke hangup in the dialplan.
> Or, you can use event socket library to call "wait_for_silence" and wait for
> "CHANNEL_EXECUTE_COMPLETE" with
> variable_current_application=wait_for_silence, upon which you can hangup.
> Helps?
>


From sos at sokhapkin.dyndns.org  Mon Jan  4 05:51:23 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 4 Jan 2010 08:51:23 -0500
Subject: [Freeswitch-users] Was call answered or not?
Message-ID: <201001040851.23977.sos@sokhapkin.dyndns.org>

Which channel variable accessible from mod_cdr_csv can be used to reliable 
find out if the call was answered or not? "billsec" can't be used - it is 
equal to 0 if the call has been answered for less than 1 second.


From mike at jerris.com  Mon Jan  4 08:01:11 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Jan 2010 11:01:11 -0500
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <6E6E6D36-6E93-49BF-9516-FC028E73D279@ukgb.net>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com>
	<B6763940-A3CB-4006-8DF1-ADEDD9C68294@ukgb.net>
	<0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org>
	<CA53F057-4A51-457C-A673-B0B6E4DF482A@ukgb.net>
	<207e7a5e0912301103h17133fdfk22430bbce5a1718b@mail.gmail.com>
	<090E289D-76EE-4F4A-82DF-513626A915E6@ukgb.net>
	<79B790D4-DE06-47FF-B9FD-52D9E1EB8BCB@ukgb.net>
	<207e7a5e1001020840y1bd3bae2y2b683a19659c9d7d@mail.gmail.com>
	<6E6E6D36-6E93-49BF-9516-FC028E73D279@ukgb.net>
Message-ID: <5BED3371-0023-4D53-BBE1-37D241B4AAFB@jerris.com>

its all in there.

Mike

On Jan 4, 2010, at 4:14 AM, Ken Gillett wrote:

> I will look into this, but in the meantime, would it not be possible to simply copy the installed files? I realise that basically this must be possible, but practically it depends on what is installed where. I believe the default prefix is /usr/local/freeswitch, but is EVERYTHING in there or is other stuff scattered in other directories?
> 
> If it's all together I'm thinking I could simply tar the folder and copy the tarball. Wouldn't this be possible?
> 
> 
> On 2 Jan 2010, at 16:40, jonathan augenstine wrote:
> 
>> A dmg install package would need to be created.  A default package does not currently exist and it would need to be created.  You would need to do this, particularly if you are going to customize the build/install.
>> 
>> On Sat, Jan 2, 2010 at 5:17 AM, Ken Gillett <ken at ukgb.net> wrote:
>> One question still outstanding:-
>> 
>> How can I compile FS on one Mac and install it onto a different Mac? This means compiling on a MacPro running Snow Leopard and then installing onto a Snow Leopard Server which doesn't have the developer tools installed (and I don't want it to).
>> 
>> Advice on this would be appreciated.
>> 



From brian at freeswitch.org  Mon Jan  4 08:05:05 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 4 Jan 2010 10:05:05 -0600
Subject: [Freeswitch-users] Was call answered or not?
In-Reply-To: <201001040851.23977.sos@sokhapkin.dyndns.org>
References: <201001040851.23977.sos@sokhapkin.dyndns.org>
Message-ID: <AA7E74A2-3C0E-44F8-A585-0BE706D763C7@freeswitch.org>

Less than one second will usually mean ZERO... you can use 


Caller-Profile-Created-Time: 1262619753748917
Caller-Channel-Created-Time: 1262619753748917
Caller-Channel-Answered-Time: 1262619754069545
Caller-Channel-Progress-Time: 0
Caller-Channel-Progress-Media-Time: 1262619754069545
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0


These will give you in nano second I think.. maybe micro second.

/b

On Jan 4, 2010, at 7:51 AM, Sergey Okhapkin wrote:

> Which channel variable accessible from mod_cdr_csv can be used to reliable 
> find out if the call was answered or not? "billsec" can't be used - it is 
> equal to 0 if the call has been answered for less than 1 second.



From devel at thom.fr.eu.org  Mon Jan  4 08:07:37 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Mon, 04 Jan 2010 17:07:37 +0100
Subject: [Freeswitch-users] Zap channel not released when voicemail starts
Message-ID: <ea085d8b43f8105108d9c1063fb56534@thom.fr.eu.org>



Hello, 

I have an issue with voicemail and openzap channels. 

When an
incoming call on an openzap channel is bridged to voicemail, if that
channel is hung up before the beginning of voicemail recording, that
channel is kept open open until 3 or 4 seconds after the voicemail started
to record the message. 

What should I do to make freeswitch/voicemail
release the channel immediately when the caller hang up ? 

Thanks in
advance 

Fran?ois
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From sos at sokhapkin.dyndns.org  Mon Jan  4 08:19:14 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 4 Jan 2010 11:19:14 -0500
Subject: [Freeswitch-users] Was call answered or not?
In-Reply-To: <AA7E74A2-3C0E-44F8-A585-0BE706D763C7@freeswitch.org>
References: <201001040851.23977.sos@sokhapkin.dyndns.org>
	<AA7E74A2-3C0E-44F8-A585-0BE706D763C7@freeswitch.org>
Message-ID: <201001041119.15011.sos@sokhapkin.dyndns.org>

Thanks for the idea, if Caller-Channel-Answered-Time is not 0, then the call 
has been answered.

On Monday 04 January 2010, Brian West wrote:
> Less than one second will usually mean ZERO... you can use
>
>
> Caller-Profile-Created-Time: 1262619753748917
> Caller-Channel-Created-Time: 1262619753748917
> Caller-Channel-Answered-Time: 1262619754069545
> Caller-Channel-Progress-Time: 0
> Caller-Channel-Progress-Media-Time: 1262619754069545
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
>
>
> These will give you in nano second I think.. maybe micro second.
>
> /b
>
> On Jan 4, 2010, at 7:51 AM, Sergey Okhapkin wrote:
> > Which channel variable accessible from mod_cdr_csv can be used to
> > reliable find out if the call was answered or not? "billsec" can't be
> > used - it is equal to 0 if the call has been answered for less than 1
> > second.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From anthony.minessale at gmail.com  Mon Jan  4 08:30:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Jan 2010 10:30:51 -0600
Subject: [Freeswitch-users] How to control call volume?
In-Reply-To: <1b46b4e81001032245k7d10e027p46b21e50bfb2264f@mail.gmail.com>
References: <1b46b4e81001032245k7d10e027p46b21e50bfb2264f@mail.gmail.com>
Message-ID: <191c3a031001040830j527bcfd4g37797edcab2466f8@mail.gmail.com>

The volume should really be set on the devices who are originally encoding
the audio (the phone or analog card)
Digital audio never changes so the server is not the right place to mess
with the volume because you will have to actually manipulate the digital
signal to do it.  We have a way but I recommend you find the real source of
your problem.

<action application="set_audio_level" data="read 1"/>

change read to write if you want to do it going the other way


On Mon, Jan 4, 2010 at 12:45 AM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Hi, is there a way of controlling the volume of a call? I'm bridging
> calls with a JS script. Sometimes the people getting the calls
> complain the volume is too low. I've recorded a few of the calls and
> most of the times, while playing the recorded wav files, the volume of
> LegB (second leg of the bridge) is pretty hard to hear, even with the
> computer and player volume to the max (ok, it's a laptop, but even
> with headphones). I saw there are some volume control options for
> conferences, but I couldn't find anything for regular "originate
> calls" or bridges. I am doing transcoding, so that might help (?).
>
> Thank you very much for your help.
>
> Best,
>
> Nicolas
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at jerris.com  Mon Jan  4 08:41:45 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Jan 2010 11:41:45 -0500
Subject: [Freeswitch-users] Hangup on silence?
In-Reply-To: <1b46b4e81001040759n1cb84b18n1cab0138a6ed05cf@mail.gmail.com>
References: <1b46b4e81001040710x4886d57ci8227c80c7fe3b796@mail.gmail.com>
	<910309031001040741x219d4085k5821045f5b5edce8@mail.gmail.com>
	<1b46b4e81001040759n1cb84b18n1cab0138a6ed05cf@mail.gmail.com>
Message-ID: <36FB269F-4416-4CFE-AF7F-0E7FBC2BC7E3@jerris.com>

you can bind events in javascript in much the same way.  Why is it normal to not get a hangup for 60 seconds?  

Mike

On Jan 4, 2010, at 10:59 AM, Nicolas Brenner wrote:

> Vinuth, thank you very much for your response. I guess the event
> socket solution is the best for me, since I'm not using the dialplan.
> for these calls Do you think there's an alternative for Javascript
> though? hehe.
> 
> 
> On Mon, Jan 4, 2010 at 12:41 PM, Vinuth Madinur
> <vinuth.madinur at gmail.com> wrote:
>> You can call wait_for_silence with timeout as 1 minute. It'll wait for max 1
>> minute, but it'll return immediately when there is silence of mentioned
>> length. You can next invoke hangup in the dialplan.
>> Or, you can use event socket library to call "wait_for_silence" and wait for
>> "CHANNEL_EXECUTE_COMPLETE" with
>> variable_current_application=wait_for_silence, upon which you can hangup.
>> Helps?
>> 



From anthony.minessale at gmail.com  Mon Jan  4 08:53:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Jan 2010 10:53:17 -0600
Subject: [Freeswitch-users] system functions silently ignored
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E906AC73@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E906A04C@prato.activenetwerx.local>
	<CFF81042AFE645CCA2D6AE3276D63283@cune.pri>
	<CA5A491E9DEFBE4CB777DE97E21575E906A13E@prato.activenetwerx.local>
	<78ED9C83-CC46-41D5-AA2F-85291381B5C3@freeswitch.org>
	<CA5A491E9DEFBE4CB777DE97E21575E906AC73@prato.activenetwerx.local>
Message-ID: <191c3a031001040853s648a774bsc254ce616465e8dd@mail.gmail.com>

The best bet is to use Sangoma cards so you do not need any Dahdi,
That makes you an Emancipated Minor I guess.


On Sun, Jan 3, 2010 at 9:09 PM, Joseph L. Casale
<jcasale at activenetwerx.com>wrote:

> >Funny Pre's no longer exist use this:
> >
> >http://latest.freeswitch.org/freeswitch-1.0.5-latest.tar.gz
>
> That's what I was on, I just tried pre10 to see if it was a bug
> in that "latest" build.
>
> This is on hold for a day or two now, when this server shuts off
> it panic's with a fault related to dahdi, so given that, I am not
> going to trouble shoot this until obviously that's resolved.
>
> I'll get back to this once we have some new hardware...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jcasale at activenetwerx.com  Mon Jan  4 09:05:00 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Mon, 4 Jan 2010 17:05:00 +0000
Subject: [Freeswitch-users] system functions silently ignored
In-Reply-To: <191c3a031001040853s648a774bsc254ce616465e8dd@mail.gmail.com>
References: <CA5A491E9DEFBE4CB777DE97E21575E906A04C@prato.activenetwerx.local>
	<CFF81042AFE645CCA2D6AE3276D63283@cune.pri>
	<CA5A491E9DEFBE4CB777DE97E21575E906A13E@prato.activenetwerx.local>
	<78ED9C83-CC46-41D5-AA2F-85291381B5C3@freeswitch.org>
	<CA5A491E9DEFBE4CB777DE97E21575E906AC73@prato.activenetwerx.local>
	<191c3a031001040853s648a774bsc254ce616465e8dd@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906B34F@prato.activenetwerx.local>

>The best bet is to use Sangoma cards so you do not need any Dahdi, 
>That makes you an Emancipated Minor I guess.

I'll make a note of this point. I soon decided after this initial purchase
ages ago that the next time I ever needed FXO/S ports, I would use an ip gateway.

Thanks,
jlc


From william.suffill at gmail.com  Mon Jan  4 09:44:29 2010
From: william.suffill at gmail.com (William Suffill)
Date: Mon, 4 Jan 2010 12:44:29 -0500
Subject: [Freeswitch-users] time stamp for sched_api
In-Reply-To: <B3DC0945-8C81-4BD5-8821-F48A3B894E83@freeswitch.org>
References: <1262598891498-4249284.post@n2.nabble.com>
	<b8ad2a5b1001040223yc7ab19dm57e0101ee335a2f1@mail.gmail.com> 
	<1262602299063-4249417.post@n2.nabble.com>
	<B3DC0945-8C81-4BD5-8821-F48A3B894E83@freeswitch.org>
Message-ID: <6b65470d1001040944u508f3b25mc08c40a93242c232@mail.gmail.com>

Many programming languages already have functions to handle converting
date/time to a unix timestamp. For example:
http://www.php.net/manual/en/function.mktime.php

takes date/time assuming the local timezone and returns a unix timestamp
that could be used with sched_api.

-- W
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From anthony.minessale at gmail.com  Mon Jan  4 09:47:39 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Jan 2010 11:47:39 -0600
Subject: [Freeswitch-users] PHP ESL Problem
In-Reply-To: <6204a6f4ea0160933ba6493a292c1b9b.squirrel@correo.nodo50.org>
References: <285BD733E19541989B31B95871BF5642@fromage>
	<6204a6f4ea0160933ba6493a292c1b9b.squirrel@correo.nodo50.org>
Message-ID: <191c3a031001040947t6cc3ea8h6a3cb2bc7c06109e@mail.gmail.com>

What you are missing is that you are parsing the results incorrectly.

The events that are showing up as CUSTOM only do so because the id of CUSTOM
is 0
there are no types in the events received with recvEvent the are only used
for transport.

When you say "events plain all"

you are asking for events to be delivered, the FS events are not the same as
the events you are using
to communicate at the lowest level.

What you should be doing is checking for content-type of text/event-plain
and then and only then, get the payload with getBody.
This will contain a serialized event in it's entirety from FS.

for clarity sake I have added a new event SOCKET_DATA to tree and from now
on you will
see those low level events with that type.



On Sat, Jan 2, 2010 at 12:08 PM, Alberto Escudero <aep.lists at it46.se> wrote:

> I do not know if really helps you but we are facing the same problem in
> one of our implementations using the ESL.so for PHP.
>
> We have only see this problem when subscribing to the CHANNEL_STATE
>
> getType() should always match EventName... but it does not
> ./aep
>
> --
> Stopping junk mailers is good for the environment
>
> > Would someone please take a look at this simple PHP event socket script
> > and
> > tell me what I am doing wrong - or tell me that this could be a bug
> > elsewhere?  Any help would be appreciated.
> >
> > When I run the script without the call to execute(), everything seems
> > fine.
> > When I include the call to execute(), the calls to getType() return
> CUSTOM
> > for a while, then later start to return the correct name.
> >
> > #!/usr/bin/php
> > <?php
> > require_once 'ESL.php';
> > $endPoint = 'sofia/internal/695%192.168.100.132';
> >
> > $eventSocket = New ESLconnection('192.168.100.132', '8021', 'ClueCon');
> > $event = $eventSocket->events('plain', 'ALL');
> >
> > // call endpoint, get uuid
> > $event = $eventSocket->api('originate', $endPoint . ' &park');
> > $serializedEvent = explode("\n", $event->serialize());
> > foreach ($serializedEvent as $eventLine) {
> >     list($dummy, $uuid) = explode('+OK ', $eventLine);
> >     if ($uuid) { break; }
> > }
> >
> > // play announcement to endpoint
> > $event = $eventSocket->execute('playback', '/opt/ann/user-busy.wav',
> > $uuid);
> >
> > // monitor events
> > while (TRUE) {
> >     echo "getType: " . $event->getType() . "\n";
> >     $serializedEvent = explode("\n", $event->serialize());
> >     foreach ($serializedEvent as $eventLine) {
> >         list($header, $value) = explode(': ', $eventLine);
> >         if ($header == "Event-Name")   { printf($eventLine . "\n"); }
> >         if ($header == "Content-Type") { printf($eventLine . "\n"); }
> >     }
> >
> >       printf("\n");
> >     $event = $eventSocket->recvEvent();
> > }?>
> >
> >
> > Run without the call to execute():
> > ==================================
> > getType: CUSTOM
> > Content-Type: api/response
> >
> > getType: CHANNEL_CREATE
> > Event-Name: CHANNEL_CREATE
> >
> > getType: CHANNEL_OUTGOING
> > Event-Name: CHANNEL_OUTGOING
> >
> > getType: CHANNEL_ORIGINATE
> > Event-Name: CHANNEL_ORIGINATE
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> > getType: PRESENCE_IN
> > Event-Name: PRESENCE_IN
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> > getType: PRESENCE_IN
> > Event-Name: PRESENCE_IN
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> > getType: CALL_UPDATE
> > Event-Name: CALL_UPDATE
> >
> > getType: CHANNEL_PROGRESS
> > Event-Name: CHANNEL_PROGRESS
> >
> > getType: HEARTBEAT
> > Event-Name: HEARTBEAT
> >
> > getType: HEARTBEAT
> > Event-Name: RE_SCHEDULE
> >
> > getType: CALL_UPDATE
> > Event-Name: CALL_UPDATE
> >
> > getType: CODEC
> > Event-Name: CODEC
> >
> > getType: CODEC
> > Event-Name: CODEC
> >
> > getType: CHANNEL_ANSWER
> > Event-Name: CHANNEL_ANSWER
> >
> > getType: PRESENCE_IN
> > Event-Name: PRESENCE_IN
> >
> > getType: API
> > Event-Name: API
> >
> > getType: PRESENCE_IN
> > Event-Name: PRESENCE_IN
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> > getType: CHANNEL_EXECUTE
> > Event-Name: CHANNEL_EXECUTE
> >
> > getType: CHANNEL_PARK
> > Event-Name: CHANNEL_PARK
> >
> > getType: CHANNEL_HANGUP
> > Event-Name: CHANNEL_HANGUP
> >
> > getType: CHANNEL_UNPARK
> > Event-Name: CHANNEL_UNPARK
> >
> > getType: CHANNEL_EXECUTE_COMPLETE
> > Event-Name: CHANNEL_EXECUTE_COMPLETE
> >
> > getType: PRESENCE_IN
> > Event-Name: PRESENCE_IN
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> > getType: PRESENCE_IN
> > Event-Name: PRESENCE_IN
> >
> > getType: CHANNEL_HANGUP_COMPLETE
> > Event-Name: CHANNEL_HANGUP_COMPLETE
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> > getType: CHANNEL_DESTROY
> > Event-Name: CHANNEL_DESTROY
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> >
> > Run with the call to execute():
> > ===============================
> > getType: CUSTOM
> > Content-Type: command/reply
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_CREATE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_OUTGOING
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_ORIGINATE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_STATE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: PRESENCE_IN
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_STATE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: PRESENCE_IN
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_STATE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CALL_UPDATE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_PROGRESS
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CALL_UPDATE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CODEC
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CODEC
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_ANSWER
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: PRESENCE_IN
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: API
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: PRESENCE_IN
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_STATE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_EXECUTE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_PARK
> >
> > getType: CHANNEL_EXECUTE
> > Event-Name: CHANNEL_EXECUTE
> >
> > getType: CHANNEL_HANGUP
> > Event-Name: CHANNEL_HANGUP
> >
> > getType: CHANNEL_EXECUTE_COMPLETE
> > Event-Name: CHANNEL_EXECUTE_COMPLETE
> >
> > getType: COMMAND
> > Event-Name: COMMAND
> >
> > getType: CHANNEL_UNPARK
> > Event-Name: CHANNEL_UNPARK
> >
> > getType: CHANNEL_EXECUTE_COMPLETE
> > Event-Name: CHANNEL_EXECUTE_COMPLETE
> >
> > getType: PRESENCE_IN
> > Event-Name: PRESENCE_IN
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> > getType: PRESENCE_IN
> > Event-Name: PRESENCE_IN
> >
> > getType: CHANNEL_HANGUP_COMPLETE
> > Event-Name: CHANNEL_HANGUP_COMPLETE
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> > getType: CHANNEL_DESTROY
> > Event-Name: CHANNEL_DESTROY
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> >
> > Thanks,
> > Ron
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From nicolas at medularis.com  Mon Jan  4 10:15:14 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 4 Jan 2010 15:15:14 -0300
Subject: [Freeswitch-users] How to control call volume?
In-Reply-To: <191c3a031001040830j527bcfd4g37797edcab2466f8@mail.gmail.com>
References: <1b46b4e81001032245k7d10e027p46b21e50bfb2264f@mail.gmail.com>
	<191c3a031001040830j527bcfd4g37797edcab2466f8@mail.gmail.com>
Message-ID: <1b46b4e81001041015g74d550b3q3714d3e1f9cd6719@mail.gmail.com>

Great! Thanks! I'll play around with those setting to see how it goes.
Can I set that variable "per leg"? (instead of globally for a call).
The devices getting the calls are regular phones and the termination
service is provided by a few different VoIP companies. I'd say the
volume problem has to do with bad or poorly configured GSM gateways
(that's how they make calls to cellphones), plus maybe some problems
in the GSM network relating to poor signal or something like that. I
can't really control the PSTN or the GSM network and have almost zero
influence with the VoIP companies, so my best bet now is to mess with
the transcoding.

Thank you very much, we'll see how it goes.

Nicolas


On Mon, Jan 4, 2010 at 1:30 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> The volume should really be set on the devices who are originally encoding
> the audio (the phone or analog card)
> Digital audio never changes so the server is not the right place to mess
> with the volume because you will have to actually manipulate the digital
> signal to do it.? We have a way but I recommend you find the real source of
> your problem.
>
> <action application="set_audio_level" data="read 1"/>
>
> change read to write if you want to do it going the other way
>


From nicolas at medularis.com  Mon Jan  4 10:21:37 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 4 Jan 2010 15:21:37 -0300
Subject: [Freeswitch-users] Hangup on silence?
In-Reply-To: <36FB269F-4416-4CFE-AF7F-0E7FBC2BC7E3@jerris.com>
References: <1b46b4e81001040710x4886d57ci8227c80c7fe3b796@mail.gmail.com>
	<910309031001040741x219d4085k5821045f5b5edce8@mail.gmail.com>
	<1b46b4e81001040759n1cb84b18n1cab0138a6ed05cf@mail.gmail.com>
	<36FB269F-4416-4CFE-AF7F-0E7FBC2BC7E3@jerris.com>
Message-ID: <1b46b4e81001041021o7bf52b3bp5f1e4c2a5bf19d80@mail.gmail.com>

Event-based javascript? like Node.js? how would I bind the end of a
wait_for_silence to some script or callback function?

Here, when you call from a regular landline to another, the person
receiving the call may hangup and the call will still be alive, this
is so the person can "transfer" the call between phones connected to
the same line (e.g. kitchen and room phones). Only if the person
originating the call hangs up, the call is terminated right away.
Maybe 60 seconds is too much, but that's the way the PSTN works here.


On Mon, Jan 4, 2010 at 1:41 PM, Michael Jerris <mike at jerris.com> wrote:
> you can bind events in javascript in much the same way. ?Why is it normal to not get a hangup for 60 seconds?
>
> Mike
>


From anthony.minessale at gmail.com  Mon Jan  4 11:16:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Jan 2010 13:16:23 -0600
Subject: [Freeswitch-users] How to control call volume?
In-Reply-To: <1b46b4e81001041015g74d550b3q3714d3e1f9cd6719@mail.gmail.com>
References: <1b46b4e81001032245k7d10e027p46b21e50bfb2264f@mail.gmail.com>
	<191c3a031001040830j527bcfd4g37797edcab2466f8@mail.gmail.com>
	<1b46b4e81001041015g74d550b3q3714d3e1f9cd6719@mail.gmail.com>
Message-ID: <191c3a031001041116x56a102f3y96294c493a30ce67@mail.gmail.com>

yes the example is per_leg per_direction

On Mon, Jan 4, 2010 at 12:15 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Great! Thanks! I'll play around with those setting to see how it goes.
> Can I set that variable "per leg"? (instead of globally for a call).
> The devices getting the calls are regular phones and the termination
> service is provided by a few different VoIP companies. I'd say the
> volume problem has to do with bad or poorly configured GSM gateways
> (that's how they make calls to cellphones), plus maybe some problems
> in the GSM network relating to poor signal or something like that. I
> can't really control the PSTN or the GSM network and have almost zero
> influence with the VoIP companies, so my best bet now is to mess with
> the transcoding.
>
> Thank you very much, we'll see how it goes.
>
> Nicolas
>
>
> On Mon, Jan 4, 2010 at 1:30 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > The volume should really be set on the devices who are originally
> encoding
> > the audio (the phone or analog card)
> > Digital audio never changes so the server is not the right place to mess
> > with the volume because you will have to actually manipulate the digital
> > signal to do it.  We have a way but I recommend you find the real source
> of
> > your problem.
> >
> > <action application="set_audio_level" data="read 1"/>
> >
> > change read to write if you want to do it going the other way
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From msc at freeswitch.org  Mon Jan  4 13:23:21 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Jan 2010 13:23:21 -0800
Subject: [Freeswitch-users] How to control call volume?
In-Reply-To: <191c3a031001041116x56a102f3y96294c493a30ce67@mail.gmail.com>
References: <1b46b4e81001032245k7d10e027p46b21e50bfb2264f@mail.gmail.com>
	<191c3a031001040830j527bcfd4g37797edcab2466f8@mail.gmail.com>
	<1b46b4e81001041015g74d550b3q3714d3e1f9cd6719@mail.gmail.com>
	<191c3a031001041116x56a102f3y96294c493a30ce67@mail.gmail.com>
Message-ID: <87f2f3b91001041323r5984942ete1f326bac4d15cc3@mail.gmail.com>

FYI, I just wikified this dp app:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_audio_level

Feel free to add/edit as needed.
-MC

On Mon, Jan 4, 2010 at 11:16 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> yes the example is per_leg per_direction
>
>
> On Mon, Jan 4, 2010 at 12:15 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> Great! Thanks! I'll play around with those setting to see how it goes.
>> Can I set that variable "per leg"? (instead of globally for a call).
>> The devices getting the calls are regular phones and the termination
>> service is provided by a few different VoIP companies. I'd say the
>> volume problem has to do with bad or poorly configured GSM gateways
>> (that's how they make calls to cellphones), plus maybe some problems
>> in the GSM network relating to poor signal or something like that. I
>> can't really control the PSTN or the GSM network and have almost zero
>> influence with the VoIP companies, so my best bet now is to mess with
>> the transcoding.
>>
>> Thank you very much, we'll see how it goes.
>>
>> Nicolas
>>
>>
>> On Mon, Jan 4, 2010 at 1:30 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > The volume should really be set on the devices who are originally
>> encoding
>> > the audio (the phone or analog card)
>> > Digital audio never changes so the server is not the right place to mess
>> > with the volume because you will have to actually manipulate the digital
>> > signal to do it.  We have a way but I recommend you find the real source
>> of
>> > your problem.
>> >
>> > <action application="set_audio_level" data="read 1"/>
>> >
>> > change read to write if you want to do it going the other way
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jerry.richards at teotech.com  Mon Jan  4 15:49:32 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 4 Jan 2010 15:49:32 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
Message-ID: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>


Hello,

I have one FS instance that is working well with a PRI and running FS
version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
upgraded it.

Now, I am trying to bring up another FS instance (basically a clone of the
first), but the PRI does not work.  When I attempt to make an
internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
both servers are using the latest Sangoma Wanpipe driver, and I copied the
conf XML file tree from the old server to the new one.  I think the problem
has to do with the openzap module, but I'm having difficulty isolating the
problem.  Could it have built the openzap module incorrectly?  Another
difference is that I installed 1.0.5pre9 from scratch on the new server
(i.e. it never had 1.0.4 running on it).

I put the FS log into the pastebin when an outbound call attempt is made:  

http://pastebin.freeswitch.org/11675

Could someone give me a pointer on what to try next?

Thanks and Best Regards,
Jerry



From anthony.minessale at gmail.com  Mon Jan  4 16:00:05 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Jan 2010 18:00:05 -0600
Subject: [Freeswitch-users] Zap channel not released when voicemail
	starts
In-Reply-To: <ea085d8b43f8105108d9c1063fb56534@thom.fr.eu.org>
References: <ea085d8b43f8105108d9c1063fb56534@thom.fr.eu.org>
Message-ID: <191c3a031001041600y1c98465cifcc522056096f746@mail.gmail.com>

hangup detection on TDM is a bitch.


On Mon, Jan 4, 2010 at 10:07 AM, Fran?ois Legal <devel at thom.fr.eu.org>wrote:

> Hello,
>
>
>
> I have an issue with voicemail and openzap channels.
>
> When an incoming call on an openzap channel is bridged to voicemail, if
> that channel is hung up before the beginning of voicemail recording, that
> channel is kept open open until 3 or 4 seconds after the voicemail started
> to record the message.
>
> What should I do to make freeswitch/voicemail release the channel
> immediately when the caller hang up ?
>
>
>
> Thanks in advance
>
>
>
> Fran?ois
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From timuckun at gmail.com  Mon Jan  4 18:42:57 2010
From: timuckun at gmail.com (Tim Uckun)
Date: Tue, 5 Jan 2010 15:42:57 +1300
Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS
	based callcenter)
In-Reply-To: <20091219014359.GA21798@hijacked.us>
References: <20091219014359.GA21798@hijacked.us>
Message-ID: <855e4dcf1001041842u46c6d36q9c2e4ece2ced7380@mail.gmail.com>

> http://opencsm.org/downloads/spice-telephony-0.9.6.tar.gz
>

There seems to be something wrong with both opencsm.org and wiki.opencsm.org.

Just thought I'd let you know.


From help at pdscc.com  Mon Jan  4 19:08:15 2010
From: help at pdscc.com (Harondel J. Sibble)
Date: Mon, 04 Jan 2010 19:08:15 -0800
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <C2EB0262-CB8A-44F5-9978-1C5F8073832F@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20091228233838.75E611694@sinclaire.sibble.net>,
	<C2EB0262-CB8A-44F5-9978-1C5F8073832F@freeswitch.org>
Message-ID: <20100105030813.8B90012F5@sinclaire.sibble.net>

Brian, Just following up on this, any news?

On 28 Dec 2009 at 17:49, Brian West wrote:

> I'm still not done with this I think we found a bug in the lib... Viktor
> fixed it today and I'm going to retry after I get done testing G729 more
> today!  ;)
> 
> /b
> 
> On Dec 28, 2009, at 5:38 PM, Harondel J. Sibble wrote:
> 
> > Hmm, okay, I went back to basics and did a full rebuild for 1.0.4 svn
> trunk, 
> > I am able to now enroll my nokia e61i running the beta 2.0.7 Tiviphone 
> > client, however I am not seeing the enrollment option popup in zfone 0.92
> > build 218 on windows in front of an x-lite client.
> > 
> > Any suggestions on what I should look at to troubleshoot this?
> > 
> > I am waiting for the Tivi folks to send a 2.0.7 beta for windows mobile,
> but 
> > until then....

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From max.bridgewater at gmail.com  Mon Jan  4 19:34:45 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 4 Jan 2010 22:34:45 -0500
Subject: [Freeswitch-users] Unable to start mod_java
Message-ID: <c55d46951001041934n49bfcbf4h42564e4548aa301a@mail.gmail.com>

Hi,

I built Freeswitch with mod_java enabled. But now, when Freeswitch starts, I
get the following error message:


2010-01-04 22:32:46.574770 [NOTICE] switch_loadable_module.c:271 Adding API
Function 'jsrun'
2010-01-04 22:32:46.574811 [NOTICE] switch_loadable_module.c:271 Adding API
Function 'jsapi'
2010-01-04 22:32:46.575306 [NOTICE] modjava.c:244 Java Framework Loading...
2010-01-04 22:32:46.575721 [ERR] modjava.c:133 Error loading
/usr/local/java/jdk1.6.0_17/jre/lib/i386/client/libjvm.so
2010-01-04 22:32:46.575743 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_java.so
**Module load routine returned an error**
2010-01-04 22:32:46.576742 [CONSOLE] switch_loadable_module.c:890
Successfully Loaded [mod_lua]
2010-01-04 22:32:46.576748 [NOTICE] switch_loadable_module.c:209 Adding
Dialplan 'LUA'
2010-01-04 22:32:46.576795 [NOTICE] switch_loadable_module.c:249 Adding
Application 'lua'


As far as I can tell the
/usr/local/java/jdk1.6.0_17/jre/lib/i386/client/libjvm.so is the correct
path.
Any idea?

max.
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From help at pdscc.com  Mon Jan  4 20:23:30 2010
From: help at pdscc.com (Harondel J. Sibble)
Date: Mon, 04 Jan 2010 20:23:30 -0800
Subject: [Freeswitch-users] encrypt voicemail emails and attachments?
Message-ID: <20100105042327.7CBF412DD@sinclaire.sibble.net>

Looking throught the wiki, I see various configs for having FS email you a 
copy of received voicemail messages, has anyone done any work with having the 
voicemail messages gpg encrypted with public prior to sending?  Or is that 
something that should pretty much be handled at the mta level leaving FS out 
of the mix altogether? I'm thinking probably so, but before I try to do this, 
I figured i'd ask first.


-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From jason at jasonjgw.net  Mon Jan  4 20:25:08 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 5 Jan 2010 15:25:08 +1100
Subject: [Freeswitch-users] Unable to start mod_java
In-Reply-To: <c55d46951001041934n49bfcbf4h42564e4548aa301a@mail.gmail.com>
References: <c55d46951001041934n49bfcbf4h42564e4548aa301a@mail.gmail.com>
Message-ID: <20100105042508.GA25483@jdc.jasonjgw.net>

Max Bridgewater <max.bridgewater at gmail.com> wrote:
 
> As far as I can tell the
> /usr/local/java/jdk1.6.0_17/jre/lib/i386/client/libjvm.so is the correct
> path.
> Any idea?

Permissions?



From mike at jerris.com  Mon Jan  4 20:51:33 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Jan 2010 23:51:33 -0500
Subject: [Freeswitch-users] encrypt voicemail emails and attachments?
In-Reply-To: <20100105042327.7CBF412DD@sinclaire.sibble.net>
References: <20100105042327.7CBF412DD@sinclaire.sibble.net>
Message-ID: <03DDC308-7EC5-46EC-8AFB-CC73417E20F5@jerris.com>

you can just make a shell script (or perl or whatever) that is called as the mailer that does this.

Mike

On Jan 4, 2010, at 11:23 PM, Harondel J. Sibble wrote:

> Looking throught the wiki, I see various configs for having FS email you a 
> copy of received voicemail messages, has anyone done any work with having the 
> voicemail messages gpg encrypted with public prior to sending?  Or is that 
> something that should pretty much be handled at the mta level leaving FS out 
> of the mix altogether? I'm thinking probably so, but before I try to do this, 
> I figured i'd ask first.
> 



From sharad at coraltele.com  Mon Jan  4 20:56:36 2010
From: sharad at coraltele.com (Sharad)
Date: Mon, 4 Jan 2010 20:56:36 -0800 (PST)
Subject: [Freeswitch-users] time stamp for sched_api
In-Reply-To: <6b65470d1001040944u508f3b25mc08c40a93242c232@mail.gmail.com>
References: <1262598891498-4249284.post@n2.nabble.com>
	<b8ad2a5b1001040223yc7ab19dm57e0101ee335a2f1@mail.gmail.com>
	<1262602299063-4249417.post@n2.nabble.com>
	<B3DC0945-8C81-4BD5-8821-F48A3B894E83@freeswitch.org>
	<6b65470d1001040944u508f3b25mc08c40a93242c232@mail.gmail.com>
Message-ID: <1262667396468-4253597.post@n2.nabble.com>




William Suffill wrote:
> 
> Many programming languages already have functions to handle converting
> date/time to a unix timestamp. For example:
> http://www.php.net/manual/en/function.mktime.php
> 
> takes date/time assuming the local timezone and returns a unix timestamp
> that could be used with sched_api.
> 
> -- W
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 


thanks /b n william for your kind answer.

regards
sharad
-- 
View this message in context: http://n2.nabble.com/time-stamp-for-sched-api-tp4249284p4253597.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sharad at coraltele.com  Mon Jan  4 20:59:18 2010
From: sharad at coraltele.com (Sharad)
Date: Mon, 4 Jan 2010 20:59:18 -0800 (PST)
Subject: [Freeswitch-users] [!! SPAM] Re: Self alarm
In-Reply-To: <207e7a5e1001012050g18e5563dpe99e55b0509c5625@mail.gmail.com>
References: <1262250725607-4235713.post@n2.nabble.com>
	<1262326847726-4238924.post@n2.nabble.com>
	<2A1E0DAF-A680-47B6-AC49-6A80FEC312A2@jerris.com>
	<002b01ca8b65$16762640$0c04a8c0@compaq77db609e>
	<207e7a5e1001012050g18e5563dpe99e55b0509c5625@mail.gmail.com>
Message-ID: <1262667558154-4253610.post@n2.nabble.com>




jonathan augenstine wrote:
> 
> I believe that the question is, what do you want to alarm?  Do you want to
> setup basic monitoring of the system?  Are you trying to track T1 alarms?
>  Your question is too vague to answer.
> 
> On Fri, Jan 1, 2010 at 8:35 PM, Sharad <sharad at coraltele.com> wrote:
> 
>> Self Alarm..
>>
>> ----- Original Message -----
>> *From:* [hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4241557&i=0>
>> *To:* [hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4241557&i=1>
>> *Sent:* Friday, January 01, 2010 9:04 PM
>> *Subject:* [!! SPAM] Re: [Freeswitch-users] Self alarm
>>
>> The same what?
>>
>> On Jan 1, 2010, at 1:20 AM, Sharad <[hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4239714&i=0>>
>> wrote:
>>
>> >
>> > Hi
>> >
>> > I am also intresting in the same.
>> >
>> > Is there any script for this functionality.
>> >
>> > Regards
>> > --
>> > View this message in context:
>> http://n2.nabble.com/Self-alarm-tp4235713p4238924.html
>> > Sent from the freeswitch-users mailing list archive at Nabble.com.
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > [hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4239714&i=1>
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> > users
>> > http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> [hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4239714&i=2>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> ------------------------------
>> View this message in context: Re: [!! SPAM] Re: [Freeswitch-users] Self
>> alarm <http://n2.nabble.com/Self-alarm-tp4235713p4241557.html>
>>
>> Sent from the freeswitch-users mailing list
>> archive<http://n2.nabble.com/freeswitch-users-f2379917.html>at
>> Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

we are writing a script for the self alarm application which will make the
reminder call to the predefined user at the defined time. It is under
testing& will take some more days.

Once it is done, I will upload the same for everyone.

regards
sharad

-- 
View this message in context: http://n2.nabble.com/Self-alarm-tp4235713p4253610.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From help at pdscc.com  Mon Jan  4 22:54:01 2010
From: help at pdscc.com (Harondel J. Sibble)
Date: Mon, 04 Jan 2010 22:54:01 -0800
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <C2EB0262-CB8A-44F5-9978-1C5F8073832F@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20091228233838.75E611694@sinclaire.sibble.net>,
	<C2EB0262-CB8A-44F5-9978-1C5F8073832F@freeswitch.org>
Message-ID: <20100105065356.AEE0612F5@sinclaire.sibble.net>

Maybe that's what's affecting me now.....

I've both phones registered (confirmed by calling 9787) on both devices and 
it says each device is already enrolled. (how does one un-enroll????). Both 
phones are running the Tivi 2.0.7 beta. 

Now however, other than the first call I made between devices after 
enrollment, the sas is not matching anymore.

I set both these options in the console

global_action application="set" data="zrtp_enrollment=true"
global_setvar zrtp_secure_media=true

What should I be looking for in the console output

On 28 Dec 2009 at 17:49, Brian West wrote:

> I'm still not done with this I think we found a bug in the lib... Viktor
> fixed it today and I'm going to retry after I get done testing G729 more
> today!  ;)
> 
> /b
> 
> On Dec 28, 2009, at 5:38 PM, Harondel J. Sibble wrote:
> 
> > Hmm, okay, I went back to basics and did a full rebuild for 1.0.4 svn
> trunk, 
> > I am able to now enroll my nokia e61i running the beta 2.0.7 Tiviphone 
> > client, however I am not seeing the enrollment option popup in zfone 0.92
> > build 218 on windows in front of an x-lite client.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From yehavi.bourvine at gmail.com  Tue Jan  5 00:38:22 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 5 Jan 2010 10:38:22 +0200
Subject: [Freeswitch-users] Polycom & TLS - help please...
In-Reply-To: <3968BF39-7A15-4F9F-8FA2-A78BC4F0F38E@freeswitch.org>
References: <d361ca7b1001030757m2f55a400n3e2b3b74185c4202@mail.gmail.com>
	<BFB88CA0-8319-4D05-B3CD-39E3450DE157@jerris.com>
	<D950E561-1B94-48E0-9221-3739BAB8F91A@freeswitch.org>
	<d361ca7b1001031932m7695150fv967c0df683083af9@mail.gmail.com>
	<3968BF39-7A15-4F9F-8FA2-A78BC4F0F38E@freeswitch.org>
Message-ID: <d361ca7b1001050038x4e7103bdke322fd8741403172@mail.gmail.com>

Thanks, I have a partial success which involved two steps:

   - The wiki says to create a root certifcate with *gentls_cert setup *with
   no other parameters; I had to add my domain's data to this command. The new
   certificate has been downloaded to the phone.
   - Replaced the registrar definitions in the phone's config files from IP
   address to the server's name.
   - The above setup worked as-is. To be sure I've added the NAPTR records
   to the DNS after the above two steps worked.
      - BTW, the wiki says that the NAPTR records are not mandatory, thus I
      did not add them at the first place.

I said "partial" because now I have a phenomenon similar (not the same, but
close to) to the one I have with the SNOMs: The TLS link is reseted after a
while and then a new (additional) registration is done. I'll continue search
it per the tips I got on the other topic.

                          Thanks! __Yehavi:

2010/1/4 Brian West <brian at freeswitch.org>

> OK to properly use TLS you have to setup NAPTR and SRV records and the DNS
> domain has to match the cert.  Did you do that?
>
> /b
>
> On Jan 3, 2010, at 9:32 PM, Yehavi Bourvine wrote:
>
> > I've built the slef-signed root certificate and server;s certificate per
> the TLS wiki, and installed the root certificate on the phone (both manually
> and via the config files). I did not enter the "== untrusted ==" instead of
> the cerificate (as the Polycom's wiki suggests) as the pone doesn't allow
> this. It accepted the certificate.
> >
> > I've tried this on 501 (running 3.1.3 which is the last supported version
> on it), and 550 & 650 running 3.2.2.
> >
> >                             Thans, __Yehavi:
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From a.alalousi at gmail.com  Tue Jan  5 01:40:14 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Tue, 5 Jan 2010 09:40:14 +0000
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
Message-ID: <cd93b3701001050140q14182667o8e90dba27fa3bc85@mail.gmail.com>

Hi Jerry,

Looked at your log and your problem is quiet simple to resolve, but first
here's what's happenning:

You copied the conf/ subtree to your new server. As such, you have also
duplicated your vars.xml. By doing so, you have set the domain on the *new
server* to the same domain used on your old server which would be fine, *but
* the default domain settings used by FS is to use your primary IPv4 IP
address as your domain.

By duplicating the conf subtree from the old server, you have effectively
bound the new instance of FS to a domain that is the IP address of the old
server, if this makes sense.

You can see this on third line of your log:

*192.168.72.29 Rejected by acl "domains"*

To resolve this, modify your vars.xml on the new server to reflect whatever
domain it is you want to route, or set the domain to the new server's IP
address like so:

 <X-PRE-PROCESS cmd="set" data="domain=IP_ADDRESS_OF_NEW_SERVER"/>

You also need to check that any other files (e.g. the conf/sip_profiles,
conf/directory/ and conf/dialplan/ hierarchy) are modified to reflect the
new server settings as well.

In the limit, resolving those conflicts will also resolve your issues,
unless there is something else that's wrong.

Let's know how you get along.

Regards,

Ahmed.


2010/1/4 Jerry Richards <jerry.richards at teotech.com>

>
> Hello,
>
> I have one FS instance that is working well with a PRI and running FS
> version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
> upgraded it.
>
> Now, I am trying to bring up another FS instance (basically a clone of the
> first), but the PRI does not work.  When I attempt to make an
> internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
> both servers are using the latest Sangoma Wanpipe driver, and I copied the
> conf XML file tree from the old server to the new one.  I think the problem
> has to do with the openzap module, but I'm having difficulty isolating the
> problem.  Could it have built the openzap module incorrectly?  Another
> difference is that I installed 1.0.5pre9 from scratch on the new server
> (i.e. it never had 1.0.4 running on it).
>
> I put the FS log into the pastebin when an outbound call attempt is made:
>
> http://pastebin.freeswitch.org/11675
>
> Could someone give me a pointer on what to try next?
>
> Thanks and Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From a.alalousi at gmail.com  Tue Jan  5 01:44:20 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Tue, 5 Jan 2010 09:44:20 +0000
Subject: [Freeswitch-users] Zap channel not released when voicemail
	starts
In-Reply-To: <191c3a031001041600y1c98465cifcc522056096f746@mail.gmail.com>
References: <ea085d8b43f8105108d9c1063fb56534@thom.fr.eu.org>
	<191c3a031001041600y1c98465cifcc522056096f746@mail.gmail.com>
Message-ID: <cd93b3701001050144m648905e0i76a0f59fa3b01db1@mail.gmail.com>

I'll second that.

My way of dealing with it has been to write a little script to detect
hangups on the TDM end, then force release the corresponding "B-leg" that is
hooked up to VM. In the process of converting this to an FS module.

Not clean .. but works. Would have liked to see the same code within FS core
and, if appropriate, the VM subsystem to achieve the same end.

Regards,

Ahmed.


2010/1/5 Anthony Minessale <anthony.minessale at gmail.com>

> hangup detection on TDM is a bitch.
>
>
> On Mon, Jan 4, 2010 at 10:07 AM, Fran?ois Legal <devel at thom.fr.eu.org>wrote:
>
>> Hello,
>>
>>
>>
>> I have an issue with voicemail and openzap channels.
>>
>> When an incoming call on an openzap channel is bridged to voicemail, if
>> that channel is hung up before the beginning of voicemail recording, that
>> channel is kept open open until 3 or 4 seconds after the voicemail started
>> to record the message.
>>
>> What should I do to make freeswitch/voicemail release the channel
>> immediately when the caller hang up ?
>>
>>
>>
>> Thanks in advance
>>
>>
>>
>> Fran?ois
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From mcampbellsmith at gmail.com  Tue Jan  5 03:07:50 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 5 Jan 2010 22:07:50 +1100
Subject: [Freeswitch-users] No audio after Remote SDP:
In-Reply-To: <33c87fa31001021519j41ead649rb4a85e6be6236254@mail.gmail.com>
References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com>
	<33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com>
	<772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org>
	<33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com>
	<191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com>
	<33c87fa30912301213g3d24b773k86b95e7ce08f6761@mail.gmail.com>
	<7D75E6D6-8AB9-429E-A5A9-1639C0D5AD09@freeswitch.org>
	<33c87fa31001012330g69fc8e88m61e648d893c4d8f7@mail.gmail.com>
	<180A53B1-EFB6-4B17-AA7C-19B007ADE04C@freeswitch.org>
	<33c87fa31001021519j41ead649rb4a85e6be6236254@mail.gmail.com>
Message-ID: <33c87fa31001050307v3cbcae01q9d7d47383a63ad33@mail.gmail.com>

OK.. I have looked at this some more...   Below is the syslog from the SPA3102:

THIS WORKS
Jan  5 21:18:17 92.xx.xx.xx [0]->124.xxx.xxx.xxx:442(544)
Jan  5 21:18:17 92.xx.xx.xx REGISTER sip:myddns.dydns.org:442 SIP/2.0
Via: SIP/2.0/TLS 92.xx.xx.xx:5069;branch=z9hG4bK-7b7ebbf9  From: 2001
<sip:2001 at myddns.dydns.org:442>;tag=b69c38c549e24c42o0  To: 2001
<sip:2001 at myddns.dydns.org:442>  Call-ID:
e6b918dc-71d58fe3 at 192.168.1.3  CSeq: 44358 REGISTER  Max-Forwards: 70
Contact: 2001 <sip:2001 at 92.xx.xx.xx:5069;transport=tls>;expires=600
User-Agent: Linksys/SPA3102-5.1.10(GW)  Content-Length: 0  Allow: ACK,
BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER  Supported:
x-sipura, replaces
Jan  5 21:18:17 92.xx.xx.xx
Jan  5 21:18:17 92.xx.xx.xx
Jan  5 21:18:17 92.xx.xx.xx [0]<<124.xxx.xxx.xxx:442(658)
Jan  5 21:18:17 92.xx.xx.xx [0]<<124.xxx.xxx.xxx:442(658)
Jan  5 21:18:17 92.xx.xx.xx SIP/2.0 401 Unauthorized  Via: SIP/2.0/TLS
92.xx.xx.xx:5069;branch=z9hG4bK-7b7ebbf9;rport=5069  From: 2001
<sip:2001 at myddns.dydns.org:442>;tag=b69c38c549e24c42o0  To: 2001
<sip:2001 at myddns.dydns.org:442>;tag=Bav1HeBr3jm3B  Call-ID:
e6b918dc-71d58fe3 at 192.168.1.3  CSeq: 44358 REGISTER  User-Agent:
FreeSWITCH-mod_sofia/1.0.trunk-16131  Allow: INVITE, ACK, BYE, CANCEL,
OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH,
SUBSCRIBE  Supported: timer, precondition, path, replaces
WWW-Authenticate: Digest realm="myddns.dydns.org",
nonce="a4128380-f9e3-11de-99eb-53ce5686ac9a", algorithm=MD5,
qop="auth"  Content-Length: 0
Jan  5 21:18:17 92.xx.xx.xx
Jan  5 21:18:17 92.xx.xx.xx
Jan  5 21:18:17 92.xx.xx.xx [0]->124.xxx.xxx.xxx:442(782)
Jan  5 21:18:17 92.xx.xx.xx [0]->124.xxx.xxx.xxx:442(782)
Jan  5 21:18:17 92.xx.xx.xx REGISTER sip:myddns.dydns.org:442 SIP/2.0
Via: SIP/2.0/TLS 92.xx.xx.xx:5069;branch=z9hG4bK-18f00822  From: 2001
<sip:2001 at myddns.dydns.org:442>;tag=b69c38c549e24c42o0  To: 2001
<sip:2001 at myddns.dydns.org:442>  Call-ID:
e6b918dc-71d58fe3 at 192.168.1.3  CSeq: 44359 REGISTER  Max-Forwards: 70
Authorization: Digest
username="2001",realm="myddns.dydns.org",nonce="a4128380-f9e3-11de-99eb-53ce5686ac9a",uri="sip:myddns.dydns.org:442",algorithm=MD5,response="324ee93184ae202be4a209f5a9255229",qop=auth,nc=00000001,cnonce="804844b"
 Contact: 2001 <sip:2001 at 92.xx.xx.xx:5069;transport=tls>;expires=600
User-Agent: Linksys/SPA3102-5.1.10(GW)  Content-Length: 0  Allow: ACK,
BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER  Supported:
x-sipura, replaces

THIS DOES NOT WORK
Jan  5 22:05:48 92.xxx.xxx.xxx REGISTER sip:myddns.dydns.org:442
SIP/2.0  Via: SIP/2.0/TLS 192.168.1.3:5070;branch=z9hG4bK-faf8477a
From: 2001 <sip:2001 at myddns.dydns.org:442>;tag=b065057e3ed0befdo0  To:
2001 <sip:2001 at myddns.dydns.org:442>  Call-ID:
e6b918dc-71d58fe3 at 192.168.1.3  CSeq: 44435 REGISTER  Max-Forwards: 70
Contact: 2001 <sip:2001 at 192.168.1.3:5070;transport=tls>;expires=600
User-Agent: Linksys/SPA3102-5.1.10(GW)  Content-Length: 0  Allow: ACK,
BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER  Supported:
x-sipura, replaces
Jan  5 22:05:48 92.xxx.xxx.xxx SIP/2.0 401 Unauthorized  Via:
SIP/2.0/TLS 192.168.1.3:5070;branch=z9hG4bK-faf8477a;received=92.xxx.xxx.xxx;rport=5070
 From: 2001 <sip:2001 at myddns.dydns.org:442>;tag=b065057e3ed0befdo0
To: 2001 <sip:2001 at myddns.dydns.org:442>;tag=N51jvyeca9Umj  Call-ID:
e6b918dc-71d58fe3 at 192.168.1.3  CSeq: 44435 REGISTER  User-Agent:
FreeSWITCH-mod_sofia/1.0.trunk-16131  Allow: INVITE, ACK, BYE, CANCEL,
OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH,
SUBSCRIBE  Supported: timer, precondition, path, replaces
WWW-Authenticate: Digest realm="myddns.dydns.org",
nonce="476097b0-f9ea-11de-99fd-53ce5686ac9a", algorithm=MD5,
qop="auth"  Content-Length: 0
Jan  5 22:05:48 92.xxx.xxx.xxx
Jan  5 22:05:48 92.xxx.xxx.xxx
Jan  5 22:05:58 92.xxx.xxx.xxx [0]->124.xxx.xxx.xxx:442(783)
Jan  5 22:05:58 92.xxx.xxx.xxx [0]->124.xxx.xxx.xxx:442(783)
Jan  5 22:05:58 92.xxx.xxx.xxx REGISTER sip:myddns.dydns.org:442
SIP/2.0  Via: SIP/2.0/TLS 92.xxx.xxx.xxx:5070;branch=z9hG4bK-bfc992b3
From: 2001 <sip:2001 at myddns.dydns.org:442>;tag=b065057e3ed0befdo0  To:
2001 <sip:2001 at myddns.dydns.org:442>  Call-ID:
e6b918dc-71d58fe3 at 192.168.1.3  CSeq: 44436 REGISTER  Max-Forwards: 70
Authorization: Digest
username="2001",realm="myddns.dydns.org",nonce="476097b0-f9ea-11de-99fd-53ce5686ac9a",uri="sip:myddns.dydns.org:442",algorithm=MD5,response="466ea78ac2ccddca991a5c3d4d021bed",qop=auth,nc=00000001,cnonce="ebd80711"
 Contact: 2001 <sip:2001 at 92.xxx.xxx.xxx:5070;transport=tls>;expires=600
 User-Agent: Linksys/SPA3102-5.1.10(GW)  Content-Length: 0  Allow:
ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER  Supported:
x-sipura, replaces


The difference was a hard coded external IP address in the first
session that worked.  However I can't have it set like this as the IP
address is not static.

The second REGISTER in the Not Working session seems to be ignored by
FreeSwitch (I don't see it in the logs of FS either).   Is there
something in this Register that causes FS to ignore it?

Thanks
On Sun, Jan 3, 2010 at 10:19 AM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> I have a Linksys SPA3102, NAT'd on the internet (remotely) and
> connected to my FS on the otherside of the world, which is also
> natted. ?A PAP2T is connected on the same subnet as the FS. ?The 3102
> registers successfully and a call can be set up from the PAP2 to the
> 3102.
>
> However, after FS receives the Remote SDP the audio stops (ring tone
> stops in my case)
>
> The FS router is upnp enabled. ?The SPA3102 router is NOT upnp enabled
> (SPA3102 does not support upnp anyway I think).
>
>
>
> On Sun, Jan 3, 2010 at 3:03 AM, Brian West <brian at freeswitch.org> wrote:
>> Are you behind a nat-pmp/upnp router?
>> /b
>> On Jan 2, 2010, at 1:30 AM, Mark Campbell-Smith wrote:
>>
>> Hi!
>>
>> Both are auto-nat:
>> ???<param name="ext-rtp-ip" value="auto-nat"/>
>> ???<param name="ext-sip-ip" value="auto-nat"/>
>>
>> FreeSWITCH Version 1.0.trunk (15490)
>>
>> However, isn't it the IP address that is reported by the remote
>> SPA3102 that is incorrect? ?Or?
>>
>> On Thu, Dec 31, 2009 at 7:21 AM, Brian West <brian at freeswitch.org> wrote:
>>
>> show me the ext-rtp-ip and ext-sip-ip settings you're using along with SVN
>> rev please.
>>
>> /b
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From Prometheus001 at gmx.net  Tue Jan  5 04:40:29 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 05 Jan 2010 13:40:29 +0100
Subject: [Freeswitch-users] Force endpoint to use rfc2833 for dtmf
In-Reply-To: <FD8E5927-9DE2-41D0-B072-5051AF7C1907@jerris.com>
References: <4B30B01B.30809@gmx.net>
	<FD8E5927-9DE2-41D0-B072-5051AF7C1907@jerris.com>
Message-ID: <4B43333D.8020801@gmx.net>

Hello Michael,

I have opened a Jira for this.

Best rgerads
Peter


Michael Jerris schrieb:
> Not sure if we have an option to disable info.  Even without this,  
> dtmf should go across the bridge fine.  Please open up a bug on jira  
> about this
>
> Mike
>
> On Dec 22, 2009, at 6:40 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>
>   
>> Hello,
>>
>> in a bigger installation with some thousand endpoints in the field we
>> see, that the endpoint equipment is always using INFO messages  
>> (standard
>> setting is auto, so the endpoint decides which method to use). I  
>> have 2
>> questions to that scenario:
>>
>>   1. Is there a way that Freeswitch forces/restricts the endpoint to
>>      use rfc2833 or not to send to allow INFO in the invite message?
>>   2. Currently INFO messages do not get forwarded from the caller
>>      through freeswitch to called endpoint. How can we enable that FS
>>      is fowarding the INFO messages?
>>
>> Best regards
>> Peter
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From aep.lists at it46.se  Tue Jan  5 06:18:47 2010
From: aep.lists at it46.se (Alberto Escudero)
Date: Tue, 5 Jan 2010 15:18:47 +0100
Subject: [Freeswitch-users] encrypt voicemail emails and attachments?
In-Reply-To: <20100105042327.7CBF412DD@sinclaire.sibble.net>
References: <20100105042327.7CBF412DD@sinclaire.sibble.net>
Message-ID: <2f9531f86fc90f8d3f4cafab0cba4eae.squirrel@correo.nodo50.org>

One way to do it, it is to use procmail to handle the mails locally before
forwarding them to a final destination. You can take a similar approach
that Spam/Antivirus software and use procmailrc to add one more filter *in
your case gpg*.

/aep
-- 
Stopping junk mailers is good for the environment

> Looking throught the wiki, I see various configs for having FS email you a
> copy of received voicemail messages, has anyone done any work with having
> the
> voicemail messages gpg encrypted with public prior to sending?  Or is that
> something that should pretty much be handled at the mta level leaving FS
> out
> of the mix altogether? I'm thinking probably so, but before I try to do
> this,
> I figured i'd ask first.
>
>
> --
> Harondel J. Sibble
> Sibble Computer Consulting
> Creating Solutions for the small and medium business computer user.
> help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
> (604) 739-3709 (voice)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>




From brian at freeswitch.org  Tue Jan  5 07:15:15 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Jan 2010 09:15:15 -0600
Subject: [Freeswitch-users] No audio after Remote SDP:
In-Reply-To: <33c87fa31001050307v3cbcae01q9d7d47383a63ad33@mail.gmail.com>
References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com>
	<33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com>
	<772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org>
	<33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com>
	<191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com>
	<33c87fa30912301213g3d24b773k86b95e7ce08f6761@mail.gmail.com>
	<7D75E6D6-8AB9-429E-A5A9-1639C0D5AD09@freeswitch.org>
	<33c87fa31001012330g69fc8e88m61e648d893c4d8f7@mail.gmail.com>
	<180A53B1-EFB6-4B17-AA7C-19B007ADE04C@freeswitch.org>
	<33c87fa31001021519j41ead649rb4a85e6be6236254@mail.gmail.com>
	<33c87fa31001050307v3cbcae01q9d7d47383a63ad33@mail.gmail.com>
Message-ID: <117A38AB-C1AE-4AF0-AD36-4165FAA94816@freeswitch.org>

This is why you set up stun correctly on the SPA.

/b

On Jan 5, 2010, at 5:07 AM, Mark Campbell-Smith wrote:

> The difference was a hard coded external IP address in the first
> session that worked.  However I can't have it set like this as the IP
> address is not static.



From david.varnes at gmail.com  Tue Jan  5 05:41:42 2010
From: david.varnes at gmail.com (david varnes)
Date: Wed, 6 Jan 2010 00:41:42 +1100
Subject: [Freeswitch-users] [contrib] a new java ESL inbound/outbound client
Message-ID: <74a861001001050541v4e02f487xd21fe8fc13e8ed0a@mail.gmail.com>

Hi all,
I had a basic java inbound ESL client kicking around that I have used
in a couple of small projects over the last year.  I needed something
a little more complete for a new project so I dusted it off and made it
less incomplete.

It still needs more work and testing, but it certainly is usable right now.
I have tested it against FS 1.0.4 and latest trunk.

I have put it in my contrib area in svn in hopes that some may find it
useful:
http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/dvarnes/java/esl-client
I would be interested in any feedback ...

Features
 * Apache License (ASL) version 2
 * Standalone Inbound client
 * Framework classes to easily create an Outbound socket client
 * based on Netty [1] nio library version 3.1.5.GA (previously was
   using Apache MINA, but this is easier)
 * logging via slf4j
 * only dependencies are slf4j-api and netty (both Apache licensed)
 * single jar which is a valid OSGi bundle
 * built using maven
 * eclipse projects
 * reasonable level of java docs

Still todo
 * Docs
 * Simple example apps
 * .. more in TODO.txt in project root.

There is no binary jar available right now since I don't know how/if
I can put files up to file.freeswitch.org.

In the meantime to build you need maven [2] installed.  If you are
unfamiliar with maven usage, I can post a simple howto.

davidv

[1] http://www.jboss.org/netty/downloads.html
[2] http://maven.apache.org


From brian at freeswitch.org  Tue Jan  5 07:53:52 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Jan 2010 09:53:52 -0600
Subject: [Freeswitch-users] Unable to start mod_java
In-Reply-To: <c55d46951001041934n49bfcbf4h42564e4548aa301a@mail.gmail.com>
References: <c55d46951001041934n49bfcbf4h42564e4548aa301a@mail.gmail.com>
Message-ID: <BC685614-4CE4-46AB-9263-DB5F7FC815A3@freeswitch.org>

Are you on a 64bit platform?  If so then you're jdk is wrong.

/b

On Jan 4, 2010, at 9:34 PM, Max Bridgewater wrote:

> Hi,
> 
> I built Freeswitch with mod_java enabled. But now, when Freeswitch starts, I get the following error message:
> 
> 
> 2010-01-04 22:32:46.574770 [NOTICE] switch_loadable_module.c:271 Adding API Function 'jsrun'
> 2010-01-04 22:32:46.574811 [NOTICE] switch_loadable_module.c:271 Adding API Function 'jsapi'
> 2010-01-04 22:32:46.575306 [NOTICE] modjava.c:244 Java Framework Loading...
> 2010-01-04 22:32:46.575721 [ERR] modjava.c:133 Error loading /usr/local/java/jdk1.6.0_17/jre/lib/i386/client/libjvm.so
> 2010-01-04 22:32:46.575743 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_java.so
> **Module load routine returned an error**
> 2010-01-04 22:32:46.576742 [CONSOLE] switch_loadable_module.c:890 Successfully Loaded [mod_lua]
> 2010-01-04 22:32:46.576748 [NOTICE] switch_loadable_module.c:209 Adding Dialplan 'LUA'
> 2010-01-04 22:32:46.576795 [NOTICE] switch_loadable_module.c:249 Adding Application 'lua'
> 
> 
> As far as I can tell the /usr/local/java/jdk1.6.0_17/jre/lib/i386/client/libjvm.so is the correct path.
> Any idea?
> 
> max.



From anthony.minessale at gmail.com  Tue Jan  5 08:11:05 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Jan 2010 10:11:05 -0600
Subject: [Freeswitch-users] Zap channel not released when voicemail
	starts
In-Reply-To: <cd93b3701001050144m648905e0i76a0f59fa3b01db1@mail.gmail.com>
References: <ea085d8b43f8105108d9c1063fb56534@thom.fr.eu.org>
	<191c3a031001041600y1c98465cifcc522056096f746@mail.gmail.com>
	<cd93b3701001050144m648905e0i76a0f59fa3b01db1@mail.gmail.com>
Message-ID: <191c3a031001050811w5c33d5f6na52bcedc039ab188@mail.gmail.com>

one way is to run tone_detect on the busy signal and map it to the hangup
app


On Tue, Jan 5, 2010 at 3:44 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:

> I'll second that.
>
> My way of dealing with it has been to write a little script to detect
> hangups on the TDM end, then force release the corresponding "B-leg" that is
> hooked up to VM. In the process of converting this to an FS module.
>
> Not clean .. but works. Would have liked to see the same code within FS
> core and, if appropriate, the VM subsystem to achieve the same end.
>
> Regards,
>
> Ahmed.
>
>
> 2010/1/5 Anthony Minessale <anthony.minessale at gmail.com>
>
> hangup detection on TDM is a bitch.
>>
>>
>> On Mon, Jan 4, 2010 at 10:07 AM, Fran?ois Legal <devel at thom.fr.eu.org>wrote:
>>
>>> Hello,
>>>
>>>
>>>
>>> I have an issue with voicemail and openzap channels.
>>>
>>> When an incoming call on an openzap channel is bridged to voicemail, if
>>> that channel is hung up before the beginning of voicemail recording, that
>>> channel is kept open open until 3 or 4 seconds after the voicemail started
>>> to record the message.
>>>
>>> What should I do to make freeswitch/voicemail release the channel
>>> immediately when the caller hang up ?
>>>
>>>
>>>
>>> Thanks in advance
>>>
>>>
>>>
>>> Fran?ois
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From devel at thom.fr.eu.org  Tue Jan  5 08:11:27 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Tue, 05 Jan 2010 17:11:27 +0100
Subject: [Freeswitch-users] How to konw who picks up in group bridge
Message-ID: <06bbe9d9d06077a04e9245c84d4cb013@thom.fr.eu.org>



Hello, 

In my diaplan, when a call arrives on some specific channel, it
is routed to an extension that tries to bridge it on multiple channels
using coma separated list (either openzap, sofia or both). 

I would like
to see in my CDR which channel did pick up which channel did pick up the
call. Which variable can I use ? 

Thanks 

Fran?ois
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From help at pdscc.com  Tue Jan  5 09:11:37 2010
From: help at pdscc.com (Harondel J. Sibble)
Date: Tue, 05 Jan 2010 09:11:37 -0800
Subject: [Freeswitch-users] encrypt voicemail emails and attachments?
In-Reply-To: <03DDC308-7EC5-46EC-8AFB-CC73417E20F5@jerris.com>
References: <20100105042327.7CBF412DD@sinclaire.sibble.net>,
	<03DDC308-7EC5-46EC-8AFB-CC73417E20F5@jerris.com>
Message-ID: <20100105171137.A433E1DB501@sinclaire.sibble.net>

That's what I suspected, thanks!

On 4 Jan 2010 at 23:51, Michael Jerris wrote:

> you can just make a shell script (or perl or whatever) that is called as the
> mailer that does this.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From ken at ukgb.net  Tue Jan  5 10:06:52 2010
From: ken at ukgb.net (Ken Gillett)
Date: Tue, 5 Jan 2010 18:06:52 +0000
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <5BED3371-0023-4D53-BBE1-37D241B4AAFB@jerris.com>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com>
	<B6763940-A3CB-4006-8DF1-ADEDD9C68294@ukgb.net>
	<0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org>
	<CA53F057-4A51-457C-A673-B0B6E4DF482A@ukgb.net>
	<207e7a5e0912301103h17133fdfk22430bbce5a1718b@mail.gmail.com>
	<090E289D-76EE-4F4A-82DF-513626A915E6@ukgb.net>
	<79B790D4-DE06-47FF-B9FD-52D9E1EB8BCB@ukgb.net>
	<207e7a5e1001020840y1bd3bae2y2b683a19659c9d7d@mail.gmail.com>
	<6E6E6D36-6E93-49BF-9516-FC028E73D279@ukgb.net>
	<5BED3371-0023-4D53-BBE1-37D241B4AAFB@jerris.com>
Message-ID: <A301DB14-66F4-4154-B9F1-17704B51BD75@ukgb.net>

Thanks for that. Now I know how to proceed.


On 4 Jan 2010, at 16:01, Michael Jerris wrote:

> its all in there.
> 
> Mike
> 
> On Jan 4, 2010, at 4:14 AM, Ken Gillett wrote:
> 
>> I will look into this, but in the meantime, would it not be possible to simply copy the installed files? I realise that basically this must be possible, but practically it depends on what is installed where. I believe the default prefix is /usr/local/freeswitch, but is EVERYTHING in there or is other stuff scattered in other directories?
>> 
>> If it's all together I'm thinking I could simply tar the folder and copy the tarball. Wouldn't this be possible?
>> 
> 



Ken G i l l e t t

_/_/_/_/_/_/_/_/





From jerry.richards at teotech.com  Tue Jan  5 10:24:18 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 5 Jan 2010 10:24:18 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <cd93b3701001050140q14182667o8e90dba27fa3bc85@mail.gmail.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<cd93b3701001050140q14182667o8e90dba27fa3bc85@mail.gmail.com>
Message-ID: <8FBE2BC8AF8C486B8E569C08D4941064@greyhawk.tonecommander.com>

Hi Ahmed,
 
My vars.xml file does not set the literal IP address (nor the server's DNS
name), rather it uses the following line (so this should not cause a
problem):
 
<X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
 
Also, there are no references to hard-coded addresses anywhere in the other
XML files.  Also, line #3 of the pastebin "IP 192.168.72.29 Rejected by acl
"domains" Falling back to Digest auth." just means it is authenticating the
INVITE and then it does proceed with the call.
 
The problem is down at line #71: "zap_io.c:1197 outgoing_call method not
implemented!".  What does this error mean?
 
Thank you and Best Regards,
Jerry
 

  _____  

From: Ahmed Naji [mailto:a.alalousi at gmail.com] 
Sent: Tuesday, January 05, 2010 1:40 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER


Hi Jerry,

Looked at your log and your problem is quiet simple to resolve, but first
here's what's happenning:

You copied the conf/ subtree to your new server. As such, you have also
duplicated your vars.xml. By doing so, you have set the domain on the new
server to the same domain used on your old server which would be fine, but
the default domain settings used by FS is to use your primary IPv4 IP
address as your domain.

By duplicating the conf subtree from the old server, you have effectively
bound the new instance of FS to a domain that is the IP address of the old
server, if this makes sense.

You can see this on third line of your log:

192.168.72.29 Rejected by acl "domains"

To resolve this, modify your vars.xml on the new server to reflect whatever
domain it is you want to route, or set the domain to the new server's IP
address like so:

 <X-PRE-PROCESS cmd="set" data="domain=IP_ADDRESS_OF_NEW_SERVER"/>

You also need to check that any other files (e.g. the conf/sip_profiles,
conf/directory/ and conf/dialplan/ hierarchy) are modified to reflect the
new server settings as well.

In the limit, resolving those conflicts will also resolve your issues,
unless there is something else that's wrong.

Let's know how you get along.

Regards,

Ahmed.
 


2010/1/4 Jerry Richards <jerry.richards at teotech.com>



Hello,

I have one FS instance that is working well with a PRI and running FS
version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
upgraded it.

Now, I am trying to bring up another FS instance (basically a clone of the
first), but the PRI does not work.  When I attempt to make an
internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
both servers are using the latest Sangoma Wanpipe driver, and I copied the
conf XML file tree from the old server to the new one.  I think the problem
has to do with the openzap module, but I'm having difficulty isolating the
problem.  Could it have built the openzap module incorrectly?  Another
difference is that I installed 1.0.5pre9 from scratch on the new server
(i.e. it never had 1.0.4 running on it).

I put the FS log into the pastebin when an outbound call attempt is made:

http://pastebin.freeswitch.org/11675

Could someone give me a pointer on what to try next?

Thanks and Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
Ahmed Naji


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From tculjaga at gmail.com  Tue Jan  5 11:25:13 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 5 Jan 2010 20:25:13 +0100
Subject: [Freeswitch-users] freeswitch and H323
In-Reply-To: <c96b95651001031636q1ecfec18x2bba1c22ff6ca9bb@mail.gmail.com>
References: <7aa8bd9d0912300514r68c90b12u7c631a649981cfa3@mail.gmail.com>
	<87f2f3b90912310920q1821fe0eu5fd74e553e4cb12@mail.gmail.com>
	<65d96fc81001010452k37068e87sc0a229cdfe481c40@mail.gmail.com>
	<c96b95651001031636q1ecfec18x2bba1c22ff6ca9bb@mail.gmail.com>
Message-ID: <65d96fc81001051125u3a6c7d96q5ed39d39c0026107@mail.gmail.com>

its already there: http://wiki.freeswitch.org/wiki/Mod_h323

T.

On Mon, Jan 4, 2010 at 1:36 AM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:

> HI,
>
> It would be really nice if you can create a wiki page.
>
> Thanks
>
>
> On Fri, Jan 1, 2010 at 1:52 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> well, mod_h323 works for me... there are still some missing things and of
>> course bugs ... e.g. incorrect releaseCause mapping, no automatic codec
>> ptime sync... but it is usable ....
>>
>>
>> if you'd like to go mod_h323 way i can help you... it builds as a charm
>> for me...
>>
>>
>> T.
>>
>>
>>
>>
>>
>> On Thu, Dec 31, 2009 at 6:20 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Are you trying to use mod_h323 or mod_opal? They are both works in
>>> progress, but the latter is farther along than the former. Use the latest
>>> FreeSWITCH trunk (or latest.freeswitch.org) and run the buildopal.sh
>>> script in the build directory. If you have any build issues then paste the
>>> log on pastebin.freeswitch.org and reply to this thread with the PB URL
>>> so that we can take a look.
>>>  -MC
>>>
>>>
>>> On Wed, Dec 30, 2009 at 5:14 AM, Pete Kay <petedao at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> has anyone been able to get H323 to work?
>>>>
>>>> I have problem trying to get it compiled with either 1.0.4 or 1.0.5.
>>>>
>>>> Thanks,
>>>> pete
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jan  5 13:02:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jan 2010 13:02:37 -0800
Subject: [Freeswitch-users] XML IVR and Javascript menu-exec-app
In-Reply-To: <7d4c9553f55e0af32200eebdd332b2f9.squirrel@correo.nodo50.org>
References: <7d4c9553f55e0af32200eebdd332b2f9.squirrel@correo.nodo50.org>
Message-ID: <87f2f3b91001051302n70d4462nc667e2c14fded790@mail.gmail.com>

On Fri, Jan 1, 2010 at 10:03 AM, Alberto Escudero <aep.lists at it46.se> wrote:

> Hi,
>
> I am writing several IVRs using Freeswitch XML
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr
>
> One of the nodes of the IVR is a Javascript application that records a
> message.
> e.g.:
> <entry action="menu-exec-app" digits="3" param="javascript foo.js"/>
>
> The Javascript application starts by issuing a
> session.answer()
>
> [records the voice message]
>
> exit();
>
> Once the Javascript exits, the channel is dropped and hence the IVR
> terminates.
> Is it possible to write a Javascript application that once is completed,
> the channel returns back to the top menu of the IVR? I want to emulate the
> same behavior that "menu-play-sound", that once the file is played, the
> IVR logic returns to the top menu.
>
> Just transfer the call to an extension in the dialplan that in turn sends
the call to the IVR in question...
-MC
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From msc at freeswitch.org  Tue Jan  5 13:11:01 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jan 2010 13:11:01 -0800
Subject: [Freeswitch-users] Polycom & TLS - help please...
In-Reply-To: <d361ca7b1001050038x4e7103bdke322fd8741403172@mail.gmail.com>
References: <d361ca7b1001030757m2f55a400n3e2b3b74185c4202@mail.gmail.com>
	<BFB88CA0-8319-4D05-B3CD-39E3450DE157@jerris.com>
	<D950E561-1B94-48E0-9221-3739BAB8F91A@freeswitch.org>
	<d361ca7b1001031932m7695150fv967c0df683083af9@mail.gmail.com>
	<3968BF39-7A15-4F9F-8FA2-A78BC4F0F38E@freeswitch.org>
	<d361ca7b1001050038x4e7103bdke322fd8741403172@mail.gmail.com>
Message-ID: <87f2f3b91001051311o3f348035x86f6383e84680291@mail.gmail.com>

On Tue, Jan 5, 2010 at 12:38 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Thanks, I have a partial success which involved two steps:
>
>    - The wiki says to create a root certifcate with *gentls_cert setup *with
>    no other parameters; I had to add my domain's data to this command. The new
>    certificate has been downloaded to the phone.
>    - Replaced the registrar definitions in the phone's config files from
>    IP address to the server's name.
>    - The above setup worked as-is. To be sure I've added the NAPTR records
>    to the DNS after the above two steps worked.
>       - BTW, the wiki says that the NAPTR records are not mandatory, thus
>       I did not add them at the first place.
>
> Did you add your specific information to the wiki? If not please do so.
Thanks!
-MC
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From anthony.minessale at gmail.com  Tue Jan  5 13:18:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Jan 2010 15:18:00 -0600
Subject: [Freeswitch-users] XML IVR and Javascript menu-exec-app
In-Reply-To: <87f2f3b91001051302n70d4462nc667e2c14fded790@mail.gmail.com>
References: <7d4c9553f55e0af32200eebdd332b2f9.squirrel@correo.nodo50.org>
	<87f2f3b91001051302n70d4462nc667e2c14fded790@mail.gmail.com>
Message-ID: <191c3a031001051318q79737971l39c6acbbc3bab818@mail.gmail.com>

or call:
 session.setAutoHangup(0);

so exiting the script will not hangup the channel

if it still happens get the log line "its blue" that shows where the call is
being hungup from"



On Tue, Jan 5, 2010 at 3:02 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Jan 1, 2010 at 10:03 AM, Alberto Escudero <aep.lists at it46.se>wrote:
>
>> Hi,
>>
>> I am writing several IVRs using Freeswitch XML
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr
>>
>> One of the nodes of the IVR is a Javascript application that records a
>> message.
>> e.g.:
>> <entry action="menu-exec-app" digits="3" param="javascript foo.js"/>
>>
>> The Javascript application starts by issuing a
>> session.answer()
>>
>> [records the voice message]
>>
>> exit();
>>
>> Once the Javascript exits, the channel is dropped and hence the IVR
>> terminates.
>> Is it possible to write a Javascript application that once is completed,
>> the channel returns back to the top menu of the IVR? I want to emulate the
>> same behavior that "menu-play-sound", that once the file is played, the
>> IVR logic returns to the top menu.
>>
>> Just transfer the call to an extension in the dialplan that in turn sends
> the call to the IVR in question...
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From a.alalousi at gmail.com  Tue Jan  5 13:58:41 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Tue, 5 Jan 2010 21:58:41 +0000
Subject: [Freeswitch-users] Zap channel not released when voicemail
	starts
In-Reply-To: <191c3a031001050811w5c33d5f6na52bcedc039ab188@mail.gmail.com>
References: <ea085d8b43f8105108d9c1063fb56534@thom.fr.eu.org>
	<191c3a031001041600y1c98465cifcc522056096f746@mail.gmail.com>
	<cd93b3701001050144m648905e0i76a0f59fa3b01db1@mail.gmail.com>
	<191c3a031001050811w5c33d5f6na52bcedc039ab188@mail.gmail.com>
Message-ID: <cd93b3701001051358h51931809o56a2ce3cfb8f7d2e@mail.gmail.com>

Good method, though this isn't this assuming too much, in the sense that we
are assuming a hangup with cause 17 or cause 16 with a forced busy tone ?

2010/1/5 Anthony Minessale <anthony.minessale at gmail.com>

> one way is to run tone_detect on the busy signal and map it to the hangup
> app
>
>
>
> On Tue, Jan 5, 2010 at 3:44 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>
>> I'll second that.
>>
>> My way of dealing with it has been to write a little script to detect
>> hangups on the TDM end, then force release the corresponding "B-leg" that is
>> hooked up to VM. In the process of converting this to an FS module.
>>
>> Not clean .. but works. Would have liked to see the same code within FS
>> core and, if appropriate, the VM subsystem to achieve the same end.
>>
>> Regards,
>>
>> Ahmed.
>>
>>
>> 2010/1/5 Anthony Minessale <anthony.minessale at gmail.com>
>>
>> hangup detection on TDM is a bitch.
>>>
>>>
>>> On Mon, Jan 4, 2010 at 10:07 AM, Fran?ois Legal <devel at thom.fr.eu.org>wrote:
>>>
>>>> Hello,
>>>>
>>>>
>>>>
>>>> I have an issue with voicemail and openzap channels.
>>>>
>>>> When an incoming call on an openzap channel is bridged to voicemail, if
>>>> that channel is hung up before the beginning of voicemail recording, that
>>>> channel is kept open open until 3 or 4 seconds after the voicemail started
>>>> to record the message.
>>>>
>>>> What should I do to make freeswitch/voicemail release the channel
>>>> immediately when the caller hang up ?
>>>>
>>>>
>>>>
>>>> Thanks in advance
>>>>
>>>>
>>>>
>>>> Fran?ois
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Ahmed Naji
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From anthony.minessale at gmail.com  Tue Jan  5 14:06:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Jan 2010 16:06:03 -0600
Subject: [Freeswitch-users] Zap channel not released when voicemail
	starts
In-Reply-To: <cd93b3701001051358h51931809o56a2ce3cfb8f7d2e@mail.gmail.com>
References: <ea085d8b43f8105108d9c1063fb56534@thom.fr.eu.org>
	<191c3a031001041600y1c98465cifcc522056096f746@mail.gmail.com>
	<cd93b3701001050144m648905e0i76a0f59fa3b01db1@mail.gmail.com>
	<191c3a031001050811w5c33d5f6na52bcedc039ab188@mail.gmail.com>
	<cd93b3701001051358h51931809o56a2ce3cfb8f7d2e@mail.gmail.com>
Message-ID: <191c3a031001051406o58ff93adk2654b475de971743@mail.gmail.com>

depends,
some telco actually expect you to react to this tone for hangup and charge
you for real hangup signaling which is probably the case here.


On Tue, Jan 5, 2010 at 3:58 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:

> Good method, though this isn't this assuming too much, in the sense that we
> are assuming a hangup with cause 17 or cause 16 with a forced busy tone ?
>
>
> 2010/1/5 Anthony Minessale <anthony.minessale at gmail.com>
>
>> one way is to run tone_detect on the busy signal and map it to the hangup
>> app
>>
>>
>>
>> On Tue, Jan 5, 2010 at 3:44 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>>
>>> I'll second that.
>>>
>>> My way of dealing with it has been to write a little script to detect
>>> hangups on the TDM end, then force release the corresponding "B-leg" that is
>>> hooked up to VM. In the process of converting this to an FS module.
>>>
>>> Not clean .. but works. Would have liked to see the same code within FS
>>> core and, if appropriate, the VM subsystem to achieve the same end.
>>>
>>> Regards,
>>>
>>> Ahmed.
>>>
>>>
>>> 2010/1/5 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>> hangup detection on TDM is a bitch.
>>>>
>>>>
>>>> On Mon, Jan 4, 2010 at 10:07 AM, Fran?ois Legal <devel at thom.fr.eu.org>wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>>
>>>>>
>>>>> I have an issue with voicemail and openzap channels.
>>>>>
>>>>> When an incoming call on an openzap channel is bridged to voicemail, if
>>>>> that channel is hung up before the beginning of voicemail recording, that
>>>>> channel is kept open open until 3 or 4 seconds after the voicemail started
>>>>> to record the message.
>>>>>
>>>>> What should I do to make freeswitch/voicemail release the channel
>>>>> immediately when the caller hang up ?
>>>>>
>>>>>
>>>>>
>>>>> Thanks in advance
>>>>>
>>>>>
>>>>>
>>>>> Fran?ois
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Ahmed Naji
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From djbinter at yahoo.com  Tue Jan  5 14:42:58 2010
From: djbinter at yahoo.com (DJB)
Date: Tue, 5 Jan 2010 14:42:58 -0800 (PST)
Subject: [Freeswitch-users] Min-SE Header
Message-ID: <77858.66861.qm@web37507.mail.mud.yahoo.com>

I am wondering whether it is possible to suppress the Min-SE Header Field in SIP INVITE message.

Thank you,
Dorn B.


      
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From nik.middleton at noblesolutions.co.uk  Tue Jan  5 15:02:15 2010
From: nik.middleton at noblesolutions.co.uk (Nik Middleton)
Date: Tue, 5 Jan 2010 23:02:15 -0000
Subject: [Freeswitch-users] Call limits (time)
Message-ID: <A2B4F48489739E439D8845296A2FD9C4147E1F@sun.Noble.local>

Hi Guys,

 

I'm looking to migrate my billing platform to use FS.  So far so good,
however, if a user is low on credit I need to limit the call length.  In
other words, from the rate card, prior to connecting the call, I know
they are going to call a mobile at say $0.10/min, but they only have $2
of credit, so I want to terminate the call after 20 mins, preferably
with a message to the originator saying they only have X mins left.  Is
there a way of achieving this with the originate command?

 

Regards

 

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From sos at sokhapkin.dyndns.org  Tue Jan  5 15:14:13 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 5 Jan 2010 18:14:13 -0500
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C4147E1F@sun.Noble.local>
References: <A2B4F48489739E439D8845296A2FD9C4147E1F@sun.Noble.local>
Message-ID: <201001051814.13887.sos@sokhapkin.dyndns.org>

See sched_hangup dialplan application.

On Tuesday 05 January 2010, Nik Middleton wrote:
> Hi Guys,
>
>
>
> I'm looking to migrate my billing platform to use FS.  So far so good,
> however, if a user is low on credit I need to limit the call length.  In
> other words, from the rate card, prior to connecting the call, I know
> they are going to call a mobile at say $0.10/min, but they only have $2
> of credit, so I want to terminate the call after 20 mins, preferably
> with a message to the originator saying they only have X mins left.  Is
> there a way of achieving this with the originate command?
>
>
>
> Regards




From ron.freeswitch at mcleodnet.com  Tue Jan  5 15:25:26 2010
From: ron.freeswitch at mcleodnet.com (Ron McLeod)
Date: Tue, 5 Jan 2010 15:25:26 -0800
Subject: [Freeswitch-users] unable to call out troug siemens hie9200 in pur
	SIP
Message-ID: <B1789BC477C54AF9AD96A33F051FBAB6@fromage>

Posted on behalf of Tayeb Meftah...


hi dear friends,
we have a siemens hie9200 softswitch
we want to interconnect freeswitch with it
to use it for service, like media, voicemail, audio conferencing and ...
call trace is atached belo
if we calls from tdm, call is passed but without rtp
i routed a number like 021000000 is a tdm number to 3001 the default
8khz fs conference
so inbound call is passed but without audio
for outbound calls, i wanted to call a tdm number, like 021298235
the call tack some long time but return a 500 internal server error
from the hie9200 softswitch:
sip.Reason == "Q.850 ;cause=47 ;text=\"Resource unavailable, unspecified\""

from fs i get normal temporary failur
so please see the trace and return for me any reply ;)
thanks for your helps and times.


Traces at:
http://siplabs.net/tracebin/fs-siemens-500.pcap
http://siplabs.net/tracebin/fs-siemens-rtp.pcap




From nik.middleton at noblesolutions.co.uk  Tue Jan  5 15:43:26 2010
From: nik.middleton at noblesolutions.co.uk (Nik Middleton)
Date: Tue, 5 Jan 2010 23:43:26 -0000
Subject: [Freeswitch-users] Migrating from asterisk to FS
Message-ID: <A2B4F48489739E439D8845296A2FD9C4147E22@sun.Noble.local>

HI Guys

 

While I've been using FS for around 18 months now, and love it to bits,
it's been a specific solution.  I'm now looking to move my customer base
across, and have on the base of it some basic and perhaps dumb
questions.

 

I currently have around 150 Sip phones attached to my systems   These
are all geographically spread, so re-configuring them is out of the
question.  They all register on port 5060.  Given that FS uses port 5080
for external clients, do I simply need to do a Port translate on my
firewall or is there a simpler solution?

 

Further, how does FS handle a call FWD?  In other words, if a SIP phone
has a divert on busy set will it account for the redirect?   Currently
in Asterisk I use the 'I' option to disable this as I can't account for
the call.

 

Regards,

 

 

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From msc at freeswitch.org  Tue Jan  5 16:33:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jan 2010 16:33:26 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>

On Mon, Jan 4, 2010 at 3:49 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> Hello,
>
> I have one FS instance that is working well with a PRI and running FS
> version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
> upgraded it.
>
> Now, I am trying to bring up another FS instance (basically a clone of the
> first), but the PRI does not work.  When I attempt to make an
> internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
> both servers are using the latest Sangoma Wanpipe driver, and I copied the
> conf XML file tree from the old server to the new one.  I think the problem
> has to do with the openzap module, but I'm having difficulty isolating the
> problem.  Could it have built the openzap module incorrectly?  Another
> difference is that I installed 1.0.5pre9 from scratch on the new server
> (i.e. it never had 1.0.4 running on it).
>
> I put the FS log into the pastebin when an outbound call attempt is made:
>
> http://pastebin.freeswitch.org/11675
>
> Could someone give me a pointer on what to try next?
>

Jerry, I noticed this line:
(OpenZAP/1:1/3491028 at g1)

Is your carrier wanting full ten digit phone numbers? Try adding the area
code on this and see what happens. The error usually would be something like
"invalid number format" but I've seen carriers do stupid things like this.
Try that first and see if it makes a difference. If not you'll need to turn
on Q931 debugging as per the Sangoma wiki. (See
http://wiki.freeswitch.org/wiki/OpenZAP#Debugging_PRI_With_wanpipemon for
the link.)

-MC
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From msc at freeswitch.org  Tue Jan  5 16:37:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jan 2010 16:37:14 -0800
Subject: [Freeswitch-users] Zap channel not released when voicemail
	starts
In-Reply-To: <191c3a031001051406o58ff93adk2654b475de971743@mail.gmail.com>
References: <ea085d8b43f8105108d9c1063fb56534@thom.fr.eu.org>
	<191c3a031001041600y1c98465cifcc522056096f746@mail.gmail.com>
	<cd93b3701001050144m648905e0i76a0f59fa3b01db1@mail.gmail.com>
	<191c3a031001050811w5c33d5f6na52bcedc039ab188@mail.gmail.com>
	<cd93b3701001051358h51931809o56a2ce3cfb8f7d2e@mail.gmail.com>
	<191c3a031001051406o58ff93adk2654b475de971743@mail.gmail.com>
Message-ID: <87f2f3b91001051637p43ad3350l7bbf3074212be845@mail.gmail.com>

On Tue, Jan 5, 2010 at 2:06 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> depends,
> some telco actually expect you to react to this tone for hangup and charge
> you for real hangup signaling which is probably the case here.


Example: CenturyLink charges $5 per line per month for "disconnect after
hangup." No joke. It's an actual "feature" that they "sell" to the people
using their analog lines. I'm sure other scumbag telcos do the same thing.

-MC
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From msc at freeswitch.org  Tue Jan  5 16:46:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jan 2010 16:46:07 -0800
Subject: [Freeswitch-users] Migrating from asterisk to FS
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C4147E22@sun.Noble.local>
References: <A2B4F48489739E439D8845296A2FD9C4147E22@sun.Noble.local>
Message-ID: <87f2f3b91001051646h481e598fp3cfe2a5c605c5de0@mail.gmail.com>

On Tue, Jan 5, 2010 at 3:43 PM, Nik Middleton <
nik.middleton at noblesolutions.co.uk> wrote:

>  HI Guys
>
>
>
> While I?ve been using FS for around 18 months now, and love it to bits,
> it?s been a specific solution.  I?m now looking to move my customer base
> across, and have on the base of it some basic and perhaps dumb questions.
>
>
>
> I currently have around 150 Sip phones attached to my systems   These are
> all geographically spread, so re-configuring them is out of the question.
> They all register on port 5060.  Given that FS uses port 5080 for external
> clients, do I simply need to do a Port translate on my firewall or is there
> a simpler solution?
>
>
>
FreeSWITCH doesn't *force* you to use port 5080 for inbound registrations.
You can, but it's not a requirement. Personally I just use the internal SIP
profile for those external phones wherever possible. The real issue is
whether or not you have a horrible NAT device in between FS and the Internet
connection. In my experience, if you have a decent NAT device that supports
UPnP (like the WRT54GL running Tomato firmware) then connecting external
phones to a FS box behind NAT using port 5060 just works. Give it a try and
let us know how it works.

> Further, how does FS handle a call FWD?  In other words, if a SIP phone has
> a divert on busy set will it account for the redirect?   Currently in
> Asterisk I use the ?I? option to disable this as I can?t account for the
> call.
>
When calling to a FS box or when FS calls another server?
-MC
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From rupa at rupa.com  Tue Jan  5 18:13:25 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 5 Jan 2010 20:13:25 -0600
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C4147E1F@sun.Noble.local>
References: <AcqOWx9O9EDYsJ/ZQJSSOHSvWxhecA==>
	<A2B4F48489739E439D8845296A2FD9C4147E1F@sun.Noble.local>
Message-ID: <b39273531001051813v5271858fpd133326559238b5e@mail.gmail.com>

Look at using mod_nibblebill

On Tue, Jan 5, 2010 at 5:02 PM, Nik Middleton <
nik.middleton at noblesolutions.co.uk> wrote:

>  Hi Guys,
>
>
>
> I?m looking to migrate my billing platform to use FS.  So far so good,
> however, if a user is low on credit I need to limit the call length.  In
> other words, from the rate card, prior to connecting the call, I know they
> are going to call a mobile at say $0.10/min, but they only have $2 of
> credit, so I want to terminate the call after 20 mins, preferably with a
> message to the originator saying they only have X mins left.  Is there a way
> of achieving this with the originate command?
>
>
>
> Regards
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From sos at sokhapkin.dyndns.org  Tue Jan  5 18:25:06 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 5 Jan 2010 21:25:06 -0500
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <b39273531001051813v5271858fpd133326559238b5e@mail.gmail.com>
References: <AcqOWx9O9EDYsJ/ZQJSSOHSvWxhecA==>
	<A2B4F48489739E439D8845296A2FD9C4147E1F@sun.Noble.local>
	<b39273531001051813v5271858fpd133326559238b5e@mail.gmail.com>
Message-ID: <201001052125.06909.sos@sokhapkin.dyndns.org>

Unfortunalely, mod_nibblebill doesn't take billing increments into account.

On Tuesday 05 January 2010, Rupa Schomaker wrote:
> Look at using mod_nibblebill
>
> On Tue, Jan 5, 2010 at 5:02 PM, Nik Middleton <
>
> nik.middleton at noblesolutions.co.uk> wrote:
> >  Hi Guys,
> >
> >
> >
> > I?m looking to migrate my billing platform to use FS.  So far so good,
> > however, if a user is low on credit I need to limit the call length.  In
> > other words, from the rate card, prior to connecting the call, I know
> > they are going to call a mobile at say $0.10/min, but they only have $2
> > of credit, so I want to terminate the call after 20 mins, preferably with
> > a message to the originator saying they only have X mins left.  Is there
> > a way of achieving this with the originate command?
> >
> >
> >
> > Regards
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From mcampbellsmith at gmail.com  Tue Jan  5 19:36:25 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 6 Jan 2010 14:36:25 +1100
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
Message-ID: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>

Hi!

If I try to call out on one of my voip providers I get
INCOMPATIBLE_DESTINATION.  Something is going wrong with codec
negotiation:

2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP:
v=0
o=Sippy 257534956 1 IN IP4 80.232.37.178
s=-
t=0 0
m=audio 47904 RTP/AVP 2 101 13
c=IN IP4 213.50.90.3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [G721:2:8000:20]/[G726-32:2:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [G721:2:8000:20]/[G726-32:2:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [G721:2:8000:20]/[PCMU:0:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [G721:2:8000:20]/[PCMA:8:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3262 Set 2833 dtmf
payload to 101
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [telephone-event:101:8000:20]/[G726-32:2:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [telephone-event:101:8000:20]/[G726-32:2:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [telephone-event:101:8000:20]/[PCMU:0:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [CN:13:8000:20]/[G726-32:2:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [CN:13:8000:20]/[G726-32:2:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [CN:13:8000:20]/[PCMU:0:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [CN:13:8000:20]/[PCMA:8:8000:20]
2010-01-05 18:57:56.845029 [NOTICE] sofia.c:3937 Hangup
sofia/external/020555500 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]

According to their website, the codec they us is G711 a-law or G729.
What codecs have name CN and telephone-event?

How do I get these to match?  I assume the format of the debug output
is: Audio Codec compare [Codec Name:Media Format:Media Name:Rate:RTP
Size?]

Thanks


From brian at freeswitch.org  Tue Jan  5 19:45:20 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Jan 2010 21:45:20 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
Message-ID: <DE3CBF01-9E0B-46FC-95C3-B27B824DC7F1@freeswitch.org>

your invite says G726-32  (thats what the 2 is in the audio line)

/b

On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:

> 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP:
> v=0
> o=Sippy 257534956 1 IN IP4 80.232.37.178
> s=-
> t=0 0
> m=audio 47904 RTP/AVP 2 101 13
> c=IN IP4 213.50.90.3
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15



From mcampbellsmith at gmail.com  Tue Jan  5 19:59:19 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 6 Jan 2010 14:59:19 +1100
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <DE3CBF01-9E0B-46FC-95C3-B27B824DC7F1@freeswitch.org>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<DE3CBF01-9E0B-46FC-95C3-B27B824DC7F1@freeswitch.org>
Message-ID: <33c87fa31001051959q42627fekc2c6a85fb68792af@mail.gmail.com>

I thought the Remote SDP comes from my SIP provider?  o=Sippy
257534956 1 IN IP4 80.232.37.178.  80.232.37.178 is not my IP address,
its the ip address of my voip provider?   But now you mention that
2=G726-32, its what I have as default.

Anyway, even if G726-32 is default, I also have PCMU and PCMA ... why
isn't negotiation used to select these codecs and isn't FS comparing
codecs?

Sorry, I guess these are basic questions ...


On Wed, Jan 6, 2010 at 2:45 PM, Brian West <brian at freeswitch.org> wrote:
> your invite says G726-32 ?(thats what the 2 is in the audio line)
>
> /b
>
> On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
>
>> 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP:
>> v=0
>> o=Sippy 257534956 1 IN IP4 80.232.37.178
>> s=-
>> t=0 0
>> m=audio 47904 RTP/AVP 2 101 13
>> c=IN IP4 213.50.90.3
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Tue Jan  5 20:02:59 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Jan 2010 22:02:59 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001051959q42627fekc2c6a85fb68792af@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<DE3CBF01-9E0B-46FC-95C3-B27B824DC7F1@freeswitch.org>
	<33c87fa31001051959q42627fekc2c6a85fb68792af@mail.gmail.com>
Message-ID: <A42B7C6C-F40B-453F-B895-EE33307DAB81@freeswitch.org>

because they aren't in the invite... it can't negotiate things that aren't in the invite... its clearly NOT in that sdp... its only CN, G726-32 and Telephony event.

/b

On Jan 5, 2010, at 9:59 PM, Mark Campbell-Smith wrote:

> Anyway, even if G726-32 is default, I also have PCMU and PCMA ... why
> isn't negotiation used to select these codecs and isn't FS comparing
> codecs?
> 
> Sorry, I guess these are basic questions ...



From mcampbellsmith at gmail.com  Tue Jan  5 20:16:40 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 6 Jan 2010 15:16:40 +1100
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <A42B7C6C-F40B-453F-B895-EE33307DAB81@freeswitch.org>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<DE3CBF01-9E0B-46FC-95C3-B27B824DC7F1@freeswitch.org>
	<33c87fa31001051959q42627fekc2c6a85fb68792af@mail.gmail.com>
	<A42B7C6C-F40B-453F-B895-EE33307DAB81@freeswitch.org>
Message-ID: <33c87fa31001052016r6d2ae072g8a210e244dfcc268@mail.gmail.com>

For my codec prefs I have:
  <X-PRE-PROCESS cmd="set"
data="global_codec_prefs=iLBC at 30i,G726-32,G722,PCMU,PCMA"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G726-32,PCMU,PCMA"/>

How many codecs are sent in an invite?  Is it only the top three of
global_codec_prefs?  Is CN = iLBC and telephony-event = G722?

Thanks

On Wed, Jan 6, 2010 at 3:02 PM, Brian West <brian at freeswitch.org> wrote:
> because they aren't in the invite... it can't negotiate things that aren't in the invite... its clearly NOT in that sdp... its only CN, G726-32 and Telephony event.
>
> /b
>
> On Jan 5, 2010, at 9:59 PM, Mark Campbell-Smith wrote:
>
>> Anyway, even if G726-32 is default, I also have PCMU and PCMA ... why
>> isn't negotiation used to select these codecs and isn't FS comparing
>> codecs?
>>
>> Sorry, I guess these are basic questions ...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Tue Jan  5 20:20:25 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Jan 2010 22:20:25 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
Message-ID: <1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>

Lets try this again.. THIS SDP IS NOT from FreeSWITCH its an inbound invite to FreeSWITCH... fix that and you'll be golden.

/b

On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:

> 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP:
> v=0
> o=Sippy 257534956 1 IN IP4 80.232.37.178
> s=-
> t=0 0
> m=audio 47904 RTP/AVP 2 101 13
> c=IN IP4 213.50.90.3
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15



From freeswitch at aastral.net  Tue Jan  5 20:55:15 2010
From: freeswitch at aastral.net (Bill W.)
Date: Tue, 05 Jan 2010 23:55:15 -0500
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <201001052125.06909.sos@sokhapkin.dyndns.org>
References: <AcqOWx9O9EDYsJ/ZQJSSOHSvWxhecA==>	<A2B4F48489739E439D8845296A2FD9C4147E1F@sun.Noble.local>	<b39273531001051813v5271858fpd133326559238b5e@mail.gmail.com>
	<201001052125.06909.sos@sokhapkin.dyndns.org>
Message-ID: <4B4417B3.9090807@aastral.net>

Hey Sergey,

But nibblebill will transfer to an extension of your choice when the
balance reaches $0. So if you set the nibble heartbeat to 60 seconds or
whatever, nibblebill will deduct the appropriate amount every
<heartbeat> seconds.   So after about 20 minutes, the call will execute
the nobal_action specified in nibblebill.conf.xml.

So that should meet your needs.

Bill W.


Sergey Okhapkin wrote:
> Unfortunalely, mod_nibblebill doesn't take billing increments into account.
> 
> On Tuesday 05 January 2010, Rupa Schomaker wrote:
>> Look at using mod_nibblebill
>>


From mcampbellsmith at gmail.com  Tue Jan  5 21:03:33 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 6 Jan 2010 16:03:33 +1100
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
Message-ID: <33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>

ahh... thats what I thought originally.

So what codecs match CN and telephony-event?  Where can I find these mappings?

m=audio 47904 RTP/AVP 2 101 13

Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20]
Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20]

Thanks Brian!

On Wed, Jan 6, 2010 at 3:20 PM, Brian West <brian at freeswitch.org> wrote:
> Lets try this again.. THIS SDP IS NOT from FreeSWITCH its an inbound invite to FreeSWITCH... fix that and you'll be golden.
>
> /b
>
> On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
>
>> 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP:
>> v=0
>> o=Sippy 257534956 1 IN IP4 80.232.37.178
>> s=-
>> t=0 0
>> m=audio 47904 RTP/AVP 2 101 13
>> c=IN IP4 213.50.90.3
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Tue Jan  5 21:19:28 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Jan 2010 23:19:28 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
Message-ID: <895DF0A8-334E-4214-916D-16446AAED1F7@freeswitch.org>

13 = CN
101 = Telephony Event
2 = G723-32

/b

On Jan 5, 2010, at 11:03 PM, Mark Campbell-Smith wrote:

> ahh... thats what I thought originally.
> 
> So what codecs match CN and telephony-event?  Where can I find these mappings?
> 
> m=audio 47904 RTP/AVP 2 101 13
> 
> Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20]
> Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20]
> 
> Thanks Brian!



From anthony.minessale at gmail.com  Tue Jan  5 21:20:04 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Jan 2010 23:20:04 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
Message-ID: <191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>

those are not audio codecs CN is just comfort noise and telephone-event is
dtmf


On Tue, Jan 5, 2010 at 11:03 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> ahh... thats what I thought originally.
>
> So what codecs match CN and telephony-event?  Where can I find these
> mappings?
>
> m=audio 47904 RTP/AVP 2 101 13
>
> Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20]
> Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20]
>
> Thanks Brian!
>
> On Wed, Jan 6, 2010 at 3:20 PM, Brian West <brian at freeswitch.org> wrote:
> > Lets try this again.. THIS SDP IS NOT from FreeSWITCH its an inbound
> invite to FreeSWITCH... fix that and you'll be golden.
> >
> > /b
> >
> > On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
> >
> >> 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP:
> >> v=0
> >> o=Sippy 257534956 1 IN IP4 80.232.37.178
> >> s=-
> >> t=0 0
> >> m=audio 47904 RTP/AVP 2 101 13
> >> c=IN IP4 213.50.90.3
> >> a=rtpmap:101 telephone-event/8000
> >> a=fmtp:101 0-15
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
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From mcampbellsmith at gmail.com  Tue Jan  5 21:40:10 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 6 Jan 2010 16:40:10 +1100
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
	<191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
Message-ID: <33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>

Thanks Brian and Anthony.

Brian: 2 = G723-32 Do you mean G721 instead of G723-32 ?

2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [G721:2:8000:20]/[G726-32:2:8000:20]

So the only codec they have offered is G723-32 (or G721), which FS
only supports as a passthrough codec.  Is that correct?


On Wed, Jan 6, 2010 at 4:20 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> those are not audio codecs CN is just comfort noise and telephone-event is
> dtmf
>
>
> On Tue, Jan 5, 2010 at 11:03 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>>
>> ahh... thats what I thought originally.
>>
>> So what codecs match CN and telephony-event? ?Where can I find these
>> mappings?
>>
>> m=audio 47904 RTP/AVP 2 101 13
>>
>> Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20]
>> Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20]
>>
>> Thanks Brian!
>>
>> On Wed, Jan 6, 2010 at 3:20 PM, Brian West <brian at freeswitch.org> wrote:
>> > Lets try this again.. THIS SDP IS NOT from FreeSWITCH its an inbound
>> > invite to FreeSWITCH... fix that and you'll be golden.
>> >
>> > /b
>> >
>> > On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
>> >
>> >> 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP:
>> >> v=0
>> >> o=Sippy 257534956 1 IN IP4 80.232.37.178
>> >> s=-
>> >> t=0 0
>> >> m=audio 47904 RTP/AVP 2 101 13
>> >> c=IN IP4 213.50.90.3
>> >> a=rtpmap:101 telephone-event/8000
>> >> a=fmtp:101 0-15
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jon at radel.com  Tue Jan  5 21:45:37 2010
From: jon at radel.com (Jon Radel)
Date: Wed, 06 Jan 2010 00:45:37 -0500
Subject: [Freeswitch-users] Codec Negotiation:
	Codec	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
Message-ID: <4B442381.9020705@radel.com>

Mark Campbell-Smith wrote:
> ahh... thats what I thought originally.
> 
> So what codecs match CN and telephony-event?  Where can I find these mappings?

http://www.iana.org/assignments/rtp-parameters

is one place to start your journey, but there's enough dynamic 
assignment, convention, and general etc. to make it an interesting 
journey.  If somebody can set me straight and provide the definitive 
documentation on payload types, should such actually exist, I'd be most 
appreciative.

-- 

--Jon Radel
jon at radel.com
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From anthony.minessale at gmail.com  Tue Jan  5 21:50:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Jan 2010 23:50:10 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
	<191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
	<33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
Message-ID: <191c3a031001052150x1fd22ed0p86d3a6f307df4b32@mail.gmail.com>

no we fully support it, just add it to your config

google for iana codec sdp for the reserved numbers and what they mean.


On Tue, Jan 5, 2010 at 11:40 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Thanks Brian and Anthony.
>
> Brian: 2 = G723-32 Do you mean G721 instead of G723-32 ?
>
> 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
> Compare [G721:2:8000:20]/[G726-32:2:8000:20]
>
> So the only codec they have offered is G723-32 (or G721), which FS
> only supports as a passthrough codec.  Is that correct?
>
>
> On Wed, Jan 6, 2010 at 4:20 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > those are not audio codecs CN is just comfort noise and telephone-event
> is
> > dtmf
> >
> >
> > On Tue, Jan 5, 2010 at 11:03 PM, Mark Campbell-Smith
> > <mcampbellsmith at gmail.com> wrote:
> >>
> >> ahh... thats what I thought originally.
> >>
> >> So what codecs match CN and telephony-event?  Where can I find these
> >> mappings?
> >>
> >> m=audio 47904 RTP/AVP 2 101 13
> >>
> >> Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20]
> >> Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20]
> >>
> >> Thanks Brian!
> >>
> >> On Wed, Jan 6, 2010 at 3:20 PM, Brian West <brian at freeswitch.org>
> wrote:
> >> > Lets try this again.. THIS SDP IS NOT from FreeSWITCH its an inbound
> >> > invite to FreeSWITCH... fix that and you'll be golden.
> >> >
> >> > /b
> >> >
> >> > On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
> >> >
> >> >> 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP:
> >> >> v=0
> >> >> o=Sippy 257534956 1 IN IP4 80.232.37.178
> >> >> s=-
> >> >> t=0 0
> >> >> m=audio 47904 RTP/AVP 2 101 13
> >> >> c=IN IP4 213.50.90.3
> >> >> a=rtpmap:101 telephone-event/8000
> >> >> a=fmtp:101 0-15
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Tue Jan  5 21:50:46 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Jan 2010 23:50:46 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
	<191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
	<33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
Message-ID: <0A8CD6A7-CA51-44F2-B530-15AE623EBA5D@freeswitch.org>

They are in fact one in the same please see ITU.

/b

On Jan 5, 2010, at 11:40 PM, Mark Campbell-Smith wrote:

> Thanks Brian and Anthony.
> 
> Brian: 2 = G723-32 Do you mean G721 instead of G723-32 ?
> 
> 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
> Compare [G721:2:8000:20]/[G726-32:2:8000:20]
> 
> So the only codec they have offered is G723-32 (or G721), which FS
> only supports as a passthrough codec.  Is that correct?



From achaloyan at yahoo.com  Tue Jan  5 23:29:10 2010
From: achaloyan at yahoo.com (Arsen Chaloyan)
Date: Tue, 5 Jan 2010 23:29:10 -0800 (PST)
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <0A8CD6A7-CA51-44F2-B530-15AE623EBA5D@freeswitch.org>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
	<191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
	<33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
	<0A8CD6A7-CA51-44F2-B530-15AE623EBA5D@freeswitch.org>
Message-ID: <637054.57565.qm@web111313.mail.gq1.yahoo.com>

The following section in RFC3551 states the same
http://tools.ietf.org/html/rfc3551#section-4.5.4


   The encodings of G.726 are here denoted as G726-40, G726-32, G726-24,
   and G726-16.  Prior to 1990, G721 described the 32 kbit/s ADPCM
   encoding, and G723 described the 40, 32, and 16 kbit/s encodings.
   Thus, G726-32 designates the same algorithm as G721 in RFC 1890.





________________________________
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Wed, January 6, 2010 9:50:46 AM
Subject: Re: [Freeswitch-users] Codec Negotiation: Codec telephone-event:101:8000:20

They are in fact one in the same please see ITU.

/b

On Jan 5, 2010, at 11:40 PM, Mark Campbell-Smith wrote:

> Thanks Brian and Anthony.
> 
> Brian: 2 = G723-32 Do you mean G721 instead of G723-32 ?
> 
> 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
> Compare [G721:2:8000:20]/[G726-32:2:8000:20]
> 
> So the only codec they have offered is G723-32 (or G721), which FS
> only supports as a passthrough codec.  Is that correct?


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From durk.debeer at isp.solcon.nl  Wed Jan  6 01:04:26 2010
From: durk.debeer at isp.solcon.nl (Durk.de Beer)
Date: Wed, 06 Jan 2010 10:04:26 +0100
Subject: [Freeswitch-users]  Detecting status of User Agent
Message-ID: <mailman.2.1262768701.2492.freeswitch-users@lists.freeswitch.org>



Ok I want to do the following thing. If an user agent (UA), some SIP-client for instance X-lite, is of line the call is to be forwarded to an phone number provided by the user how was called.
I succeed doing this if there is no registration for the number dialled. FS is then reporting an USER_NOT_REGISTERED so I am able to alter the dial plan accordingly and redirect the call. So far so good. Now my problem arises when there is an registration on FS but the UA is not online for what ever reason (my cat seems to like to chew on CAT5 cable). If this occurs FS is trying to bridge but it is unable to because there is no response from the UA called. It ends by a RECOVERY_ON_TIMER_EXPIRE but the time it takes for this to happen is far to long. I've tried to set this variable but failed to do so. Other channel variables seem to be only effective if there's an successful bridge. So how do I set the RECOVERY_ON_TIMER_EXPIRE?




Durk 


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From sos at sokhapkin.dyndns.org  Wed Jan  6 03:20:13 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 6 Jan 2010 06:20:13 -0500
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <4B4417B3.9090807@aastral.net>
References: <AcqOWx9O9EDYsJ/ZQJSSOHSvWxhecA==>
	<201001052125.06909.sos@sokhapkin.dyndns.org>
	<4B4417B3.9090807@aastral.net>
Message-ID: <201001060620.13735.sos@sokhapkin.dyndns.org>

nibblebill has no concept of billing blocks. What if I want to bill customer 
30 seconds minimum and 6 seconds increment thereafter?

On Tuesday 05 January 2010, Bill W. wrote:
> Hey Sergey,
>
> But nibblebill will transfer to an extension of your choice when the
> balance reaches $0. So if you set the nibble heartbeat to 60 seconds or
> whatever, nibblebill will deduct the appropriate amount every
> <heartbeat> seconds.   So after about 20 minutes, the call will execute
> the nobal_action specified in nibblebill.conf.xml.
>
> So that should meet your needs.
>
> Bill W.
>
> Sergey Okhapkin wrote:
> > Unfortunalely, mod_nibblebill doesn't take billing increments into
> > account.
> >
> > On Tuesday 05 January 2010, Rupa Schomaker wrote:
> >> Look at using mod_nibblebill
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jbr at consiglia.dk  Wed Jan  6 03:55:14 2010
From: jbr at consiglia.dk (Jon Bruel)
Date: Wed, 6 Jan 2010 12:55:14 +0100
Subject: [Freeswitch-users] Is there support for custom fields for all events
Message-ID: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE09AC568438@SBS2008SERVER.consiglia.local>

I would like to add a custom field to all events sent from the FS. The value of the field should be the value of a global variable, the name should preferably be the name of this variable. Is there any way to set this up? /Jon
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From jcasale at activenetwerx.com  Wed Jan  6 03:56:14 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 6 Jan 2010 11:56:14 +0000
Subject: [Freeswitch-users] Dahdi Saga continues
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906D5C2@prato.activenetwerx.local>

So when the dahdi config is left empty, just defaultzone=us and
loadzone=us, it works except obviously there is no echo canceller.

Once I add an echo canceller and set fxsks=1 then callerid fails,
it just says OpenZap on the handsets and the audio doesn't start
working after some time after the call is answered.

Any ideas?

Thanks!
jlc


From Russell.Mosemann at cune.org  Wed Jan  6 04:17:15 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Wed, 6 Jan 2010 06:17:15 -0600
Subject: [Freeswitch-users] Dahdi Saga continues
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E906D5C2@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E906D5C2@prato.activenetwerx.local>
Message-ID: <45529145881D44A89C1B0437D78B2712@cune.pri>

Joseph L. Casale wrote:

> Once I add an echo canceller and set fxsks=1 then callerid fails,

You could try building DAHDI with OSLEC. Don't put any echo cancel statements in the config file. The steps you want are under "Install OSLEC with DAHDI".

http://www.rowetel.com/ucasterisk/oslec.html

--
Russell Mosemann



From jbr at consiglia.dk  Wed Jan  6 05:33:07 2010
From: jbr at consiglia.dk (Jon Bruel)
Date: Wed, 6 Jan 2010 14:33:07 +0100
Subject: [Freeswitch-users] Detecting status of User Agent
In-Reply-To: <9ad1f5d7-ad02-49df-81dd-42e4f7d5f1cd@SBS2008SERVER.consiglia.local>
References: <9ad1f5d7-ad02-49df-81dd-42e4f7d5f1cd@SBS2008SERVER.consiglia.local>
Message-ID: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE09AC568439@SBS2008SERVER.consiglia.local>

Hi Durk

I tried the situation mentioned by unplugging the UA. FS then reports back (via the channel variable originate_disposition): NORMAL_TEMPORARY_FAILURE, which can then be used to take action. Before bridging to the phone, I have set <action application="set" data="continue_on_fail=true"/>. Hope it assists you.

Anyhow, your cat must be very intelligent since it knows it's a CAT cable.

/Jon

Jon Br?el
Consiglia Telecommunications
DK-2960 Rungsted Kyst
Tel: +45 45 16 1000
Mob: +45 26 15 30 60
CVR: 27047882

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Durk.de Beer
Sent: 6. januar 2010 10:04
To: Freeswitch-Users
Subject: [Freeswitch-users] Detecting status of User Agent




Ok I want to do the following thing. If an user agent (UA), some SIP-client for instance X-lite, is of line the call is to be forwarded to an phone number provided by the user how was called.

I succeed doing this if there is no registration for the number dialled. FS is then reporting an USER_NOT_REGISTERED so I am able to alter the dial plan accordingly and redirect the call. So far so good. Now my problem arises when there is an registration on FS but the UA is not online for what ever reason (my cat seems to like to chew on CAT5 cable). If this occurs FS is trying to bridge but it is unable to because there is no response from the UA called. It ends by a RECOVERY_ON_TIMER_EXPIRE but the time it takes for this to happen is far to long. I've tried to set this variable but failed to do so. Other channel variables seem to be only effective if there's an successful bridge. So how do I set the RECOVERY_ON_TIMER_EXPIRE?





Durk

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From linux4michelle at tamay-dogan.net  Wed Jan  6 05:52:10 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Wed, 6 Jan 2010 14:52:10 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <20100103184636.GW5547@tamay-dogan.net>
References: <20100103184636.GW5547@tamay-dogan.net>
Message-ID: <20100106135210.GG5547@tamay-dogan.net>

Realy no one who use FreeSwitch as GSM PBX?


Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From jcasale at activenetwerx.com  Wed Jan  6 06:29:38 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 6 Jan 2010 14:29:38 +0000
Subject: [Freeswitch-users] Dahdi Saga continues
In-Reply-To: <45529145881D44A89C1B0437D78B2712@cune.pri>
References: <CA5A491E9DEFBE4CB777DE97E21575E906D5C2@prato.activenetwerx.local>
	<45529145881D44A89C1B0437D78B2712@cune.pri>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906E0F1@prato.activenetwerx.local>

>You could try building DAHDI with OSLEC. Don't put any echo cancel statements in the config file. The steps you want >are under "Install OSLEC with DAHDI".
>
>http://www.rowetel.com/ucasterisk/oslec.html

You still need to specify the echo canceller w/ oslec as well
though. What do you think the lack of configuration making it
work is indicative of? I think for for the price, I might just
buy an SPA3102 and be done w/ this nightmare...


From Russell.Mosemann at cune.org  Wed Jan  6 06:38:52 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Wed, 6 Jan 2010 14:38:52 -0000
Subject: [Freeswitch-users] Dahdi Saga continues
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E906E0F1@prato.activenetwerx.local>
Message-ID: <20100106143852.7E98C2C83EC@cuneorg-email.cune.pri>

"Joseph L. Casale" <jcasale at activenetwerx.com> said:

> You still need to specify the echo canceller w/ oslec as well
> though.

Are you sure? I was under the impression that OSLEC was built in and that
there was no choice to turn it on or off.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From brian at freeswitch.org  Wed Jan  6 07:04:39 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jan 2010 09:04:39 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <637054.57565.qm@web111313.mail.gq1.yahoo.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
	<191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
	<33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
	<0A8CD6A7-CA51-44F2-B530-15AE623EBA5D@freeswitch.org>
	<637054.57565.qm@web111313.mail.gq1.yahoo.com>
Message-ID: <965454EF-F779-4563-AC64-082CAB07CCA5@freeswitch.org>

w00t! :)

/b

On Jan 6, 2010, at 1:29 AM, Arsen Chaloyan wrote:

> The following section in RFC3551 states the same
> http://tools.ietf.org/html/rfc3551#section-4.5.4
> 
> 
>    The encodings of G.726 are here denoted as G726-40, G726-32, G726-24,
>    and G726-16.  Prior to 1990, G721 described the 32 kbit/s ADPCM
>    encoding, and G723 described the 40, 32, and 16 kbit/s encodings.
>    Thus, G726-32 designates the same algorithm as G721 in RFC 1890.
> 
> 
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wed, January 6, 2010 9:50:46 AM
> Subject: Re: [Freeswitch-users] Codec Negotiation: Codec telephone-event:101:8000:20
> 
> They are in fact one in the same please see ITU.
> 
> /b

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From mike at jerris.com  Wed Jan  6 07:19:19 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 6 Jan 2010 10:19:19 -0500
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <20100106135210.GG5547@tamay-dogan.net>
References: <20100103184636.GW5547@tamay-dogan.net>
	<20100106135210.GG5547@tamay-dogan.net>
Message-ID: <CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>

I was hoping Giovanni, who is working on code as an endpoint module for gsm devices, would answer this.  But he has not

Mie

On Jan 6, 2010, at 8:52 AM, Michelle Konzack wrote:

> Realy no one who use FreeSwitch as GSM PBX?



From gmaruzz at celliax.org  Wed Jan  6 07:59:14 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 6 Jan 2010 16:59:14 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>
References: <20100103184636.GW5547@tamay-dogan.net>
	<20100106135210.GG5547@tamay-dogan.net> 
	<CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>
Message-ID: <7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com>

hehehe,

MikeJ, you're right!

I'm taking opportunity from holidays (in Italy we still in holidays
until tomorrow), to make ready for testing the endpoint for GSM
devices.

Short blurb:
- can use as phisical interface high end GSM modules, or gsm modems,
or gsm cellphones (with cables)
- can send/receive SMSs and voice calls
- SMSs generates events
- will use standard CHAT api interface for SMSs (like Jingle and sofia/SIMPLE)
- will be possible to compile it with sound support and a c++ library
for PDU access (for who that wants maximum SMS details plus voice
calls)
-will be possible to compile it as bare C (no c++) without sound
support and without PDU support for maximum embeddability in low end
machines (will act as an SMS gateway, sending/receiving SMSs, without
voice calls)

will soon be ported to work on windoz too (at the moment, works only
on Linux, maybe on *BSD too)

You can see old wikipage (no more reliable, to be thoroughly updated)
here: http://wiki.freeswitch.org/wiki/GSMopen

-giovanni


On Wed, Jan 6, 2010 at 4:19 PM, Michael Jerris <mike at jerris.com> wrote:
> I was hoping Giovanni, who is working on code as an endpoint module for gsm devices, would answer this. ?But he has not
>
> Mie
>
> On Jan 6, 2010, at 8:52 AM, Michelle Konzack wrote:
>
>> Realy no one who use FreeSwitch as GSM PBX?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Wed Jan  6 08:08:42 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 6 Jan 2010 17:08:42 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com>
References: <20100103184636.GW5547@tamay-dogan.net>
	<20100106135210.GG5547@tamay-dogan.net> 
	<CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>
	<7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com>
Message-ID: <7b197bef1001060808m1139bd96o3d86bfe89601399d@mail.gmail.com>

Michelle,

sorry, after thorough search I find your first two messages in the
Spam box (while MikeJ was passing through).

(I've now set the filter on Freeswitch-users to "never" go in Spam)

If you need any info, don't hesitate to write to the mailing list or
catch me in IRC @freeswitch channel as gmaruzz (or gmaruzz1, at the
will of net splits).

-giovanni


On Wed, Jan 6, 2010 at 4:59 PM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
> hehehe,
>
> MikeJ, you're right!
>
> I'm taking opportunity from holidays (in Italy we still in holidays
> until tomorrow), to make ready for testing the endpoint for GSM
> devices.
>
> Short blurb:
> - can use as phisical interface high end GSM modules, or gsm modems,
> or gsm cellphones (with cables)
> - can send/receive SMSs and voice calls
> - SMSs generates events
> - will use standard CHAT api interface for SMSs (like Jingle and sofia/SIMPLE)
> - will be possible to compile it with sound support and a c++ library
> for PDU access (for who that wants maximum SMS details plus voice
> calls)
> -will be possible to compile it as bare C (no c++) without sound
> support and without PDU support for maximum embeddability in low end
> machines (will act as an SMS gateway, sending/receiving SMSs, without
> voice calls)
>
> will soon be ported to work on windoz too (at the moment, works only
> on Linux, maybe on *BSD too)
>
> You can see old wikipage (no more reliable, to be thoroughly updated)
> here: http://wiki.freeswitch.org/wiki/GSMopen
>
> -giovanni
>
>
> On Wed, Jan 6, 2010 at 4:19 PM, Michael Jerris <mike at jerris.com> wrote:
>> I was hoping Giovanni, who is working on code as an endpoint module for gsm devices, would answer this. ?But he has not
>>
>> Mie
>>
>> On Jan 6, 2010, at 8:52 AM, Michelle Konzack wrote:
>>
>>> Realy no one who use FreeSwitch as GSM PBX?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From freeswitch at aastral.net  Wed Jan  6 08:25:18 2010
From: freeswitch at aastral.net (Bill W)
Date: Wed, 06 Jan 2010 11:25:18 -0500
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <201001060620.13735.sos@sokhapkin.dyndns.org>
References: <AcqOWx9O9EDYsJ/ZQJSSOHSvWxhecA==>	<201001052125.06909.sos@sokhapkin.dyndns.org>	<4B4417B3.9090807@aastral.net>
	<201001060620.13735.sos@sokhapkin.dyndns.org>
Message-ID: <4B44B96E.2040408@aastral.net>

True, there is no inherent support for that, but you might be able to 
get close by doing it in the dialplan.  Establish the call, pause 
nibblebill, deduct a specific amount (nibblebill adjust), and when 30 
seconds are up, unpause nibblebill.

More than likely you'd have to do this in a script rather than in XML.

The issue would be the last interval after the last heartbeat. 
Nibblebill won't round up to the next 6 seconds.  It will just bill for 
the actual call time. (If I understand things correctly).

Or you could put in a bounty to have billing block support added to 
nibblebill.


Hope this helps.
Bill


Sergey Okhapkin wrote:
> nibblebill has no concept of billing blocks. What if I want to bill customer 
> 30 seconds minimum and 6 seconds increment thereafter?
> 


From oscav at hotmail.fr  Wed Jan  6 08:37:21 2010
From: oscav at hotmail.fr (Oscav)
Date: Wed, 6 Jan 2010 08:37:21 -0800 (PST)
Subject: [Freeswitch-users] re lease an outbound call when caller sends
 digits like ##
Message-ID: <27026910.post@talk.nabble.com>


Hi,

How can we cancel an outbound call if the caller digits some DTMF like ## ??
There is the bind_meta_app but it only handles 1 digit.

Thanks.
-- 
View this message in context: http://old.nabble.com/release-an-outbound-call-when-caller-sends-digits-like----tp27026910p27026910.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From mike at jerris.com  Wed Jan  6 08:41:08 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 6 Jan 2010 11:41:08 -0500
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <201001060620.13735.sos@sokhapkin.dyndns.org>
References: <AcqOWx9O9EDYsJ/ZQJSSOHSvWxhecA==>
	<201001052125.06909.sos@sokhapkin.dyndns.org>
	<4B4417B3.9090807@aastral.net>
	<201001060620.13735.sos@sokhapkin.dyndns.org>
Message-ID: <EF51CEC4-232C-4026-AAEF-451AB18CAB01@jerris.com>

This is open source, you can create a patch to add this functionality and contribute it back.

Mike

On Jan 6, 2010, at 6:20 AM, Sergey Okhapkin wrote:

> nibblebill has no concept of billing blocks. What if I want to bill customer 
> 30 seconds minimum and 6 seconds increment thereafter?
> 



From dome at tel.co.th  Wed Jan  6 08:59:44 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 6 Jan 2010 23:59:44 +0700
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <201001060620.13735.sos@sokhapkin.dyndns.org>
References: <201001052125.06909.sos@sokhapkin.dyndns.org>
	<4B4417B3.9090807@aastral.net>
	<201001060620.13735.sos@sokhapkin.dyndns.org>
Message-ID: <8ccbff061001060859p59aa6b5bw6f52275650f9138@mail.gmail.com>

2010/1/6 Sergey Okhapkin <sos at sokhapkin.dyndns.org>:
> nibblebill has no concept of billing blocks. What if I want to bill customer
> 30 seconds minimum and 6 seconds increment thereafter?

I have billing (in house develop) and customize  nibble_bill update
cdr table (in my billing) i use postgresql trigger to update account
balance. i have many increment rule 1/1 30/6 60/60

It's work well for me :)


BG

Dome C.
>
> On Tuesday 05 January 2010, Bill W. wrote:
>> Hey Sergey,
>>
>> But nibblebill will transfer to an extension of your choice when the
>> balance reaches $0. So if you set the nibble heartbeat to 60 seconds or
>> whatever, nibblebill will deduct the appropriate amount every
>> <heartbeat> seconds. ? So after about 20 minutes, the call will execute
>> the nobal_action specified in nibblebill.conf.xml.
>>
>> So that should meet your needs.
>>
>> Bill W.
>>
>> Sergey Okhapkin wrote:
>> > Unfortunalely, mod_nibblebill doesn't take billing increments into
>> > account.
>> >
>> > On Tuesday 05 January 2010, Rupa Schomaker wrote:
>> >> Look at using mod_nibblebill
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From linux4michelle at tamay-dogan.net  Wed Jan  6 09:14:13 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Wed, 6 Jan 2010 18:14:13 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com>
References: <20100103184636.GW5547@tamay-dogan.net>
	<20100106135210.GG5547@tamay-dogan.net>
	<CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>
	<7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com>
Message-ID: <20100106171413.GI5547@tamay-dogan.net>

Chiao Giovanni,

thankyoufor you efforts to get this running...

Do you have a list of High-End GSM-Modules?

If not my co-worker and me would develop one but we are not sure, which
End-Points we need in the Modem...  The biggest  problem  is,  that  we
need the HD*PA part too, to get the Internet connection.

If I see it right, the "Nokia 76120 classic" has  tonns  of  End-Points
and it seems, irt support Data+Voice in the same time...

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant


Am 2010-01-06 16:59:14, schrieb Giovanni Maruzzelli:
> hehehe,
> 
> MikeJ, you're right!
> 
> I'm taking opportunity from holidays (in Italy we still in holidays
> until tomorrow), to make ready for testing the endpoint for GSM
> devices.
> 
> Short blurb:
> - can use as phisical interface high end GSM modules, or gsm modems,
> or gsm cellphones (with cables)
> - can send/receive SMSs and voice calls
> - SMSs generates events
> - will use standard CHAT api interface for SMSs (like Jingle and sofia/SIMPLE)
> - will be possible to compile it with sound support and a c++ library
> for PDU access (for who that wants maximum SMS details plus voice
> calls)
> -will be possible to compile it as bare C (no c++) without sound
> support and without PDU support for maximum embeddability in low end
> machines (will act as an SMS gateway, sending/receiving SMSs, without
> voice calls)
> 
> will soon be ported to work on windoz too (at the moment, works only
> on Linux, maybe on *BSD too)
> 
> You can see old wikipage (no more reliable, to be thoroughly updated)
> here: http://wiki.freeswitch.org/wiki/GSMopen
> 
> -giovanni
------------------------ END OF REPLIED MESSAGE ------------------------



-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From jcasale at activenetwerx.com  Wed Jan  6 09:26:02 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 6 Jan 2010 17:26:02 +0000
Subject: [Freeswitch-users] Dahdi Saga continues
In-Reply-To: <20100106143852.7E98C2C83EC@cuneorg-email.cune.pri>
References: <CA5A491E9DEFBE4CB777DE97E21575E906E0F1@prato.activenetwerx.local>
	<20100106143852.7E98C2C83EC@cuneorg-email.cune.pri>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906E729@prato.activenetwerx.local>

>Are you sure? I was under the impression that OSLEC was built in and that
>there was no choice to turn it on or off.

All the digium dahdi docs and my previous experience with it reference it
being activated like any other canceller.

Dahdi specifically states all echo cancellers must be manually activated
per channel where dahdi_echocan_* => echocanceller=*,1... and the oslec
module builds with the same naming convention...


From anthony.minessale at gmail.com  Wed Jan  6 09:34:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Jan 2010 11:34:37 -0600
Subject: [Freeswitch-users] re lease an outbound call when caller sends
	digits like ##
In-Reply-To: <27026910.post@talk.nabble.com>
References: <27026910.post@talk.nabble.com>
Message-ID: <191c3a031001060934n3f33a977m1820d6e491364f16@mail.gmail.com>

well it's 1 digit plus the meta so its really 2

*0 for instance


On Wed, Jan 6, 2010 at 10:37 AM, Oscav <oscav at hotmail.fr> wrote:

>
> Hi,
>
> How can we cancel an outbound call if the caller digits some DTMF like ##
> ??
> There is the bind_meta_app but it only handles 1 digit.
>
> Thanks.
> --
> View this message in context:
> http://old.nabble.com/release-an-outbound-call-when-caller-sends-digits-like----tp27026910p27026910.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From gmaruzz at celliax.org  Wed Jan  6 09:39:22 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 6 Jan 2010 18:39:22 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <20100106171413.GI5547@tamay-dogan.net>
References: <20100103184636.GW5547@tamay-dogan.net>
	<20100106135210.GG5547@tamay-dogan.net> 
	<CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>
	<7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com> 
	<20100106171413.GI5547@tamay-dogan.net>
Message-ID: <7b197bef1001060939rd1e08f3h2cb82cd89e50225e@mail.gmail.com>

Hello Michelle,

it supports all modules that accepts standard ETSI AT-GSM commands
(so, let's say all of them).

Maybe I do not understand the second question, what do you means for Endpoints?

If you're talking usb endpoints, you'll need a modem endpoint (that
can be seen as a serial port), and (if you need audio, eg not just
SMSs but voice calls too) you need an audio endpoint (that can be seen
as a soundcard).

Many modules and cellphones can be seen as HDSPA or GPRS modems, just
check their specs.

For audio, if the module/cellphone/modem does not offer an audio usb
endpoint (eg cannot be seen as a soundcard) one trick is to connect
the headset jack to an usb soundcard (you can find soundcard with for
factor like a dongle based on cm-108 chipset for under $10). I'll
publish the schema of the cable needed from hadset jack in the
phone/module to the usb soundcard).

If I have not get what you asked, please explain more your question.

-giovanni

On Wed, Jan 6, 2010 at 6:14 PM, Michelle Konzack
<linux4michelle at tamay-dogan.net> wrote:
> Chiao Giovanni,
>
> thankyoufor you efforts to get this running...
>
> Do you have a list of High-End GSM-Modules?
>
> If not my co-worker and me would develop one but we are not sure, which
> End-Points we need in the Modem... ?The biggest ?problem ?is, ?that ?we
> need the HD*PA part too, to get the Internet connection.
>
> If I see it right, the "Nokia 76120 classic" has ?tonns ?of ?End-Points
> and it seems, irt support Data+Voice in the same time...
>
> Thanks, Greetings and nice Day/Evening
> ? ?Michelle Konzack
> ? ?Electronic Engineer
> ? ?Tamay Dogan Network
> ? ?Debian GNU/Linux Consultant
>
>
> Am 2010-01-06 16:59:14, schrieb Giovanni Maruzzelli:
>> hehehe,
>>
>> MikeJ, you're right!
>>
>> I'm taking opportunity from holidays (in Italy we still in holidays
>> until tomorrow), to make ready for testing the endpoint for GSM
>> devices.
>>
>> Short blurb:
>> - can use as phisical interface high end GSM modules, or gsm modems,
>> or gsm cellphones (with cables)
>> - can send/receive SMSs and voice calls
>> - SMSs generates events
>> - will use standard CHAT api interface for SMSs (like Jingle and sofia/SIMPLE)
>> - will be possible to compile it with sound support and a c++ library
>> for PDU access (for who that wants maximum SMS details plus voice
>> calls)
>> -will be possible to compile it as bare C (no c++) without sound
>> support and without PDU support for maximum embeddability in low end
>> machines (will act as an SMS gateway, sending/receiving SMSs, without
>> voice calls)
>>
>> will soon be ported to work on windoz too (at the moment, works only
>> on Linux, maybe on *BSD too)
>>
>> You can see old wikipage (no more reliable, to be thoroughly updated)
>> here: http://wiki.freeswitch.org/wiki/GSMopen
>>
>> -giovanni
> ------------------------ END OF REPLIED MESSAGE ------------------------
>
>
>
> --
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
> ##################### Debian GNU/Linux Consultant #####################
> <http://www.tamay-dogan.net/> ? ? ? ? ? ? ? ? Michelle Konzack
> <http://www.can4linux.org/> ? ? ? ? ? ? ? ? ? Apt. 917
> <http://www.flexray4linux.org/> ? ? ? ? ? ? ? 50, rue de Soultz
> Jabber linux4michelle at jabber.ccc.de ? ? ? ? ? 67100 Strabourg/France
> IRC ? ?#Debian (irc.icq.com) ? ? ? ? ? ? ? ? ?Tel. DE: +49 177 9351947
> ICQ ? ?#328449886 ? ? ? ? ? ? ? ? ? ? ? ? ? ? Tel. FR: +33 ?6 ?61925193
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From jerry.richards at teotech.com  Wed Jan  6 09:59:35 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 6 Jan 2010 09:59:35 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>
Message-ID: <BE608CAF54594CA48BB8B4DB82A8BFD6@greyhawk.tonecommander.com>

When I attempt an internal-to-PSTN call, there are no Q931 packets sent out
the PRI (I confirmed this using the Sangoma wanpipemon utility).  I suspect
this has something to do with my XML configuration.  Below are my
openzap/wanpipe configurations (which should all be defaulted).  Do you seen
anything wrong with these defaults, which might cause the following FS
console error?
 
zap_io.c:1197 outgoing_call method not implemented!
 
openzap.conf:
[span wanpipe smg_prid]
name => smg_prid
trunk_type =>t1
b-channel => 1:1-23
 
autoload_configs/openzap.conf.xml:
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="9"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
  <boost_spans>
    <span id="smg_prid">
      <!--<param name="hold-music" value="$${moh_uri}"/>-->
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <!-- regex to stop dialing when it matches -->
      <!--<param name="dial-regex" value="5555"/>-->
      <!-- regex to stop dialing when it does not match -->
      <!--<param name="fail-dial-regex" value="^5"/>-->
    </span>
  </boost_spans>
  <analog_spans>
  </analog_spans>
</configuration>
 
wanpipe.conf:
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
 
Thanks and Best Regards,
Jerry
 


  _____  

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Tuesday, January 05, 2010 4:33 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER




On Mon, Jan 4, 2010 at 3:49 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:



Hello,

I have one FS instance that is working well with a PRI and running FS
version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
upgraded it.

Now, I am trying to bring up another FS instance (basically a clone of the
first), but the PRI does not work.  When I attempt to make an
internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
both servers are using the latest Sangoma Wanpipe driver, and I copied the
conf XML file tree from the old server to the new one.  I think the problem
has to do with the openzap module, but I'm having difficulty isolating the
problem.  Could it have built the openzap module incorrectly?  Another
difference is that I installed 1.0.5pre9 from scratch on the new server
(i.e. it never had 1.0.4 running on it).

I put the FS log into the pastebin when an outbound call attempt is made:

http://pastebin.freeswitch.org/11675

Could someone give me a pointer on what to try next?



Jerry, I noticed this line:
(OpenZAP/1:1/3491028 at g1)

Is your carrier wanting full ten digit phone numbers? Try adding the area
code on this and see what happens. The error usually would be something like
"invalid number format" but I've seen carriers do stupid things like this.
Try that first and see if it makes a difference. If not you'll need to turn
on Q931 debugging as per the Sangoma wiki. (See
http://wiki.freeswitch.org/wiki/OpenZAP#Debugging_PRI_With_wanpipemon for
the link.)

-MC



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From anthony.minessale at gmail.com  Wed Jan  6 11:51:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Jan 2010 13:51:23 -0600
Subject: [Freeswitch-users] Is there support for custom fields for all
	events
In-Reply-To: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE09AC568438@SBS2008SERVER.consiglia.local>
References: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE09AC568438@SBS2008SERVER.consiglia.local>
Message-ID: <191c3a031001061151o756149d4x9e08ceea5cb263ff@mail.gmail.com>

not exactly, but you can set channel variables in the dialplan and those
will be in every event about that channel.
so you can make a global exten in your dialplan to set the variable before
anything else.

the hostname of the box is also in every event.
finally you could post a bounty request for some list of special variables
to add to every event on the bounty page or in jira under bounties. probably
about $500 by my estimation.



On Wed, Jan 6, 2010 at 5:55 AM, Jon Bruel <jbr at consiglia.dk> wrote:

>  *I would like to add a custom field to all events sent from the FS. The
> value of the field should be the value of a global variable, the name should
> preferably be the name of this variable. Is there any way to set this up?
> /Jon*
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jerry.richards at teotech.com  Wed Jan  6 11:52:21 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 6 Jan 2010 11:52:21 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com> 
Message-ID: <8441EC83BCD54223A5DAE91BEEC9C1FF@greyhawk.tonecommander.com>

I capured logs of my FS startup and put them into the pastebin
(http://pastebin.freeswitch.org/11692).  At line 722, I see some errors:
 
2010-01-06 11:42:49.907861 [ERR] zap_io.c:2562 Error loading
/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
[/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
file: No such file or directory]
2010-01-06 11:42:49.907883 [ERR] zap_io.c:2722 can't find 'sangoma_boost
2010-01-06 11:42:49.907902 [ERR] mod_openzap.c:2379 Error starting OpenZAP
span 1 error
2010-01-06 11:42:49.907929 [CONSOLE] switch_loadable_module.c:890
Successfully Loaded [mod_openzap]
 
Do you know why I would get this?  Where is the ozmod_sangoma_boost.so file
supposed to come from?
 
Thanks,
Jerry
 
 


  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, January 06, 2010 10:00 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] DESTINATION_OUT_OF_ORDER


When I attempt an internal-to-PSTN call, there are no Q931 packets sent out
the PRI (I confirmed this using the Sangoma wanpipemon utility).  I suspect
this has something to do with my XML configuration.  Below are my
openzap/wanpipe configurations (which should all be defaulted).  Do you seen
anything wrong with these defaults, which might cause the following FS
console error?
 
zap_io.c:1197 outgoing_call method not implemented!
 
openzap.conf:
[span wanpipe smg_prid]
name => smg_prid
trunk_type =>t1
b-channel => 1:1-23
 
autoload_configs/openzap.conf.xml:
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="9"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
  <boost_spans>
    <span id="smg_prid">
      <!--<param name="hold-music" value="$${moh_uri}"/>-->
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <!-- regex to stop dialing when it matches -->
      <!--<param name="dial-regex" value="5555"/>-->
      <!-- regex to stop dialing when it does not match -->
      <!--<param name="fail-dial-regex" value="^5"/>-->
    </span>
  </boost_spans>
  <analog_spans>
  </analog_spans>
</configuration>
 
wanpipe.conf:
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
 
Thanks and Best Regards,
Jerry
 


  _____  

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Tuesday, January 05, 2010 4:33 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER




On Mon, Jan 4, 2010 at 3:49 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:



Hello,

I have one FS instance that is working well with a PRI and running FS
version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
upgraded it.

Now, I am trying to bring up another FS instance (basically a clone of the
first), but the PRI does not work.  When I attempt to make an
internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
both servers are using the latest Sangoma Wanpipe driver, and I copied the
conf XML file tree from the old server to the new one.  I think the problem
has to do with the openzap module, but I'm having difficulty isolating the
problem.  Could it have built the openzap module incorrectly?  Another
difference is that I installed 1.0.5pre9 from scratch on the new server
(i.e. it never had 1.0.4 running on it).

I put the FS log into the pastebin when an outbound call attempt is made:

http://pastebin.freeswitch.org/11675

Could someone give me a pointer on what to try next?



Jerry, I noticed this line:
(OpenZAP/1:1/3491028 at g1)

Is your carrier wanting full ten digit phone numbers? Try adding the area
code on this and see what happens. The error usually would be something like
"invalid number format" but I've seen carriers do stupid things like this.
Try that first and see if it makes a difference. If not you'll need to turn
on Q931 debugging as per the Sangoma wiki. (See
http://wiki.freeswitch.org/wiki/OpenZAP#Debugging_PRI_With_wanpipemon for
the link.)

-MC



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From msc at freeswitch.org  Wed Jan  6 11:58:06 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Jan 2010 11:58:06 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <BE608CAF54594CA48BB8B4DB82A8BFD6@greyhawk.tonecommander.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>
	<BE608CAF54594CA48BB8B4DB82A8BFD6@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91001061158h4420961aj2878dc813d5637aa@mail.gmail.com>

I have an A104D on a box here. I'll need some time but I will see if I can
reproduce your symptoms. I'll let you know what I find out.

-MC

On Wed, Jan 6, 2010 at 9:59 AM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  When I attempt an internal-to-PSTN call, there are no Q931 packets sent
> out the PRI (I confirmed this using the Sangoma wanpipemon utility).  I
> suspect this has something to do with my XML configuration.  Below are my
> openzap/wanpipe configurations (which should all be defaulted).  Do you
> seen anything wrong with these defaults, which might cause the following FS
> console error?
>
> zap_io.c:1197 outgoing_call method not implemented!
>

>
> *openzap.conf:
> *[span wanpipe smg_prid]
> name => smg_prid
> trunk_type =>t1
> b-channel => 1:1-23
>
> *autoload_configs/openzap.conf.xml:*
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>   <settings>
>     <param name="debug" value="9"/>
>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>     <!--<param name="enable-analog-option" value="3-way"/>-->
>   </settings>
>   <boost_spans>
>     <span id="smg_prid">
>       <!--<param name="hold-music" value="$${moh_uri}"/>-->
>       <param name="dialplan" value="XML"/>
>       <param name="context" value="default"/>
>       <!-- regex to stop dialing when it matches -->
>       <!--<param name="dial-regex" value="5555"/>-->
>       <!-- regex to stop dialing when it does not match -->
>       <!--<param name="fail-dial-regex" value="^5"/>-->
>     </span>
>   </boost_spans>
>   <analog_spans>
>   </analog_spans>
> </configuration>
>
> *wanpipe.conf:*
> [defaults]
> codec_ms => 20
> wink_ms => 150
> flash_ms => 750
>
> Thanks and Best Regards,
> Jerry
>
>
>  ------------------------------
> *From:* Michael Collins [mailto:msc at freeswitch.org]
> *Sent:* Tuesday, January 05, 2010 4:33 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
>
>
>
> On Mon, Jan 4, 2010 at 3:49 PM, Jerry Richards <jerry.richards at teotech.com
> > wrote:
>
>>
>> Hello,
>>
>> I have one FS instance that is working well with a PRI and running FS
>> version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
>> upgraded it.
>>
>> Now, I am trying to bring up another FS instance (basically a clone of the
>> first), but the PRI does not work.  When I attempt to make an
>> internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
>> both servers are using the latest Sangoma Wanpipe driver, and I copied the
>> conf XML file tree from the old server to the new one.  I think the
>> problem
>> has to do with the openzap module, but I'm having difficulty isolating the
>> problem.  Could it have built the openzap module incorrectly?  Another
>> difference is that I installed 1.0.5pre9 from scratch on the new server
>> (i.e. it never had 1.0.4 running on it).
>>
>> I put the FS log into the pastebin when an outbound call attempt is made:
>>
>> http://pastebin.freeswitch.org/11675
>>
>> Could someone give me a pointer on what to try next?
>>
>
> Jerry, I noticed this line:
> (OpenZAP/1:1/3491028 at g1)
>
> Is your carrier wanting full ten digit phone numbers? Try adding the area
> code on this and see what happens. The error usually would be something like
> "invalid number format" but I've seen carriers do stupid things like this.
> Try that first and see if it makes a difference. If not you'll need to turn
> on Q931 debugging as per the Sangoma wiki. (See
> http://wiki.freeswitch.org/wiki/OpenZAP#Debugging_PRI_With_wanpipemon for
> the link.)
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Jan  6 11:58:08 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jan 2010 13:58:08 -0600
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <8441EC83BCD54223A5DAE91BEEC9C1FF@greyhawk.tonecommander.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>
	<8441EC83BCD54223A5DAE91BEEC9C1FF@greyhawk.tonecommander.com>
Message-ID: <7C331461-5692-47E7-B86F-81DD19904185@freeswitch.org>

You need to make sure you have SCTP libs installed and dev env to compile it.

/b

On Jan 6, 2010, at 1:52 PM, Jerry Richards wrote:

> I capured logs of my FS startup and put them into the pastebin (http://pastebin.freeswitch.org/11692).  At line 722, I see some errors:
>  
> 2010-01-06 11:42:49.907861 [ERR] zap_io.c:2562 Error loading /usr/local/freeswitch/mod/ozmod_sangoma_boost.so [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object file: No such file or directory]
> 2010-01-06 11:42:49.907883 [ERR] zap_io.c:2722 can't find 'sangoma_boost
> 2010-01-06 11:42:49.907902 [ERR] mod_openzap.c:2379 Error starting OpenZAP span 1 error
> 2010-01-06 11:42:49.907929 [CONSOLE] switch_loadable_module.c:890 Successfully Loaded [mod_openzap]
>  
> Do you know why I would get this?  Where is the ozmod_sangoma_boost.so file supposed to come from?
>  
> Thanks,
> Jerry
>  

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From Russell.Mosemann at cune.org  Wed Jan  6 12:04:26 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Wed, 6 Jan 2010 14:04:26 -0600
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <87f2f3b91001061158h4420961aj2878dc813d5637aa@mail.gmail.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com><87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com><BE608CAF54594CA48BB8B4DB82A8BFD6@greyhawk.tonecommander.com>
	<87f2f3b91001061158h4420961aj2878dc813d5637aa@mail.gmail.com>
Message-ID: <C2934F2F571143B5BD99457FB3C0475D@cune.pri>

> 	openzap.conf:
> 	[span wanpipe smg_prid]
> 	name => smg_prid
> 	trunk_type =>t1
> 	b-channel => 1:1-23

Do you miss a line when you copied the lines, or is a D channel not defined?

--
Russell Mosemann



From jmesquita at freeswitch.org  Wed Jan  6 14:16:44 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 6 Jan 2010 20:16:44 -0200
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <8ccbff061001060859p59aa6b5bw6f52275650f9138@mail.gmail.com>
References: <201001052125.06909.sos@sokhapkin.dyndns.org>
	<4B4417B3.9090807@aastral.net>
	<201001060620.13735.sos@sokhapkin.dyndns.org>
	<8ccbff061001060859p59aa6b5bw6f52275650f9138@mail.gmail.com>
Message-ID: <ebbc2d551001061416l25c17143gafe62de35cc6dee0@mail.gmail.com>

Why can't someone just sponsor some love to the module? The author has his
email on the header or open a Jira asking stuff so we have nibblebill more
mature.

Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927


On Wed, Jan 6, 2010 at 2:59 PM, Dome Charoenyost <dome at tel.co.th> wrote:

> 2010/1/6 Sergey Okhapkin <sos at sokhapkin.dyndns.org>:
> > nibblebill has no concept of billing blocks. What if I want to bill
> customer
> > 30 seconds minimum and 6 seconds increment thereafter?
>
> I have billing (in house develop) and customize  nibble_bill update
> cdr table (in my billing) i use postgresql trigger to update account
> balance. i have many increment rule 1/1 30/6 60/60
>
> It's work well for me :)
>
>
> BG
>
> Dome C.
> >
> > On Tuesday 05 January 2010, Bill W. wrote:
> >> Hey Sergey,
> >>
> >> But nibblebill will transfer to an extension of your choice when the
> >> balance reaches $0. So if you set the nibble heartbeat to 60 seconds or
> >> whatever, nibblebill will deduct the appropriate amount every
> >> <heartbeat> seconds.   So after about 20 minutes, the call will execute
> >> the nobal_action specified in nibblebill.conf.xml.
> >>
> >> So that should meet your needs.
> >>
> >> Bill W.
> >>
> >> Sergey Okhapkin wrote:
> >> > Unfortunalely, mod_nibblebill doesn't take billing increments into
> >> > account.
> >> >
> >> > On Tuesday 05 January 2010, Rupa Schomaker wrote:
> >> >> Look at using mod_nibblebill
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jerry.richards at teotech.com  Wed Jan  6 14:22:21 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 6 Jan 2010 14:22:21 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com> 
Message-ID: <4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>

By the way, I posted another pastebin log
(http://pastebin.freeswitch.org/11694) that shows the output of the "make"
and "make install sounds-install moh-install..." commands.  Just prior to
these makes I executed "make clean" and "make distclean".  You will notice
in the makefile output that ozmod_sangoma_boost.c never appears to get
compiled.  Shouldn't everything be compiled in this case?  I also confirmed
that my working server does have the ozmod_sangoma_boost.so file located in
the right place, which is why it's working.
 
Could version 1.0.5pre9 introduced this bug?  Could that be why the older
server (which original ran with 1.0.4) works and the new one doesn't?
 
Thanks and Best Regards,
Jerry
 
 


  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, January 06, 2010 11:52 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] DESTINATION_OUT_OF_ORDER


I capured logs of my FS startup and put them into the pastebin
(http://pastebin.freeswitch.org/11692).  At line 722, I see some errors:
 
2010-01-06 11:42:49.907861 [ERR] zap_io.c:2562 Error loading
/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
[/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
file: No such file or directory]
2010-01-06 11:42:49.907883 [ERR] zap_io.c:2722 can't find 'sangoma_boost
2010-01-06 11:42:49.907902 [ERR] mod_openzap.c:2379 Error starting OpenZAP
span 1 error
2010-01-06 11:42:49.907929 [CONSOLE] switch_loadable_module.c:890
Successfully Loaded [mod_openzap]
 
Do you know why I would get this?  Where is the ozmod_sangoma_boost.so file
supposed to come from?
 
Thanks,
Jerry
 
 


  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, January 06, 2010 10:00 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] DESTINATION_OUT_OF_ORDER


When I attempt an internal-to-PSTN call, there are no Q931 packets sent out
the PRI (I confirmed this using the Sangoma wanpipemon utility).  I suspect
this has something to do with my XML configuration.  Below are my
openzap/wanpipe configurations (which should all be defaulted).  Do you seen
anything wrong with these defaults, which might cause the following FS
console error?
 
zap_io.c:1197 outgoing_call method not implemented!
 
openzap.conf:
[span wanpipe smg_prid]
name => smg_prid
trunk_type =>t1
b-channel => 1:1-23
 
autoload_configs/openzap.conf.xml:
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="9"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
  <boost_spans>
    <span id="smg_prid">
      <!--<param name="hold-music" value="$${moh_uri}"/>-->
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <!-- regex to stop dialing when it matches -->
      <!--<param name="dial-regex" value="5555"/>-->
      <!-- regex to stop dialing when it does not match -->
      <!--<param name="fail-dial-regex" value="^5"/>-->
    </span>
  </boost_spans>
  <analog_spans>
  </analog_spans>
</configuration>
 
wanpipe.conf:
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
 
Thanks and Best Regards,
Jerry
 


  _____  

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Tuesday, January 05, 2010 4:33 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER




On Mon, Jan 4, 2010 at 3:49 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:



Hello,

I have one FS instance that is working well with a PRI and running FS
version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
upgraded it.

Now, I am trying to bring up another FS instance (basically a clone of the
first), but the PRI does not work.  When I attempt to make an
internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
both servers are using the latest Sangoma Wanpipe driver, and I copied the
conf XML file tree from the old server to the new one.  I think the problem
has to do with the openzap module, but I'm having difficulty isolating the
problem.  Could it have built the openzap module incorrectly?  Another
difference is that I installed 1.0.5pre9 from scratch on the new server
(i.e. it never had 1.0.4 running on it).

I put the FS log into the pastebin when an outbound call attempt is made:

http://pastebin.freeswitch.org/11675

Could someone give me a pointer on what to try next?



Jerry, I noticed this line:
(OpenZAP/1:1/3491028 at g1)

Is your carrier wanting full ten digit phone numbers? Try adding the area
code on this and see what happens. The error usually would be something like
"invalid number format" but I've seen carriers do stupid things like this.
Try that first and see if it makes a difference. If not you'll need to turn
on Q931 debugging as per the Sangoma wiki. (See
http://wiki.freeswitch.org/wiki/OpenZAP#Debugging_PRI_With_wanpipemon for
the link.)

-MC



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From brian at freeswitch.org  Wed Jan  6 14:32:37 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jan 2010 16:32:37 -0600
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>
	<4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
Message-ID: <407CE59D-5591-4838-B3D7-A0EC674844F7@freeswitch.org>

Again if you did not have the SCTP libs and dev headers installed when you did ./configure you'll never get this working SCTP is required to have boost work.  Also don't use Pre9 get the latest from latest.freeswitch.org and run with that please.

/b

On Jan 6, 2010, at 4:22 PM, Jerry Richards wrote:

> By the way, I posted another pastebin log (http://pastebin.freeswitch.org/11694) that shows the output of the "make" and "make install sounds-install moh-install..." commands.  Just prior to these makes I executed "make clean" and "make distclean".  You will notice in the makefile output that ozmod_sangoma_boost.c never appears to get compiled.  Shouldn't everything be compiled in this case?  I also confirmed that my working server does have the ozmod_sangoma_boost.so file located in the right place, which is why it's working.
>  
> Could version 1.0.5pre9 introduced this bug?  Could that be why the older server (which original ran with 1.0.4) works and the new one doesn't?
>  
> Thanks and Best Regards,
> Jerry
>  

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From brian at freeswitch.org  Wed Jan  6 14:33:34 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jan 2010 16:33:34 -0600
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <ebbc2d551001061416l25c17143gafe62de35cc6dee0@mail.gmail.com>
References: <201001052125.06909.sos@sokhapkin.dyndns.org>
	<4B4417B3.9090807@aastral.net>
	<201001060620.13735.sos@sokhapkin.dyndns.org>
	<8ccbff061001060859p59aa6b5bw6f52275650f9138@mail.gmail.com>
	<ebbc2d551001061416l25c17143gafe62de35cc6dee0@mail.gmail.com>
Message-ID: <5F37CA2E-129F-4082-B304-D50D0E1A4FAF@freeswitch.org>

Or better yet take over the maint. of the module if its making you money give back by providing some help to the project... its the ultimate way to give back.

/b

On Jan 6, 2010, at 4:16 PM, Jo?o Mesquita wrote:

> Why can't someone just sponsor some love to the module? The author has his email on the header or open a Jira asking stuff so we have nibblebill more mature.
> 
> Jo?o Mesquita
> FreeSWITCH? Solutions
> t: +1 (646) 4959927
> 



From jerry.richards at teotech.com  Wed Jan  6 15:37:48 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 6 Jan 2010 15:37:48 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com><87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>
	<4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
Message-ID: <536764FFF8C1495DB281DFABC9F02438@greyhawk.tonecommander.com>

Thank you for your suggestions.  Yes, I have the series of three
lksctp-tools-1.0.6 (SCTP) packages installed, and I do have the Development
Tools installed because the rest of the system is building okay.
 
Regarding the openzap.conf file, we have only 1 D-channel and 23 B-channels
and the openzap.conf file is auto-generated without a D-channel line, so I
don't think this is the issue.  Just to be sure, I did try adding a
D-channel line (d-channel => 1:1), but this produced another error (failed
to open wanpipe device span 1 channel 1) and the original error still
exists.
 
So I am trying to figure out why the openzap make is not compiling the
ozmod_sangoma_boost.c source file (http://pastebin.freeswitch.org/11694).
And I don't know why yet?
 
Thank You and Regards,
Jerry
 


  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, January 06, 2010 2:22 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER


By the way, I posted another pastebin log
(http://pastebin.freeswitch.org/11694) that shows the output of the "make"
and "make install sounds-install moh-install..." commands.  Just prior to
these makes I executed "make clean" and "make distclean".  You will notice
in the makefile output that ozmod_sangoma_boost.c never appears to get
compiled.  Shouldn't everything be compiled in this case?  I also confirmed
that my working server does have the ozmod_sangoma_boost.so file located in
the right place, which is why it's working.
 
Could version 1.0.5pre9 introduced this bug?  Could that be why the older
server (which original ran with 1.0.4) works and the new one doesn't?
 
Thanks and Best Regards,
Jerry
 
 


  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, January 06, 2010 11:52 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] DESTINATION_OUT_OF_ORDER


I capured logs of my FS startup and put them into the pastebin
(http://pastebin.freeswitch.org/11692).  At line 722, I see some errors:
 
2010-01-06 11:42:49.907861 [ERR] zap_io.c:2562 Error loading
/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
[/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
file: No such file or directory]
2010-01-06 11:42:49.907883 [ERR] zap_io.c:2722 can't find 'sangoma_boost
2010-01-06 11:42:49.907902 [ERR] mod_openzap.c:2379 Error starting OpenZAP
span 1 error
2010-01-06 11:42:49.907929 [CONSOLE] switch_loadable_module.c:890
Successfully Loaded [mod_openzap]
 
Do you know why I would get this?  Where is the ozmod_sangoma_boost.so file
supposed to come from?
 
Thanks,
Jerry
 
 


  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, January 06, 2010 10:00 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] DESTINATION_OUT_OF_ORDER


When I attempt an internal-to-PSTN call, there are no Q931 packets sent out
the PRI (I confirmed this using the Sangoma wanpipemon utility).  I suspect
this has something to do with my XML configuration.  Below are my
openzap/wanpipe configurations (which should all be defaulted).  Do you seen
anything wrong with these defaults, which might cause the following FS
console error?
 
zap_io.c:1197 outgoing_call method not implemented!
 
openzap.conf:
[span wanpipe smg_prid]
name => smg_prid
trunk_type =>t1
b-channel => 1:1-23
 
autoload_configs/openzap.conf.xml:
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="9"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
  <boost_spans>
    <span id="smg_prid">
      <!--<param name="hold-music" value="$${moh_uri}"/>-->
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <!-- regex to stop dialing when it matches -->
      <!--<param name="dial-regex" value="5555"/>-->
      <!-- regex to stop dialing when it does not match -->
      <!--<param name="fail-dial-regex" value="^5"/>-->
    </span>
  </boost_spans>
  <analog_spans>
  </analog_spans>
</configuration>
 
wanpipe.conf:
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
 
Thanks and Best Regards,
Jerry
 


  _____  

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Tuesday, January 05, 2010 4:33 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER




On Mon, Jan 4, 2010 at 3:49 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:



Hello,

I have one FS instance that is working well with a PRI and running FS
version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
upgraded it.

Now, I am trying to bring up another FS instance (basically a clone of the
first), but the PRI does not work.  When I attempt to make an
internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
both servers are using the latest Sangoma Wanpipe driver, and I copied the
conf XML file tree from the old server to the new one.  I think the problem
has to do with the openzap module, but I'm having difficulty isolating the
problem.  Could it have built the openzap module incorrectly?  Another
difference is that I installed 1.0.5pre9 from scratch on the new server
(i.e. it never had 1.0.4 running on it).

I put the FS log into the pastebin when an outbound call attempt is made:

http://pastebin.freeswitch.org/11675

Could someone give me a pointer on what to try next?



Jerry, I noticed this line:
(OpenZAP/1:1/3491028 at g1)

Is your carrier wanting full ten digit phone numbers? Try adding the area
code on this and see what happens. The error usually would be something like
"invalid number format" but I've seen carriers do stupid things like this.
Try that first and see if it makes a difference. If not you'll need to turn
on Q931 debugging as per the Sangoma wiki. (See
http://wiki.freeswitch.org/wiki/OpenZAP#Debugging_PRI_With_wanpipemon for
the link.)

-MC



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From anthony.minessale at gmail.com  Wed Jan  6 15:49:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Jan 2010 17:49:24 -0600
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <536764FFF8C1495DB281DFABC9F02438@greyhawk.tonecommander.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>
	<4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
	<536764FFF8C1495DB281DFABC9F02438@greyhawk.tonecommander.com>
Message-ID: <191c3a031001061549n2b67cbcax89d93a89fc62b55e@mail.gmail.com>

ya, you should not use make distclean with any of our code, we do not have
it properly implemented since we have a very large and challenging build
system/


On Wed, Jan 6, 2010 at 5:37 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  Thank you for your suggestions.  Yes, I have the series of three
> lksctp-tools-1.0.6 (SCTP) packages installed, and I do have the
> Development Tools installed because the rest of the system is building
> okay.
>
> Regarding the openzap.conf file, we have only 1 D-channel and 23 B-channels
> and the openzap.conf file is auto-generated without a D-channel line, so I
> don't think this is the issue.  Just to be sure, I did try adding a
> D-channel line (d-channel => 1:1), but this produced another error (failed
> to open wanpipe device span 1 channel 1) and the original error still
> exists.
>
> So I am trying to figure out why the openzap make is not compiling the
> ozmod_sangoma_boost.c source file (http://pastebin.freeswitch.org/11694).
> And I don't know why yet?
>
> Thank You and Regards,
> Jerry
>
>
>  ------------------------------
> *From:* Jerry Richards [mailto:jerry.richards at teotech.com]
> *Sent:* Wednesday, January 06, 2010 2:22 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
>
>  By the way, I posted another pastebin log (
> http://pastebin.freeswitch.org/11694) that shows the output of the "make"
> and "make install sounds-install moh-install..." commands.  Just prior to
> these makes I executed "make clean" and "make distclean".  You will notice
> in the makefile output that ozmod_sangoma_boost.c never appears to get
> compiled.  Shouldn't everything be compiled in this case?  I also confirmed
> that my working server does have the ozmod_sangoma_boost.so file located in
> the right place, which is why it's working.
>
> Could version 1.0.5pre9 introduced this bug?  Could that be why the older
> server (which original ran with 1.0.4) works and the new one doesn't?
>
> Thanks and Best Regards,
> Jerry
>
>
>
>  ------------------------------
> *From:* Jerry Richards [mailto:jerry.richards at teotech.com]
> *Sent:* Wednesday, January 06, 2010 11:52 AM
> *To:* 'freeswitch-users at lists.freeswitch.org'
> *Subject:* RE: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
>
>  I capured logs of my FS startup and put them into the pastebin (
> http://pastebin.freeswitch.org/11692).  At line 722, I see some errors:
>
> 2010-01-06 11:42:49.907861 [ERR] zap_io.c:2562 Error loading
> /usr/local/freeswitch/mod/ozmod_sangoma_boost.so [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so:
> cannot open shared object file: No such file or directory]
> 2010-01-06 11:42:49.907883 [ERR] zap_io.c:2722 can't find 'sangoma_boost
> 2010-01-06 11:42:49.907902 [ERR] mod_openzap.c:2379 Error starting OpenZAP
> span 1 error
> 2010-01-06 11:42:49.907929 [CONSOLE] switch_loadable_module.c:890Successfully Loaded
> [mod_openzap]
>
> Do you know why I would get this?  Where is the ozmod_sangoma_boost.so file
> supposed to come from?
>
> Thanks,
> Jerry
>
>
>
>  ------------------------------
> *From:* Jerry Richards [mailto:jerry.richards at teotech.com]
> *Sent:* Wednesday, January 06, 2010 10:00 AM
> *To:* 'freeswitch-users at lists.freeswitch.org'
> *Subject:* RE: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
>
>  When I attempt an internal-to-PSTN call, there are no Q931 packets sent
> out the PRI (I confirmed this using the Sangoma wanpipemon utility).  I
> suspect this has something to do with my XML configuration.  Below are my
> openzap/wanpipe configurations (which should all be defaulted).  Do you
> seen anything wrong with these defaults, which might cause the following FS
> console error?
>
> zap_io.c:1197 outgoing_call method not implemented!
>
> *openzap.conf:
> *[span wanpipe smg_prid]
> name => smg_prid
> trunk_type =>t1
> b-channel => 1:1-23
>
> *autoload_configs/openzap.conf.xml:*
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>   <settings>
>     <param name="debug" value="9"/>
>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>     <!--<param name="enable-analog-option" value="3-way"/>-->
>   </settings>
>   <boost_spans>
>     <span id="smg_prid">
>       <!--<param name="hold-music" value="$${moh_uri}"/>-->
>       <param name="dialplan" value="XML"/>
>       <param name="context" value="default"/>
>       <!-- regex to stop dialing when it matches -->
>       <!--<param name="dial-regex" value="5555"/>-->
>       <!-- regex to stop dialing when it does not match -->
>       <!--<param name="fail-dial-regex" value="^5"/>-->
>     </span>
>   </boost_spans>
>   <analog_spans>
>   </analog_spans>
> </configuration>
>
> *wanpipe.conf:*
> [defaults]
> codec_ms => 20
> wink_ms => 150
> flash_ms => 750
>
> Thanks and Best Regards,
> Jerry
>
>
>  ------------------------------
> *From:* Michael Collins [mailto:msc at freeswitch.org]
> *Sent:* Tuesday, January 05, 2010 4:33 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
>
>
>
> On Mon, Jan 4, 2010 at 3:49 PM, Jerry Richards <jerry.richards at teotech.com
> > wrote:
>
>>
>> Hello,
>>
>> I have one FS instance that is working well with a PRI and running FS
>> version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
>> upgraded it.
>>
>> Now, I am trying to bring up another FS instance (basically a clone of the
>> first), but the PRI does not work.  When I attempt to make an
>> internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
>> both servers are using the latest Sangoma Wanpipe driver, and I copied the
>> conf XML file tree from the old server to the new one.  I think the
>> problem
>> has to do with the openzap module, but I'm having difficulty isolating the
>> problem.  Could it have built the openzap module incorrectly?  Another
>> difference is that I installed 1.0.5pre9 from scratch on the new server
>> (i.e. it never had 1.0.4 running on it).
>>
>> I put the FS log into the pastebin when an outbound call attempt is made:
>>
>> http://pastebin.freeswitch.org/11675
>>
>> Could someone give me a pointer on what to try next?
>>
>
> Jerry, I noticed this line:
> (OpenZAP/1:1/3491028 at g1)
>
> Is your carrier wanting full ten digit phone numbers? Try adding the area
> code on this and see what happens. The error usually would be something like
> "invalid number format" but I've seen carriers do stupid things like this.
> Try that first and see if it makes a difference. If not you'll need to turn
> on Q931 debugging as per the Sangoma wiki. (See
> http://wiki.freeswitch.org/wiki/OpenZAP#Debugging_PRI_With_wanpipemon for
> the link.)
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mcampbellsmith at gmail.com  Wed Jan  6 16:43:42 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 7 Jan 2010 11:43:42 +1100
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <965454EF-F779-4563-AC64-082CAB07CCA5@freeswitch.org>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
	<191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
	<33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
	<0A8CD6A7-CA51-44F2-B530-15AE623EBA5D@freeswitch.org>
	<637054.57565.qm@web111313.mail.gq1.yahoo.com>
	<965454EF-F779-4563-AC64-082CAB07CCA5@freeswitch.org>
Message-ID: <33c87fa31001061643p4829e176leb294bec11e8b6c9@mail.gmail.com>

Thanks guys for your response.  I'll have a read through the links sent to me.

On the INVITE my FS box sends to Phonzo (the SIP provider) the codecs
it supports are G726-32, PCMU, PCMA.

 v=0
 o=FreeSWITCH 1262742568 1262742569 IN IP4 xxx.xxx.xxx.xx
 s=FreeSWITCH
 c=IN IP4 xxx.xxx.xxx.xxx
 t=0 0
 m=audio 31566 RTP/AVP 2 0 8 101 13
 a=rtpmap:2 G726-32/8000
 a=rtpmap:0 PCMU/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=rtpmap:13 CN/8000
 a=ptime:20

However, Phonzo responds with the following in the Session Progress message:

  v=0
  o=Sippy 158698636 1 IN IP4 80.232.37.178
  s=-
  t=0 0
  m=audio 61812 RTP/AVP 2 101 13
  c=IN IP4 213.50.91.3
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-15

Does that make sense?   The codec rtpmap only included DTMF ...

On Thu, Jan 7, 2010 at 2:04 AM, Brian West <brian at freeswitch.org> wrote:
> w00t! :)
> /b
> On Jan 6, 2010, at 1:29 AM, Arsen Chaloyan wrote:
>
> The following section in RFC3551 states the same
> http://tools.ietf.org/html/rfc3551#section-4.5.4
>
>
>    The encodings of G.726 are here denoted as G726-40, G726-32, G726-24,
>    and G726-16.  Prior to 1990, G721 described the 32 kbit/s ADPCM
>    encoding, and G723 described the 40, 32, and 16 kbit/s encodings.
>    Thus, G726-32 designates the same algorithm as G721 in RFC 1890.
>
>
> ________________________________
> From:?Brian West <brian at freeswitch.org>
> To:?freeswitch-users at lists.freeswitch.org
> Sent:?Wed, January 6, 2010 9:50:46 AM
> Subject:?Re: [Freeswitch-users] Codec Negotiation: Codec
> telephone-event:101:8000:20
>
> They are in fact one in the same please see ITU.
>
> /b
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Wed Jan  6 16:57:48 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jan 2010 18:57:48 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001061643p4829e176leb294bec11e8b6c9@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
	<191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
	<33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
	<0A8CD6A7-CA51-44F2-B530-15AE623EBA5D@freeswitch.org>
	<637054.57565.qm@web111313.mail.gq1.yahoo.com>
	<965454EF-F779-4563-AC64-082CAB07CCA5@freeswitch.org>
	<33c87fa31001061643p4829e176leb294bec11e8b6c9@mail.gmail.com>
Message-ID: <4DDDB005-6482-4495-881E-7242DB157082@freeswitch.org>

NO the codec map includes 2 aka g726-32 which you don't have to list in the map if its defined in the standard... what I need is a pcap of the whole process.

/b

On Jan 6, 2010, at 6:43 PM, Mark Campbell-Smith wrote:

> Thanks guys for your response.  I'll have a read through the links sent to me.
> 
> On the INVITE my FS box sends to Phonzo (the SIP provider) the codecs
> it supports are G726-32, PCMU, PCMA.
> 
> v=0
> o=FreeSWITCH 1262742568 1262742569 IN IP4 xxx.xxx.xxx.xx
> s=FreeSWITCH
> c=IN IP4 xxx.xxx.xxx.xxx
> t=0 0
> m=audio 31566 RTP/AVP 2 0 8 101 13
> a=rtpmap:2 G726-32/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
> 
> However, Phonzo responds with the following in the Session Progress message:
> 
>  v=0
>  o=Sippy 158698636 1 IN IP4 80.232.37.178
>  s=-
>  t=0 0
>  m=audio 61812 RTP/AVP 2 101 13
>  c=IN IP4 213.50.91.3
>  a=rtpmap:101 telephone-event/8000
>  a=fmtp:101 0-15
> 
> Does that make sense?   The codec rtpmap only included DTMF ...



From anthony.minessale at gmail.com  Wed Jan  6 17:51:12 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Jan 2010 19:51:12 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <4DDDB005-6482-4495-881E-7242DB157082@freeswitch.org>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
	<191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
	<33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
	<0A8CD6A7-CA51-44F2-B530-15AE623EBA5D@freeswitch.org>
	<637054.57565.qm@web111313.mail.gq1.yahoo.com>
	<965454EF-F779-4563-AC64-082CAB07CCA5@freeswitch.org>
	<33c87fa31001061643p4829e176leb294bec11e8b6c9@mail.gmail.com>
	<4DDDB005-6482-4495-881E-7242DB157082@freeswitch.org>
Message-ID: <191c3a031001061751w445324ebqbb34fd4b0fba08cf@mail.gmail.com>

when they only return 1, that means that is what they want you to use.
The call should establish in that case with g726-32


On Wed, Jan 6, 2010 at 6:57 PM, Brian West <brian at freeswitch.org> wrote:

> NO the codec map includes 2 aka g726-32 which you don't have to list in the
> map if its defined in the standard... what I need is a pcap of the whole
> process.
>
> /b
>
> On Jan 6, 2010, at 6:43 PM, Mark Campbell-Smith wrote:
>
> > Thanks guys for your response.  I'll have a read through the links sent
> to me.
> >
> > On the INVITE my FS box sends to Phonzo (the SIP provider) the codecs
> > it supports are G726-32, PCMU, PCMA.
> >
> > v=0
> > o=FreeSWITCH 1262742568 1262742569 IN IP4 xxx.xxx.xxx.xx
> > s=FreeSWITCH
> > c=IN IP4 xxx.xxx.xxx.xxx
> > t=0 0
> > m=audio 31566 RTP/AVP 2 0 8 101 13
> > a=rtpmap:2 G726-32/8000
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=rtpmap:13 CN/8000
> > a=ptime:20
> >
> > However, Phonzo responds with the following in the Session Progress
> message:
> >
> >  v=0
> >  o=Sippy 158698636 1 IN IP4 80.232.37.178
> >  s=-
> >  t=0 0
> >  m=audio 61812 RTP/AVP 2 101 13
> >  c=IN IP4 213.50.91.3
> >  a=rtpmap:101 telephone-event/8000
> >  a=fmtp:101 0-15
> >
> > Does that make sense?   The codec rtpmap only included DTMF ...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From Russell.Mosemann at cune.org  Wed Jan  6 18:16:33 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Wed, 6 Jan 2010 20:16:33 -0600
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <536764FFF8C1495DB281DFABC9F02438@greyhawk.tonecommander.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com><87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com><4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
	<536764FFF8C1495DB281DFABC9F02438@greyhawk.tonecommander.com>
Message-ID: <F677BB8D0A424932889541C56C96C78A@cune.pri>

Jerry Richards wrote:

> Regarding the openzap.conf file, we have only 1 D-channel and 23 B-
> channels and the openzap.conf file is auto-generated without a D-channel
> line, so I don't think this is the issue.  Just to be sure, I did try
> adding a D-channel line (d-channel => 1:1), but this produced another
> error (failed to open wanpipe device span 1 channel 1) and the original
> error still exists.

Channel 1 is already assigned to a B channel.

> b-channel => 1:1-23

You can't also assign it to a D channel. You can assign it to channel 24.

d-channel => 1:24

That's the same example that you will find on the wiki under "Wanpipe mode".

http://wiki.freeswitch.org/wiki/OpenZAP

--
Russell Mosemann



From brian at freeswitch.org  Wed Jan  6 18:25:53 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jan 2010 20:25:53 -0600
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <F677BB8D0A424932889541C56C96C78A@cune.pri>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com><87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com><4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
	<536764FFF8C1495DB281DFABC9F02438@greyhawk.tonecommander.com>
	<F677BB8D0A424932889541C56C96C78A@cune.pri>
Message-ID: <33827AB6-8B14-4F07-9C3E-36E155939ED2@freeswitch.org>

You don't define any d-channels when using boost.

/b

On Jan 6, 2010, at 8:16 PM, Russell Mosemann wrote:

> You can't also assign it to a D channel. You can assign it to channel 24.
> 
> d-channel => 1:24



From Russell.Mosemann at cune.org  Wed Jan  6 18:30:18 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Wed, 6 Jan 2010 20:30:18 -0600
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <33827AB6-8B14-4F07-9C3E-36E155939ED2@freeswitch.org>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com><87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com><4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com><536764FFF8C1495DB281DFABC9F02438@greyhawk.tonecommander.com><F677BB8D0A424932889541C56C96C78A@cune.pri>
	<33827AB6-8B14-4F07-9C3E-36E155939ED2@freeswitch.org>
Message-ID: <F5A693D296B54119AB9F51B5C05BEB21@cune.pri>

> You don't define any d-channels when using boost.

Thanks for the clarification.

--
Russell Mosemann



From darklion11 at yahoo.com  Wed Jan  6 21:04:31 2010
From: darklion11 at yahoo.com (Edmar Cruz)
Date: Wed, 6 Jan 2010 21:04:31 -0800 (PST)
Subject: [Freeswitch-users]  Script for Presence using PHP
Message-ID: <27050100.post@talk.nabble.com>


Dear Sir,

       Is there a script to trigger whether a user is online or offline
using a script or a PHP code?

       Presence for freeswitch is for command line only?


       Can you give me an example?

Thanks,
Edmar
-- 
View this message in context: http://old.nabble.com/Script-for-Presence-using-PHP-tp27050100p27050100.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From brian at freeswitch.org  Wed Jan  6 21:18:17 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jan 2010 23:18:17 -0600
Subject: [Freeswitch-users] Script for Presence using PHP
In-Reply-To: <27050100.post@talk.nabble.com>
References: <27050100.post@talk.nabble.com>
Message-ID: <A15C41BC-3548-4A2E-9BB6-FC79EB7A15C6@freeswitch.org>

see libs/esl/perl/*.pl should be similar to php in esl.

/b

On Jan 6, 2010, at 11:04 PM, Edmar Cruz wrote:

> 
> Dear Sir,
> 
>       Is there a script to trigger whether a user is online or offline
> using a script or a PHP code?
> 
>       Presence for freeswitch is for command line only?
> 
> 
>       Can you give me an example?
> 
> Thanks,
> Edmar



From msc at freeswitch.org  Wed Jan  6 22:59:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Jan 2010 22:59:10 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <F5A693D296B54119AB9F51B5C05BEB21@cune.pri>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>
	<4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
	<536764FFF8C1495DB281DFABC9F02438@greyhawk.tonecommander.com>
	<F677BB8D0A424932889541C56C96C78A@cune.pri>
	<33827AB6-8B14-4F07-9C3E-36E155939ED2@freeswitch.org>
	<F5A693D296B54119AB9F51B5C05BEB21@cune.pri>
Message-ID: <87f2f3b91001062259j7edbf029tacb7cdfa9156350@mail.gmail.com>

On Wed, Jan 6, 2010 at 6:30 PM, Russell Mosemann
<Russell.Mosemann at cune.org>wrote:

> > You don't define any d-channels when using boost.
>
> Thanks for the clarification.
>
FYI, I started the boost section under Sangoma on the wiki but got
distracted before finishing. I will get on that ASAP as soon as I have a
minute to get a boost setup all configured and running. Also, a tip o' the
hat to Moy at Sangoma who has been extremely responsive to my boost
questions in the past.

Thanks,
MC
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From msc at freeswitch.org  Wed Jan  6 23:03:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Jan 2010 23:03:16 -0800
Subject: [Freeswitch-users] encrypt voicemail emails and attachments?
In-Reply-To: <20100105171137.A433E1DB501@sinclaire.sibble.net>
References: <20100105042327.7CBF412DD@sinclaire.sibble.net>
	<03DDC308-7EC5-46EC-8AFB-CC73417E20F5@jerris.com>
	<20100105171137.A433E1DB501@sinclaire.sibble.net>
Message-ID: <87f2f3b91001062303n41871ff4x8c220d0564b27a50@mail.gmail.com>

On Tue, Jan 5, 2010 at 9:11 AM, Harondel J. Sibble <help at pdscc.com> wrote:

> That's what I suspected, thanks!
>

Let us know what you do so we can document it on the wiki.
Thanks,
MC
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From ken at ukgb.net  Thu Jan  7 01:17:21 2010
From: ken at ukgb.net (Ken Gillett)
Date: Thu, 7 Jan 2010 09:17:21 +0000
Subject: [Freeswitch-users] video
In-Reply-To: <90AC63C4-48ED-43F7-A534-CB90C208604F@ukgb.net>
References: <90AC63C4-48ED-43F7-A534-CB90C208604F@ukgb.net>
Message-ID: <9BFCBAD9-742A-41CD-8AB2-18E93B4A2A14@ukgb.net>

I'm still rather in the dark about what might loosely be termed 'SIP Video Phone' and in particular how it can relate to FreeSwitch.

	How does a video call work?
	Is it really a standard that governs this?
	What happens if the destination has no display?
	Does the originating camera only start streaming video when the call is started?
	Is the above hardware dependent?
	What would be a good video softphone for the Mac?

I have no prior knowledge of the working of such a video phone, but am trying to gain a better understanding. At least I want to know in what direction to research further, so hope someone can offer some basic ideas here.



Ken G i l l e t t

_/_/_/_/_/_/_/_/





From devel at thom.fr.eu.org  Thu Jan  7 01:55:17 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Thu, 07 Jan 2010 10:55:17 +0100
Subject: [Freeswitch-users] Zap channel not released when voicemail
	starts
Message-ID: <045f8df05054638c4a8e62e87026c060@thom.fr.eu.org>

 

Thanks. This works perfectly. 

Fran?ois 

On Tue, 5 Jan 2010 10:11:05
-0600, Anthony Minessale wrote:  

one way is to run tone_detect on the
busy signal and map it to the hangup app

 On Tue, Jan 5, 2010 at 3:44 AM,
Ahmed Naji  wrote:
 I'll second that.

My way of dealing with it has been
to write a little script to detect hangups on the TDM end, then force
release the corresponding "B-leg" that is hooked up to VM. In the process
of converting this to an FS module.

Not clean .. but works. Would have
liked to see the same code within FS core and, if appropriate, the VM
subsystem to achieve the same end.

Regards,

Ahmed.

 2010/1/5 Anthony
Minessale   
 hangup detection on TDM is a bitch.

 On Mon, Jan 4, 2010 at
10:07 AM, Fran?ois Legal  wrote:

Hello, 

I have an issue with voicemail
and openzap channels. 

When an incoming call on an openzap channel is
bridged to voicemail, if that channel is hung up before the beginning of
voicemail recording, that channel is kept open open until 3 or 4 seconds
after the voicemail started to record the message. 

What should I do to
make freeswitch/voicemail release the channel immediately when the caller
hang up ? 

Thanks in advance 

Fran?ois

_______________________________________________
 FreeSWITCH-users mailing
list
FreeSWITCH-users at lists.freeswitch.org
[4]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [5]

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[6]
http://www.freeswitch.org [7]

-- 
Anthony Minessale II

FreeSWITCH
http://www.freeswitch.org/ [8]
ClueCon http://www.cluecon.com/ [9]

Twitter: http://twitter.com/FreeSWITCH_wire [10]

AIM:
anthm
MSN:anthony_minessale at hotmail.com [11]

GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [12]
 IRC: irc.freenode.net
[13] #freeswitch

FreeSWITCH Developer
Conference
sip:888 at conference.freeswitch.org
[14]
iax:guest at conference.freeswitch.org/888
[15]
googletalk:conf+888 at conference.freeswitch.org
[16]
pstn:+19193869900

_______________________________________________

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
[17]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [18]

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[19]
http://www.freeswitch.org [20]

-- 
Ahmed
Naji

_______________________________________________
 FreeSWITCH-users
mailing list
FreeSWITCH-users at lists.freeswitch.org
[21]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [22]

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[23]
http://www.freeswitch.org [24]

-- 
Anthony Minessale II

FreeSWITCH
http://www.freeswitch.org/ [25]
ClueCon http://www.cluecon.com/ [26]

Twitter: http://twitter.com/FreeSWITCH_wire [27]

AIM:
anthm
MSN:anthony_minessale at hotmail.com
[28]
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [29]
 IRC:
irc.freenode.net [30] #freeswitch

FreeSWITCH Developer
Conference
sip:888 at conference.freeswitch.org
[31]
iax:guest at conference.freeswitch.org/888
[32]
googletalk:conf+888 at conference.freeswitch.org [33]
pstn:+19193869900




Links:
------
[1] mailto:a.alalousi at gmail.com
[2]
mailto:anthony.minessale at gmail.com
[3] mailto:devel at thom.fr.eu.org
[4]
mailto:FreeSWITCH-users at lists.freeswitch.org
[5]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[6]
http://lists.freeswitch.org/mailman/options/freeswitch-users
[7]
http://www.freeswitch.org
[8] http://www.freeswitch.org/
[9]
http://www.cluecon.com/
[10] http://twitter.com/FreeSWITCH_wire
[11]
mailto:MSN%3Aanthony_minessale at hotmail.com
[12]
mailto:PAYPAL%3Aanthony.minessale at gmail.com
[13]
http://irc.freenode.net
[14]
mailto:sip%3A888 at conference.freeswitch.org
[15]
http://iax:guest at conference.freeswitch.org/888
[16]
mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
[17]
mailto:FreeSWITCH-users at lists.freeswitch.org
[18]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[19]
http://lists.freeswitch.org/mailman/options/freeswitch-users
[20]
http://www.freeswitch.org
[21]
mailto:FreeSWITCH-users at lists.freeswitch.org
[22]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[23]
http://lists.freeswitch.org/mailman/options/freeswitch-users
[24]
http://www.freeswitch.org
[25] http://www.freeswitch.org/
[26]
http://www.cluecon.com/
[27] http://twitter.com/FreeSWITCH_wire
[28]
mailto:MSN%3Aanthony_minessale at hotmail.com
[29]
mailto:PAYPAL%3Aanthony.minessale at gmail.com
[30]
http://irc.freenode.net
[31]
mailto:sip%3A888 at conference.freeswitch.org
[32]
http://iax:guest at conference.freeswitch.org/888
[33]
mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
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From tzury.by at reguluslabs.com  Thu Jan  7 02:18:07 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Thu, 7 Jan 2010 12:18:07 +0200
Subject: [Freeswitch-users] a weired problem when calling pstn number(s)
	with openzap and libpri
Message-ID: <10128ef11001070218g1367dc9bvcb5576c9d1a1b4dd@mail.gmail.com>

Hi all,

I got a weired problem and I suspect it has to do with either openzap or
libpri.

My setup is simple FS box, and a Sangoma E1 (A101) card.

Everything seems to be well configured and working. Yet, sometimes, when I
call from the SIP the call failed, on the client I get "call ended" message.

On the server when looking at the log I see UNALLOCATED_NUMBER (line:274)

I can certainly tell that this number exists and alive and it is not a wrong
number.

to make it even more complicated, calling that problematic number over and
over, yields a statistics of working rate 5/1
that is every 5th or 4th call works

I placed here the logs of the working case and the not working case

http://gist.github.com/raw/271113/b8f13ef292669a1ab69878bfe205fc41d3722d0a/libpri-openzap-bridge-outbound-calls-working.cs<http://gist.github.com/raw/271113/b8f13ef292669a1ab69878bfe205fc41d3722d0a/libpri-openzap-bridge-outbound-calls-working.log>
http://gist.github.com/raw/271113/8b8b55922ebd8faffe1a850681f6a642dbdf4f04/libpri-openzap-bridge-outbound-calls-not-working.cs


please help,
Tzury
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From saeedahmad1981 at gmail.com  Thu Jan  7 02:35:31 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Thu, 7 Jan 2010 11:35:31 +0100
Subject: [Freeswitch-users] freeswitch and H323
In-Reply-To: <65d96fc81001051125u3a6c7d96q5ed39d39c0026107@mail.gmail.com>
References: <7aa8bd9d0912300514r68c90b12u7c631a649981cfa3@mail.gmail.com> 
	<87f2f3b90912310920q1821fe0eu5fd74e553e4cb12@mail.gmail.com> 
	<65d96fc81001010452k37068e87sc0a229cdfe481c40@mail.gmail.com> 
	<c96b95651001031636q1ecfec18x2bba1c22ff6ca9bb@mail.gmail.com> 
	<65d96fc81001051125u3a6c7d96q5ed39d39c0026107@mail.gmail.com>
Message-ID: <c96b95651001070235j2fe1a4b4m26da30aeb1861bb4@mail.gmail.com>

Thanks

On Tue, Jan 5, 2010 at 8:25 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> its already there: http://wiki.freeswitch.org/wiki/Mod_h323
>
> T.
>
>
> On Mon, Jan 4, 2010 at 1:36 AM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:
>
>> HI,
>>
>> It would be really nice if you can create a wiki page.
>>
>> Thanks
>>
>>
>> On Fri, Jan 1, 2010 at 1:52 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> well, mod_h323 works for me... there are still some missing things and of
>>> course bugs ... e.g. incorrect releaseCause mapping, no automatic codec
>>> ptime sync... but it is usable ....
>>>
>>>
>>> if you'd like to go mod_h323 way i can help you... it builds as a charm
>>> for me...
>>>
>>>
>>> T.
>>>
>>>
>>>
>>>
>>>
>>> On Thu, Dec 31, 2009 at 6:20 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Are you trying to use mod_h323 or mod_opal? They are both works in
>>>> progress, but the latter is farther along than the former. Use the latest
>>>> FreeSWITCH trunk (or latest.freeswitch.org) and run the buildopal.sh
>>>> script in the build directory. If you have any build issues then paste the
>>>> log on pastebin.freeswitch.org and reply to this thread with the PB URL
>>>> so that we can take a look.
>>>>  -MC
>>>>
>>>>
>>>> On Wed, Dec 30, 2009 at 5:14 AM, Pete Kay <petedao at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> has anyone been able to get H323 to work?
>>>>>
>>>>> I have problem trying to get it compiled with either 1.0.4 or 1.0.5.
>>>>>
>>>>> Thanks,
>>>>> pete
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From saeedahmad1981 at gmail.com  Thu Jan  7 03:00:50 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Thu, 7 Jan 2010 12:00:50 +0100
Subject: [Freeswitch-users] Installing freeswitch on CentOS
Message-ID: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>

Hi,

Since CentOS is recommend for FS but i can't see a CentOS specific
installation guide on wiki as we have a separate guide for Ubuntu.

Do we have similar guide like this one
http://wiki.freeswitch.org/wiki/SBC_Setup* *actually its for debian but good
thing is that it also explains which extra services should be stopped or
removed for better performance.

Do we have similar for CentOS?

Thanks
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From jcasale at activenetwerx.com  Thu Jan  7 04:02:56 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Thu, 7 Jan 2010 12:02:56 +0000
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>

>Since CentOS is recommend for FS but i can't see a CentOS specific installation guide on wiki as we have a separate >guide for Ubuntu.?
>
>Do we have similar guide like this one?http://wiki.freeswitch.org/wiki/SBC_Setup?actually its for debian but good thing >is that it also explains which extra >services should be stopped or removed for better performance.?
>
>Do we have similar for CentOS?

Check out the http://wiki.freeswitch.org/wiki/Installation_Guide




From freeswitch at peely.com  Thu Jan  7 06:56:00 2010
From: freeswitch at peely.com (peely)
Date: Thu, 7 Jan 2010 06:56:00 -0800 (PST)
Subject: [Freeswitch-users] Call through gateway without register > sends to
 gateway name?
Message-ID: <26979541.post@talk.nabble.com>


Hi,

I have a problem where I'm trying to send calls to a gateway that does not
support registration.

In my external sip profile directory I have a file containing:

<include>
    <gateway name="mygateway">
      
      
      
      
      
    </gateway>
</include>

Then in my dialplan I have a dialplan with an action of:

<action application="bridge" data="sofia/gateway/mygateway/1234567890"/>

However, when I call out, the Sofia diag shows:

sres_send_dns_query(0x7f19ac011150, 0x7f19a401cdf0) id=20898 NAPTR mygateway
(to [172.16.1.1]:53)

Could somebody please tell me how I get the gateway config to send INVITEs
to a specific IP?



Thanks,


Neil.

-- 
View this message in context: http://old.nabble.com/Call-through-gateway-without-register-%3E-sends-to-gateway-name--tp26979541p26979541.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From anthony.minessale at gmail.com  Thu Jan  7 07:14:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Jan 2010 09:14:50 -0600
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
Message-ID: <191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>

the big rule for the time being is stick with 5.3 5.4 appears to have some
bugs in the toolchain and libc

On Thu, Jan 7, 2010 at 6:02 AM, Joseph L. Casale
<jcasale at activenetwerx.com>wrote:

> >Since CentOS is recommend for FS but i can't see a CentOS specific
> installation guide on wiki as we have a separate >guide for Ubuntu.
> >
> >Do we have similar guide like this one
> http://wiki.freeswitch.org/wiki/SBC_Setup actually its for debian but good
> thing >is that it also explains which extra >services should be stopped or
> removed for better performance.
> >
> >Do we have similar for CentOS?
>
> Check out the http://wiki.freeswitch.org/wiki/Installation_Guide
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From nicolas at medularis.com  Thu Jan  7 07:43:11 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Thu, 7 Jan 2010 12:43:11 -0300
Subject: [Freeswitch-users] Calls getting queued?
Message-ID: <1b46b4e81001070743v5389ef3ewb75dbeb06b402c9a@mail.gmail.com>

Hi, I'm having a strange problem with FS. I'm using a few JS scripts
to generate calls and bridge them together. Usually everything works
just fine, but them at some point it's like if FS choked, calls for
the first leg of the bridges are apparently made, but the second leg
is never called. The call is not hanged up for several minutes and the
system keeps opening new channels but never connecting a call.

For example, right now, doing 'show channels' on the console, I get a
list of 72 open channels (it's adding up, it was 40 a couple minutes
ago), but doing a 'show calls' gives me 0 active calls. The usual
behavior, when everything's working fine, is to get twice as many
channels as there are active calls and no channels at all when there
are no calls, unless they haven't been bridged yet.

The originate string is something like this:

var stUsRing = "%(2000,4000,440,480)";
var timeout = 45;
originate_str1 = "{api_hangup_hook=jsapi::callback.js
l1,execute_on_answer=lua answered.lua 1
c2c_call,ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,ringback='"+stUsRing+"',medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]"+dialstr1;

Where diasltr1 has the phonenumber and and gateway info. The
callback.js has a curl request to update some call info on an external
database and answered.lua calls a ruby script through the os.execute()
function (I know, I should be doing all this through the event socket,
I was doing that but had trouble and had to come up with a quick
solution).

The system is not loaded at all, at least not for what I think and
read that FS can handle. We are having at most 10 concurrent calls (20
channels), with maybe 5 to 10 calls per minute.

What worries me is not only that I don't know where the problem is,
but that I have no clue how to debug it or send you guys more
"lowlevel" and detailed information to give you an insight about
what's going on. Any help would be greatly appreciated!

Thanks!

Nico


From dujinfang at gmail.com  Thu Jan  7 08:11:21 2010
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 8 Jan 2010 00:11:21 +0800
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org>
	<2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org>
	<B732197E-1123-4738-A69F-F84206FCE38A@freeswitch.org>
	<0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org>
	<0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org>
	<23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com>
	<60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org>
	<23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com>
	<ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com>
Message-ID: <23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com>

:lol. I do like to involve into this. I saw you have done a lot of
works. I read some code and here are some questions:

1) what's your nick on IRC? I'm seven(or seven_ ?)
2) Are you developing on Windows? How can I compile on Mac(I have no
experience on QT)?
3) Does is needs fsGUI? I downloaded fsGUI dmg file and it complains
no mods on start.


2009/12/30, Jo?o Mesquita <jmesquita at freeswitch.org>:
> What is JM is not the question but rather WHO is JM and that would be me.
> :-)
>
> I have already stripped down the config handler based on mod_xml_curl. I
> have been discussing with Brian how to make it happen and I am conducting a
> couple of tests with Qt. Today I might be able to have it properly linked
> with Qt and the core spawn on its own thread inside the Qt event loop. I'll
> keep you posted.
>
> Jo?o Mesquita A.K.A -> JM
>
>
> On Wed, Dec 30, 2009 at 12:17 PM, Seven Du <dujinfang at gmail.com> wrote:
>
>> I rarely joined in IRC, becuase I live in China, timezone +8000 ....
>> I really would like to start the official softphone, btw, what is JM?
>>
>> 2009/12/30, Brian West <brian at freeswitch.org>:
>> > You should join IRC and join in JM and really start the official
>> softphone
>> > project.
>> >
>> > /b
>> >
>> > On Dec 29, 2009, at 7:36 PM, Seven Du wrote:
>> >
>> >> I had wrote a Air based GUI, is it make sense?
>> >>
>> >> http://wiki.freeswitch.org/wiki/FsAir
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From brian at freeswitch.org  Thu Jan  7 08:25:11 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jan 2010 10:25:11 -0600
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org>
	<2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org>
	<B732197E-1123-4738-A69F-F84206FCE38A@freeswitch.org>
	<0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org>
	<0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org>
	<23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com>
	<60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org>
	<23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com>
	<ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com>
	<23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com>
Message-ID: <B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>

Lets schedule FSComm on the weekly conference call... We need people to step up and take some roles in both FreeSWITCH and FSComm projects... Even if its just testing bugs and collecting info.

Thanks,
Brian

On Jan 7, 2010, at 10:11 AM, Seven Du wrote:

> :lol. I do like to involve into this. I saw you have done a lot of
> works. I read some code and here are some questions:
> 
> 1) what's your nick on IRC? I'm seven(or seven_ ?)
> 2) Are you developing on Windows? How can I compile on Mac(I have no
> experience on QT)?
> 3) Does is needs fsGUI? I downloaded fsGUI dmg file and it complains
> no mods on start.



From oscav at hotmail.fr  Thu Jan  7 08:34:22 2010
From: oscav at hotmail.fr (Oscav)
Date: Thu, 7 Jan 2010 08:34:22 -0800 (PST)
Subject: [Freeswitch-users]  Failed to connect to a SKYPE API
Message-ID: <27062783.post@talk.nabble.com>


Hi,

I'm trying to use to SkypeIAX. When I load the mod_skypiax, I got the
following error :

 Failed to connect to a SKYPE API for interface_id=1, no SKYPE client
running, please (re)start Skype client. Skypiax exiting

Skype is running with the correct account and skypiax.conf use the same
account. I was expecting a permission request from the Skype user but
nothing happens.

Somebody knows how I can solve this ??

Many thanks.
-- 
View this message in context: http://old.nabble.com/Failed-to-connect-to-a-SKYPE-API-tp27062783p27062783.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From Claudio.Cavalera at italtel.it  Thu Jan  7 08:46:24 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Thu, 7 Jan 2010 17:46:24 +0100
Subject: [Freeswitch-users] Codecs and things
In-Reply-To: <4B3229C2.4080109@coppice.org>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F362B8@BESONE.corp.dom>

> The G.729 codec for FS is in testing, and should be out so. If you
> really want to implement your own, TI DSP code is unlikely to
> be a good
> starting point. I assume that code is fixed point. You really need a
> floating point codec to get any decent speed on a PC. Pentiums and
> Athlons lack saturating arithmetic (MMX actually has partial
> saturating arithmetic, but it isn't much use for anything but image
> processing), so a fixed point implementation ends up very slow.
> 

Hello Steve,
from what you have written it seems very unlikely that we are gonna buy
the official G.729 codec for embedded hardware?
I don't know much about it but would a MIPS32 24kf be enough? Just
speculating from here
http://www.mips.com/products/processors/32-64-bit-cores/mips32-24k/

Thanks,
Claudio


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From brian at freeswitch.org  Thu Jan  7 08:56:55 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jan 2010 10:56:55 -0600
Subject: [Freeswitch-users] Codecs and things
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0F362B8@BESONE.corp.dom>
References: <A2167AE7633A5C4DAE56EEBD1428ECC0F362B8@BESONE.corp.dom>
Message-ID: <7A65B817-E490-4582-8D43-8531FFA61CC4@freeswitch.org>

You usually still have to pay a license even if you buy a DSP that is capable of doing it.  

/b

On Jan 7, 2010, at 10:46 AM, Cavalera Claudio Luigi wrote:

> Hello Steve,
> from what you have written it seems very unlikely that we are gonna buy
> the official G.729 codec for embedded hardware?
> I don't know much about it but would a MIPS32 24kf be enough? Just
> speculating from here
> http://www.mips.com/products/processors/32-64-bit-cores/mips32-24k/
> 
> Thanks,
> Claudio

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From msc at freeswitch.org  Thu Jan  7 09:20:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Jan 2010 09:20:07 -0800
Subject: [Freeswitch-users] Call through gateway without register >
	sends to gateway name?
In-Reply-To: <26979541.post@talk.nabble.com>
References: <26979541.post@talk.nabble.com>
Message-ID: <87f2f3b91001070920r12028425k266ba050d9cec7f8@mail.gmail.com>

On Thu, Jan 7, 2010 at 6:56 AM, peely <freeswitch at peely.com> wrote:

>
> Hi,
>
> I have a problem where I'm trying to send calls to a gateway that does not
> support registration.
>
> In my external sip profile directory I have a file containing:
>
> <include>
>    <gateway name="mygateway">
>
>
>
>
>
>    </gateway>
> </include>
>
> Then in my dialplan I have a dialplan with an action of:
>
> <action application="bridge" data="sofia/gateway/mygateway/1234567890"/>
>
> However, when I call out, the Sofia diag shows:
>
> sres_send_dns_query(0x7f19ac011150, 0x7f19a401cdf0) id=20898 NAPTR
> mygateway
> (to [172.16.1.1]:53)
>
> Could somebody please tell me how I get the gateway config to send INVITEs
> to a specific IP?
>
>
Is that your actual gateway definition? I see a bunch of blank lines and I
don't know if that's a typo or what. Be sure that you have this in your
gateway:
<param name="register" value="false"/>

Of course, you still need the username and password fields, even if they
just have dummy values, plus you should have the realm specified. If you're
still having issues then pastebin your whole gateway file (redacting private
info) and also a capture of a debug trace of a failed call from start to
finish.

-MC
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From msc at freeswitch.org  Thu Jan  7 09:26:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Jan 2010 09:26:31 -0800
Subject: [Freeswitch-users] Calls getting queued?
In-Reply-To: <1b46b4e81001070743v5389ef3ewb75dbeb06b402c9a@mail.gmail.com>
References: <1b46b4e81001070743v5389ef3ewb75dbeb06b402c9a@mail.gmail.com>
Message-ID: <87f2f3b91001070926j60139f2cxe0ae9687e25a302e@mail.gmail.com>

On Thu, Jan 7, 2010 at 7:43 AM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Hi, I'm having a strange problem with FS. I'm using a few JS scripts
> to generate calls and bridge them together. Usually everything works
> just fine, but them at some point it's like if FS choked, calls for
> the first leg of the bridges are apparently made, but the second leg
> is never called. The call is not hanged up for several minutes and the
> system keeps opening new channels but never connecting a call.
>
> For example, right now, doing 'show channels' on the console, I get a
> list of 72 open channels (it's adding up, it was 40 a couple minutes
> ago), but doing a 'show calls' gives me 0 active calls. The usual
> behavior, when everything's working fine, is to get twice as many
> channels as there are active calls and no channels at all when there
> are no calls, unless they haven't been bridged yet.
>
> The originate string is something like this:
>
> var stUsRing = "%(2000,4000,440,480)";
> var timeout = 45;
> originate_str1 = "{api_hangup_hook=jsapi::callback.js
> l1,execute_on_answer=lua answered.lua 1
>
> c2c_call,ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,ringback='"+stUsRing+"',medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]"+dialstr1;
>
> Where diasltr1 has the phonenumber and and gateway info. The
> callback.js has a curl request to update some call info on an external
> database and answered.lua calls a ruby script through the os.execute()
> function (I know, I should be doing all this through the event socket,
> I was doing that but had trouble and had to come up with a quick
> solution).
>
> The system is not loaded at all, at least not for what I think and
> read that FS can handle. We are having at most 10 concurrent calls (20
> channels), with maybe 5 to 10 calls per minute.
>
> What worries me is not only that I don't know where the problem is,
> but that I have no clue how to debug it or send you guys more
> "lowlevel" and detailed information to give you an insight about
> what's going on. Any help would be greatly appreciated!
>
> Thanks!
>
> Nico
>
> First off you'll want to get familiar with the resources mentioned here:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

It has good tips on how to collect and report information.

Second, I recommend that you pastebin your relevant portion of the dialplan
and the whole javascript program that you are using so that others can take
a look.

Last thing: if you restart FreeSWITCH does everything work fine for a while
but then eventually it breaks down and exhibits the behavior that you are
reporting?

-MC
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From neil.burgess at redmatter.com  Thu Jan  7 07:52:52 2010
From: neil.burgess at redmatter.com (Neil Burgess)
Date: Thu, 7 Jan 2010 15:52:52 +0000
Subject: [Freeswitch-users] UNIMRCP RECOGNIZE headers
Message-ID: <787302A89ACCE24DA8F56DA101E77C841392F8C2D8@THHS2E12BE1X.hostedservice2.net>

Hello,

Wondering if anyone can help with a unimrcp question.

We are using the UNIMRCP client in FreeSwitch to communicate with a commercial UNIMRCP server (Loquendo). We need to control such items as Speech Timeouts that are occurring on the server side, however we have been advised that the only way these can be affected is via  headers in the MRCP requests. So, we need to be able to set headers such as Speech-Complete-Timeout, Speech-Incomplete-Timeout, Recognition-Timeout, etc in the MRCP RECOGNIZE request.  Is there a mechanism in FreeSwitch which we can use to pass these down before, (or as) we issue the "detect_speech" command. We are happy to use a jscript interface, or whatever if such a capability is available.

Many thanks,
Neil
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From nicolas at medularis.com  Thu Jan  7 10:12:25 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Thu, 7 Jan 2010 15:12:25 -0300
Subject: [Freeswitch-users] Calls getting queued?
In-Reply-To: <87f2f3b91001070926j60139f2cxe0ae9687e25a302e@mail.gmail.com>
References: <1b46b4e81001070743v5389ef3ewb75dbeb06b402c9a@mail.gmail.com>
	<87f2f3b91001070926j60139f2cxe0ae9687e25a302e@mail.gmail.com>
Message-ID: <1b46b4e81001071012y237fb8c0jd68232137b7d8e96@mail.gmail.com>

Michael,

Thanks for your help. Yes, if I restart FS things go back to normal
for a while and then the same thing starts happening again.

The weird thing is, it started only 2 days ago, and happened only once
or twice. Before that I had no trouble, and I only made 1 change,
which I reverted, but it wasn't that. Today it's happening all the
time, if I restart FS things will work for maybe an hour and then it
will start doing the same thing.

I'm guessing it might be something external to FS, like curl calls not
finishing properly because of the url they are requesting or something
like that.

What kind of info should I collect? I don't think it has to do with
sofia or any sip-related problems. I'm also using the default
dialplan, no changes at all, I'm doing everything through JS, well and
one really small lua script.

This is the main JS file:
It originates 2 calls and bridges them.

- http://pastebin.freeswitch.org/11706


This is another JS script which gets called when each call is hanged up:
It gets some info and then requests a url using curl to update call
status on an external db.

- http://pastebin.freeswitch.org/11707


This lua script calls a ruby script to do some other stuff when a call
is answered:

- http://pastebin.freeswitch.org/11708


Thanks!


Nico



On Thu, Jan 7, 2010 at 2:26 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Thu, Jan 7, 2010 at 7:43 AM, Nicolas Brenner <nicolas at medularis.com>
> wrote:
>>
>> Hi, I'm having a strange problem with FS. I'm using a few JS scripts
>> to generate calls and bridge them together. Usually everything works
>> just fine, but them at some point it's like if FS choked, calls for
>> the first leg of the bridges are apparently made, but the second leg
>> is never called. The call is not hanged up for several minutes and the
>> system keeps opening new channels but never connecting a call.
>>
>> For example, right now, doing 'show channels' on the console, I get a
>> list of 72 open channels (it's adding up, it was 40 a couple minutes
>> ago), but doing a 'show calls' gives me 0 active calls. The usual
>> behavior, when everything's working fine, is to get twice as many
>> channels as there are active calls and no channels at all when there
>> are no calls, unless they haven't been bridged yet.
>>
>> The originate string is something like this:
>>
>> var stUsRing = "%(2000,4000,440,480)";
>> var timeout = 45;
>> originate_str1 = "{api_hangup_hook=jsapi::callback.js
>> l1,execute_on_answer=lua answered.lua 1
>>
>> c2c_call,ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,ringback='"+stUsRing+"',medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]"+dialstr1;
>>
>> Where diasltr1 has the phonenumber and and gateway info. The
>> callback.js has a curl request to update some call info on an external
>> database and answered.lua calls a ruby script through the os.execute()
>> function (I know, I should be doing all this through the event socket,
>> I was doing that but had trouble and had to come up with a quick
>> solution).
>>
>> The system is not loaded at all, at least not for what I think and
>> read that FS can handle. We are having at most 10 concurrent calls (20
>> channels), with maybe 5 to 10 calls per minute.
>>
>> What worries me is not only that I don't know where the problem is,
>> but that I have no clue how to debug it or send you guys more
>> "lowlevel" and detailed information to give you an insight about
>> what's going on. Any help would be greatly appreciated!
>>
>> Thanks!
>>
>> Nico
>>
> First off you'll want to get familiar with the resources mentioned here:
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>
> It has good tips on how to collect and report information.
>
> Second, I recommend that you pastebin your relevant portion of the dialplan
> and the whole javascript program that you are using so that others can take
> a look.
>
> Last thing: if you restart FreeSWITCH does everything work fine for a while
> but then eventually it breaks down and exhibits the behavior that you are
> reporting?
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Thu Jan  7 10:22:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Jan 2010 12:22:03 -0600
Subject: [Freeswitch-users] Calls getting queued?
In-Reply-To: <1b46b4e81001071012y237fb8c0jd68232137b7d8e96@mail.gmail.com>
References: <1b46b4e81001070743v5389ef3ewb75dbeb06b402c9a@mail.gmail.com>
	<87f2f3b91001070926j60139f2cxe0ae9687e25a302e@mail.gmail.com>
	<1b46b4e81001071012y237fb8c0jd68232137b7d8e96@mail.gmail.com>
Message-ID: <191c3a031001071022j1e5796fdvf63900f5968bc01b@mail.gmail.com>

try setting the timeout in curl

conf/autoload_configs/xml_curl.conf.xml:
<param name="timeout" value="5"/>

On Thu, Jan 7, 2010 at 12:12 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Michael,
>
> Thanks for your help. Yes, if I restart FS things go back to normal
> for a while and then the same thing starts happening again.
>
> The weird thing is, it started only 2 days ago, and happened only once
> or twice. Before that I had no trouble, and I only made 1 change,
> which I reverted, but it wasn't that. Today it's happening all the
> time, if I restart FS things will work for maybe an hour and then it
> will start doing the same thing.
>
> I'm guessing it might be something external to FS, like curl calls not
> finishing properly because of the url they are requesting or something
> like that.
>
> What kind of info should I collect? I don't think it has to do with
> sofia or any sip-related problems. I'm also using the default
> dialplan, no changes at all, I'm doing everything through JS, well and
> one really small lua script.
>
> This is the main JS file:
> It originates 2 calls and bridges them.
>
> - http://pastebin.freeswitch.org/11706
>
>
> This is another JS script which gets called when each call is hanged up:
> It gets some info and then requests a url using curl to update call
> status on an external db.
>
> - http://pastebin.freeswitch.org/11707
>
>
> This lua script calls a ruby script to do some other stuff when a call
> is answered:
>
> - http://pastebin.freeswitch.org/11708
>
>
> Thanks!
>
>
> Nico
>
>
>
> On Thu, Jan 7, 2010 at 2:26 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >
> >
> > On Thu, Jan 7, 2010 at 7:43 AM, Nicolas Brenner <nicolas at medularis.com>
> > wrote:
> >>
> >> Hi, I'm having a strange problem with FS. I'm using a few JS scripts
> >> to generate calls and bridge them together. Usually everything works
> >> just fine, but them at some point it's like if FS choked, calls for
> >> the first leg of the bridges are apparently made, but the second leg
> >> is never called. The call is not hanged up for several minutes and the
> >> system keeps opening new channels but never connecting a call.
> >>
> >> For example, right now, doing 'show channels' on the console, I get a
> >> list of 72 open channels (it's adding up, it was 40 a couple minutes
> >> ago), but doing a 'show calls' gives me 0 active calls. The usual
> >> behavior, when everything's working fine, is to get twice as many
> >> channels as there are active calls and no channels at all when there
> >> are no calls, unless they haven't been bridged yet.
> >>
> >> The originate string is something like this:
> >>
> >> var stUsRing = "%(2000,4000,440,480)";
> >> var timeout = 45;
> >> originate_str1 = "{api_hangup_hook=jsapi::callback.js
> >> l1,execute_on_answer=lua answered.lua 1
> >>
> >>
> c2c_call,ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,ringback='"+stUsRing+"',medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]"+dialstr1;
> >>
> >> Where diasltr1 has the phonenumber and and gateway info. The
> >> callback.js has a curl request to update some call info on an external
> >> database and answered.lua calls a ruby script through the os.execute()
> >> function (I know, I should be doing all this through the event socket,
> >> I was doing that but had trouble and had to come up with a quick
> >> solution).
> >>
> >> The system is not loaded at all, at least not for what I think and
> >> read that FS can handle. We are having at most 10 concurrent calls (20
> >> channels), with maybe 5 to 10 calls per minute.
> >>
> >> What worries me is not only that I don't know where the problem is,
> >> but that I have no clue how to debug it or send you guys more
> >> "lowlevel" and detailed information to give you an insight about
> >> what's going on. Any help would be greatly appreciated!
> >>
> >> Thanks!
> >>
> >> Nico
> >>
> > First off you'll want to get familiar with the resources mentioned here:
> > http://wiki.freeswitch.org/wiki/Reporting_Bugs
> >
> > It has good tips on how to collect and report information.
> >
> > Second, I recommend that you pastebin your relevant portion of the
> dialplan
> > and the whole javascript program that you are using so that others can
> take
> > a look.
> >
> > Last thing: if you restart FreeSWITCH does everything work fine for a
> while
> > but then eventually it breaks down and exhibits the behavior that you are
> > reporting?
> >
> > -MC
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From linux4michelle at tamay-dogan.net  Thu Jan  7 10:27:31 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Thu, 7 Jan 2010 19:27:31 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <7b197bef1001060939rd1e08f3h2cb82cd89e50225e@mail.gmail.com>
References: <20100103184636.GW5547@tamay-dogan.net>
	<20100106135210.GG5547@tamay-dogan.net>
	<CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>
	<7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com>
	<20100106171413.GI5547@tamay-dogan.net>
	<7b197bef1001060939rd1e08f3h2cb82cd89e50225e@mail.gmail.com>
Message-ID: <20100107182731.GL5547@tamay-dogan.net>

Good evening Giovanni,

Am 2010-01-06 18:39:22, schrieb Giovanni Maruzzelli:
> Hello Michelle,
> 
> it supports all modules that accepts standard ETSI AT-GSM commands
> (so, let's say all of them).

Currently I havepayed over 20.000 Euro for ETSI and ANS specs  and  have
currently not the money to buy more...  Are the ETSI  AT-GSM  specs  are
free available?  --  If yes, I need them!

> Maybe I do not understand the second question, what do you means for
> Endpoints?
> 
> If you're talking usb endpoints, you'll need a modem endpoint (that
> can be seen as a serial port), and (if you need audio, eg not just
> SMSs but voice calls too) you need an audio endpoint (that can be seen
> as a soundcard).

I mean the USB-Endpoints...

If you have an USB-Microcontroller where the USB port is  a  device,  it
identify it self over the Endpoint 0 and is for us non usable.

And no it comes, where I haveproblems to understnd HOW the GSM Modem  is
working but I will assume some things:

The EP1 of the USB-pot is configured for bidirectional Data transmission
andwill controll out Device and is normaly  configured  as  /dev/ttyUSB0
and this is, where we use the AT commands to  get  infos  from  the  GSM
modem/cellphone and send/receive SMS.

Now We need EP2 and configure it as streaming output for the Audio port.

EP3 would be the streaming input for the Audio Port.

is this right up to here?

Then, EP4 would be the bidirectinal dataport  for  the  UMTS  and  HS*PA
Tranceiver, since it is entirely independant from the rest  of  the  GSM
modem/cellphone.

Is this right?

If yes, then it is easier as I was thinking...


> Many modules and cellphones can be seen as HDSPA or GPRS modems, just
> check their specs.

My "Nokia 6120 classic" has in total 13 endpoints...  Hell, where can  I
get an USB-Microcontroller which support this mass of USB Endpoints?

Most ARM9/11 support not more then 7 or 9.  :-(

> For audio, if the module/cellphone/modem does not offer an audio usb
> endpoint (eg cannot be seen as a soundcard) one trick is to connect
> the headset jack to an usb soundcard (you can find soundcard with for
> factor like a dongle based on cm-108 chipset for under $10). I'll
> publish the schema of the cable needed from hadset jack in the
> phone/module to the usb soundcard).
> 
> If I have not get what you asked, please explain more your question.

Most important things are the above desibed understanding  problem  with
the USB Endpoints

I have the HSPA and GS frontends (Maxim and Infineon chips) here and  my
selfemade simple GSM/GPRS cell-phone is already working,  but  has  less
functionality as the cell-phones from Year 2000  :-D

Hey, ist is my first experience of developing GSM stuff.

I an to develop a VERY simple GSM/UMTS/HSPA USB-Modem which do  its  job
without balast.

So if someone can help me with infos, I am very open...

I prefer FreeSwitch over Asterisk which froze in the  last  2  years  to
many times in situations where it should not freeze, exspecialy if  I  a
call a Chip-Manfacturer (Maxim/TI) Tech-Support.  --  It is not funny!

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From cmrienzo at gmail.com  Thu Jan  7 11:22:46 2010
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Thu, 7 Jan 2010 14:22:46 -0500
Subject: [Freeswitch-users] UNIMRCP RECOGNIZE headers
In-Reply-To: <E1D3989424711B43A00C5476DE5E1A7001E300D1@hqmail01.soundbite.com>
References: <E1D3989424711B43A00C5476DE5E1A7001E300D1@hqmail01.soundbite.com>
Message-ID: <7C265966-14D1-48E9-870F-2EE7B52A15BC@gmail.com>

Neil,

In general, you can set most MRCP params in mod_unimrcp like this:

detect_speech unimrcp {speech-complete-timeout=5000,speech-incomplete-timeout=5000}grammar grammar-name

These params will remain set on the speech handle until freeswitch closes it.  For ASR, that's when the call ends.




> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neil Burgess
> Sent: Thursday, January 07, 2010 10:53 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] UNIMRCP RECOGNIZE headers
>  
> Hello,
>  
> Wondering if anyone can help with a unimrcp question.
>  
> We are using the UNIMRCP client in FreeSwitch to communicate with a commercial UNIMRCP server (Loquendo). We need to control such items as Speech Timeouts that are occurring on the server side, however we have been advised that the only way these can be affected is via  headers in the MRCP requests. So, we need to be able to set headers such as Speech-Complete-Timeout, Speech-Incomplete-Timeout, Recognition-Timeout, etc in the MRCP RECOGNIZE request.  Is there a mechanism in FreeSwitch which we can use to pass these down before, (or as) we issue the ?detect_speech? command. We are happy to use a jscript interface, or whatever if such a capability is available.
>  
> Many thanks,
> Neil
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From achaloyan at yahoo.com  Thu Jan  7 11:51:48 2010
From: achaloyan at yahoo.com (Arsen Chaloyan)
Date: Thu, 7 Jan 2010 11:51:48 -0800 (PST)
Subject: [Freeswitch-users] UNIMRCP RECOGNIZE headers
In-Reply-To: <787302A89ACCE24DA8F56DA101E77C841392F8C2D8@THHS2E12BE1X.hostedservice2.net>
References: <787302A89ACCE24DA8F56DA101E77C841392F8C2D8@THHS2E12BE1X.hostedservice2.net>
Message-ID: <594856.45588.qm@web111302.mail.gq1.yahoo.com>

Hello Neil,


I guess you are using a commercial MRCP server such as Loquendo, at least I'm not aware of UniMRCP server based Loquendo ASR :)
Though someone asked me about such a solution a few months ago.

As of actual request, looking at the code of mod_unimrcp, I'd say such an option exists. See recog_channel_set_params().
Looking through mod_unimrcp wiki examples, I'd say the following should do what you need

switch_ivr_detect_speech(session, "unimrcp", "{recognition-timeout=15000}yesno", "yesno-name", "", ah);

-- 
Arsen Chaloyan
The author of UniMRCP 
http://www.unimrcp.org




________________________________
From: Neil Burgess <neil.burgess at redmatter.com>
To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
Sent: Thu, January 7, 2010 7:52:52 PM
Subject: [Freeswitch-users] UNIMRCP RECOGNIZE headers

 
Hello,
 
Wondering if anyone can help with a unimrcp question. 
 
We are using the UNIMRCP client in FreeSwitch to communicate
with a commercial UNIMRCP server (Loquendo). We need to control such items as Speech
Timeouts that are occurring on the server side, however we have been advised
that the only way these can be affected is via  headers in the MRCP
requests. So, we need to be able to set headers such as Speech-Complete-Timeout,
Speech-Incomplete-Timeout, Recognition-Timeout, etc in the MRCP RECOGNIZE request. 
Is there a mechanism in FreeSwitch which we can use to pass these down before,
(or as) we issue the ?detect_speech? command. We are happy to use a
jscript interface, or whatever if such a capability is available.
 
Many thanks,
Neil 
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From msc at freeswitch.org  Thu Jan  7 12:01:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Jan 2010 12:01:00 -0800
Subject: [Freeswitch-users] Announcement: FSComm - The FreeSWITCH-based
	softphone
Message-ID: <87f2f3b91001071201vb759713w3f37b192cd09020f@mail.gmail.com>

We are happy to announce a new project: FSComm, a FreeSWITCH-based
softphone. Read the story here <http://bit.ly/4HLF71>. We look forward to
watching this project grow and become a truly useful tool for VoIP users
everywhere.

-Michael
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From jmesquita at freeswitch.org  Thu Jan  7 12:27:01 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 7 Jan 2010 18:27:01 -0200
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<B732197E-1123-4738-A69F-F84206FCE38A@freeswitch.org>
	<0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org>
	<0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org>
	<23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com>
	<60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org>
	<23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com>
	<ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com>
	<23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com>
	<B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>
Message-ID: <ebbc2d551001071227y22d6eaa8y43742fb4328a67b2@mail.gmail.com>

There is a wiki page up now.

http://wiki.freeswitch.org/wiki/FSComm

It's a bit poor at the moment, but I will fill in more stuff when I feel
better (really sick now).

Regards,
Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927


On Thu, Jan 7, 2010 at 2:25 PM, Brian West <brian at freeswitch.org> wrote:

> Lets schedule FSComm on the weekly conference call... We need people to
> step up and take some roles in both FreeSWITCH and FSComm projects... Even
> if its just testing bugs and collecting info.
>
> Thanks,
> Brian
>
> On Jan 7, 2010, at 10:11 AM, Seven Du wrote:
>
> > :lol. I do like to involve into this. I saw you have done a lot of
> > works. I read some code and here are some questions:
> >
> > 1) what's your nick on IRC? I'm seven(or seven_ ?)
> > 2) Are you developing on Windows? How can I compile on Mac(I have no
> > experience on QT)?
> > 3) Does is needs fsGUI? I downloaded fsGUI dmg file and it complains
> > no mods on start.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Thu Jan  7 12:36:11 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jan 2010 14:36:11 -0600
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <ebbc2d551001071227y22d6eaa8y43742fb4328a67b2@mail.gmail.com>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<B732197E-1123-4738-A69F-F84206FCE38A@freeswitch.org>
	<0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org>
	<0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org>
	<23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com>
	<60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org>
	<23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com>
	<ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com>
	<23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com>
	<B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>
	<ebbc2d551001071227y22d6eaa8y43742fb4328a67b2@mail.gmail.com>
Message-ID: <D03B492C-74B9-4B38-B67B-017690FF372C@freeswitch.org>

Everyone should get JM's paypal and toss him some cash for all the good work he's doing... Without him this project wouldn't have become a reality.

/b

On Jan 7, 2010, at 2:27 PM, Jo?o Mesquita wrote:

> There is a wiki page up now.
> 
> http://wiki.freeswitch.org/wiki/FSComm
> 
> It's a bit poor at the moment, but I will fill in more stuff when I feel better (really sick now).
> 
> Regards,
> Jo?o Mesquita
> FreeSWITCH? Solutions
> t: +1 (646) 4959927

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From jerry.richards at teotech.com  Thu Jan  7 13:42:34 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 7 Jan 2010 13:42:34 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <407CE59D-5591-4838-B3D7-A0EC674844F7@freeswitch.org>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com><87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com><4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
	<407CE59D-5591-4838-B3D7-A0EC674844F7@freeswitch.org>
Message-ID: <5F093406C6D045CAB941A284E3A17178@greyhawk.tonecommander.com>

Brian,
 
Thank you.  The issue did have to do with SCTP package installation.
Apparently, after I installed the SCTP packages yesterday, I did not re-run
./confgure.  Anyway, now my PRI is working on the new server.
 
Thank You and Best Regards,
Jerry
 


  _____  

From: Brian West [mailto:brian at freeswitch.org] 
Sent: Wednesday, January 06, 2010 2:33 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER


Again if you did not have the SCTP libs and dev headers installed when you
did ./configure you'll never get this working SCTP is required to have boost
work.  Also don't use Pre9 get the latest from latest.freeswitch.org and run
with that please. 

/b

On Jan 6, 2010, at 4:22 PM, Jerry Richards wrote:



By the way, I posted another pastebin log
(http://pastebin.freeswitch.org/11694) that shows the output of the "make"
and "make install sounds-install moh-install..." commands.  Just prior to
these makes I executed "make clean" and "make distclean".  You will notice
in the makefile output that ozmod_sangoma_boost.c never appears to get
compiled.  Shouldn't everything be compiled in this case?  I also confirmed
that my working server does have the ozmod_sangoma_boost.so file located in
the right place, which is why it's working.
 
Could version 1.0.5pre9 introduced this bug?  Could that be why the older
server (which original ran with 1.0.4) works and the new one doesn't?
 
Thanks and Best Regards,
Jerry
 


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From larclap at yahoo.com  Thu Jan  7 14:15:14 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 7 Jan 2010 14:15:14 -0800
Subject: [Freeswitch-users] Compile error fscomm?
Message-ID: <012901ca8fe6$e36b71c0$aa425540$@com>

I just downloaded the fscomm project and loaded it into vs2008. I've never
programmed in C++ (or c), just C#, so I can't make anything of the following
two messages:

 

Error      1              error PRJ0019: A tool returned an error code from
"RCC resources.qrc"  FSComm              FSComm

 

Warning               2              The following environment variables
were not found: $(QTDIR)                Project FSComm

 

Any suggestions? I do see the file "resources.qrc" in the fscomm folder.

 

The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)

 

Lars

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From msc at freeswitch.org  Thu Jan  7 14:35:54 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Jan 2010 14:35:54 -0800
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <012901ca8fe6$e36b71c0$aa425540$@com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>
Message-ID: <87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:

>  I just downloaded the fscomm project and loaded it into vs2008. I?ve
> never programmed in C++ (or c), just C#, so I can?t make anything of the
> following two messages:
>
>
>
> Error      1              error PRJ0019: A tool returned an error code from
> "RCC resources.qrc"  FSComm              FSComm
>
>
>
> Warning               2              The following environment variables
> were not found: $(QTDIR)                Project FSComm
>
>
>
> Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.
>
>
>
> The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)
>
>
>
> Lars
>
Do you have Qt 4.6 installed?
-MC
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From larclap at yahoo.com  Thu Jan  7 15:11:15 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 7 Jan 2010 15:11:15 -0800
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>
Message-ID: <014801ca8fee$b75f8780$261e9680$@com>

No Qt installed. I just checked out from
http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into
VS2008. 

Do I need to get
http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

 

 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:

I just downloaded the fscomm project and loaded it into vs2008. I've never
programmed in C++ (or c), just C#, so I can't make anything of the following
two messages:

 

Error      1              error PRJ0019: A tool returned an error code from
"RCC resources.qrc"  FSComm              FSComm

 

Warning               2              The following environment variables
were not found: $(QTDIR)                Project FSComm

 

Any suggestions? I do see the file "resources.qrc" in the fscomm folder.

 

The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)

 

Lars

Do you have Qt 4.6 installed?
-MC

 

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From jcasale at activenetwerx.com  Thu Jan  7 15:24:15 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Thu, 7 Jan 2010 23:24:15 +0000
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9070B8B@prato.activenetwerx.local>

>the big rule for the time being is stick with 5.3 5.4 appears to have some bugs in the toolchain and libc

OMG, I have been messing with my broken fax and zap for like two weeks?
Someone shoot me...

Unless you install of a dvd and avoid using public repo's, that's kind of
hard? Do you guys have any idea when this could be resolved, I am going to
hold off on my migration then.

Thanks!
jlc


From brian at freeswitch.org  Thu Jan  7 15:30:59 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jan 2010 17:30:59 -0600
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4 (fixed)
In-Reply-To: <20100105065356.AEE0612F5@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20091228233838.75E611694@sinclaire.sibble.net>,
	<C2EB0262-CB8A-44F5-9978-1C5F8073832F@freeswitch.org>
	<20100105065356.AEE0612F5@sinclaire.sibble.net>
Message-ID: <8C20E0B1-0E56-4892-9D36-4D6DF3331244@freeswitch.org>

Harondel,
	Please update your FreeSWITCH source rev 16204 fixes the SAS passing in all cases from my testing.  You'll need the latest ZRTP Lib and zfone application to make this work... I'm not too sure Tiviphone does this yet as I don't have one to test with.  This also fixes the issue when both sides are enrolled.  Next we will fix the video portion so both video and audio will go thru zrtp.

Please try it and let me know.

Thanks,
Brian

On Jan 5, 2010, at 12:54 AM, Harondel J. Sibble wrote:

> Maybe that's what's affecting me now.....
> 
> I've both phones registered (confirmed by calling 9787) on both devices and 
> it says each device is already enrolled. (how does one un-enroll????). Both 
> phones are running the Tivi 2.0.7 beta. 
> 
> Now however, other than the first call I made between devices after 
> enrollment, the sas is not matching anymore.
> 
> I set both these options in the console
> 
> global_action application="set" data="zrtp_enrollment=true"
> global_setvar zrtp_secure_media=true
> 
> What should I be looking for in the console output
> 
> On 28 Dec 2009 at 17:49, Brian West wrote:
> 
>> I'm still not done with this I think we found a bug in the lib... Viktor
>> fixed it today and I'm going to retry after I get done testing G729 more
>> today!  ;)
>> 
>> /b
>> 
>> On Dec 28, 2009, at 5:38 PM, Harondel J. Sibble wrote:
>> 
>>> Hmm, okay, I went back to basics and did a full rebuild for 1.0.4 svn
>> trunk, 
>>> I am able to now enroll my nokia e61i running the beta 2.0.7 Tiviphone 
>>> client, however I am not seeing the enrollment option popup in zfone 0.92
>>> build 218 on windows in front of an x-lite client.
> 
> -- 
> Harondel J. Sibble 
> Sibble Computer Consulting
> Creating Solutions for the small and medium business computer user.
> help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
> (604) 739-3709 (voice)
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jmesquita at freeswitch.org  Thu Jan  7 15:45:56 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 7 Jan 2010 21:45:56 -0200
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <014801ca8fee$b75f8780$261e9680$@com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>
	<014801ca8fee$b75f8780$261e9680$@com>
Message-ID: <ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>

I don't have a Windows machine to test that. Maybe jlenk could give us a
hand since he is the one who has created the visual studio project?

Regards,
Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927


On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:

>  No Qt installed. I just checked out from
> http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into
> VS2008.
>
> Do I need to get
> http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, January 07, 2010 2:36 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Compile error fscomm?
>
>
>
>
>
> On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:
>
> I just downloaded the fscomm project and loaded it into vs2008. I?ve never
> programmed in C++ (or c), just C#, so I can?t make anything of the following
> two messages:
>
>
>
> Error      1              error PRJ0019: A tool returned an error code from
> "RCC resources.qrc"  FSComm              FSComm
>
>
>
> Warning               2              The following environment variables
> were not found: $(QTDIR)                Project FSComm
>
>
>
> Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.
>
>
>
> The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)
>
>
>
> Lars
>
> Do you have Qt 4.6 installed?
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Jan  7 16:01:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Jan 2010 18:01:56 -0600
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9070B8B@prato.activenetwerx.local>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E9070B8B@prato.activenetwerx.local>
Message-ID: <191c3a031001071601if302c1bsd8a9a2af7c166cf7@mail.gmail.com>

if you get centos5.3 or 5.2 it will be resolved because it's not centos5.4
which is the only bad one atm, newer is not always better.



On Thu, Jan 7, 2010 at 5:24 PM, Joseph L. Casale
<jcasale at activenetwerx.com>wrote:

> >the big rule for the time being is stick with 5.3 5.4 appears to have some
> bugs in the toolchain and libc
>
> OMG, I have been messing with my broken fax and zap for like two weeks?
> Someone shoot me...
>
> Unless you install of a dvd and avoid using public repo's, that's kind of
> hard? Do you guys have any idea when this could be resolved, I am going to
> hold off on my migration then.
>
> Thanks!
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Thu Jan  7 16:03:07 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jan 2010 18:03:07 -0600
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>
	<014801ca8fee$b75f8780$261e9680$@com>
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
Message-ID: <F31651BB-C5E4-429F-ADAF-512B641F2CE1@freeswitch.org>

Remember we have #fscomm on irc.freenode.net please join there if you wish to get involved... help out...   ;)

Thanks,
Brian

On Jan 7, 2010, at 5:45 PM, Jo?o Mesquita wrote:

> I don't have a Windows machine to test that. Maybe jlenk could give us a hand since he is the one who has created the visual studio project?
> 
> Regards,
> Jo?o Mesquita
> FreeSWITCH? Solutions
> t: +1 (646) 4959927



From freeswitch at aastral.net  Thu Jan  7 16:15:42 2010
From: freeswitch at aastral.net (Bill W.)
Date: Thu, 07 Jan 2010 19:15:42 -0500
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
Message-ID: <4B46792E.6090805@aastral.net>

Personally, I'm migrating to openSUSE, because I'm tired of RedHat's
non-standard kernel backports and outdated packages.

Others in the VoIP industry (ViciDial) recommend SuSE because of their
experience with with RedHat's long-standing perl bug, process preemption
set to desktop, version number mismatch on packages and non-standard
libraries.

But SuSE is not without it's problems.  The perl module for FreeSWITCH
will compile on SuSE, but FreeSWITCH coredumps when trying to use it.
Haven't had time to research that one.

Also, if you're going to use wanpipe with Sangoma cards on SuSE, check
with Sangoma to make sure wanpipe will compile on 11.2.   It wouldn't
compile for me and I reported the bug to Sangoma, but I haven't heard
anything back yet.  Wanpipe compiles just fine on 11.1.

Having said that, the Suse installation is very similar to
CentOS/RedHat.  I've run FreeSWITCH on both, and it works just fine on
both.

Hope this helps,
Bill



Anthony Minessale wrote:
> the big rule for the time being is stick with 5.3 5.4 appears to have
> some bugs in the toolchain and libc
> 
> On Thu, Jan 7, 2010 at 6:02 AM, Joseph L. Casale
> <jcasale at activenetwerx.com <mailto:jcasale at activenetwerx.com>> wrote:
> 
>     >Since CentOS is recommend for FS but i can't see a CentOS specific
>     installation guide on wiki as we have a separate >guide for Ubuntu. 
>     >
>     >Do we have similar guide like this
>     one http://wiki.freeswitch.org/wiki/SBC_Setup actually its for
>     debian but good thing >is that it also explains which extra
>     >services should be stopped or removed for better performance. 
>     >
>     >Do we have similar for CentOS?
> 
>     Check out the http://wiki.freeswitch.org/wiki/Installation_Guide
> 
> 
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Thu Jan  7 16:49:11 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Jan 2010 18:49:11 -0600
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <4B46792E.6090805@aastral.net>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
	<4B46792E.6090805@aastral.net>
Message-ID: <191c3a031001071649g26ad7257pb9ba6e7045224f5a@mail.gmail.com>

The last guy using SUSE had filesystem problems so bad he almost cried while
he was apologizing for how much time he wasted asking for help about it once
he tried a different one.  Not to say you can't use whatever you want but we
are going to tread carefully about which distros we support.


On Thu, Jan 7, 2010 at 6:15 PM, Bill W. <freeswitch at aastral.net> wrote:

> Personally, I'm migrating to openSUSE, because I'm tired of RedHat's
> non-standard kernel backports and outdated packages.
>
> Others in the VoIP industry (ViciDial) recommend SuSE because of their
> experience with with RedHat's long-standing perl bug, process preemption
> set to desktop, version number mismatch on packages and non-standard
> libraries.
>
> But SuSE is not without it's problems.  The perl module for FreeSWITCH
> will compile on SuSE, but FreeSWITCH coredumps when trying to use it.
> Haven't had time to research that one.
>
> Also, if you're going to use wanpipe with Sangoma cards on SuSE, check
> with Sangoma to make sure wanpipe will compile on 11.2.   It wouldn't
> compile for me and I reported the bug to Sangoma, but I haven't heard
> anything back yet.  Wanpipe compiles just fine on 11.1.
>
> Having said that, the Suse installation is very similar to
> CentOS/RedHat.  I've run FreeSWITCH on both, and it works just fine on
> both.
>
> Hope this helps,
> Bill
>
>
>
> Anthony Minessale wrote:
> > the big rule for the time being is stick with 5.3 5.4 appears to have
> > some bugs in the toolchain and libc
> >
> > On Thu, Jan 7, 2010 at 6:02 AM, Joseph L. Casale
> > <jcasale at activenetwerx.com <mailto:jcasale at activenetwerx.com>> wrote:
> >
> >     >Since CentOS is recommend for FS but i can't see a CentOS specific
> >     installation guide on wiki as we have a separate >guide for Ubuntu.
> >     >
> >     >Do we have similar guide like this
> >     one http://wiki.freeswitch.org/wiki/SBC_Setup actually its for
> >     debian but good thing >is that it also explains which extra
> >     >services should be stopped or removed for better performance.
> >     >
> >     >Do we have similar for CentOS?
> >
> >     Check out the http://wiki.freeswitch.org/wiki/Installation_Guide
> >
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:+19193869900
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Thu Jan  7 16:52:25 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jan 2010 18:52:25 -0600
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <4B46792E.6090805@aastral.net>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
	<4B46792E.6090805@aastral.net>
Message-ID: <E5EFC626-353A-4B54-9F22-59C0C48E3B95@freeswitch.org>

Good luck with that you'll have an ass load of problems.  The reason its stable is the backports and outdated packages.  Bleeding edge will only screw you over... just ask KJV... He was on OpenSuSE and had nothing but weird problems.

/b

On Jan 7, 2010, at 6:15 PM, Bill W. wrote:

> Personally, I'm migrating to openSUSE, because I'm tired of RedHat's
> non-standard kernel backports and outdated packages.
> 
> Others in the VoIP industry (ViciDial) recommend SuSE because of their
> experience with with RedHat's long-standing perl bug, process preemption
> set to desktop, version number mismatch on packages and non-standard
> libraries.
> 
> But SuSE is not without it's problems.  The perl module for FreeSWITCH
> will compile on SuSE, but FreeSWITCH coredumps when trying to use it.
> Haven't had time to research that one.
> 
> Also, if you're going to use wanpipe with Sangoma cards on SuSE, check
> with Sangoma to make sure wanpipe will compile on 11.2.   It wouldn't
> compile for me and I reported the bug to Sangoma, but I haven't heard
> anything back yet.  Wanpipe compiles just fine on 11.1.
> 
> Having said that, the Suse installation is very similar to
> CentOS/RedHat.  I've run FreeSWITCH on both, and it works just fine on
> both.
> 
> Hope this helps,
> Bill



From jcasale at activenetwerx.com  Thu Jan  7 17:02:46 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 8 Jan 2010 01:02:46 +0000
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <191c3a031001071649g26ad7257pb9ba6e7045224f5a@mail.gmail.com>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
	<4B46792E.6090805@aastral.net>
	<191c3a031001071649g26ad7257pb9ba6e7045224f5a@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9072253@prato.activenetwerx.local>

>The last guy using SUSE had filesystem problems so bad he almost cried while
>he was apologizing for how much time he wasted asking for help about it once
>he tried a different one.? Not to say you can't use whatever you want but we
>are going to tread carefully about which distros we support.

Which begs the question, which distro do _you_ run on and suggest behind the
scenes:)


From jason at jasonjgw.net  Thu Jan  7 17:04:57 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 8 Jan 2010 12:04:57 +1100
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <191c3a031001071649g26ad7257pb9ba6e7045224f5a@mail.gmail.com>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
	<4B46792E.6090805@aastral.net>
	<191c3a031001071649g26ad7257pb9ba6e7045224f5a@mail.gmail.com>
Message-ID: <20100108010457.GA23508@jdc.jasonjgw.net>

Anthony Minessale <anthony.minessale at gmail.com> wrote:
> The last guy using SUSE had filesystem problems so bad he almost cried while
> he was apologizing for how much time he wasted asking for help about it once
> he tried a different one.  

Are they still using reiserfs by default? If I were using Reiserfs (which I'm
not, and I'm not using Suse either), I would be moving to EXT3, EXT4, or XFS.



From a.alalousi at gmail.com  Thu Jan  7 17:05:23 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Fri, 8 Jan 2010 01:05:23 +0000
Subject: [Freeswitch-users] FAS detection with FS
Message-ID: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>

Hi everyone,

Was just wondering what/if anyone is doing any work on FAS detection and
spoofed ring tones. Be great to discuss some ideas.

Regards,

Ahmed.

-- 
Ahmed Naji
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From freeswitch at aastral.net  Thu Jan  7 17:05:55 2010
From: freeswitch at aastral.net (Bill W.)
Date: Thu, 07 Jan 2010 20:05:55 -0500
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <E5EFC626-353A-4B54-9F22-59C0C48E3B95@freeswitch.org>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>	<4B46792E.6090805@aastral.net>
	<E5EFC626-353A-4B54-9F22-59C0C48E3B95@freeswitch.org>
Message-ID: <4B4684F3.8030504@aastral.net>

Wow, I haven't heard of these issues.  Obviously this concerns me.  Are
these documented anywhere so I can research this?  How do I get in touch
with KJV?

Thanks!
Bill



Brian West wrote:
> Good luck with that you'll have an ass load of problems.  The reason its stable is the backports and outdated packages.  Bleeding edge will only screw you over... just ask KJV... He was on OpenSuSE and had nothing but weird problems.
> 
> 


From anthony.minessale at gmail.com  Thu Jan  7 17:11:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Jan 2010 19:11:34 -0600
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9072253@prato.activenetwerx.local>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
	<4B46792E.6090805@aastral.net>
	<191c3a031001071649g26ad7257pb9ba6e7045224f5a@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E9072253@prato.activenetwerx.local>
Message-ID: <191c3a031001071711g776488e7t474806fb3527f194@mail.gmail.com>

We use CentOS 5.3 64 bit and have very few issues,
The problem is since it's an older kernel there may be some real time
improvements we could benefit from so we will probably play with some newer
bleeding kernels to compare and stay ahead of the curve but still rely on
CentOS to keep the users happy.


On Thu, Jan 7, 2010 at 7:02 PM, Joseph L. Casale
<jcasale at activenetwerx.com>wrote:

> >The last guy using SUSE had filesystem problems so bad he almost cried
> while
> >he was apologizing for how much time he wasted asking for help about it
> once
> >he tried a different one.  Not to say you can't use whatever you want but
> we
> >are going to tread carefully about which distros we support.
>
> Which begs the question, which distro do _you_ run on and suggest behind
> the
> scenes:)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From dujinfang at gmail.com  Thu Jan  7 17:21:30 2010
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 8 Jan 2010 09:21:30 +0800
Subject: [Freeswitch-users] Failed to connect to a SKYPE API
In-Reply-To: <27062783.post@talk.nabble.com>
References: <27062783.post@talk.nabble.com>
Message-ID: <23f91031001071721g3f220722o5a07e8e7c74ae339@mail.gmail.com>

To make others help you easier, you'd better include more informations

1) what's your FS version/OS?
2) did you followed all steps as the wiki page said to generate
.Skype/ conf files ?
3) there is a client.c in source code, can you compile and make sure
it works like this?

./client :101   #where 101 is your display no.
4) what your skypiax.conf looks like?

2010/1/8 Oscav <oscav at hotmail.fr>:
>
> Hi,
>
> I'm trying to use to SkypeIAX. When I load the mod_skypiax, I got the
> following error :
>
> ?Failed to connect to a SKYPE API for interface_id=1, no SKYPE client
> running, please (re)start Skype client. Skypiax exiting
>
> Skype is running with the correct account and skypiax.conf use the same
> account. I was expecting a permission request from the Skype user but
> nothing happens.
>
> Somebody knows how I can solve this ??
>
> Many thanks.
> --
> View this message in context: http://old.nabble.com/Failed-to-connect-to-a-SKYPE-API-tp27062783p27062783.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Thu Jan  7 17:23:57 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jan 2010 19:23:57 -0600
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <4B4684F3.8030504@aastral.net>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>	<4B46792E.6090805@aastral.net>
	<E5EFC626-353A-4B54-9F22-59C0C48E3B95@freeswitch.org>
	<4B4684F3.8030504@aastral.net>
Message-ID: <74C18F18-89EB-49DF-B5CE-C8D56856ABCD@freeswitch.org>

He's on the list Karl J. Vesterling

/b

On Jan 7, 2010, at 7:05 PM, Bill W. wrote:

> Wow, I haven't heard of these issues.  Obviously this concerns me.  Are
> these documented anywhere so I can research this?  How do I get in touch
> with KJV?
> 
> Thanks!
> Bill
> 
> 
> 
> Brian West wrote:
>> Good luck with that you'll have an ass load of problems.  The reason its stable is the backports and outdated packages.  Bleeding edge will only screw you over... just ask KJV... He was on OpenSuSE and had nothing but weird problems.
>> 
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From max.bridgewater at gmail.com  Thu Jan  7 18:36:37 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Thu, 7 Jan 2010 21:36:37 -0500
Subject: [Freeswitch-users] Skypiax on CentOS
Message-ID: <c55d46951001071836i55516863r359bb56695744054@mail.gmail.com>

Hi,

Has anybody installed Skypiax on CentOS 5 lately? The documentation is based
on Skype 2.0.0 (skype-2.0.0.72-centos.i586.rpm) which apparently is not
available online anymore. I am hoping that you guys can help me either 1)
get Skypiax to run with the latest Skype or 2) share this old version so i
can follow the existing documentation. My preference would be the second
option though ;)

Thanks,
Max.
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From brian at freeswitch.org  Thu Jan  7 18:49:57 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jan 2010 20:49:57 -0600
Subject: [Freeswitch-users] Skypiax on CentOS
In-Reply-To: <c55d46951001071836i55516863r359bb56695744054@mail.gmail.com>
References: <c55d46951001071836i55516863r359bb56695744054@mail.gmail.com>
Message-ID: <581E0571-986A-450D-98E3-D3AAF34E21C6@freeswitch.org>

Just download the static binary build for linux... problem solved.

/b

On Jan 7, 2010, at 8:36 PM, Max Bridgewater wrote:

> Hi,
> 
> Has anybody installed Skypiax on CentOS 5 lately? The documentation is based on Skype 2.0.0 (skype-2.0.0.72-centos.i586.rpm) which apparently is not available online anymore. I am hoping that you guys can help me either   1) get Skypiax to run with the latest  Skype or 2) share this old version so i can follow the existing documentation. My preference would be the second option though ;)
> 
> Thanks,
> Max.
> _________

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From william.suffill at gmail.com  Thu Jan  7 18:52:49 2010
From: william.suffill at gmail.com (William Suffill)
Date: Thu, 7 Jan 2010 21:52:49 -0500
Subject: [Freeswitch-users] Skypiax on CentOS
In-Reply-To: <c55d46951001071836i55516863r359bb56695744054@mail.gmail.com>
References: <c55d46951001071836i55516863r359bb56695744054@mail.gmail.com>
Message-ID: <6b65470d1001071852i752b8306r51a48576090b4be@mail.gmail.com>

Skype appears to be pushing a new version on the linux side but the old
packages are still available but not linked anywhere.

http://download.skype.com/linux/skype-2.0.0.72-centos.i586.rpm
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From max.bridgewater at gmail.com  Thu Jan  7 19:03:24 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Thu, 7 Jan 2010 22:03:24 -0500
Subject: [Freeswitch-users] Skypiax on CentOS
In-Reply-To: <6b65470d1001071852i752b8306r51a48576090b4be@mail.gmail.com>
References: <c55d46951001071836i55516863r359bb56695744054@mail.gmail.com>
	<6b65470d1001071852i752b8306r51a48576090b4be@mail.gmail.com>
Message-ID: <c55d46951001071903q5a23639ehee09da7640351a74@mail.gmail.com>

Thanks you very much guys.

Ad Brian: I tried the static binary build but was blocked by a qt4-x11
missing and I couldn't find an obvious solution to that. Hence my  inquiry
in the group before spending time on things that other people potentially
already solved.

max.

On Thu, Jan 7, 2010 at 9:52 PM, William Suffill
<william.suffill at gmail.com>wrote:

> Skype appears to be pushing a new version on the linux side but the old
> packages are still available but not linked anywhere.
>
> http://download.skype.com/linux/skype-2.0.0.72-centos.i586.rpm
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jeff at jefflenk.com  Thu Jan  7 19:16:40 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 7 Jan 2010 21:16:40 -0600
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>,
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>,
	<014801ca8fee$b75f8780$261e9680$@com>,
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
Message-ID: <COL117-W486989B811B372C35935AAB9700@phx.gbl>


The windows support is very experimental at this time!

 

You must manually install http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe

 

Then set the environment variable QTDIR in the environment variables. This can be set from the Computer/Properties/Advanced system settings/Environment Variables/User Variables settings screen.

 

QTDIR=c:\qt\4.6.0 - or wherever you installed it

 

then restart VS

 


 


Date: Thu, 7 Jan 2010 21:45:56 -0200
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

I don't have a Windows machine to test that. Maybe jlenk could give us a hand since he is the one who has created the visual studio project?


Regards,Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927



On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:




No Qt installed. I just checked out from http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into VS2008. 
Do I need to get http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?
 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?



 
 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:


I just downloaded the fscomm project and loaded it into vs2008. I?ve never programmed in C++ (or c), just C#, so I can?t make anything of the following two messages:
 
Error      1              error PRJ0019: A tool returned an error code from "RCC resources.qrc"  FSComm              FSComm
 
Warning               2              The following environment variables were not found: $(QTDIR)                Project FSComm
 
Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.
 
The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)
 
Lars

Do you have Qt 4.6 installed?
-MC
 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


 		 	   		  
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From pekkis50 at gmail.com  Thu Jan  7 19:25:56 2010
From: pekkis50 at gmail.com (Pekka Kurki)
Date: Fri, 08 Jan 2010 04:25:56 +0100
Subject: [Freeswitch-users] really no installer for w2k anywhere?
Message-ID: <4B46A5C4.9040809@gmail.com>

all installer versions fail with  missing  getnameinfo/getaddressinfo 
support in w2k.



From jmesquita at freeswitch.org  Thu Jan  7 19:28:31 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 8 Jan 2010 01:28:31 -0200
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <COL117-W486989B811B372C35935AAB9700@phx.gbl>
References: <012901ca8fe6$e36b71c0$aa425540$@com>
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>
	<014801ca8fee$b75f8780$261e9680$@com>
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>
Message-ID: <ebbc2d551001071928u422b1884r6ba556b8b7a06e80@mail.gmail.com>

Jeff, any chance we can get this on the wiki?

We have created a page here: http://wiki.freeswitch.org/wiki/FSComm

I am looking for a Windows machine to do testing on it as well.

Regards,
Jo?o Mesquita


On Fri, Jan 8, 2010 at 1:16 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

>  The windows support is very experimental at this time!
>
> You must manually install
> http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
>
> Then set the environment variable QTDIR in the environment variables. This
> can be set from the Computer/Properties/Advanced system settings/Environment
> Variables/User Variables settings screen.
>
> QTDIR=c:\qt\4.6.0 - or wherever you installed it
>
> then restart VS
>
>
>
> ------------------------------
> Date: Thu, 7 Jan 2010 21:45:56 -0200
> From: jmesquita at freeswitch.org
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Compile error fscomm?
>
> I don't have a Windows machine to test that. Maybe jlenk could give us a
> hand since he is the one who has created the visual studio project?
>
> Regards,
> Jo?o Mesquita
> FreeSWITCH? Solutions
> t: +1 (646) 4959927
>
>
> On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:
>
>  No Qt installed. I just checked out from
> http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into
> VS2008.
>
> Do I need to get
> http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, January 07, 2010 2:36 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Compile error fscomm?
>
>
>
>
>
> On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:
>
> I just downloaded the fscomm project and loaded it into vs2008. I?ve never
> programmed in C++ (or c), just C#, so I can?t make anything of the following
> two messages:
>
>
>
> Error      1              error PRJ0019: A tool returned an error code from
> "RCC resources.qrc"  FSComm              FSComm
>
>
>
> Warning               2              The following environment variables
> were not found: $(QTDIR)                Project FSComm
>
>
>
> Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.
>
>
>
> The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)
>
>
>
> Lars
>
> Do you have Qt 4.6 installed?
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up
> now. <http://clk.atdmt.com/GBL/go/196390709/direct/01/>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jeff at jefflenk.com  Thu Jan  7 19:36:52 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 7 Jan 2010 21:36:52 -0600
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <COL117-W486989B811B372C35935AAB9700@phx.gbl>
References: <012901ca8fe6$e36b71c0$aa425540$@com>, ,
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>, ,
	<014801ca8fee$b75f8780$261e9680$@com>, ,
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>,
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>
Message-ID: <COL117-W45E166F06B72FC096E4ACB9700@phx.gbl>


I will update the project files tommorow to account for the new project files/locations in the last 24 hours too.
 


From: jeff at jefflenk.com
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 7 Jan 2010 21:16:40 -0600
Subject: Re: [Freeswitch-users] Compile error fscomm?



The windows support is very experimental at this time!
 
You must manually install http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
 
Then set the environment variable QTDIR in the environment variables. This can be set from the Computer/Properties/Advanced system settings/Environment Variables/User Variables settings screen.
 
QTDIR=c:\qt\4.6.0 - or wherever you installed it
 
then restart VS
 

 


Date: Thu, 7 Jan 2010 21:45:56 -0200
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

I don't have a Windows machine to test that. Maybe jlenk could give us a hand since he is the one who has created the visual studio project? 


Regards,Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927



On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:




No Qt installed. I just checked out from http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into VS2008. 
Do I need to get http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?
 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?



 
 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:


I just downloaded the fscomm project and loaded it into vs2008. I?ve never programmed in C++ (or c), just C#, so I can?t make anything of the following two messages:
 
Error      1              error PRJ0019: A tool returned an error code from "RCC resources.qrc"  FSComm              FSComm
 
Warning               2              The following environment variables were not found: $(QTDIR)                Project FSComm
 
Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.
 
The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)
 
Lars

Do you have Qt 4.6 installed?
-MC
 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. 		 	   		  
_________________________________________________________________
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From jeff at jefflenk.com  Thu Jan  7 19:39:41 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 7 Jan 2010 21:39:41 -0600
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <ebbc2d551001071928u422b1884r6ba556b8b7a06e80@mail.gmail.com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>,
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>,
	<014801ca8fee$b75f8780$261e9680$@com>,
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>,
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>,
	<ebbc2d551001071928u422b1884r6ba556b8b7a06e80@mail.gmail.com>
Message-ID: <COL117-W15F7382FBF45603F4000E1B9700@phx.gbl>


yep sure thing I will update/add some of this information tom. after updating the project files.
 


Date: Fri, 8 Jan 2010 01:28:31 -0200
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

Jeff, any chance we can get this on the wiki?


We have created a page here: http://wiki.freeswitch.org/wiki/FSComm


I am looking for a Windows machine to do testing on it as well.


Regards,Jo?o Mesquita



On Fri, Jan 8, 2010 at 1:16 AM, Jeff Lenk <jeff at jefflenk.com> wrote:


The windows support is very experimental at this time!
 
You must manually install http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
 
Then set the environment variable QTDIR in the environment variables. This can be set from the Computer/Properties/Advanced system settings/Environment Variables/User Variables settings screen.
 
QTDIR=c:\qt\4.6.0 - or wherever you installed it
 
then restart VS
 

 


Date: Thu, 7 Jan 2010 21:45:56 -0200
From: jmesquita at freeswitch.org



To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

I don't have a Windows machine to test that. Maybe jlenk could give us a hand since he is the one who has created the visual studio project? 


Regards,Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927



On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:




No Qt installed. I just checked out from http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into VS2008. 

Do I need to get http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?

 


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?




 

 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:



I just downloaded the fscomm project and loaded it into vs2008. I?ve never programmed in C++ (or c), just C#, so I can?t make anything of the following two messages:

 

Error      1              error PRJ0019: A tool returned an error code from "RCC resources.qrc"  FSComm              FSComm

 

Warning               2              The following environment variables were not found: $(QTDIR)                Project FSComm

 

Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.

 

The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)

 

Lars


Do you have Qt 4.6 installed?
-MC

 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now.
_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From mike at jerris.com  Thu Jan  7 19:43:44 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 7 Jan 2010 22:43:44 -0500
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
Message-ID: <9AC8A302-DAC3-42FE-B1BB-91EB8A717248@jerris.com>

what is FAS ?

Mike

On Jan 7, 2010, at 8:05 PM, Ahmed Naji wrote:

> Was just wondering what/if anyone is doing any work on FAS detection and spoofed ring tones. Be great to discuss some ideas.



From max.bridgewater at gmail.com  Thu Jan  7 19:48:45 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Thu, 7 Jan 2010 22:48:45 -0500
Subject: [Freeswitch-users] hw:dummy not visible
Message-ID: <c55d46951001071948s558060fera2b08fecd667c544@mail.gmail.com>

Hi,

I got a few more Skypiax questions. Please bear with me.

1) Skype should use hw:dummy as audio device. But where do I set this on
Skype? In Options>Sound Devices, the only devices I see are; "Default
device" and "hdmi". My guess was that alsa-utils or some other ALSA related
lib would install this. But it seems this is not happening. Am i missing
something?

2) To create the configuration directory, I connect to the FreeNX XServer
running on my remote machine. I use NX to connect to the remote Xserver. I
can start all sort of GUI applications this way. But when I try to run
skypiax_auth, I get the following error message: "Cannot open X Display
':0.0', exiting". Any idea?

Thanks again,

Max.
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From nicolas at medularis.com  Thu Jan  7 19:58:26 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 8 Jan 2010 00:58:26 -0300
Subject: [Freeswitch-users] Ruby ESL missing pthread
Message-ID: <1b46b4e81001071958i17ccb6bey6eeecd9d94eb8438@mail.gmail.com>

I followed the instructions on the wiki to compile the Ruby version of
ESL, but then when I tried to run the examples, I kept getting a
"undefined symbol: pthread_mutexattr_init" error. I ran ldd on the
ESL.so file in the libs/esl/ruby folder and found it wasn't linked
against pthread, so I manually added -lpthread to the Makefile,
recompiled and got it to work. The only other version I tried was the
Perl one and it compiled and worked without needing anything.


From mrene_lists at avgs.ca  Thu Jan  7 20:38:30 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 8 Jan 2010 05:38:30 +0100
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <9AC8A302-DAC3-42FE-B1BB-91EB8A717248@jerris.com>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
	<9AC8A302-DAC3-42FE-B1BB-91EB8A717248@jerris.com>
Message-ID: <065B7BF9-3D3E-4FB5-84DC-ABB7BC24D16E@avgs.ca>

False Answer Supervision.. 200 ok while it's still ringing

Sent from my iPhone

On 2010-01-08, at 4:43 AM, Michael Jerris <mike at jerris.com> wrote:

> what is FAS ?
>
> Mike
>
> On Jan 7, 2010, at 8:05 PM, Ahmed Naji wrote:
>
>> Was just wondering what/if anyone is doing any work on FAS  
>> detection and spoofed ring tones. Be great to discuss some ideas.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From jcasale at activenetwerx.com  Thu Jan  7 20:57:10 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 8 Jan 2010 04:57:10 +0000
Subject: [Freeswitch-users] SPA3102 Help
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9072618@prato.activenetwerx.local>

Reading the wiki, I have Line 1 (the fxs port) configured as ext 1001
and PSTN User (the fxo port) configured as ext 1000. I am a bit unsure
of the dial plan part of the wiki's config?

Under "VoIP-To-PSTN Gateway Setup" I set "Line 1 VoIP Caller DP:" to a
dial plan with (xx.) which dials whatever is sent to it?

Under "PSTN-To-VoIP Gateway Setup" I set "PSTN Caller Default DP:" to a
dial plan with (<:ABCD>S0) where ABCD could be a group dial for example
like 2000 as in the default config?

Do I understand this correctly? I would try this out, but I don't have a
pstn here and want to pass this off configured for delivery tomorrow and
be prepared for any adjustments via ssh if need be...

Thanks!
jlc


From jmesquita at freeswitch.org  Thu Jan  7 21:17:53 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 8 Jan 2010 02:17:53 -0300
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <COL117-W15F7382FBF45603F4000E1B9700@phx.gbl>
References: <012901ca8fe6$e36b71c0$aa425540$@com>
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>
	<014801ca8fee$b75f8780$261e9680$@com>
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>
	<ebbc2d551001071928u422b1884r6ba556b8b7a06e80@mail.gmail.com>
	<COL117-W15F7382FBF45603F4000E1B9700@phx.gbl>
Message-ID: <ebbc2d551001072117i5759a8efn680024657964c964@mail.gmail.com>

You are tha man! Thank you!

On Friday, January 8, 2010, Jeff Lenk <jeff at jefflenk.com> wrote:
>
>
>
>
>
> yep sure thing I will?update/add some of this information?tom. after updating the project files.
>
>
> Date: Fri, 8 Jan 2010 01:28:31 -0200
> From: jmesquita at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Compile error fscomm?
>
> Jeff, any chance we can get this on the wiki?
>
>
> We have created a page here: http://wiki.freeswitch.org/wiki/FSComm
>
>
> I am looking for a Windows machine to do testing on it as well.
>
>
> Regards,Jo?o Mesquita
>
>
> On Fri, Jan 8, 2010 at 1:16 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>
> The windows support is very experimental at this time!
>
> You must manually install http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
>
> Then set the environment variable QTDIR in the environment variables. This can be set from the Computer/Properties/Advanced system settings/Environment Variables/User Variables settings screen.
>
> QTDIR=c:\qt\4.6.0 - or wherever you installed it
>
> then restart VS
>
>
>
>
> Date: Thu, 7 Jan 2010 21:45:56 -0200
> From: jmesquita at freeswitch.org
>
>
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Compile error fscomm?
>
> I don't have a Windows machine to test that. Maybe jlenk could give us a hand since he is the one who has created the visual studio project?
>
>
> Regards,Jo?o Mesquita
> FreeSWITCH? Solutions
> t: +1 (646) 4959927
>
>
> On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:
>
>
>
> No Qt installed. I just checked out from http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into VS2008.
> Do I need to get http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
> Sent: Thursday, January 07, 2010 2:36 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Compile error fscomm?
>
>
>
>
>
>
>
> On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:
>
>
> I just downloaded the fscomm project and loaded it into vs2008. I?ve never programmed in C++ (or c), just C#, so I can?t make anything of the following two messages:
>
> Error????? 1????????????? error PRJ0019: A tool returned an error code from "RCC resources.qrc"? FSComm????????????? FSComm
>
> Warning?????????????? 2????????????? The following environment variables were not found: $(QTDIR)??????????????? Project FSComm
>
> Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.
>
> The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)
>
> Lars
>
>
> Do you have Qt 4.6 installed?
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now.?<http://clk.atdmt.com/GBL/go/196390709/direct/01/>
> _______________________________________________
> FreeSWITCH-users mailing list
> Hotmail: Trusted email with powerful SPAM protection. Sign up now.?<http://clk.atdmt.com/GBL/go/196390707/direct/01/>
>

-- 
Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927


From dujinfang at gmail.com  Thu Jan  7 21:29:14 2010
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 8 Jan 2010 13:29:14 +0800
Subject: [Freeswitch-users] hw:dummy not visible
In-Reply-To: <c55d46951001071948s558060fera2b08fecd667c544@mail.gmail.com>
References: <c55d46951001071948s558060fera2b08fecd667c544@mail.gmail.com>
Message-ID: <23f91031001072129v6a229573t79b7a6c17c89486b@mail.gmail.com>

2010/1/8 Max Bridgewater <max.bridgewater at gmail.com>:
> Hi,
> I got a few more Skypiax questions. Please bear with me.
> 1) Skype should use?hw:dummy?as audio device. But where do I set this on
> Skype? In Options>Sound Devices, the only devices I see are; "Default
> device" and "hdmi". My guess was that?alsa-utils or some other ALSA related
> lib would install this. But it seems this is not happening. Am i missing
> something?

lsmod ?
modprobe snd_dummy ?


> 2) To create the configuration directory, I connect to the FreeNX XServer
> running on my remote machine. I use NX to connect to the remote Xserver. I
> can start all sort of GUI applications this way. But when I try to run
> skypiax_auth, I get the following error message: "Cannot open X Display
> ':0.0', exiting". Any idea?

skypiax_auth :101  #or other display number skype running on

> Thanks again,
> Max.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From nicolas at medularis.com  Thu Jan  7 21:33:30 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 8 Jan 2010 02:33:30 -0300
Subject: [Freeswitch-users] Need to fake ringback
Message-ID: <1b46b4e81001072133h3f4c9bf1g68e25d27fd71edc1@mail.gmail.com>

I'm trying to fake a ringback for leg1 of a two-legged call without success.

I'm doing this with a JS script, originating one leg first, then the
second and executing the bridge application on both afterwards. I
would like to fake a ringback on the first call while they wait for
the second call to connect.

I tried originating the call and then setting the ringback (like the
example here: http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones),
and I also tried setting the ringback in the originate command, but
none of those worked. I even tried using the variable
instant_ringback, but it didn't work either.


This is the code for the first case:

ostr = "{ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]sofia/gateway/mygw/123456789";
session1 = new Session(ostr);
if (session1.ready())
{
session1.execute("set","instant_ringback=%(2000,4000,440.0,480.0)");
}


And this is the originate string for the second case:

var stUsRing = "%(2000,4000,440,480)";
ostr = "{ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,ringback='"+stUsRing+"',medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]sofia/gateway/mygw/123456789";


Thanks for your help!


Nico


From scott.torr.fs at letterboxes.org  Thu Jan  7 22:24:56 2010
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Fri, 08 Jan 2010 17:24:56 +1100
Subject: [Freeswitch-users] Segmentation fault (core dumped) on "shutdown"
	if mod_skypiax loaded
Message-ID: <1262931896.12045.1353566301@webmail.messagingengine.com>

Hi,

I'm getting a Segmentation fault on "shutdown" if mod_skypiax is loaded
on the current build.
The audio on skypiax calls is also extremely choppy.

This only occurred after a "make current" from 15787.

Has anyone else experienced this problem?

VMware Server Version 2.0.1
ubuntu-8.04.3-server-amd64.iso (update/upgrade)
FreeSWITCH Version 1.0.trunk (16172)
skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb
mod_skypiax 

http://jira.freeswitch.org/browse/MODSKYPIAX-68

Regards,
Scott

 


From pmhshz at gmail.com  Thu Jan  7 22:46:41 2010
From: pmhshz at gmail.com (shehzad p)
Date: Thu, 7 Jan 2010 22:46:41 -0800 (PST)
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<B98B6921358D0C49BD21CD023F1F52C302623BE0@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
Message-ID: <27071973.post@talk.nabble.com>




Brian West-3 wrote:
> 
> You could but I think you want to stream RTP to a multicast it would  
> be better off building an rtp format mod so you can record rtp:// 
> x.x.x.x:5000 and play from rtp://y.y.y.y:5000
> 
> /b
> 
> 

I was looking for such functionality, but unfortunately it seems not present
right now, I am willing to build the rtp format mod as described by Brian.,
and will provide back to trunk. 

Although I have modified mod_skel application for use in dialplan in custom
application, I need to have a basic understanding regarding format mod. 
Will anybody please guide me from where to starts?
-- 
View this message in context: http://old.nabble.com/stream-a-file-multicast-with-mod_esf-tp21976696p27071973.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From pmhshz at gmail.com  Thu Jan  7 22:51:42 2010
From: pmhshz at gmail.com (shehzad p)
Date: Thu, 7 Jan 2010 22:51:42 -0800 (PST)
Subject: [Freeswitch-users] Re cording call into existing file
Message-ID: <26975973.post@talk.nabble.com>


Hi,

while recording a file using session_record, can i continue the existing
recorded file?  So that the existing record will remain as it is and new
recording will be added into that file?

Thanks
msp
-- 
View this message in context: http://old.nabble.com/Recording-call-into-existing-file-tp26975973p26975973.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From pmhshz at gmail.com  Thu Jan  7 22:57:30 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Fri, 8 Jan 2010 12:27:30 +0530
Subject: [Freeswitch-users] Re cording call into existing file
In-Reply-To: <191c3a030912310704g5c57d296pbbe553aa46c63c4e@mail.gmail.com>
References: <26975973.post@talk.nabble.com>
	<191c3a030912310704g5c57d296pbbe553aa46c63c4e@mail.gmail.com>
Message-ID: <a792304b1001072257r7c29a43fo4a772c15a092616f@mail.gmail.com>

On Thu, Dec 31, 2009 at 8:34 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> set RECORD_APPEND=true on the channel and all recordings will behave this
> way to formats which support it
> (curently mod_sndfile for WAV etc)
>
>
> On Thu, Dec 31, 2009 at 12:49 AM, shehzad p <pmhshz at gmail.com> wrote:
>
>>
>> Hi,
>>
>> while recording a file using session_record, can i continue the existing
>> recorded file?  So that the existing record will remain as it is and new
>> recording will be added into that file?
>>
>> Thanks
>> msp
>> --
>> View this message in context:
>> http://old.nabble.com/Recording-call-into-existing-file-tp26975973p26975973.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
Great, Thanks Anthony.
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From jingwei.yang at gmail.com  Fri Jan  8 00:42:20 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Fri, 8 Jan 2010 16:42:20 +0800
Subject: [Freeswitch-users]  IVR and TTS
Message-ID: <13529f9d1001080042w420c9f0vdbc5b7a57edd1e6c@mail.gmail.com>

Hi Guys,

I need to implement a function using IVR and TTS. Here's the scenario.

1. User A calls in
2. FS plays a welcome message and directs A to press '1' to continue
3. FS detects A's number and finds A's address from the database and plays
another piece of voice message including the address info just found

I understand this logic can be implemented using javascript. However, in
this scenario, the database is a remote one and the local js has no access
to it. What I'm planning to do is write a Java program, talking to FS via
ESL. Could someone please tell me what event FS will trigger after user A
selects a certain option and how to inform the FS to continue the rest of
IVR menu after finding the address?

Thanks,
-Jingwei
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From mattdfong at gmail.com  Fri Jan  8 01:29:32 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Fri, 8 Jan 2010 16:29:32 +0700
Subject: [Freeswitch-users] Connecting GSM Cards to FreeSWITCH
Message-ID: <4256bf831001080129m69c6b5d1s20557da27f040bca@mail.gmail.com>

Does anyone have any hardware recommendations for setting up 4+ GSM cell
phone lines to work with FreeSWITCH?
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From jonas.gauffin at gmail.com  Fri Jan  8 01:39:46 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Fri, 8 Jan 2010 10:39:46 +0100
Subject: [Freeswitch-users] hangup cause: NORMAL_UNSPECIFIED
Message-ID: <d021d3be1001080139m5b4cd7c6ge87b19e4ad6a62fe@mail.gmail.com>

Hello,

Is it possible to get a more detailed reason (in the log) to
why NORMAL_UNSPECIFIED was returned as hang up cause?

2010-01-08 09:30:50.987200 [DEBUG] sofia.c:3831 Channel
sofia/external/070738xxxx entering state [terminated][904]
2010-01-08 09:30:50.987200 [NOTICE] sofia.c:4461 Hangup
sofia/external/070738xxxx [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED]

Regards,
  Jonas
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From darklion11 at yahoo.com  Fri Jan  8 02:25:10 2010
From: darklion11 at yahoo.com (Edmar Cruz)
Date: Fri, 8 Jan 2010 02:25:10 -0800 (PST)
Subject: [Freeswitch-users] Personal Greeting
In-Reply-To: <1262066443847-4226681.post@n2.nabble.com>
References: <1262066443847-4226681.post@n2.nabble.com>
Message-ID: <27073953.post@talk.nabble.com>


You can set it in the dialplan

<action application="playback" data="yourgreeting.wav" />

For some cases softphones has its own greeting :working:

Hope it can help you..


sharad-5 wrote:
> 
> 
> 
> Hi
> 
> I am new to Freeswitch so my question may be a stupid question.
> 
> I just want to know how to disable the personal greeting to the default
> one. 
> One user has recorded his personal greeting now how can he make this
> default.
> 
> I could not find any option for the same.
> 
> Plz advice.
> 
> Thanks & regards
> Sharad garg
> -- 
> View this message in context:
> http://n2.nabble.com/Personal-Greeting-tp4226681p4226681.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://old.nabble.com/Personal-Greeting-tp26951471p27073953.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From gavin.henry at gmail.com  Fri Jan  8 03:31:57 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 8 Jan 2010 11:31:57 +0000
Subject: [Freeswitch-users] FSComm builds OK on Fedora F-12 with QT4 4.5.3
	but doesn't save SIP account
Message-ID: <13ca621c1001080331l363122a5l3b093adb5782538e@mail.gmail.com>

Hi,

Just a quick one to say this builds ok with:

Compiled FSComm version: 1.0.trunk (16209M)
FreeSWITCH Version 1.0.trunk (16209M)

But it doesn't want to save my SIP account details. How to debug?

Thanks.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From vinuth.madinur at gmail.com  Fri Jan  8 04:16:20 2010
From: vinuth.madinur at gmail.com (Vinuth Madinur)
Date: Fri, 8 Jan 2010 17:46:20 +0530
Subject: [Freeswitch-users] IVR and TTS
In-Reply-To: <13529f9d1001080042w420c9f0vdbc5b7a57edd1e6c@mail.gmail.com>
References: <13529f9d1001080042w420c9f0vdbc5b7a57edd1e6c@mail.gmail.com>
Message-ID: <910309031001080416r6064670bic4e6285494f8d25b@mail.gmail.com>

Hi,

"DTMF" event will be raised in ESL, when "A" presses a key. It'll be raised
for each key pressed.

Alternatively you can use play_and_get_digits.

To continue FS execution after you fetch the address, you just need to
invoke the "speak" command on that socket.

Since FS is handling inbound calls, you can use the outbound event socket,
where a new connection will be opened per call from FS to your java program.

One way to know what events are raised in ESL, you can telnet to 8021 port,
authenticate and send "events plain all" command. Configure FS dialplan for
an extension which will just answer a call when it comes in. Then call this
extension from a softphone, press a key and you'll see the corresponding
event in the telnet console.


Thanks,
Vinuth.



On Fri, Jan 8, 2010 at 2:12 PM, Jingwei Yang <jingwei.yang at gmail.com> wrote:

> Hi Guys,
>
> I need to implement a function using IVR and TTS. Here's the scenario.
>
> 1. User A calls in
> 2. FS plays a welcome message and directs A to press '1' to continue
> 3. FS detects A's number and finds A's address from the database and plays
> another piece of voice message including the address info just found
>
> I understand this logic can be implemented using javascript. However, in
> this scenario, the database is a remote one and the local js has no access
> to it. What I'm planning to do is write a Java program, talking to FS via
> ESL. Could someone please tell me what event FS will trigger after user A
> selects a certain option and how to inform the FS to continue the rest of
> IVR menu after finding the address?
>
> Thanks,
> -Jingwei
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vhatz at kinetix.gr  Fri Jan  8 04:44:21 2010
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Fri, 08 Jan 2010 14:44:21 +0200
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
Message-ID: <4B4728A5.4040805@kinetix.gr>

Hello Ahmed,

I don't think there is a reliable way to detect FAS on a per call basis. 
Even audio detection software can be confused by strange ringtones, 
answering machines, etc.

The most reliable way we have found is to use statistics from CDRs and 
see if the INVITE-to-200(OK) delay averaged over a number of calls 
appears to be too small. If it is, then it is possible that you got FAS 
for those calls.

But this can only tell you if you have been experiencing FAS in past 
calls, ie you will not know you are getting FAS in real time.

It is still useful however: after you detect a possible FAS case via 
statistics you can place a few test calls yourself to verify that there 
actually exists FAS (and this is the only information that a carrier 
will accept in a trouble ticket to prove to them that they actually give 
you FAS).

I hope this helps.

Best regards,
Vlasis Hatzistavrou.


On 8/1/10 3:05 ??, Ahmed Naji wrote:
> Hi everyone,
>
> Was just wondering what/if anyone is doing any work on FAS detection 
> and spoofed ring tones. Be great to discuss some ideas.
>
> Regards,
>
> Ahmed.
>
> -- 
> Ahmed Naji
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From codecomplete at free.fr  Fri Jan  8 05:03:00 2010
From: codecomplete at free.fr (Fred-145)
Date: Fri, 8 Jan 2010 05:03:00 -0800 (PST)
Subject: [Freeswitch-users]  IP PBX and NAT firewalls
Message-ID: <27075600.post@talk.nabble.com>


Hello,

    I read a couple of thorough articles on SIP, and I'd like to make sure I
got things right when it comes to using SIP with NAT routers.
   
I know that, ideally, the IP PBX should be located in the DMZ to void
NAT-related issues in SIP, but SOHO routers don't necessarily support this,
so I'll assume that the SIP caller "Alice" and the IP PBX (eg. Freeswitch or
Asterisk) server are located in a non-routable, private LAN, while the
remote callee "Bob" is located on the Internet (either behind their own NAT
router, or connected with a public, routable address). The SIP phone of
Alice and Bob are both logged on to the Freeswitch server:

http://img46.imageshack.us/img46/5120/sipnatrouters.jpg

1. When Alice wants to call Bob, her SIP phone sends an SIP packet to the
Freeswitch server with her private IP address and a UDP port that it opened
to let incoming RTP packets from Bob

2. Freeswitch rings Bob's phone through the UDP port is used to register
with Freeswitch (usually, UDP5060). Bob's phone replies to Freeswitch with
his public IP address and the RTP port it chose to receive voice packets
from Alice

3. Once Bob picks up the phone, RTP voice packets flow directly between
Alice and Bob, while Freeswitch remains in the loop to handle call signaling
such as closing the connection when someone hangs up the call.

Provided this is how things work... there are three issues when one or all
SIP end-points are located in a (different) private LAN:
1. End-points use their private IP and a private UDP port for RTP. A server
has to translate this into a routable IP address, and...
2. it must negotiate with the NAT firewall to make sure this RTP port is
available, and if not, open some other port, and...
3. the server must rewrite the SDP packet to use this public port

I have a couple of questions:

1. Can Freeswitch/Asterisk handle this rewriting/negotiation?

2. Provided the NAT firewall doesn't support UPnP/NAT-PMP, does it mean I
must a) enable STUN in Freeswitch, b) set SIP end-points so that they use a
fixed port for RTP, and c) configure the NAT firewall to map this UDP port
to point to the SIP end-point?

3. Should SIP end-points be configured to use STUN/NAT, or should I let the
server handle the IP/port rewriting itself?

Thank you for any help.
-- 
View this message in context: http://old.nabble.com/IP-PBX-and-NAT-firewalls-tp27075600p27075600.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From steve at justfone.com  Fri Jan  8 05:17:53 2010
From: steve at justfone.com (Steven Brown)
Date: Fri, 8 Jan 2010 13:17:53 +0000
Subject: [Freeswitch-users] Connecting GSM Cards to FreeSWITCH
Message-ID: <3e6d7b0c1001080517i7af2a275oda7baac9429a9c29@mail.gmail.com>

Hi,

I have used Portech gateways with good results, you can get cards and
external gateways with various numbers of GSM channels, see

http://www.portech.com.tw/p3-product1_1.asp?Pid=13

Regards

Steve


From gmaruzz at celliax.org  Fri Jan  8 05:23:46 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 8 Jan 2010 14:23:46 +0100
Subject: [Freeswitch-users] Connecting GSM Cards to FreeSWITCH
In-Reply-To: <3e6d7b0c1001080517i7af2a275oda7baac9429a9c29@mail.gmail.com>
References: <3e6d7b0c1001080517i7af2a275oda7baac9429a9c29@mail.gmail.com>
Message-ID: <7b197bef1001080523g31c28424i3a1ba13049421642@mail.gmail.com>

Hi Mattew,

in a short while (before monday) will be available mod_gsmopen that
maybe can fit your needs.

http://wiki.freeswitch.org/wiki/GSMopen

-giovanni

On Fri, Jan 8, 2010 at 2:17 PM, Steven Brown <steve at justfone.com> wrote:
> Hi,
>
> I have used Portech gateways with good results, you can get cards and
> external gateways with various numbers of GSM channels, see
>
> http://www.portech.com.tw/p3-product1_1.asp?Pid=13
>
> Regards
>
> Steve
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Fri Jan  8 05:38:19 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 8 Jan 2010 14:38:19 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <20100107182731.GL5547@tamay-dogan.net>
References: <20100103184636.GW5547@tamay-dogan.net>
	<20100106135210.GG5547@tamay-dogan.net> 
	<CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>
	<7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com> 
	<20100106171413.GI5547@tamay-dogan.net>
	<7b197bef1001060939rd1e08f3h2cb82cd89e50225e@mail.gmail.com> 
	<20100107182731.GL5547@tamay-dogan.net>
Message-ID: <7b197bef1001080538u5ed67082r6df84d287c0c06c1@mail.gmail.com>

On Thu, Jan 7, 2010 at 7:27 PM, Michelle Konzack
<linux4michelle at tamay-dogan.net> wrote:
>> it supports all modules that accepts standard ETSI AT-GSM commands
>> (so, let's say all of them).
>
> Currently I havepayed over 20.000 Euro for ETSI and ANS specs ?and ?have
> currently not the money to buy more... ?Are the ETSI ?AT-GSM ?specs ?are
> free available? ?-- ?If yes, I need them!

yes is part of the GSM specifications that are freely available. Just
google for them. Anyway, any GSM modem or module (but not all
cellphones) supports at least a fair share of those specs.

>
>> Maybe I do not understand the second question, what do you means for
>> Endpoints?
>>
>> If you're talking usb endpoints, you'll need a modem endpoint (that
>> can be seen as a serial port), and (if you need audio, eg not just
>> SMSs but voice calls too) you need an audio endpoint (that can be seen
>> as a soundcard).
>
> I mean the USB-Endpoints...
>
> If you have an USB-Microcontroller where the USB port is ?a ?device, ?it
> identify it self over the Endpoint 0 and is for us non usable.
>
> And no it comes, where I haveproblems to understnd HOW the GSM Modem ?is
> working but I will assume some things:
>
> The EP1 of the USB-pot is configured for bidirectional Data transmission
> andwill controll out Device and is normaly ?configured ?as ?/dev/ttyUSB0
> and this is, where we use the AT commands to ?get ?infos ?from ?the ?GSM
> modem/cellphone and send/receive SMS.
>
> Now We need EP2 and configure it as streaming output for the Audio port.
>
> EP3 would be the streaming input for the Audio Port.
>
> is this right up to here?
>
> Then, EP4 would be the bidirectinal dataport ?for ?the ?UMTS ?and ?HS*PA
> Tranceiver, since it is entirely independant from the rest ?of ?the ?GSM
> modem/cellphone.
>
> Is this right?

Is completely up to the implementation.

I suggest you use lsusb with full debug/verbosity turned on, it will
tell you (almost) all.


>
> If yes, then it is easier as I was thinking...
>
>
>> Many modules and cellphones can be seen as HDSPA or GPRS modems, just
>> check their specs.
>
> My "Nokia 6120 classic" has in total 13 endpoints... ?Hell, where can ?I
> get an USB-Microcontroller which support this mass of USB Endpoints?
>
> Most ARM9/11 support not more then 7 or 9. ?:-(

I think you just needs to interface the modem endpoint and the audio
endpoint. The others are probably Human Interfaces for volume, color,
keyboard, whatever. You control the full phone features through AT
commands exchanged through the modem endpoint.


>
>> For audio, if the module/cellphone/modem does not offer an audio usb
>> endpoint (eg cannot be seen as a soundcard) one trick is to connect
>> the headset jack to an usb soundcard (you can find soundcard with for
>> factor like a dongle based on cm-108 chipset for under $10). I'll
>> publish the schema of the cable needed from hadset jack in the
>> phone/module to the usb soundcard).
>>
>> If I have not get what you asked, please explain more your question.
>
> Most important things are the above desibed understanding ?problem ?with
> the USB Endpoints
>
> I have the HSPA and GS frontends (Maxim and Infineon chips) here and ?my
> selfemade simple GSM/GPRS cell-phone is already working, ?but ?has ?less
> functionality as the cell-phones from Year 2000 ?:-D

you don't need them all (at least if you don't want to make dirty pics
with a kludge full of wires... hey, that can be arousing! :))

I suggest you use a ready-made GSM/GPRS/HDSPA/whatever module, that
contains all that you need and is available from Chinese suppliers for
very low prices. You can find some module that allows you to directly
tap in the GSM pcm audio stream, that would means you will not need to
sample and convert from analog to digital (so, no cpu power at all, no
dsp, no nothing).


If you would like to keep me in the loop, I would like to know how you progress.

Happy hacking

-giovanni

>
> Hey, ist is my first experience of developing GSM stuff.
>
> I an to develop a VERY simple GSM/UMTS/HSPA USB-Modem which do ?its ?job
> without balast.
>
> So if someone can help me with infos, I am very open...
>
> I prefer FreeSwitch over Asterisk which froze in the ?last ?2 ?years ?to
> many times in situations where it should not freeze, exspecialy if ?I ?a
> call a Chip-Manfacturer (Maxim/TI) Tech-Support. ?-- ?It is not funny!
>
> Thanks, Greetings and nice Day/Evening
> ? ?Michelle Konzack
> ? ?Systemadministrator
> ? ?Electronic Engineer
> ? ?Tamay Dogan Network
> ? ?Debian GNU/Linux Consultant
>
> --
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
> ##################### Debian GNU/Linux Consultant #####################
> <http://www.tamay-dogan.net/> ? ? ? ? ? ? ? ? Michelle Konzack
> <http://www.can4linux.org/> ? ? ? ? ? ? ? ? ? Apt. 917
> <http://www.flexray4linux.org/> ? ? ? ? ? ? ? 50, rue de Soultz
> Jabber linux4michelle at jabber.ccc.de ? ? ? ? ? 67100 Strabourg/France
> IRC ? ?#Debian (irc.icq.com) ? ? ? ? ? ? ? ? ?Tel. DE: +49 177 9351947
> ICQ ? ?#328449886 ? ? ? ? ? ? ? ? ? ? ? ? ? ? Tel. FR: +33 ?6 ?61925193
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From kdjakovic at hotmail.com  Fri Jan  8 06:03:55 2010
From: kdjakovic at hotmail.com (katarina djakovic)
Date: Fri, 8 Jan 2010 15:03:55 +0100
Subject: [Freeswitch-users] How to suspend certain users from the
	system	using flush_inbound_reg
In-Reply-To: <59CAAB6B-5259-4B78-9C9C-676ACF04D6B1@jerris.com>
References: <SNT137-w54C41FF82F0E62B26415E9B3750@phx.gbl>,
	<59CAAB6B-5259-4B78-9C9C-676ACF04D6B1@jerris.com>
Message-ID: <SNT137-w3221FCED6EE7F190EDCC65B3700@phx.gbl>


Dear all,

 

thanks for your suggestions, they helped us to understand what was happening. 

 

The mistake was ours, we had auth-calls parameter in the profile set to false, and that was the cause of the problem, since the calls went through regardless of the directory settings.

 

Thanks again,

Katarina
 


From: mike at jerris.com
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 3 Jan 2010 20:24:15 -0500
Subject: Re: [Freeswitch-users] How to suspend certain users from the system using flush_inbound_reg


First a note.  Registration and authentication are completely different. Removing the registration has to do with the switch knowin where to send the calls and nothing to do with auth for receiving calls.  There is one caveat to this.  We do support nonce count, and it could be using the auth from the previous registration that is still valid.  Double check the nc from the registrations and the call and see if that rings true.  We may want to add something to explicitly expire the nonce when youflush reg but I need some confirmation on that first.  Otherwise the other responces seem to cover the possibilities.  Crank up the debug and check sip trace for more details on what is allowing the call through and report back.


Mike

On Jan 3, 2010, at 11:50 AM, katarina djakovic <kdjakovic at hotmail.com> wrote:




Hi,
 
we are trying to figure out how to suspend certain subscribers from our system and we have some problems with removing thier 
registrations. The UAs are ATAs.
 
This is what we do:
 
1) We remove the subscriber extension from the conf\directory .xml files
2) We do reloadxml
3) We flush user's registration with flush_inbound_reg 
 
but, the users are still able to make calls as if they were still registered. To make it clearer, their registrations are removed from 
the registration list (checked with sofia status), but they system still accepts the calls from them. From this, it seems that if ATA is never rebooted - we are not able to ban these users from the system.
 
Only after the ATA is rebooted user is not able to make calls any more, as the ATA can not register any more - since they users are 
removed from the directory. But before we reboot ATA everything works as nothing had been done.
 
Does anyone have an idea what are we doing wrong? We expect that after the registration is removed from the FS the UA should not be 
able to make a call but this is not what happnes.
 
Can anybody help please?
Thanks,
Katarina



Keep your friends updated? even when you?re not signed in. 

_______________________________________________
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_________________________________________________________________
Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you.
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From max.bridgewater at gmail.com  Fri Jan  8 06:14:39 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 8 Jan 2010 09:14:39 -0500
Subject: [Freeswitch-users] hw:dummy not visible
In-Reply-To: <23f91031001072129v6a229573t79b7a6c17c89486b@mail.gmail.com>
References: <c55d46951001071948s558060fera2b08fecd667c544@mail.gmail.com>
	<23f91031001072129v6a229573t79b7a6c17c89486b@mail.gmail.com>
Message-ID: <c55d46951001080614u3024b2a3t71f8d8e066ebe13a@mail.gmail.com>

Wow, it works like a charm. And of course, your two suggestions were spot
on.

Gracie Mille.

Max.


On Fri, Jan 8, 2010 at 12:29 AM, Seven Du <dujinfang at gmail.com> wrote:

> 2010/1/8 Max Bridgewater <max.bridgewater at gmail.com>:
> > Hi,
> > I got a few more Skypiax questions. Please bear with me.
> > 1) Skype should use hw:dummy as audio device. But where do I set this on
> > Skype? In Options>Sound Devices, the only devices I see are; "Default
> > device" and "hdmi". My guess was that alsa-utils or some other ALSA
> related
> > lib would install this. But it seems this is not happening. Am i missing
> > something?
>
> lsmod ?
> modprobe snd_dummy ?
>
>
> > 2) To create the configuration directory, I connect to the FreeNX XServer
> > running on my remote machine. I use NX to connect to the remote Xserver.
> I
> > can start all sort of GUI applications this way. But when I try to run
> > skypiax_auth, I get the following error message: "Cannot open X Display
> > ':0.0', exiting". Any idea?
>
> skypiax_auth :101  #or other display number skype running on
>
> > Thanks again,
> > Max.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at celliax.org  Fri Jan  8 06:26:14 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 8 Jan 2010 15:26:14 +0100
Subject: [Freeswitch-users] hw:dummy not visible
In-Reply-To: <c55d46951001080614u3024b2a3t71f8d8e066ebe13a@mail.gmail.com>
References: <c55d46951001071948s558060fera2b08fecd667c544@mail.gmail.com> 
	<23f91031001072129v6a229573t79b7a6c17c89486b@mail.gmail.com> 
	<c55d46951001080614u3024b2a3t71f8d8e066ebe13a@mail.gmail.com>
Message-ID: <7b197bef1001080626y4ca13f2br6d5aceed9c2db9a8@mail.gmail.com>

On Fri, Jan 8, 2010 at 3:14 PM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Wow, it works like a charm. And of course, your two suggestions were spot
> on.

Seven is an authority on using mod_skypiax!

-giovanni


>
> Gracie Mille.
>
> Max.
>
>
> On Fri, Jan 8, 2010 at 12:29 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> 2010/1/8 Max Bridgewater <max.bridgewater at gmail.com>:
>> > Hi,
>> > I got a few more Skypiax questions. Please bear with me.
>> > 1) Skype should use?hw:dummy?as audio device. But where do I set this on
>> > Skype? In Options>Sound Devices, the only devices I see are; "Default
>> > device" and "hdmi". My guess was that?alsa-utils or some other ALSA
>> > related
>> > lib would install this. But it seems this is not happening. Am i missing
>> > something?
>>
>> lsmod ?
>> modprobe snd_dummy ?
>>
>>
>> > 2) To create the configuration directory, I connect to the FreeNX
>> > XServer
>> > running on my remote machine. I use NX to connect to the remote Xserver.
>> > I
>> > can start all sort of GUI applications this way. But when I try to run
>> > skypiax_auth, I get the following error message: "Cannot open X Display
>> > ':0.0', exiting". Any idea?
>>
>> skypiax_auth :101 ?#or other display number skype running on
>>
>> > Thanks again,
>> > Max.
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Fri Jan  8 06:29:51 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 8 Jan 2010 15:29:51 +0100
Subject: [Freeswitch-users] Skypiax on CentOS
In-Reply-To: <c55d46951001071903q5a23639ehee09da7640351a74@mail.gmail.com>
References: <c55d46951001071836i55516863r359bb56695744054@mail.gmail.com> 
	<6b65470d1001071852i752b8306r51a48576090b4be@mail.gmail.com> 
	<c55d46951001071903q5a23639ehee09da7640351a74@mail.gmail.com>
Message-ID: <7b197bef1001080629h6b812e0dt67d66dcb740d21a7@mail.gmail.com>

I would use the version signaled by William (
http://download.skype.com/linux/skype-2.0.0.72-centos.i586.rpm ) or,
better yet, if you can find the static version of 2.0.0.72.

If you, or others, find that static version, please post the link here.

Thanks,

-giovanni


On Fri, Jan 8, 2010 at 4:03 AM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Thanks you very much guys.
> Ad Brian: I tried the static binary build but was blocked by a qt4-x11
> missing and I couldn't find an obvious solution to that. Hence my ?inquiry
> in the group before spending time on things that other people potentially
> already solved.
> max.
>
> On Thu, Jan 7, 2010 at 9:52 PM, William Suffill <william.suffill at gmail.com>
> wrote:
>>
>> Skype appears to be pushing a new version on the linux side but the old
>> packages are still available but not linked anywhere.
>>
>> http://download.skype.com/linux/skype-2.0.0.72-centos.i586.rpm
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From Claudio.Cavalera at italtel.it  Fri Jan  8 06:30:05 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Fri, 8 Jan 2010 15:30:05 +0100
Subject: [Freeswitch-users] Codecs and things
In-Reply-To: <7A65B817-E490-4582-8D43-8531FFA61CC4@freeswitch.org>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F362C3@BESONE.corp.dom>

Hi Bkw,
my doubt is not the price but about the availability of the official
mod_G729 to be used on embedded hardware.
Thanks,
Claudio


________________________________

	From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West
	Sent: Thursday, January 07, 2010 5:57 PM
	To: freeswitch-users at lists.freeswitch.org
	Subject: Re: [Freeswitch-users] Codecs and things
	
	
	You usually still have to pay a license even if you buy a DSP
that is capable of doing it.   

	/b

	On Jan 7, 2010, at 10:46 AM, Cavalera Claudio Luigi wrote:


		Hello Steve,
		from what you have written it seems very unlikely that
we are gonna buy
		the official G.729 codec for embedded hardware?
		I don't know much about it but would a MIPS32 24kf be
enough? Just
		speculating from here
	
http://www.mips.com/products/processors/32-64-bit-cores/mips32-24k/
		
		Thanks,
		Claudio



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From brian at freeswitch.org  Fri Jan  8 06:41:55 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 08:41:55 -0600
Subject: [Freeswitch-users] Need to fake ringback
In-Reply-To: <1b46b4e81001072133h3f4c9bf1g68e25d27fd71edc1@mail.gmail.com>
References: <1b46b4e81001072133h3f4c9bf1g68e25d27fd71edc1@mail.gmail.com>
Message-ID: <91E856D5-B499-4C5F-A6A3-E2BB36C926C6@freeswitch.org>

Are you using proxy media?

/b

On Jan 7, 2010, at 11:33 PM, Nicolas Brenner wrote:

> I'm trying to fake a ringback for leg1 of a two-legged call without success.



From brian at freeswitch.org  Fri Jan  8 06:43:07 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 08:43:07 -0600
Subject: [Freeswitch-users] hangup cause: NORMAL_UNSPECIFIED
In-Reply-To: <d021d3be1001080139m5b4cd7c6ge87b19e4ad6a62fe@mail.gmail.com>
References: <d021d3be1001080139m5b4cd7c6ge87b19e4ad6a62fe@mail.gmail.com>
Message-ID: <583FB3F4-9CFC-48A8-ADE0-30FD43F8F1F1@freeswitch.org>

Not without the siptrace and sofia loglevel all 9

/b

On Jan 8, 2010, at 3:39 AM, Jonas Gauffin wrote:

> Hello,
> 
> Is it possible to get a more detailed reason (in the log) to why NORMAL_UNSPECIFIED was returned as hang up cause?
> 
> 2010-01-08 09:30:50.987200 [DEBUG] sofia.c:3831 Channel sofia/external/070738xxxx entering state [terminated][904]
> 2010-01-08 09:30:50.987200 [NOTICE] sofia.c:4461 Hangup sofia/external/070738xxxx [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED]



From brian at freeswitch.org  Fri Jan  8 06:51:44 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 08:51:44 -0600
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <27071973.post@talk.nabble.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<B98B6921358D0C49BD21CD023F1F52C302623BE0@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
Message-ID: <378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>

www.bkw.org/mod_rtp_stream.tgz is a skel I started on.. You can take that and finish it up if you like.

/b

On Jan 8, 2010, at 12:46 AM, shehzad p wrote:

> I was looking for such functionality, but unfortunately it seems not present
> right now, I am willing to build the rtp format mod as described by Brian.,
> and will provide back to trunk. 
> 
> Although I have modified mod_skel application for use in dialplan in custom
> application, I need to have a basic understanding regarding format mod. 
> Will anybody please guide me from where to starts?



From max.bridgewater at gmail.com  Fri Jan  8 07:15:00 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 8 Jan 2010 10:15:00 -0500
Subject: [Freeswitch-users] Skypiax on CentOS
In-Reply-To: <7b197bef1001080629h6b812e0dt67d66dcb740d21a7@mail.gmail.com>
References: <c55d46951001071836i55516863r359bb56695744054@mail.gmail.com>
	<6b65470d1001071852i752b8306r51a48576090b4be@mail.gmail.com>
	<c55d46951001071903q5a23639ehee09da7640351a74@mail.gmail.com>
	<7b197bef1001080629h6b812e0dt67d66dcb740d21a7@mail.gmail.com>
Message-ID: <c55d46951001080715x1401b6f5tdde6c99b3c60655c@mail.gmail.com>

That's indeed what I used. Thanks everybody.

On Fri, Jan 8, 2010 at 9:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> I would use the version signaled by William (
> http://download.skype.com/linux/skype-2.0.0.72-centos.i586.rpm ) or,
> better yet, if you can find the static version of 2.0.0.72.
>
> If you, or others, find that static version, please post the link here.
>
> Thanks,
>
> -giovanni
>
>
> On Fri, Jan 8, 2010 at 4:03 AM, Max Bridgewater
> <max.bridgewater at gmail.com> wrote:
> > Thanks you very much guys.
> > Ad Brian: I tried the static binary build but was blocked by a qt4-x11
> > missing and I couldn't find an obvious solution to that. Hence my
>  inquiry
> > in the group before spending time on things that other people potentially
> > already solved.
> > max.
> >
> > On Thu, Jan 7, 2010 at 9:52 PM, William Suffill <
> william.suffill at gmail.com>
> > wrote:
> >>
> >> Skype appears to be pushing a new version on the linux side but the old
> >> packages are still available but not linked anywhere.
> >>
> >> http://download.skype.com/linux/skype-2.0.0.72-centos.i586.rpm
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.bridgewater at gmail.com  Fri Jan  8 07:19:52 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 8 Jan 2010 10:19:52 -0500
Subject: [Freeswitch-users] Skypiax and Socket API
Message-ID: <c55d46951001080719p743df4faufe6d3964060392cb@mail.gmail.com>

Hey,

I see that it's possible to send chat messages to Skype users using sk on
top of socket api and skypiax. But how do I received chat messages via the
socket api? Are there events generated by Skypiax? If so, do we have a list
somewhere?

Thanks,
Max.
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From pmhshz at gmail.com  Fri Jan  8 07:25:55 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Fri, 8 Jan 2010 20:55:55 +0530
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<B98B6921358D0C49BD21CD023F1F52C302623BE0@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
Message-ID: <a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>

Thanks Brian,

Great... Let me start the developing it.
Although I am analyzing the mod_local_stream, to understand exact working of
format module, Will anybody please let me know any similar thing, I can take
reference?


On Fri, Jan 8, 2010 at 8:21 PM, Brian West <brian at freeswitch.org> wrote:

> www.bkw.org/mod_rtp_stream.tgz is a skel I started on.. You can take that
> and finish it up if you like.
>
> /b
>
> On Jan 8, 2010, at 12:46 AM, shehzad p wrote:
>
> > I was looking for such functionality, but unfortunately it seems not
> present
> > right now, I am willing to build the rtp format mod as described by
> Brian.,
> > and will provide back to trunk.
> >
> > Although I have modified mod_skel application for use in dialplan in
> custom
> > application, I need to have a basic understanding regarding format mod.
> > Will anybody please guide me from where to starts?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Fri Jan  8 07:33:48 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 8 Jan 2010 10:33:48 -0500
Subject: [Freeswitch-users] Ruby ESL missing pthread
In-Reply-To: <1b46b4e81001071958i17ccb6bey6eeecd9d94eb8438@mail.gmail.com>
References: <1b46b4e81001071958i17ccb6bey6eeecd9d94eb8438@mail.gmail.com>
Message-ID: <8A3B5711-727D-4BDB-97AE-7AD20C4528F7@jerris.com>

Please open up a bug on jira for me for this issue.

Mike

On Jan 7, 2010, at 10:58 PM, Nicolas Brenner <nicolas at medularis.com>  
wrote:

> I followed the instructions on the wiki to compile the Ruby version of
> ESL, but then when I tried to run the examples, I kept getting a
> "undefined symbol: pthread_mutexattr_init" error. I ran ldd on the
> ESL.so file in the libs/esl/ruby folder and found it wasn't linked
> against pthread, so I manually added -lpthread to the Makefile,
> recompiled and got it to work. The only other version I tried was the
> Perl one and it compiled and worked without needing anything.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From mike at jerris.com  Fri Jan  8 07:47:34 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 8 Jan 2010 10:47:34 -0500
Subject: [Freeswitch-users] Personal Greeting
In-Reply-To: <27073953.post@talk.nabble.com>
References: <1262066443847-4226681.post@n2.nabble.com>
	<27073953.post@talk.nabble.com>
Message-ID: <53FBFF55-D94D-4BCA-81EC-230F00DDC344@jerris.com>

Huh?  What does this have to do with his question?

On Jan 8, 2010, at 5:25 AM, Edmar Cruz <darklion11 at yahoo.com> wrote:

>
> You can set it in the dialplan
>
> <action application="playback" data="yourgreeting.wav" />
>
> For some cases softphones has its own greeting :working:
>
> Hope it can help you..
>
>
> sharad-5 wrote:
>>
>>
>>
>> Hi
>>
>> I am new to Freeswitch so my question may be a stupid question.
>>
>> I just want to know how to disable the personal greeting to the  
>> default
>> one.
>> One user has recorded his personal greeting now how can he make this
>> default.
>>
>> I could not find any option for the same.
>>
>> Plz advice.
>>
>> Thanks & regards
>> Sharad garg
>> -- 
>> View this message in context:
>> http://n2.nabble.com/Personal-Greeting-tp4226681p4226681.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>>
>
> -- 
> View this message in context: http://old.nabble.com/Personal-Greeting-tp26951471p27073953.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From gmaruzz at celliax.org  Fri Jan  8 07:47:43 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 8 Jan 2010 16:47:43 +0100
Subject: [Freeswitch-users] Skypiax and Socket API
In-Reply-To: <c55d46951001080719p743df4faufe6d3964060392cb@mail.gmail.com>
References: <c55d46951001080719p743df4faufe6d3964060392cb@mail.gmail.com>
Message-ID: <7b197bef1001080747u7ea7852l4cf776e02ec07f16@mail.gmail.com>

yes, it uses standard MESSAGE messages (chat api)

to test it:

telnet localhost 8021

-> auth ClueCon

-> events plain message

or, if you prefere xml:

-> events xml message

this will subscribe to the events of type message.

You can send messages using the standard chat API of FS (like with
JINGLE and sofia/SIP/SIMPLE)

-giovanni


On Fri, Jan 8, 2010 at 4:19 PM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Hey,
>
> I see that it's possible to send chat messages to Skype users using sk on
> top of socket api and skypiax. But how do I received chat messages via the
> socket api? Are there events generated by Skypiax? If so, do we have a list
> somewhere?
>
> Thanks,
> Max.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From a.alalousi at gmail.com  Fri Jan  8 08:07:52 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Fri, 8 Jan 2010 16:07:52 +0000
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <4B4728A5.4040805@kinetix.gr>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
	<4B4728A5.4040805@kinetix.gr>
Message-ID: <cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>

Hi Valsis,

Thanks for this.

My line of thought is to tone-detect secondary ringing tones post 200(OK) to
detect FAS (False Answer Supervision). This should eliminate at least a good
proportion of calls, and it can be done real time through a
script/modules/...etc.

Working along this thought, at least you are minimising the hit cost-wise to
a few seconds at most.

As to answering machines and fake conferences, fake network messages ...etc,
one can possibly use voice detection, perhaps with heuristic and statistical
training.

Just a thought ..

Regards,

Ahmed.


2010/1/8 Vlasis Hatzistavrou (KTI) <vhatz at kinetix.gr>

>  Hello Ahmed,
>
> I don't think there is a reliable way to detect FAS on a per call basis.
> Even audio detection software can be confused by strange ringtones,
> answering machines, etc.
>
> The most reliable way we have found is to use statistics from CDRs and see
> if the INVITE-to-200(OK) delay averaged over a number of calls appears to be
> too small. If it is, then it is possible that you got FAS for those calls.
>
> But this can only tell you if you have been experiencing FAS in past calls,
> ie you will not know you are getting FAS in real time.
>
> It is still useful however: after you detect a possible FAS case via
> statistics you can place a few test calls yourself to verify that there
> actually exists FAS (and this is the only information that a carrier will
> accept in a trouble ticket to prove to them that they actually give you
> FAS).
>
> I hope this helps.
>
> Best regards,
> Vlasis Hatzistavrou.
>
>
> On 8/1/10 3:05 ??, Ahmed Naji wrote:
>
> Hi everyone,
>
> Was just wondering what/if anyone is doing any work on FAS detection and
> spoofed ring tones. Be great to discuss some ideas.
>
> Regards,
>
> Ahmed.
>
> --
> Ahmed Naji
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From oscav at hotmail.fr  Fri Jan  8 08:14:26 2010
From: oscav at hotmail.fr (Oscav)
Date: Fri, 8 Jan 2010 08:14:26 -0800 (PST)
Subject: [Freeswitch-users] Failed to connect to a SKYPE API
In-Reply-To: <27062783.post@talk.nabble.com>
References: <27062783.post@talk.nabble.com>
Message-ID: <27078464.post@talk.nabble.com>


Im' running FS on windows server 2003 64bits


Oscav wrote:
> 
> Hi,
> 
> I'm trying to use to SkypeIAX. When I load the mod_skypiax, I got the
> following error :
> 
>  Failed to connect to a SKYPE API for interface_id=1, no SKYPE client
> running, please (re)start Skype client. Skypiax exiting
> 
> Skype is running with the correct account and skypiax.conf use the same
> account. I was expecting a permission request from the Skype user but
> nothing happens.
> 
> Somebody knows how I can solve this ??
> 
> Many thanks.
> 

-- 
View this message in context: http://old.nabble.com/Failed-to-connect-to-a-SKYPE-API-tp27062783p27078464.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From max.bridgewater at gmail.com  Fri Jan  8 08:15:56 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 8 Jan 2010 11:15:56 -0500
Subject: [Freeswitch-users] Skypiax and Socket API
In-Reply-To: <7b197bef1001080747u7ea7852l4cf776e02ec07f16@mail.gmail.com>
References: <c55d46951001080719p743df4faufe6d3964060392cb@mail.gmail.com>
	<7b197bef1001080747u7ea7852l4cf776e02ec07f16@mail.gmail.com>
Message-ID: <c55d46951001080815j22388bcai471b9986747e631b@mail.gmail.com>

Awesome; I will try it and report next week on my experience.

On Fri, Jan 8, 2010 at 10:47 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> yes, it uses standard MESSAGE messages (chat api)
>
> to test it:
>
> telnet localhost 8021
>
> -> auth ClueCon
>
> -> events plain message
>
> or, if you prefere xml:
>
> -> events xml message
>
> this will subscribe to the events of type message.
>
> You can send messages using the standard chat API of FS (like with
> JINGLE and sofia/SIP/SIMPLE)
>
> -giovanni
>
>
> On Fri, Jan 8, 2010 at 4:19 PM, Max Bridgewater
> <max.bridgewater at gmail.com> wrote:
> > Hey,
> >
> > I see that it's possible to send chat messages to Skype users using sk on
> > top of socket api and skypiax. But how do I received chat messages via
> the
> > socket api? Are there events generated by Skypiax? If so, do we have a
> list
> > somewhere?
> >
> > Thanks,
> > Max.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Fri Jan  8 08:20:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Jan 2010 10:20:14 -0600
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
	<4B4728A5.4040805@kinetix.gr>
	<cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>
Message-ID: <191c3a031001080820i426194cw58b8a2cf8b123493@mail.gmail.com>

you could use the existing tone_detect app after the false answer to xfer to
an extension of your choice if ring tone was detected.


2010/1/8 Ahmed Naji <a.alalousi at gmail.com>

> Hi Valsis,
>
> Thanks for this.
>
> My line of thought is to tone-detect secondary ringing tones post 200(OK)
> to detect FAS (False Answer Supervision). This should eliminate at least a
> good proportion of calls, and it can be done real time through a
> script/modules/...etc.
>
> Working along this thought, at least you are minimising the hit cost-wise
> to a few seconds at most.
>
> As to answering machines and fake conferences, fake network messages
> ...etc, one can possibly use voice detection, perhaps with heuristic and
> statistical training.
>
> Just a thought ..
>
> Regards,
>
> Ahmed.
>
>
> 2010/1/8 Vlasis Hatzistavrou (KTI) <vhatz at kinetix.gr>
>
>  Hello Ahmed,
>>
>> I don't think there is a reliable way to detect FAS on a per call basis.
>> Even audio detection software can be confused by strange ringtones,
>> answering machines, etc.
>>
>> The most reliable way we have found is to use statistics from CDRs and see
>> if the INVITE-to-200(OK) delay averaged over a number of calls appears to be
>> too small. If it is, then it is possible that you got FAS for those calls.
>>
>> But this can only tell you if you have been experiencing FAS in past
>> calls, ie you will not know you are getting FAS in real time.
>>
>> It is still useful however: after you detect a possible FAS case via
>> statistics you can place a few test calls yourself to verify that there
>> actually exists FAS (and this is the only information that a carrier will
>> accept in a trouble ticket to prove to them that they actually give you
>> FAS).
>>
>> I hope this helps.
>>
>> Best regards,
>> Vlasis Hatzistavrou.
>>
>>
>> On 8/1/10 3:05 ??, Ahmed Naji wrote:
>>
>> Hi everyone,
>>
>> Was just wondering what/if anyone is doing any work on FAS detection and
>> spoofed ring tones. Be great to discuss some ideas.
>>
>> Regards,
>>
>> Ahmed.
>>
>> --
>> Ahmed Naji
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From gmaruzz at celliax.org  Fri Jan  8 08:20:47 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 8 Jan 2010 17:20:47 +0100
Subject: [Freeswitch-users] Failed to connect to a SKYPE API
In-Reply-To: <27078464.post@talk.nabble.com>
References: <27062783.post@talk.nabble.com> <27078464.post@talk.nabble.com>
Message-ID: <7b197bef1001080820m371dd494v4c8fd55a07e0c6a0@mail.gmail.com>

you are probably running skype and FS as different windows users
(maybe skype as yourself and fs as administrator, or local_account, or
whatever).

both skype client AND freeSWITCH need to be running as the same
windows user, so FS can find the Skype client

-giovanni


On Fri, Jan 8, 2010 at 5:14 PM, Oscav <oscav at hotmail.fr> wrote:
>
> Im' running FS on windows server 2003 64bits
>
>
> Oscav wrote:
>>
>> Hi,
>>
>> I'm trying to use to SkypeIAX. When I load the mod_skypiax, I got the
>> following error :
>>
>> ?Failed to connect to a SKYPE API for interface_id=1, no SKYPE client
>> running, please (re)start Skype client. Skypiax exiting
>>
>> Skype is running with the correct account and skypiax.conf use the same
>> account. I was expecting a permission request from the Skype user but
>> nothing happens.
>>
>> Somebody knows how I can solve this ??
>>
>> Many thanks.
>>
>
> --
> View this message in context: http://old.nabble.com/Failed-to-connect-to-a-SKYPE-API-tp27062783p27078464.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From jonas.gauffin at gmail.com  Fri Jan  8 08:25:14 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Fri, 8 Jan 2010 17:25:14 +0100
Subject: [Freeswitch-users] hangup cause: NORMAL_UNSPECIFIED
In-Reply-To: <583FB3F4-9CFC-48A8-ADE0-30FD43F8F1F1@freeswitch.org>
References: <d021d3be1001080139m5b4cd7c6ge87b19e4ad6a62fe@mail.gmail.com>
	<583FB3F4-9CFC-48A8-ADE0-30FD43F8F1F1@freeswitch.org>
Message-ID: <d021d3be1001080825m1948cf9drbfb3a1f3ebd99b90@mail.gmail.com>

Consider it a feature request then :) A one liner would be enough, just a
bit more about why the call failed.

On Fri, Jan 8, 2010 at 3:43 PM, Brian West <brian at freeswitch.org> wrote:

> Not without the siptrace and sofia loglevel all 9
>
> /b
>
> On Jan 8, 2010, at 3:39 AM, Jonas Gauffin wrote:
>
> > Hello,
> >
> > Is it possible to get a more detailed reason (in the log) to why
> NORMAL_UNSPECIFIED was returned as hang up cause?
> >
> > 2010-01-08 09:30:50.987200 [DEBUG] sofia.c:3831 Channel
> sofia/external/070738xxxx entering state [terminated][904]
> > 2010-01-08 09:30:50.987200 [NOTICE] sofia.c:4461 Hangup
> sofia/external/070738xxxx [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED]
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Fri Jan  8 08:35:36 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 10:35:36 -0600
Subject: [Freeswitch-users] hangup cause: NORMAL_UNSPECIFIED
In-Reply-To: <d021d3be1001080825m1948cf9drbfb3a1f3ebd99b90@mail.gmail.com>
References: <d021d3be1001080139m5b4cd7c6ge87b19e4ad6a62fe@mail.gmail.com>
	<583FB3F4-9CFC-48A8-ADE0-30FD43F8F1F1@freeswitch.org>
	<d021d3be1001080825m1948cf9drbfb3a1f3ebd99b90@mail.gmail.com>
Message-ID: <3872821A-B842-40C8-ADD7-1079C35EBC6C@freeswitch.org>

A one liner can't solve this... their are so many things interacting... the 90X usually means you have given something to sofia it didn't like.

/b

On Jan 8, 2010, at 10:25 AM, Jonas Gauffin wrote:

> Consider it a feature request then :) A one liner would be enough, just a bit more about why the call failed.
> 
> On Fri, Jan 8, 2010 at 3:43 PM, Brian West <brian at freeswitch.org> wrote:
> Not without the siptrace and sofia loglevel all 9
> 
> /b

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From vhatz at kinetix.gr  Fri Jan  8 08:48:21 2010
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Fri, 08 Jan 2010 18:48:21 +0200
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>	<4B4728A5.4040805@kinetix.gr>
	<cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>
Message-ID: <4B4761D5.5010907@kinetix.gr>

Hello Ahmed,

On 8/1/10 6:07 ??, Ahmed Naji wrote:
> My line of thought is to tone-detect secondary ringing tones post 
> 200(OK) to detect FAS (False Answer Supervision). This should 
> eliminate at least a good proportion of calls, and it can be done real 
> time through a script/modules/...etc.
>

Tone detection can work only in cases where actual tones are sent over 
the audio. There are some routes (especially mobile routes) where 
different ringtones or music can be used instead of ringback tones, 
which render the whole tone detection effort difficult. There are also 
cases where no audio at all is sent until the called party answers the 
phone. Not to mention that the "FASed" route's behavior can change over 
time, rendering an effort for permanent FAS detection futile.

> Working along this thought, at least you are minimising the hit 
> cost-wise to a few seconds at most.

Well, even if you manage to detect FAS given to you by your termination 
provider, you cannot really minimize loss, as typically calls are 
charged by the time difference between 200(OK) and BYE. You can't really 
dispute anything even if you manage to detect FAS in real time, because 
you cannot prove this in a court or to anyone else by presenting CDRs 
only. You can't really pinpoint in a list of calls which ones had FAS. 
And of course, the terminating partner who gives FAS will deny it most 
of the times.

>
> As to answering machines and fake conferences, fake network messages 
> ...etc, one can possibly use voice detection, perhaps with heuristic 
> and statistical training.

You will need to combine tones detection, music detection, answering 
machine detection, which in the end makes the whole effort pointless. 
You will need to use DSP in software or in hardware to even try 
accomplish this (which means additional cost), and for what? For trying 
to detect FAS on a route which is not suitable for production use in the 
first place? And for today's wholesale margins?

IMHO the best solution is policy:

1) gather statistics from your CDRs (FreeSWITCH has lots of variables 
that you can use in your CDRs)
2) place test calls in cases where the stats show a possible problem
3) send a trouble ticket to the offending carrier in cases of FAS
and
4) stop using the route until the "problem" is fixed.

There are "carriers" who give FAS issues all the time and you can't 
really deal with them... It's best to just avoid them completely after 
repeated FAS infractions.

Best regards,
Vlasis Hatzistavrou.


From larclap at yahoo.com  Fri Jan  8 08:53:25 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 8 Jan 2010 08:53:25 -0800
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <COL117-W486989B811B372C35935AAB9700@phx.gbl>
References: <012901ca8fe6$e36b71c0$aa425540$@com>,
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>,
	<014801ca8fee$b75f8780$261e9680$@com>,
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>
Message-ID: <013a01ca9083$19a10920$4ce31b60$@com>

Thanks for the instructions Jeff.

 

After installing QT and setting the environmental variable, I get the
following error on build:

 

Error      1              error PRJ0019: A tool returned an error code from
"MOC prefportaudio.h"          FSComm              FSComm

 

Do I need to have checked-out the entire FreeSWITCH trunk in order to build
fscomm? I did not, just
http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm.

 

Lars

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Thursday, January 07, 2010 7:17 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

 

The windows support is very experimental at this time!
 
You must manually install
http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
 
Then set the environment variable QTDIR in the environment variables. This
can be set from the Computer/Properties/Advanced system settings/Environment
Variables/User Variables settings screen.
 
QTDIR=c:\qt\4.6.0 - or wherever you installed it
 
then restart VS
 

 

  _____  

Date: Thu, 7 Jan 2010 21:45:56 -0200
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

I don't have a Windows machine to test that. Maybe jlenk could give us a
hand since he is the one who has created the visual studio project? 

 

Regards,
Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927



On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:

No Qt installed. I just checked out from
http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into
VS2008. 

Do I need to get
http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

 

 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:

I just downloaded the fscomm project and loaded it into vs2008. I?ve never
programmed in C++ (or c), just C#, so I can?t make anything of the following
two messages:

 

Error      1              error PRJ0019: A tool returned an error code from
"RCC resources.qrc"  FSComm              FSComm

 

Warning               2              The following environment variables
were not found: $(QTDIR)                Project FSComm

 

Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.

 

The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)

 

Lars

Do you have Qt 4.6 installed?
-MC

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

 

  _____  

Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up
<http://clk.atdmt.com/GBL/go/196390709/direct/01/>  now.

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From msc at freeswitch.org  Fri Jan  8 08:57:19 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 8 Jan 2010 08:57:19 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly!
Message-ID: <87f2f3b91001080857l2f61903cw7f92a850f6c27718@mail.gmail.com>

Please call in! We'll mingle for a bit and then get started. Agenda is here:
http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_08

-Michael
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From lists at redbonez.net  Fri Jan  8 08:59:09 2010
From: lists at redbonez.net (Adam Ford)
Date: Fri, 8 Jan 2010 09:59:09 -0700
Subject: [Freeswitch-users] Delay in connecting inbound calls
Message-ID: <016e01ca9083$e6de14f0$b49a3ed0$@net>

I have setup a FreeSWITCH system using OpenZAP and Redfone>foneBridge2 for
my connection to the PSTN and there seems to be a 3-5 second delay between
when the incoming call is answered and the parties are able to hear each
other.  This is only with inbound calls, outbound audio connection is
instant.  

 

Any suggestions on where to start troubleshooting this issue?

 

-AF

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From brian at freeswitch.org  Fri Jan  8 09:05:19 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 11:05:19 -0600
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <013a01ca9083$19a10920$4ce31b60$@com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>,
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>,
	<014801ca8fee$b75f8780$261e9680$@com>,
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>
	<013a01ca9083$19a10920$4ce31b60$@com>
Message-ID: <68D59A25-D24F-4D1D-A548-67CE33D1EEEB@freeswitch.org>

You have to have freeswitch built and installed... and you need to do it all right now.

/b

On Jan 8, 2010, at 10:53 AM, Lars Zeb wrote:

> Thanks for the instructions Jeff.
>  
> After installing QT and setting the environmental variable, I get the following error on build:
>  
> Error      1              error PRJ0019: A tool returned an error code from "MOC prefportaudio.h"          FSComm              FSComm
>  
> Do I need to have checked-out the entire FreeSWITCH trunk in order to build fscomm? I did not, just http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm.
>  
> Lars

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From dujinfang at gmail.com  Fri Jan  8 10:00:58 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 9 Jan 2010 02:00:58 +0800
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<B732197E-1123-4738-A69F-F84206FCE38A@freeswitch.org>
	<0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org>
	<0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org>
	<23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com>
	<60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org>
	<23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com>
	<ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com>
	<23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com>
	<B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>
Message-ID: <23f91031001081000t26bfebfcy222a1ffc13b95a1e@mail.gmail.com>

Bad luck, I had trouble to join the conference due to my bad internet.
Good thing is that I compiled and run on Mac and dialed into the
conference successfully.

Definitely I can help test and reporting bugs, and might be more
helpful on small bug fixes and implement new features though I need to
learn QT first. But first of all I think it would be better if we have
a feature list for the new phone.

2010/1/8, Brian West <brian at freeswitch.org>:
> Lets schedule FSComm on the weekly conference call... We need people to step
> up and take some roles in both FreeSWITCH and FSComm projects... Even if its
> just testing bugs and collecting info.
>
> Thanks,
> Brian
>
> On Jan 7, 2010, at 10:11 AM, Seven Du wrote:
>
>> :lol. I do like to involve into this. I saw you have done a lot of
>> works. I read some code and here are some questions:
>>
>> 1) what's your nick on IRC? I'm seven(or seven_ ?)
>> 2) Are you developing on Windows? How can I compile on Mac(I have no
>> experience on QT)?
>> 3) Does is needs fsGUI? I downloaded fsGUI dmg file and it complains
>> no mods on start.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From jcasale at activenetwerx.com  Fri Jan  8 10:29:48 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 8 Jan 2010 18:29:48 +0000
Subject: [Freeswitch-users] Purge Stale Registrations
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90734F9@prato.activenetwerx.local>

Looking through the cli, I thought this might be attainable with
sofia status profile internal flush_inbound_reg|rescan for example
but it's not removing stake registrations from UA's I am not expecting
to see. Anyway to do this?

Thanks!
jlc


From brian at freeswitch.org  Fri Jan  8 10:40:32 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 12:40:32 -0600
Subject: [Freeswitch-users] Purge Stale Registrations
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E90734F9@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E90734F9@prato.activenetwerx.local>
Message-ID: <72E53884-A014-43AF-93C7-A33EE240A221@freeswitch.org>

how do we know they are stale?  They remove when they expire or your endpoint unregisters on shutdown.

/b

On Jan 8, 2010, at 12:29 PM, Joseph L. Casale wrote:

> Looking through the cli, I thought this might be attainable with
> sofia status profile internal flush_inbound_reg|rescan for example
> but it's not removing stake registrations from UA's I am not expecting
> to see. Anyway to do this?
> 
> Thanks!
> jlc



From jcasale at activenetwerx.com  Fri Jan  8 10:53:52 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 8 Jan 2010 18:53:52 +0000
Subject: [Freeswitch-users] Purge Stale Registrations
In-Reply-To: <72E53884-A014-43AF-93C7-A33EE240A221@freeswitch.org>
References: <CA5A491E9DEFBE4CB777DE97E21575E90734F9@prato.activenetwerx.local>
	<72E53884-A014-43AF-93C7-A33EE240A221@freeswitch.org>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90735AE@prato.activenetwerx.local>

>how do we know they are stale?  They remove when they expire or your endpoint unregisters on shutdown.

Right, should have been more specific: I made changes to a few UA's and
I see the new UA's but wanted to remove the old reg's forcibly to clean
up the status so it's more readable. Only wanted to use this during testing
of course...


From nicolas at medularis.com  Fri Jan  8 11:02:08 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 8 Jan 2010 16:02:08 -0300
Subject: [Freeswitch-users] Need to fake ringback
In-Reply-To: <91E856D5-B499-4C5F-A6A3-E2BB36C926C6@freeswitch.org>
References: <1b46b4e81001072133h3f4c9bf1g68e25d27fd71edc1@mail.gmail.com>
	<91E856D5-B499-4C5F-A6A3-E2BB36C926C6@freeswitch.org>
Message-ID: <1b46b4e81001081102k36c76792n5619426d05cca8cd@mail.gmail.com>

Not that I'm aware.


On Fri, Jan 8, 2010 at 11:41 AM, Brian West <brian at freeswitch.org> wrote:
> Are you using proxy media?
>
> /b
>
> On Jan 7, 2010, at 11:33 PM, Nicolas Brenner wrote:
>
>> I'm trying to fake a ringback for leg1 of a two-legged call without success.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From nicolas at medularis.com  Fri Jan  8 11:02:46 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 8 Jan 2010 16:02:46 -0300
Subject: [Freeswitch-users] Need to fake ringback
In-Reply-To: <1b46b4e81001081102k36c76792n5619426d05cca8cd@mail.gmail.com>
References: <1b46b4e81001072133h3f4c9bf1g68e25d27fd71edc1@mail.gmail.com>
	<91E856D5-B499-4C5F-A6A3-E2BB36C926C6@freeswitch.org>
	<1b46b4e81001081102k36c76792n5619426d05cca8cd@mail.gmail.com>
Message-ID: <1b46b4e81001081102h2f644d28r86e14d7ba151e5c4@mail.gmail.com>

I'm using the default dialplan.


On Fri, Jan 8, 2010 at 4:02 PM, Nicolas Brenner <nicolas at medularis.com> wrote:
> Not that I'm aware.
>
>
> On Fri, Jan 8, 2010 at 11:41 AM, Brian West <brian at freeswitch.org> wrote:
>> Are you using proxy media?
>>
>> /b
>>
>> On Jan 7, 2010, at 11:33 PM, Nicolas Brenner wrote:
>>
>>> I'm trying to fake a ringback for leg1 of a two-legged call without success.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From kristian.kielhofner at gmail.com  Fri Jan  8 11:03:35 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 8 Jan 2010 14:03:35 -0500
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <D03B492C-74B9-4B38-B67B-017690FF372C@freeswitch.org>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org>
	<23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com>
	<60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org>
	<23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com>
	<ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com>
	<23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com>
	<B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>
	<ebbc2d551001071227y22d6eaa8y43742fb4328a67b2@mail.gmail.com>
	<D03B492C-74B9-4B38-B67B-017690FF372C@freeswitch.org>
Message-ID: <2d9149cd1001081103m4d2b6852l9848ce4c82005fd3@mail.gmail.com>

What is his PayPal?  I've been looking through FSComm and it looks
sick...  I love the idea of being able to tweak my SOFTPHONE params
just like I would tweak FS.  Sick, just sick.

Speaking of sick, I hope JM gets better soon :).

On Thu, Jan 7, 2010 at 3:36 PM, Brian West <brian at freeswitch.org> wrote:
> Everyone should get JM's paypal and toss him some cash for all the good work
> he's doing... Without him this project wouldn't have become a reality.
> /b


-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From nicolas at medularis.com  Fri Jan  8 11:13:52 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 8 Jan 2010 16:13:52 -0300
Subject: [Freeswitch-users] Ruby ESL missing pthread
In-Reply-To: <8A3B5711-727D-4BDB-97AE-7AD20C4528F7@jerris.com>
References: <1b46b4e81001071958i17ccb6bey6eeecd9d94eb8438@mail.gmail.com>
	<8A3B5711-727D-4BDB-97AE-7AD20C4528F7@jerris.com>
Message-ID: <1b46b4e81001081113j40ed02fehd34eb2b99d1a5ef6@mail.gmail.com>

Thanks!

http://jira.freeswitch.org/browse/ESL-29

On Fri, Jan 8, 2010 at 12:33 PM, Michael Jerris <mike at jerris.com> wrote:
> Please open up a bug on jira for me for this issue.
>
> Mike


From linux4michelle at tamay-dogan.net  Fri Jan  8 11:32:45 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Fri, 8 Jan 2010 20:32:45 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <7b197bef1001080538u5ed67082r6df84d287c0c06c1@mail.gmail.com>
References: <20100103184636.GW5547@tamay-dogan.net>
	<20100106135210.GG5547@tamay-dogan.net>
	<CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>
	<7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com>
	<20100106171413.GI5547@tamay-dogan.net>
	<7b197bef1001060939rd1e08f3h2cb82cd89e50225e@mail.gmail.com>
	<20100107182731.GL5547@tamay-dogan.net>
	<7b197bef1001080538u5ed67082r6df84d287c0c06c1@mail.gmail.com>
Message-ID: <20100108193244.GR5547@tamay-dogan.net>

Hello Giovanni,

thankyou forthe answer...

> you don't need them all (at least if you don't want to make dirty pics
> with a kludge full of wires... hey, that can be arousing! :))

My Longsun L580 can be put into Web-Cam mode...

But it is cheal chinese Dual-SIM/Standby Cell-Phone which  can  not  use
anything in parallel.  :-/

> I suggest you use a ready-made GSM/GPRS/HDSPA/whatever module, that
> contains all that you need and is available from Chinese suppliers for
> very low prices. You can find some module that allows you to directly
> tap in the GSM pcm audio stream, that would means you will not need to
> sample and convert from analog to digital (so, no cpu power at all, no
> dsp, no nothing).

I will see, they cost arround 110 to 230 US$...

> If you would like to keep me in the loop, I would like to know how you
> progress.

If my ISP would repair my ADSL I had my website back online...

    <http://www.electronica.tamay-dogan.net/>

Fully "Open Hardware Development" in the sense of "Open Source" with some
small limitations...  or advantages.

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From jerry.richards at teotech.com  Fri Jan  8 11:38:06 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 8 Jan 2010 11:38:06 -0800
Subject: [Freeswitch-users] http://jira.freeswitch.org/browse/FSCORE-262
Message-ID: <39ED1D5111554D33AF502370BC34D288@greyhawk.tonecommander.com>

Is there a plan to fix this JIRA issue:
http://jira.freeswitch.org/browse/FSCORE-262

This is causing a problem in sharing presence data between FS and another
gateway.

Thanks,
Jerry



From brian at freeswitch.org  Fri Jan  8 11:43:40 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 13:43:40 -0600
Subject: [Freeswitch-users] http://jira.freeswitch.org/browse/FSCORE-262
In-Reply-To: <39ED1D5111554D33AF502370BC34D288@greyhawk.tonecommander.com>
References: <39ED1D5111554D33AF502370BC34D288@greyhawk.tonecommander.com>
Message-ID: <5817BC3F-C091-42E3-AAAE-2CF4A21A9531@freeswitch.org>

Do you happen to have a patch for that?

/b

On Jan 8, 2010, at 1:38 PM, Jerry Richards wrote:

> Is there a plan to fix this JIRA issue:
> http://jira.freeswitch.org/browse/FSCORE-262
> 
> This is causing a problem in sharing presence data between FS and another
> gateway.
> 
> Thanks,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From william.suffill at gmail.com  Fri Jan  8 12:09:04 2010
From: william.suffill at gmail.com (William Suffill)
Date: Fri, 8 Jan 2010 15:09:04 -0500
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <2d9149cd1001081103m4d2b6852l9848ce4c82005fd3@mail.gmail.com>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> 
	<23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com> 
	<60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org>
	<23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com> 
	<ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com> 
	<23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com> 
	<B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>
	<ebbc2d551001071227y22d6eaa8y43742fb4328a67b2@mail.gmail.com> 
	<D03B492C-74B9-4B38-B67B-017690FF372C@freeswitch.org>
	<2d9149cd1001081103m4d2b6852l9848ce4c82005fd3@mail.gmail.com>
Message-ID: <6b65470d1001081209v64d20f9ehdd7a7b4c6516c540@mail.gmail.com>

jmesquita at gmail.com is his Paypal account. Ya hope he gets better as well.

-- W
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From a.alalousi at gmail.com  Fri Jan  8 13:43:59 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Fri, 8 Jan 2010 21:43:59 +0000
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <191c3a031001080820i426194cw58b8a2cf8b123493@mail.gmail.com>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
	<4B4728A5.4040805@kinetix.gr>
	<cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>
	<191c3a031001080820i426194cw58b8a2cf8b123493@mail.gmail.com>
Message-ID: <cd93b3701001081343o35b6752wdd44f4f97155bae6@mail.gmail.com>

That's what I had in mind + some other techniques to address things to help
with answering machines and recorded messages.

Vasilis rightly states some of the challenges in doing this, and he is also
right on the legal and effort/benefit ratio.

My aim behind all of this is not to really prosecute or dispute issues with
carriers - we have an active policy of barring FAS routes after the second
offence, so that's covered. The aim is to be able to detect such routes as
an when they arise, and be proactive rather than reactive in dealing with
this horrendous and disgusting practice that plagues the telecom industry.

Best regards,

Ahmed.


2010/1/8 Anthony Minessale <anthony.minessale at gmail.com>

> you could use the existing tone_detect app after the false answer to xfer
> to an extension of your choice if ring tone was detected.
>
>
> 2010/1/8 Ahmed Naji <a.alalousi at gmail.com>
>
> Hi Valsis,
>>
>> Thanks for this.
>>
>> My line of thought is to tone-detect secondary ringing tones post 200(OK)
>> to detect FAS (False Answer Supervision). This should eliminate at least a
>> good proportion of calls, and it can be done real time through a
>> script/modules/...etc.
>>
>> Working along this thought, at least you are minimising the hit cost-wise
>> to a few seconds at most.
>>
>> As to answering machines and fake conferences, fake network messages
>> ...etc, one can possibly use voice detection, perhaps with heuristic and
>> statistical training.
>>
>> Just a thought ..
>>
>> Regards,
>>
>> Ahmed.
>>
>>
>> 2010/1/8 Vlasis Hatzistavrou (KTI) <vhatz at kinetix.gr>
>>
>>  Hello Ahmed,
>>>
>>> I don't think there is a reliable way to detect FAS on a per call basis.
>>> Even audio detection software can be confused by strange ringtones,
>>> answering machines, etc.
>>>
>>> The most reliable way we have found is to use statistics from CDRs and
>>> see if the INVITE-to-200(OK) delay averaged over a number of calls appears
>>> to be too small. If it is, then it is possible that you got FAS for those
>>> calls.
>>>
>>> But this can only tell you if you have been experiencing FAS in past
>>> calls, ie you will not know you are getting FAS in real time.
>>>
>>> It is still useful however: after you detect a possible FAS case via
>>> statistics you can place a few test calls yourself to verify that there
>>> actually exists FAS (and this is the only information that a carrier will
>>> accept in a trouble ticket to prove to them that they actually give you
>>> FAS).
>>>
>>> I hope this helps.
>>>
>>> Best regards,
>>> Vlasis Hatzistavrou.
>>>
>>>
>>> On 8/1/10 3:05 ??, Ahmed Naji wrote:
>>>
>>> Hi everyone,
>>>
>>> Was just wondering what/if anyone is doing any work on FAS detection and
>>> spoofed ring tones. Be great to discuss some ideas.
>>>
>>> Regards,
>>>
>>> Ahmed.
>>>
>>> --
>>> Ahmed Naji
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Ahmed Naji
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From a.alalousi at gmail.com  Fri Jan  8 13:57:16 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Fri, 8 Jan 2010 21:57:16 +0000
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <4B4761D5.5010907@kinetix.gr>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
	<4B4728A5.4040805@kinetix.gr>
	<cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>
	<4B4761D5.5010907@kinetix.gr>
Message-ID: <cd93b3701001081357i56c39e15j262ed05e5351a31a@mail.gmail.com>

Hi Vasilis,

Good points .. my comments inline.

Regards,

Ahmed.

2010/1/8 Vlasis Hatzistavrou (KTI) <vhatz at kinetix.gr>

> Hello Ahmed,
>
> On 8/1/10 6:07 ??, Ahmed Naji wrote:
> > My line of thought is to tone-detect secondary ringing tones post
> > 200(OK) to detect FAS (False Answer Supervision). This should
> > eliminate at least a good proportion of calls, and it can be done real
> > time through a script/modules/...etc.
> >
>
> Tone detection can work only in cases where actual tones are sent over
> the audio. There are some routes (especially mobile routes) where
> different ringtones or music can be used instead of ringback tones,
> which render the whole tone detection effort difficult. There are also
> cases where no audio at all is sent until the called party answers the
> phone. Not to mention that the "FASed" route's behavior can change over
> time, rendering an effort for permanent FAS detection futile.
>
> True. One way of dealing with coloured ringing tones is quiet straight
forward with acoustics libraries for C++ and such like. No audio is also
possible to deal with in this fashion.

> Working along this thought, at least you are minimising the hit
> > cost-wise to a few seconds at most.
>
> Well, even if you manage to detect FAS given to you by your termination
> provider, you cannot really minimize loss, as typically calls are
> charged by the time difference between 200(OK) and BYE.

True, but the idea is to minimise this duration, and hangup the call ASAP.

You can't really
> dispute anything even if you manage to detect FAS in real time, because
> you cannot prove this in a court or to anyone else by presenting CDRs
> only. You can't really pinpoint in a list of calls which ones had FAS.
> And of course, the terminating partner who gives FAS will deny it most
> of the times.
>
Absolutely right on all accounts. At least in doing so, you are minimising
the financial hit, particularly on busy routes that carry a high premium.


>
> >
> > As to answering machines and fake conferences, fake network messages
> > ...etc, one can possibly use voice detection, perhaps with heuristic
> > and statistical training.
>
> You will need to combine tones detection, music detection, answering
> machine detection, which in the end makes the whole effort pointless.
>
Debatable. Hard, yes, but pointless ? I don't really agree.


> You will need to use DSP in software or in hardware to even try
> accomplish this (which means additional cost),

DSP techniques - to some extent, if you really want to go that far. The idea
is to do it all in FS, and not really employ expensive software/hardware
techniques as you so rightly state.


> and for what? For trying
> to detect FAS on a route which is not suitable for production use in the
> first place? And for today's wholesale margins?
>
Here's the thing: even on those terms, it is worth it. For so many
destinations (e.g. Central/Sub-Saharan Africa and such like), there really
isn't much of a distinction between "reliable" wholesale routes and retail
routes. You use the best you could to carry your traffic and my concern is
for my retail customers. People like Vodafone, Orange and BT don't take too
kindly to being FASed, and if you carry their traffic, then you by default
are under contractual obligation to cushion the hit where they can prove a
case. One case, for one of those routes that carry 250K minutes/day will
bankrupt a small/medium carrier, even if you end up winning the legal
argument.

What I'm trying to do is not make an unusable route usable. Rather, to
detect an unusable route ASAP, place the terminating carrier on a very short
leash, and disconnect them as early as possible should they re-offend.


> IMHO the best solution is policy:
>
> 1) gather statistics from your CDRs (FreeSWITCH has lots of variables
> that you can use in your CDRs)
> 2) place test calls in cases where the stats show a possible problem
> 3) send a trouble ticket to the offending carrier in cases of FAS
> and
> 4) stop using the route until the "problem" is fixed.
>
> I hear what you're saying. And how much is the time and man effort involved
in manually dealing with this ?


> There are "carriers" who give FAS issues all the time and you can't
> really deal with them... It's best to just avoid them completely after
> repeated FAS infractions.
>
> Best regards,
> Vlasis Hatzistavrou.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From freeswitch at aastral.net  Fri Jan  8 15:39:54 2010
From: freeswitch at aastral.net (Bill W)
Date: Fri, 08 Jan 2010 18:39:54 -0500
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <74C18F18-89EB-49DF-B5CE-C8D56856ABCD@freeswitch.org>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>	<4B46792E.6090805@aastral.net>	<E5EFC626-353A-4B54-9F22-59C0C48E3B95@freeswitch.org>	<4B4684F3.8030504@aastral.net>
	<74C18F18-89EB-49DF-B5CE-C8D56856ABCD@freeswitch.org>
Message-ID: <4B47C24A.3080305@aastral.net>

Hey Everyone,

I spoke with Karl, and for the sake of completing this thread I'm 
posting the results.

Ultimately the problem Karl found with OpenSUSE was the SQLite libraries 
leading to database corruption, and freeswitch misbehaving because of 
that.  He also was trying to do traffic shaping for his application and 
ran into problems with shaping not working right on multi-core x86_64 
kernels in SuSE.

I asked him about using odbc in the core to get around the sqlite bug 
but he didn't bother because he needed traffic shaping that worked.

So it's not so much that SuSE was a bad distro, but rather that it 
didn't work well with freeswitch and traffic shaping on multi-core 
x86_64.  He did mention that SuSE 11.1 is nice in general but to stay 
away from 11.2 because gcc segfaults on any significant build.

Hope this helps the community!
Bill



His response:
====================
The big thing you should learn from the investment of my time in the lab 
here is simply this, "listen to Brian".  From now on, I'm considering 
him the EF Hutton on #freeswitch.

(Editors Note: For you youngsters out there, EF Hutton's tagline was 
"When EF Hutton talks, people listen.")

When I added about 60 phones to the system, it essentially "blew up" 
whereby phones wouldn't register, the database would get corrupted, etc...

We worked around the problem by making the freeswitch "db" directory a 
ramdisk, but that only mitigated the problem, and didn't entirely fix it.

Oh yeah, and although mod_perl will compile in freeswitch, it will bomb 
out and segfault when attempting to run any mod_perl scripts in 
freeswitch.  The fix is to recompile perl from source.  Even then, I 
still had problems.

Other problems with the remaining Suse installations still are:

1.) When you answer, it will hang up.  No rhyme, reason, or otherwise.
2.) Occasionally (rarely) we have issues with audio not passing through 
correctly.  One way audio, or none at all.
3.) When you dial, it will take 10 seconds to go through enum lookups 
and the like before finally hitting the PSTN gateway.

Oh yeah, about RedHat...  Don't bother with it's sister CentOS either. 
I tested it here in the lab, and it was like getting in a time machine 
and going back to 2007.   Also, Centos is busted.  You'll find that 
Linux kernel will re-transmit IP packets from processes long since dead. 
  Suffice it to say, when it does this to RTP traffic it drives things 
bugnutz. (Ed. His test was to run 20mbit/sec through centos with a 
gigabit card overnight and it dropped 150 packets.  He did try different 
cards.)

Mandrake - recommend this to people you dislike.  If they're ignorant 
enough you'll find them thanking you for it.

Debian - almost awesome, but if failed miserably in the lab with packet 
shaping.  I mean, it thought it was working, but the overall quality was 
hit & miss.  Other than that, no complaints.  What was really attractive 
was it's got Cyrus 2.3 out of the box, so if you're using Cyrus and not 
using it for packet shaping you might consider Debian an option.  Be 
advised, since it flunked packet shaping we never bothered to 
compile/test freeswitch on it, so do your own research.

Ubuntu, make no mistake...
We tested Ubuntu pretty heavily here in the lab.
Even the packet shaping works (HTB & SFQ).
With Suse I has to custom-compile the kernel, and packet shaping ONLY 
worked with 2.6.28.8 - 2.6.28.10, the rest were buggy and the problems 
manifested themselves in ways that you'd think would be totally 
unrelated.  I should be more specific and state x86_64 multi-core. 
x86_64 single core seemed to do packet shaping (somewhat nicely) in 
2.6.18 on up.  Previous to that it was rather "interesting"...

Save yourself the headache and go with Ubuntu.

==================================




Brian West wrote:
> He's on the list Karl J. Vesterling
> 
> /b
> 
> On Jan 7, 2010, at 7:05 PM, Bill W. wrote:
> 
>> Wow, I haven't heard of these issues.  Obviously this concerns me.  Are
>> these documented anywhere so I can research this?  How do I get in touch
>> with KJV?
>>
>> Thanks!
>> Bill
>>
>>
>>
>> Brian West wrote:
>>> Good luck with that you'll have an ass load of problems.  The reason its stable is the backports and outdated packages.  Bleeding edge will only screw you over... just ask KJV... He was on OpenSuSE and had nothing but weird problems.
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From vhatz at kinetix.gr  Fri Jan  8 16:02:05 2010
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Sat, 09 Jan 2010 02:02:05 +0200
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <cd93b3701001081357i56c39e15j262ed05e5351a31a@mail.gmail.com>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>	<4B4728A5.4040805@kinetix.gr>	<cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>	<4B4761D5.5010907@kinetix.gr>
	<cd93b3701001081357i56c39e15j262ed05e5351a31a@mail.gmail.com>
Message-ID: <4B47C77D.4080502@kinetix.gr>

Hello Ahmed,

>
>     IMHO the best solution is policy:
>
>     1) gather statistics from your CDRs (FreeSWITCH has lots of variables
>     that you can use in your CDRs)
>     2) place test calls in cases where the stats show a possible problem
>     3) send a trouble ticket to the offending carrier in cases of FAS
>     and
>     4) stop using the route until the "problem" is fixed.
>
> I hear what you're saying. And how much is the time and man effort 
> involved in manually dealing with this ?
>  
>
Unfortunately, it is a lot of work. In fact it is a 24x7 manual work... 
What makes things a bit easier is that a FAS route will have a short 
setup-connect delay, and statistics will definitely show a heavily FASed 
route, although a route with a small percentage of FAS will still go 
under the radar. And of course, using statistical methods doesn't give 
you results in real time...

We have experimented with tone_detect app as Anthony Minesalle suggested 
for detecting tones to apply to FAS problems and it works very well. But 
after a while we dropped the whole FAS detection idea because of the so 
many different FAS scenarios out there... Plus, quite a few methods that 
monitor audio involve decompressing the incoming audio before processing 
it. This means that you need to have audio transcoding on FS, which a 
bad things in terms of perceived audio quality and CPU power/scaling. In 
that case you would almost certainly need quite a few G729 codec 
licenses, as most carriers use this codec to send you their traffic.

Nonetheless if you want to pursue this and you need help I'd be happy to 
assist, as FAS is a huge problem for carriers.

Best regards,
Vlasis.
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From msc at freeswitch.org  Fri Jan  8 16:17:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 8 Jan 2010 16:17:26 -0800
Subject: [Freeswitch-users] Delay in connecting inbound calls
In-Reply-To: <016e01ca9083$e6de14f0$b49a3ed0$@net>
References: <016e01ca9083$e6de14f0$b49a3ed0$@net>
Message-ID: <87f2f3b91001081617t1844119bp1ece7ae951c68836@mail.gmail.com>

On Fri, Jan 8, 2010 at 8:59 AM, Adam Ford <lists at redbonez.net> wrote:

>  I have setup a FreeSWITCH system using OpenZAP and Redfone>foneBridge2
> for my connection to the PSTN and there seems to be a 3-5 second delay
> between when the incoming call is answered and the parties are able to hear
> each other.  This is only with inbound calls, outbound audio connection is
> instant.
>
>
>
> Any suggestions on where to start troubleshooting this issue?
>
Start by getting a debug log of an incoming call and putting it on pastebin
so others can take a look.
-MC
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From jmesquita at freeswitch.org  Fri Jan  8 17:13:32 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 8 Jan 2010 22:13:32 -0300
Subject: [Freeswitch-users] FSComm builds OK on Fedora F-12 with QT4
	4.5.3 but doesn't save SIP account
In-Reply-To: <13ca621c1001080331l363122a5l3b093adb5782538e@mail.gmail.com>
References: <13ca621c1001080331l363122a5l3b093adb5782538e@mail.gmail.com>
Message-ID: <ebbc2d551001081713r70da95e3i115769116a17390a@mail.gmail.com>

Gavin,

http://wiki.freeswitch.org/wiki/FSComm#Configuration

Sofia configuration is not yet persisted. I have it kinda working on my
local copy and I hope to have it commited by the end of tomorrow. Till then,
you will have to do the configuration manually like described on the wiki.

Regards,

Jo?o Mesquita


On Fri, Jan 8, 2010 at 8:31 AM, Gavin Henry <gavin.henry at gmail.com> wrote:

> Hi,
>
> Just a quick one to say this builds ok with:
>
> Compiled FSComm version: 1.0.trunk (16209M)
> FreeSWITCH Version 1.0.trunk (16209M)
>
> But it doesn't want to save my SIP account details. How to debug?
>
> Thanks.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dave at 3c.co.uk  Fri Jan  8 17:17:24 2010
From: dave at 3c.co.uk (David Knell)
Date: Sat, 09 Jan 2010 01:17:24 +0000
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <4B47C77D.4080502@kinetix.gr>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
	<4B4728A5.4040805@kinetix.gr>
	<cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>
	<4B4761D5.5010907@kinetix.gr>
	<cd93b3701001081357i56c39e15j262ed05e5351a31a@mail.gmail.com>
	<4B47C77D.4080502@kinetix.gr>
Message-ID: <1262999844.21753.52.camel@local.freepabx.com>

> Nonetheless if you want to pursue this and you need help I'd be happy
> to assist, as FAS is a huge problem for carriers.

FYI - here's some guys advertising "multilingual FAS" - presumably for
those crooked enough to want to inflate their margins in this manner and
too thick to do it for themselves:
http://www.calltermination.com/forums/thread270514.html

--Dave






From jcasale at activenetwerx.com  Fri Jan  8 17:39:56 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 9 Jan 2010 01:39:56 +0000
Subject: [Freeswitch-users] rxfax ending dialplan
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9074638@prato.activenetwerx.local>

Hey Guys,
In my initial testing with an old version of fs, my dialplan with a system
call after the rxfax app would execute, but now the dialplan ends at rxfax
after it writes the file out.

Anyone know what var is different by default now so I can allow this to
continue on to hit the script and then end after?

Thanks,
jlc


From brian at freeswitch.org  Fri Jan  8 17:46:51 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 19:46:51 -0600
Subject: [Freeswitch-users] rxfax ending dialplan
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9074638@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E9074638@prato.activenetwerx.local>
Message-ID: <CCB7B570-7845-4BA2-985B-DC40922E5E6E@freeswitch.org>

use api_hangup_hook variable.

/b

On Jan 8, 2010, at 7:39 PM, Joseph L. Casale wrote:

> Anyone know what var is different by default now so I can allow this to
> continue on to hit the script and then end after?



From jingwei.yang at gmail.com  Fri Jan  8 17:58:07 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Sat, 9 Jan 2010 09:58:07 +0800
Subject: [Freeswitch-users] IVR and TTS
In-Reply-To: <910309031001080416r6064670bic4e6285494f8d25b@mail.gmail.com>
References: <13529f9d1001080042w420c9f0vdbc5b7a57edd1e6c@mail.gmail.com>
	<910309031001080416r6064670bic4e6285494f8d25b@mail.gmail.com>
Message-ID: <13529f9d1001081758h4a32fae7q83401def09f2f34e@mail.gmail.com>

Hi Vinuth,

Thanks a lot for your great response!

Regards,
-Jingwei

On Fri, Jan 8, 2010 at 8:16 PM, Vinuth Madinur <vinuth.madinur at gmail.com>wrote:

> Hi,
>
> "DTMF" event will be raised in ESL, when "A" presses a key. It'll be raised
> for each key pressed.
>
> Alternatively you can use play_and_get_digits.
>
> To continue FS execution after you fetch the address, you just need to
> invoke the "speak" command on that socket.
>
> Since FS is handling inbound calls, you can use the outbound event socket,
> where a new connection will be opened per call from FS to your java program.
>
> One way to know what events are raised in ESL, you can telnet to 8021 port,
> authenticate and send "events plain all" command. Configure FS dialplan for
> an extension which will just answer a call when it comes in. Then call this
> extension from a softphone, press a key and you'll see the corresponding
> event in the telnet console.
>
>
> Thanks,
> Vinuth.
>
>
>
> On Fri, Jan 8, 2010 at 2:12 PM, Jingwei Yang <jingwei.yang at gmail.com>
>  wrote:
>
>> Hi Guys,
>>
>> I need to implement a function using IVR and TTS. Here's the scenario.
>>
>> 1. User A calls in
>> 2. FS plays a welcome message and directs A to press '1' to continue
>> 3. FS detects A's number and finds A's address from the database and plays
>> another piece of voice message including the address info just found
>>
>> I understand this logic can be implemented using javascript. However, in
>> this scenario, the database is a remote one and the local js has no access
>> to it. What I'm planning to do is write a Java program, talking to FS via
>> ESL. Could someone please tell me what event FS will trigger after user A
>> selects a certain option and how to inform the FS to continue the rest of
>> IVR menu after finding the address?
>>
>> Thanks,
>> -Jingwei
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jcasale at activenetwerx.com  Fri Jan  8 18:23:47 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 9 Jan 2010 02:23:47 +0000
Subject: [Freeswitch-users] rxfax ending dialplan
In-Reply-To: <CCB7B570-7845-4BA2-985B-DC40922E5E6E@freeswitch.org>
References: <CA5A491E9DEFBE4CB777DE97E21575E9074638@prato.activenetwerx.local>
	<CCB7B570-7845-4BA2-985B-DC40922E5E6E@freeswitch.org>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90746C8@prato.activenetwerx.local>

>use api_hangup_hook variable.

Thanks Brian!
I'll update the wiki for the fax related stuff once I am positive
I have it nailed...
jlc


From anthony.minessale at gmail.com  Fri Jan  8 18:46:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Jan 2010 20:46:48 -0600
Subject: [Freeswitch-users] Need to fake ringback
In-Reply-To: <1b46b4e81001081102h2f644d28r86e14d7ba151e5c4@mail.gmail.com>
References: <1b46b4e81001072133h3f4c9bf1g68e25d27fd71edc1@mail.gmail.com>
	<91E856D5-B499-4C5F-A6A3-E2BB36C926C6@freeswitch.org>
	<1b46b4e81001081102k36c76792n5619426d05cca8cd@mail.gmail.com>
	<1b46b4e81001081102h2f644d28r86e14d7ba151e5c4@mail.gmail.com>
Message-ID: <191c3a031001081846i2c31b73bp714813d71a21b611@mail.gmail.com>

ringback variable must be set on the originating leg (the A leg aka inbound
leg of the call)
then that channel must be provided as the 2nd arg to the session constructor
after the dial string or used with bridge.

session.setVariable("ringback", "%(2000,4000,440,480)");
session.execute("bridge", "sofia/internal/foo at bar.com");

or
// passing session as the 2nd arg allows the ringback to play on session
while session2 is established.
session2 = new Session("sofia/internal/foo at bar.com", session);
bridge(session, session2); //or whatever


On Fri, Jan 8, 2010 at 1:02 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> I'm using the default dialplan.
>
>
> On Fri, Jan 8, 2010 at 4:02 PM, Nicolas Brenner <nicolas at medularis.com>
> wrote:
> > Not that I'm aware.
> >
> >
> > On Fri, Jan 8, 2010 at 11:41 AM, Brian West <brian at freeswitch.org>
> wrote:
> >> Are you using proxy media?
> >>
> >> /b
> >>
> >> On Jan 7, 2010, at 11:33 PM, Nicolas Brenner wrote:
> >>
> >>> I'm trying to fake a ringback for leg1 of a two-legged call without
> success.
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Jan  8 18:48:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Jan 2010 20:48:30 -0600
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<B98B6921358D0C49BD21CD023F1F52C302623BE0@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
Message-ID: <191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>

look at mod_tone_stream its fairly basic
but really look at any mod in formats dir


On Fri, Jan 8, 2010 at 9:25 AM, MohammedShehzad <pmhshz at gmail.com> wrote:

> Thanks Brian,
>
> Great... Let me start the developing it.
> Although I am analyzing the mod_local_stream, to understand exact working
> of format module, Will anybody please let me know any similar thing, I can
> take reference?
>
>
>
> On Fri, Jan 8, 2010 at 8:21 PM, Brian West <brian at freeswitch.org> wrote:
>
>> www.bkw.org/mod_rtp_stream.tgz is a skel I started on.. You can take that
>> and finish it up if you like.
>>
>> /b
>>
>> On Jan 8, 2010, at 12:46 AM, shehzad p wrote:
>>
>> > I was looking for such functionality, but unfortunately it seems not
>> present
>> > right now, I am willing to build the rtp format mod as described by
>> Brian.,
>> > and will provide back to trunk.
>> >
>> > Although I have modified mod_skel application for use in dialplan in
>> custom
>> > application, I need to have a basic understanding regarding format mod.
>> > Will anybody please guide me from where to starts?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Fri Jan  8 19:31:18 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 21:31:18 -0600
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<B98B6921358D0C49BD21CD023F1F52C302623BE0@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
Message-ID: <85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>

Yah the skel of mod_rtp_stream was basted on mod_tone_stream so its a great jumping off point.

/b

On Jan 8, 2010, at 8:48 PM, Anthony Minessale wrote:

> look at mod_tone_stream its fairly basic
> but really look at any mod in formats dir
> 



From red.rain.seven at gmail.com  Sat Jan  9 00:47:09 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 9 Jan 2010 16:47:09 +0800
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <4B47C24A.3080305@aastral.net>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
	<4B46792E.6090805@aastral.net>
	<E5EFC626-353A-4B54-9F22-59C0C48E3B95@freeswitch.org>
	<4B4684F3.8030504@aastral.net>
	<74C18F18-89EB-49DF-B5CE-C8D56856ABCD@freeswitch.org>
	<4B47C24A.3080305@aastral.net>
Message-ID: <59ad9ca11001090047u7e3b150bpa27ff1936f41993@mail.gmail.com>

Which version of Ubuntu was tested against?
It's surprising to find such testing result about CentOS.




On Sat, Jan 9, 2010 at 7:39 AM, Bill W <freeswitch at aastral.net> wrote:

> Hey Everyone,
>
> I spoke with Karl, and for the sake of completing this thread I'm
> posting the results.
>
> Ultimately the problem Karl found with OpenSUSE was the SQLite libraries
> leading to database corruption, and freeswitch misbehaving because of
> that.  He also was trying to do traffic shaping for his application and
> ran into problems with shaping not working right on multi-core x86_64
> kernels in SuSE.
>
> I asked him about using odbc in the core to get around the sqlite bug
> but he didn't bother because he needed traffic shaping that worked.
>
> So it's not so much that SuSE was a bad distro, but rather that it
> didn't work well with freeswitch and traffic shaping on multi-core
> x86_64.  He did mention that SuSE 11.1 is nice in general but to stay
> away from 11.2 because gcc segfaults on any significant build.
>
> Hope this helps the community!
> Bill
>
>
>
> His response:
> ====================
> The big thing you should learn from the investment of my time in the lab
> here is simply this, "listen to Brian".  From now on, I'm considering
> him the EF Hutton on #freeswitch.
>
> (Editors Note: For you youngsters out there, EF Hutton's tagline was
> "When EF Hutton talks, people listen.")
>
> When I added about 60 phones to the system, it essentially "blew up"
> whereby phones wouldn't register, the database would get corrupted, etc...
>
> We worked around the problem by making the freeswitch "db" directory a
> ramdisk, but that only mitigated the problem, and didn't entirely fix it.
>
> Oh yeah, and although mod_perl will compile in freeswitch, it will bomb
> out and segfault when attempting to run any mod_perl scripts in
> freeswitch.  The fix is to recompile perl from source.  Even then, I
> still had problems.
>
> Other problems with the remaining Suse installations still are:
>
> 1.) When you answer, it will hang up.  No rhyme, reason, or otherwise.
> 2.) Occasionally (rarely) we have issues with audio not passing through
> correctly.  One way audio, or none at all.
> 3.) When you dial, it will take 10 seconds to go through enum lookups
> and the like before finally hitting the PSTN gateway.
>
> Oh yeah, about RedHat...  Don't bother with it's sister CentOS either.
> I tested it here in the lab, and it was like getting in a time machine
> and going back to 2007.   Also, Centos is busted.  You'll find that
> Linux kernel will re-transmit IP packets from processes long since dead.
>  Suffice it to say, when it does this to RTP traffic it drives things
> bugnutz. (Ed. His test was to run 20mbit/sec through centos with a
> gigabit card overnight and it dropped 150 packets.  He did try different
> cards.)
>
> Mandrake - recommend this to people you dislike.  If they're ignorant
> enough you'll find them thanking you for it.
>
> Debian - almost awesome, but if failed miserably in the lab with packet
> shaping.  I mean, it thought it was working, but the overall quality was
> hit & miss.  Other than that, no complaints.  What was really attractive
> was it's got Cyrus 2.3 out of the box, so if you're using Cyrus and not
> using it for packet shaping you might consider Debian an option.  Be
> advised, since it flunked packet shaping we never bothered to
> compile/test freeswitch on it, so do your own research.
>
> Ubuntu, make no mistake...
> We tested Ubuntu pretty heavily here in the lab.
> Even the packet shaping works (HTB & SFQ).
> With Suse I has to custom-compile the kernel, and packet shaping ONLY
> worked with 2.6.28.8 - 2.6.28.10, the rest were buggy and the problems
> manifested themselves in ways that you'd think would be totally
> unrelated.  I should be more specific and state x86_64 multi-core.
> x86_64 single core seemed to do packet shaping (somewhat nicely) in
> 2.6.18 on up.  Previous to that it was rather "interesting"...
>
> Save yourself the headache and go with Ubuntu.
>
> ==================================
>
>
>
>
> Brian West wrote:
> > He's on the list Karl J. Vesterling
> >
> > /b
> >
> > On Jan 7, 2010, at 7:05 PM, Bill W. wrote:
> >
> >> Wow, I haven't heard of these issues.  Obviously this concerns me.  Are
> >> these documented anywhere so I can research this?  How do I get in touch
> >> with KJV?
> >>
> >> Thanks!
> >> Bill
> >>
> >>
> >>
> >> Brian West wrote:
> >>> Good luck with that you'll have an ass load of problems.  The reason
> its stable is the backports and outdated packages.  Bleeding edge will only
> screw you over... just ask KJV... He was on OpenSuSE and had nothing but
> weird problems.
> >>>
> >>>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From a.afzali2003 at gmail.com  Sat Jan  9 02:29:28 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sat, 9 Jan 2010 13:59:28 +0330
Subject: [Freeswitch-users] mod_xml_ldap compile error
Message-ID: <debff5a81001090229p40d0ad7yc642cb84f2174fde@mail.gmail.com>

Hi,

I'm using CentOS 5.4 x86-64 and just mod_xml_ldap causes this compile error.
I've found an exactly same issue in the list by Keith Laaks that responded
by Patrick. Patrick says :

I had the same issue and MikeJ (one of the core developers) looked at
it. Conclusion was that it is an openldap issue and iirc the solution is
to libtoolize libraries/liblutil/Makefile.in so that when running
configure a Makefile with proper compiler flags is generated in
libraries/liblutil/

My point is :
1) As I found in openldap 2.4.19, the package already is libtool support. if
I try libtoolize on openldap package , then :

You should update your `aclocal.m4' by running aclocal.
Putting files in AC_CONFIG_AUX_DIR, `build'.
libtoolize: `config.guess' exists: use `--force' to overwrite
libtoolize: `config.sub' exists: use `--force' to overwrite
libtoolize: `ltmain.sh' exists: use `--force' to overwrite

2) Anthony says : "CentOS 5.4 appears to have some bugs in the toolchain and
libc"

Is this issue because of those bugs? How I can workaround that?

Regards,
-- afshin

making all mod_xml_ldap
Creating mod_xml_ldap.so...
/usr/bin/ld:
/root/freeswitch-1.0.5-20100106-0400/libs/openldap-2.4.19/libraries/liblutil/liblutil.a(sasl.o):
relocation R_X86_64_32S against `a local symbol' can not be used when making
a shared object; recompile with -fPIC
/root/freeswitch-1.0.5-20100106-0400/libs/openldap-2.4.19/libraries/liblutil/liblutil.a:
could not read symbols: Bad value
collect2: ld returned 1 exit status
gcc -DWITH_OPENLDAP -DLDAP_DEPRECATED
-I/root/freeswitch-1.0.5-20100106-0400/libs/openldap-2.4.19/include
-I/root/freeswitch-1.0.5-20100106-0400/src/include
-I/root/freeswitch-1.0.5-20100106-0400/src/include
-I/root/freeswitch-1.0.5-20100106-0400/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
-g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE
-shared -o .libs/mod_xml_ldap.so -shared -Wl,-x .libs/mod_xml_ldap.o
/root/freeswitch-1.0.5-20100106-0400/.libs/libfreeswitch.so
-L/root/freeswitch-1.0.5-20100106-0400/libs/apr-util/xml/expat/lib
/root/freeswitch-1.0.5-20100106-0400/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/root/freeswitch-1.0.5-20100106-0400/libs/apr/.libs/libapr-1.a -luuid -lrt
-lcrypt -lpthread -L/root/freeswitch-1.0.5-20100106-0400/libs/srtp
-L/usr/kerberos/lib64 -ldl -lz -lncurses
/root/freeswitch-1.0.5-20100106-0400/libs/openldap-2.4.19/libraries/libldap_r/.libs/libldap_r.a
-lsasl2 -lssl -lcrypto -pthread
/root/freeswitch-1.0.5-20100106-0400/libs/openldap-2.4.19/libraries/liblber/.libs/liblber.a
-lm -lresolv
/root/freeswitch-1.0.5-20100106-0400/libs/openldap-2.4.19/libraries/liblutil/liblutil.a
-Wl,--rpath -Wl,/usr/local/freeswitch/lib -Wl,--rpath
-Wl,/usr/local/freeswitch/mod
make[5]: *** [mod_xml_ldap.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_xml_ldap-all] Error 1
make[2]: *** [all-recursive] Error 1
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From nicolas at medularis.com  Sat Jan  9 04:46:45 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Sat, 9 Jan 2010 09:46:45 -0300
Subject: [Freeswitch-users] Need to fake ringback
In-Reply-To: <191c3a031001081846i2c31b73bp714813d71a21b611@mail.gmail.com>
References: <1b46b4e81001072133h3f4c9bf1g68e25d27fd71edc1@mail.gmail.com>
	<91E856D5-B499-4C5F-A6A3-E2BB36C926C6@freeswitch.org>
	<1b46b4e81001081102k36c76792n5619426d05cca8cd@mail.gmail.com>
	<1b46b4e81001081102h2f644d28r86e14d7ba151e5c4@mail.gmail.com>
	<191c3a031001081846i2c31b73bp714813d71a21b611@mail.gmail.com>
Message-ID: <1b46b4e81001090446kc5cdd0fi21816dcc3c4d8f5a@mail.gmail.com>

Thanks!

I thought I was doing exactly the same, but I was creating the 2nd
session like this:

session2 = new Session(ostr2); // ostr is the corresponding originate command

So it was missing the "link" to the first leg. Doing it like this solved it:

session2 = new Session(ostr2, session1);

Now it works great!



On Fri, Jan 8, 2010 at 11:46 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> ringback variable must be set on the originating leg (the A leg aka inbound
> leg of the call)
> then that channel must be provided as the 2nd arg to the session constructor
> after the dial string or used with bridge.
>
> session.setVariable("ringback", "%(2000,4000,440,480)");
> session.execute("bridge", "sofia/internal/foo at bar.com");
>
> or
> // passing session as the 2nd arg allows the ringback to play on session
> while session2 is established.
> session2 = new Session("sofia/internal/foo at bar.com", session);
> bridge(session, session2); //or whatever
>
>
> On Fri, Jan 8, 2010 at 1:02 PM, Nicolas Brenner <nicolas at medularis.com>
> wrote:
>>
>> I'm using the default dialplan.
>>
>>
>> On Fri, Jan 8, 2010 at 4:02 PM, Nicolas Brenner <nicolas at medularis.com>
>> wrote:
>> > Not that I'm aware.
>> >
>> >
>> > On Fri, Jan 8, 2010 at 11:41 AM, Brian West <brian at freeswitch.org>
>> > wrote:
>> >> Are you using proxy media?
>> >>
>> >> /b
>> >>
>> >> On Jan 7, 2010, at 11:33 PM, Nicolas Brenner wrote:
>> >>
>> >>> I'm trying to fake a ringback for leg1 of a two-legged call without
>> >>> success.
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From craigesmith at gmail.com  Sat Jan  9 03:59:14 2010
From: craigesmith at gmail.com (Craig Smith)
Date: Sat, 9 Jan 2010 06:59:14 -0500
Subject: [Freeswitch-users] Voicemail - Is it possible?
Message-ID: <14dd859f1001090359y3761d773of56c0118d1da124@mail.gmail.com>

Hi!

I have a legacy PBX, Mitel to be specific, and an ailing AVST voice
mail system that I'm looking to replace with Freeswitch.

I know I can toggle the MWI on or off for a particular extension on
the Mitel by dialing a feature code plus extension.

Is there a way to use voicemail so that I can turn on/off MWI based on
the status of the mailbox.

Turning the light on seems like it would be the easy part, shutting it
off when all messages have been read has me perplexed.

If I used MySQL, could I scan flags on messages in mailboxes to toggle my MWI?

Thanks,

Craig


From a.alalousi at gmail.com  Sat Jan  9 07:30:28 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Sat, 9 Jan 2010 15:30:28 +0000
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <1262999844.21753.52.camel@local.freepabx.com>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
	<4B4728A5.4040805@kinetix.gr>
	<cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>
	<4B4761D5.5010907@kinetix.gr>
	<cd93b3701001081357i56c39e15j262ed05e5351a31a@mail.gmail.com>
	<4B47C77D.4080502@kinetix.gr>
	<1262999844.21753.52.camel@local.freepabx.com>
Message-ID: <cd93b3701001090730p48c9e98vf14b6ee65225d1ea@mail.gmail.com>

Hi David,

Oh God, yes, them.

They are Russians, based in Moscow. Personally, I think such flagrant abuse
of technology for fraud should be pursued to the maximum possible legal
extent, but alas .. Russian telcos, like the profitable mineral and carbon
industries are Mafia territory.

Regards,

Ahmed.

2010/1/9 David Knell <dave at 3c.co.uk>

> > Nonetheless if you want to pursue this and you need help I'd be happy
> > to assist, as FAS is a huge problem for carriers.
>
> FYI - here's some guys advertising "multilingual FAS" - presumably for
> those crooked enough to want to inflate their margins in this manner and
> too thick to do it for themselves:
> http://www.calltermination.com/forums/thread270514.html
>
> --Dave
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From a.alalousi at gmail.com  Sat Jan  9 07:49:43 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Sat, 9 Jan 2010 15:49:43 +0000
Subject: [Freeswitch-users] Rewriting hangup causes
Message-ID: <cd93b3701001090749t36c750c9k55b8ab36e72dacfe@mail.gmail.com>

Dear All,

I have a need to remap hangup causes returned to customers on one of our
clusters. Is there an easy way of achieving this in FS ?

Regards,

Ahmed.
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From anthony.minessale at gmail.com  Sat Jan  9 08:47:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 9 Jan 2010 10:47:52 -0600
Subject: [Freeswitch-users] Voicemail - Is it possible?
In-Reply-To: <14dd859f1001090359y3761d773of56c0118d1da124@mail.gmail.com>
References: <14dd859f1001090359y3761d773of56c0118d1da124@mail.gmail.com>
Message-ID: <191c3a031001090847s15d79443ibdce45b33e3a3017@mail.gmail.com>

Use esl to listen for mwi events on fs then make the call to turn on the
light manually with originate

On Jan 9, 2010 7:36 AM, "Craig Smith" <craigesmith at gmail.com> wrote:

Hi!

I have a legacy PBX, Mitel to be specific, and an ailing AVST voice
mail system that I'm looking to replace with Freeswitch.

I know I can toggle the MWI on or off for a particular extension on
the Mitel by dialing a feature code plus extension.

Is there a way to use voicemail so that I can turn on/off MWI based on
the status of the mailbox.

Turning the light on seems like it would be the easy part, shutting it
off when all messages have been read has me perplexed.

If I used MySQL, could I scan flags on messages in mailboxes to toggle my
MWI?

Thanks,

Craig

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From freeswitch at aastral.net  Sat Jan  9 08:48:20 2010
From: freeswitch at aastral.net (Bill W.)
Date: Sat, 09 Jan 2010 11:48:20 -0500
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <59ad9ca11001090047u7e3b150bpa27ff1936f41993@mail.gmail.com>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>	<4B46792E.6090805@aastral.net>	<E5EFC626-353A-4B54-9F22-59C0C48E3B95@freeswitch.org>	<4B4684F3.8030504@aastral.net>	<74C18F18-89EB-49DF-B5CE-C8D56856ABCD@freeswitch.org>	<4B47C24A.3080305@aastral.net>
	<59ad9ca11001090047u7e3b150bpa27ff1936f41993@mail.gmail.com>
Message-ID: <4B48B354.7000907@aastral.net>

Hey Henry,

I'm not sure what version of Ubuntu he tested against.  The other thing
to note is that he tested the different distros in VMware virtual
machines first.  And only the ones that passed his test there, went on
to testing on real hardware (Debian and Ubuntu).  So there may have been
an an interaction with VMWare that wasn't accounted for.  Again his
focus was fixing the sqlite bug, and reliable traffic shaping.


So here are the results of my initial testing of Ubuntu in MY environment.

I installed Ubuntu server 9.10 i386 on real hardware.

First off, there was an install bug where grub wouldn't install itself
into the MBR on my disks that were already partitioned.  (Raid 1 /boot
partition).  I had to blow away all partitions and re-create.  Then it
installed okay.  It did, however, have a nice feature where you could
specify a hot-spare for RAID 1 during setup.

Secondly, balance-alb mode appears broken for bonded interfaces with my
switch (HP Procurve).  Ubuntu just sat there drooling and not
transmitting packets.  As soon as I changed over to balance-tlb, things
worked.

Thirdly, the clustering tools (openais/pacemaker) are in their infancy.
Openais doesn't even come with an init script to get it started.

And as a general distro thing, Ubuntu doesn't have any tool as elegant
as chkconfig for managing the init scripts.  The administrative features
of the OS don't feel nearly as refined as CentOS or SuSE.

Since I'm trying to create a highly-available clustered freeswitch
environment I need a cluster-ready distro to handle that.  Not having
OpenAIS/pacemaker work out-of-the-box is a show-stopper for me.

So it looks like my choices are CentOS or SuSE.  Right now, I'm leaning
towards using Suse 11.1 and just using ODBC for the freeswitch core
database to get around the SQLite bug.  That way, at least I can use a
current kernel and current software.  True, I can't use the freeswitch
perl module.  But RedHat has had a long-standing perl bug as well.
(Google redhat perl bug).

Also, Karl reported GCC segfaulting during big compiles on SuSE 11.2 but
I haven't run into any.  True, I haven't compiled any kernels, but
freeswitch compiles just fine.

Hope this helps,
Bill



Henry Huang wrote:
> Which version of Ubuntu was tested against?
> It's surprising to find such testing result about CentOS.
> 
> 
> 


From max.bridgewater at gmail.com  Sat Jan  9 09:00:54 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sat, 9 Jan 2010 12:00:54 -0500
Subject: [Freeswitch-users] Help with Portech <-> Freeswitch
Message-ID: <c55d46951001090900p36df42e3ga91e61812d5b8d03@mail.gmail.com>

Hi Guys,

It appears quite a few people in the list are using Portech. Can you please
help me connect Freeswitch to  it for termination puposes?

Here is what I've done so far but without success.

In Freeswitch I created a profile and stored it in under
/usr/local/freeswitch/conf/sip_profiles/external/. Here is the content:

<include>
  <gateway name="portech">
    <param name="username" value="noauthentication" />
    <param name="password" value="noauthentication" />
    <param name="proxy" value="74.24.22.59" />
    <param name="realm" value="74.24.22.59" />
    <param name="register" value="false" />
  </gateway>
</include>

Then, in the /usr/local/freeswitch/confi/dialplan/default.xml, I added the
following extension:

   <extension name="portech">
      <condition field="destination_number" expression="^2801$">
        <action application="bridge"
data="sofia/gateway/portech/5147237479"/>
      </condition>
    </extension>

In Portech MV374, what I did is simply adding one entry in the Mobile/Lan to
mobile table  that consists of  URL: 74.24.22.59 and call Num: #.

Now, when I connect to Freeswitch with Xlite and try to dial extension 2801
it simply says User Busy. i don't even see that attempts are being made to
connect to the Portech gateway.

Any idea?
Thanks in advance.

Max.
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From jcasale at activenetwerx.com  Sat Jan  9 09:40:21 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 9 Jan 2010 17:40:21 +0000
Subject: [Freeswitch-users] Gateway Configuration
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9074E14@prato.activenetwerx.local>

It seems there are two ways to configure an spa3102's fxo port w/
pbx's, you can set the dial plan to <phone_#_of_pstn>@<freeswitch_ip>
or <ext_to_dial>@<freeswitch_ip>.

>From fs's perspective, what exactly is the difference here?
Are there any significant differences between the two methods?
Are there any best practices that should be considered?

Incoming sip did's and a zap line I had all were configured so that
they entered the public context filtered by <phone_#>.

Most of the examples I see for setting up the spa don't function like this
but a couple do?

Thanks!
jlc


From lortas at freenet.de  Sun Jan 10 12:11:16 2010
From: lortas at freenet.de (Holger von Rhein)
Date: Sun, 10 Jan 2010 21:11:16 +0100
Subject: [Freeswitch-users] looking for supported hardware
Message-ID: <4B4A3464.7070503@freenet.de>

Hi,

I plan to set up a telephone system at home. At the moment I have just 
one analogue hardware phone connected to my DSL-Splitter. To build a 
system capable to be my telephone system, I have looked for a list of 
supported hardware by freeswitch, but I could not find one. :(

Especial, I want to know which PCI(e)-Cards to connect an analogue 
hardware phone are well supported / recommended by freeswitch.

Does http://www.asterisk.org/astdocs/node12.html also apply for freeswitch?


On the other side, are there any features of an analogue modem, I have 
to pay attention for? Or will any Linux supported modem be okay to dial out?


Thanks for any hint.


Holger


From larclap at yahoo.com  Sun Jan 10 13:45:59 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Sun, 10 Jan 2010 13:45:59 -0800
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <COL117-W486989B811B372C35935AAB9700@phx.gbl>
References: <012901ca8fe6$e36b71c0$aa425540$@com>,
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>,
	<014801ca8fee$b75f8780$261e9680$@com>,
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>
Message-ID: <010001ca923e$4cd49bb0$e67dd310$@com>

Jeff,

 

I downloaded and install QT as you suggested, and created an environmental
variable QTDIR as C:\Qt\4.6.0 (the installation directory).

 

I then downloaded FreeSWITCH source via svn (v16230) and built it
successfully. I then opened the FSCOMM project within the FreeSWITCH
directory and attempted to build it. It failed with the messages below. Any
ideas on what I might have done wrong? The environment is Windows 7/64.

 

Error  1      error LNK2019: unresolved external symbol
__imp__switch_core_set_globals at 0 referenced in function "public: __thiscall
FSHost::FSHost(class QObject *)" (??0FSHost@@QAE at PAVQObject@@@Z)
fshost.obj       FSComm

Error  2      error LNK2019: unresolved external symbol
__imp__switch_core_setrlimits at 0 referenced in function "public: __thiscall
FSHost::FSHost(class QObject *)" (??0FSHost@@QAE at PAVQObject@@@Z)
fshost.obj       FSComm

Error  3      error LNK2019: unresolved external symbol
__imp__switch_core_destroy at 0 referenced in function "protected: virtual void
__thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj
FSComm

Error  4      error LNK2019: unresolved external symbol
__imp__switch_event_unbind_callback at 4 referenced in function "protected:
virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)
fshost.obj       FSComm

Error  5      error LNK2019: unresolved external symbol
__imp__switch_core_runtime_loop at 4 referenced in function "protected: virtual
void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)    fshost.obj
FSComm

Error  6      error LNK2019: unresolved external symbol
__imp__switch_core_init_and_modload at 12 referenced in function "protected:
virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ) fshost.obj
FSComm

Error  7      error LNK2019: unresolved external symbol
__imp__switch_log_printf referenced in function "protected: virtual void
__thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj
FSComm

Error  8      error LNK2001: unresolved external symbol
__imp__switch_log_printf   mod_qsettings.obj       FSComm

Error  9      error LNK2001: unresolved external symbol
__imp__switch_log_printf   prefportaudio.obj       FSComm

Error  10     error LNK2019: unresolved external symbol
__imp__switch_event_bind at 20 referenced in function "protected: virtual void
__thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj
FSComm

Error  11     error LNK2019: unresolved external symbol
__imp__switch_core_init at 12 referenced in function "protected: virtual void
__thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj
FSComm

Error  12     error LNK2001: unresolved external symbol
__imp__SWITCH_GLOBAL_dirs  fshost.obj    FSComm

Error  13     error LNK2019: unresolved external symbol
__imp__switch_event_dup at 8 referenced in function "void __cdecl
eventHandlerCallback(struct switch_event *)"
(?eventHandlerCallback@@YAXPAUswitch_event@@@Z)       fshost.obj    FSComm

Error  14     error LNK2019: unresolved external symbol
__imp__switch_event_name at 4 referenced in function "private: enum
switch_status_t __thiscall FSHost::processAlegEvent(struct switch_event
*,class QString)"
(?processAlegEvent at FSHost@@AAE?AW4switch_status_t@@PAUswitch_event@@VQString
@@@Z) fshost.obj    FSComm

Error  15     error LNK2019: unresolved external symbol
__imp__switch_event_get_header at 8 referenced in function "private: enum
switch_status_t __thiscall FSHost::processAlegEvent(struct switch_event
*,class QString)"
(?processAlegEvent at FSHost@@AAE?AW4switch_status_t@@PAUswitch_event@@VQString
@@@Z)     fshost.obj       FSComm

Error  16     error LNK2019: unresolved external symbol
__imp__switch_api_execute at 16 referenced in function "public: enum
switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class
QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)
fshost.obj    FSComm

Error  17     error LNK2019: unresolved external symbol
__imp__switch_console_stream_raw_write referenced in function "public: enum
switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class
QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)
fshost.obj    FSComm

Error  18     error LNK2019: unresolved external symbol
__imp__switch_console_stream_write referenced in function "public: enum
switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class
QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)
fshost.obj    FSComm

Error  19     error LNK2019: unresolved external symbol
__imp__switch_xml_parse_str at 8 referenced in function "public: struct
switch_xml * __thiscall XMLBinding::getConfigXML(class QString)"
(?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)
mod_qsettings.obj    FSComm

Error  20     error LNK2019: unresolved external symbol
__imp__switch_event_expand_headers at 8 referenced in function "public: struct
switch_xml * __thiscall XMLBinding::getConfigXML(class QString)"
(?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)
mod_qsettings.obj    FSComm

Error  21     error LNK2019: unresolved external symbol
__imp__switch_event_add_header_string at 16 referenced in function "public:
struct switch_xml * __thiscall XMLBinding::getConfigXML(class QString)"
(?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)
mod_qsettings.obj    FSComm

Error  22     error LNK2019: unresolved external symbol
__imp__switch_event_create_subclass_detailed at 24 referenced in function
_switch_event_create_plain      mod_qsettings.obj    FSComm

Error  23     error LNK2019: unresolved external symbol
__imp__switch_xml_bind_search_function_ret at 16 referenced in function "enum
switch_status_t __cdecl do_config(void)"
(?do_config@@YA?AW4switch_status_t@@XZ)    mod_qsettings.obj    FSComm

Error  24     error LNK2019: unresolved external symbol
__imp__switch_xml_parse_section_string at 4 referenced in function "enum
switch_status_t __cdecl do_config(void)"
(?do_config@@YA?AW4switch_status_t@@XZ)       mod_qsettings.obj    FSComm

Error  25     error LNK2019: unresolved external symbol
__imp__switch_xml_attr_soft at 8 referenced in function "enum switch_status_t
__cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)
mod_qsettings.obj    FSComm

Error  26     error LNK2019: unresolved external symbol
__imp__switch_xml_free at 4 referenced in function "enum switch_status_t
__cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)
mod_qsettings.obj       FSComm

Error  27     error LNK2019: unresolved external symbol
__imp__switch_xml_child at 8 referenced in function "enum switch_status_t
__cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)
mod_qsettings.obj       FSComm

Error  28     error LNK2019: unresolved external symbol
__imp__switch_xml_open_cfg at 12 referenced in function "enum switch_status_t
__cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)
mod_qsettings.obj    FSComm

Error  29     error LNK2019: unresolved external symbol
__imp__switch_find_local_ip at 16 referenced in function "public: void
__thiscall PrefSofia::readConfig(void)" (?readConfig at PrefSofia@@QAEXXZ)
prefsofia.obj FSComm

Error  30     fatal error LNK1120: 27 unresolved externals
debug\FSComm.exe     FSComm

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Thursday, January 07, 2010 7:17 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

 

The windows support is very experimental at this time!
 
You must manually install
http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
 
Then set the environment variable QTDIR in the environment variables. This
can be set from the Computer/Properties/Advanced system settings/Environment
Variables/User Variables settings screen.
 
QTDIR=c:\qt\4.6.0 - or wherever you installed it
 
then restart VS
 

 

  _____  

Date: Thu, 7 Jan 2010 21:45:56 -0200
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

I don't have a Windows machine to test that. Maybe jlenk could give us a
hand since he is the one who has created the visual studio project? 

 

Regards,
Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927



On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:

No Qt installed. I just checked out from
http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into
VS2008. 

Do I need to get
http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

 

 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:

I just downloaded the fscomm project and loaded it into vs2008. I?ve never
programmed in C++ (or c), just C#, so I can?t make anything of the following
two messages:

 

Error      1              error PRJ0019: A tool returned an error code from
"RCC resources.qrc"  FSComm              FSComm

 

Warning               2              The following environment variables
were not found: $(QTDIR)                Project FSComm

 

Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.

 

The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)

 

Lars

Do you have Qt 4.6 installed?
-MC

 


_______________________________________________
FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From jeff at jefflenk.com  Sun Jan 10 14:16:01 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sun, 10 Jan 2010 16:16:01 -0600
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <010001ca923e$4cd49bb0$e67dd310$@com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>, ,
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>, ,
	<014801ca8fee$b75f8780$261e9680$@com>, ,
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>,
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>,
	<010001ca923e$4cd49bb0$e67dd310$@com>
Message-ID: <COL117-W6092C7A9A05EF45B305C54B96E0@phx.gbl>


you must also add FreeSwitchCoreLib as a dependency of FSComm

 

please update http://wiki.freeswitch.org/wiki/FSComm with any additional information you found helpful


 


From: larclap at yahoo.com
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 10 Jan 2010 13:45:59 -0800
Subject: Re: [Freeswitch-users] Compile error fscomm?







Jeff,
 
I downloaded and install QT as you suggested, and created an environmental variable QTDIR as C:\Qt\4.6.0 (the installation directory).
 
I then downloaded FreeSWITCH source via svn (v16230) and built it successfully. I then opened the FSCOMM project within the FreeSWITCH directory and attempted to build it. It failed with the messages below. Any ideas on what I might have done wrong? The environment is Windows 7/64.
 
Error  1      error LNK2019: unresolved external symbol __imp__switch_core_set_globals at 0 referenced in function "public: __thiscall FSHost::FSHost(class QObject *)" (??0FSHost@@QAE at PAVQObject@@@Z)   fshost.obj       FSComm
Error  2      error LNK2019: unresolved external symbol __imp__switch_core_setrlimits at 0 referenced in function "public: __thiscall FSHost::FSHost(class QObject *)" (??0FSHost@@QAE at PAVQObject@@@Z)   fshost.obj       FSComm
Error  3      error LNK2019: unresolved external symbol __imp__switch_core_destroy at 0 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj    FSComm
Error  4      error LNK2019: unresolved external symbol __imp__switch_event_unbind_callback at 4 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)    fshost.obj       FSComm
Error  5      error LNK2019: unresolved external symbol __imp__switch_core_runtime_loop at 4 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)    fshost.obj       FSComm
Error  6      error LNK2019: unresolved external symbol __imp__switch_core_init_and_modload at 12 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ) fshost.obj       FSComm
Error  7      error LNK2019: unresolved external symbol __imp__switch_log_printf referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj    FSComm
Error  8      error LNK2001: unresolved external symbol __imp__switch_log_printf   mod_qsettings.obj       FSComm
Error  9      error LNK2001: unresolved external symbol __imp__switch_log_printf   prefportaudio.obj       FSComm
Error  10     error LNK2019: unresolved external symbol __imp__switch_event_bind at 20 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj    FSComm
Error  11     error LNK2019: unresolved external symbol __imp__switch_core_init at 12 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj    FSComm
Error  12     error LNK2001: unresolved external symbol __imp__SWITCH_GLOBAL_dirs  fshost.obj    FSComm
Error  13     error LNK2019: unresolved external symbol __imp__switch_event_dup at 8 referenced in function "void __cdecl eventHandlerCallback(struct switch_event *)" (?eventHandlerCallback@@YAXPAUswitch_event@@@Z)       fshost.obj    FSComm
Error  14     error LNK2019: unresolved external symbol __imp__switch_event_name at 4 referenced in function "private: enum switch_status_t __thiscall FSHost::processAlegEvent(struct switch_event *,class QString)" (?processAlegEvent at FSHost@@AAE?AW4switch_status_t@@PAUswitch_event@@VQString@@@Z) fshost.obj    FSComm
Error  15     error LNK2019: unresolved external symbol __imp__switch_event_get_header at 8 referenced in function "private: enum switch_status_t __thiscall FSHost::processAlegEvent(struct switch_event *,class QString)" (?processAlegEvent at FSHost@@AAE?AW4switch_status_t@@PAUswitch_event@@VQString@@@Z)     fshost.obj       FSComm
Error  16     error LNK2019: unresolved external symbol __imp__switch_api_execute at 16 referenced in function "public: enum switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)   fshost.obj    FSComm
Error  17     error LNK2019: unresolved external symbol __imp__switch_console_stream_raw_write referenced in function "public: enum switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)  fshost.obj    FSComm
Error  18     error LNK2019: unresolved external symbol __imp__switch_console_stream_write referenced in function "public: enum switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)    fshost.obj    FSComm
Error  19     error LNK2019: unresolved external symbol __imp__switch_xml_parse_str at 8 referenced in function "public: struct switch_xml * __thiscall XMLBinding::getConfigXML(class QString)" (?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)      mod_qsettings.obj    FSComm
Error  20     error LNK2019: unresolved external symbol __imp__switch_event_expand_headers at 8 referenced in function "public: struct switch_xml * __thiscall XMLBinding::getConfigXML(class QString)" (?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)      mod_qsettings.obj    FSComm
Error  21     error LNK2019: unresolved external symbol __imp__switch_event_add_header_string at 16 referenced in function "public: struct switch_xml * __thiscall XMLBinding::getConfigXML(class QString)" (?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)      mod_qsettings.obj    FSComm
Error  22     error LNK2019: unresolved external symbol __imp__switch_event_create_subclass_detailed at 24 referenced in function _switch_event_create_plain      mod_qsettings.obj    FSComm
Error  23     error LNK2019: unresolved external symbol __imp__switch_xml_bind_search_function_ret at 16 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)    mod_qsettings.obj    FSComm
Error  24     error LNK2019: unresolved external symbol __imp__switch_xml_parse_section_string at 4 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)       mod_qsettings.obj    FSComm
Error  25     error LNK2019: unresolved external symbol __imp__switch_xml_attr_soft at 8 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)       mod_qsettings.obj    FSComm
Error  26     error LNK2019: unresolved external symbol __imp__switch_xml_free at 4 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)   mod_qsettings.obj       FSComm
Error  27     error LNK2019: unresolved external symbol __imp__switch_xml_child at 8 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)   mod_qsettings.obj       FSComm
Error  28     error LNK2019: unresolved external symbol __imp__switch_xml_open_cfg at 12 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)       mod_qsettings.obj    FSComm
Error  29     error LNK2019: unresolved external symbol __imp__switch_find_local_ip at 16 referenced in function "public: void __thiscall PrefSofia::readConfig(void)" (?readConfig at PrefSofia@@QAEXXZ)       prefsofia.obj FSComm
Error  30     fatal error LNK1120: 27 unresolved externals    debug\FSComm.exe     FSComm
 


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk
Sent: Thursday, January 07, 2010 7:17 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?
 
The windows support is very experimental at this time!
 
You must manually install http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
 
Then set the environment variable QTDIR in the environment variables. This can be set from the Computer/Properties/Advanced system settings/Environment Variables/User Variables settings screen.
 
QTDIR=c:\qt\4.6.0 - or wherever you installed it
 
then restart VS
 

 



Date: Thu, 7 Jan 2010 21:45:56 -0200
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

I don't have a Windows machine to test that. Maybe jlenk could give us a hand since he is the one who has created the visual studio project? 

 

Regards,Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927



On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:


No Qt installed. I just checked out from http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into VS2008. 
Do I need to get http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?
 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?


 
 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:


I just downloaded the fscomm project and loaded it into vs2008. I?ve never programmed in C++ (or c), just C#, so I can?t make anything of the following two messages:
 
Error      1              error PRJ0019: A tool returned an error code from "RCC resources.qrc"  FSComm              FSComm
 
Warning               2              The following environment variables were not found: $(QTDIR)                Project FSComm
 
Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.
 
The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)
 
Lars

Do you have Qt 4.6 installed?
-MC
 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 
 



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From jcasale at activenetwerx.com  Sun Jan 10 15:08:25 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sun, 10 Jan 2010 23:08:25 +0000
Subject: [Freeswitch-users] looking for supported hardware
In-Reply-To: <4B4A3464.7070503@freenet.de>
References: <4B4A3464.7070503@freenet.de>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9076377@prato.activenetwerx.local>

>Especial, I want to know which PCI(e)-Cards to connect an analogue 
>hardware phone are well supported / recommended by freeswitch.
>
>Does http://www.asterisk.org/astdocs/node12.html also apply for freeswitch?

Those are expensive, and a IMHO the OpenZAP implementation is still in its
infancy and I can't see it getting a strong movement behind it to make it better.

External IP based gateway's are much more easier to deal with. I have heard
good things about Patton's and AudioCodes and both make some cheap models
that are way less money than the digium cards. I set up an SPA2102 for a pair
of FXS's I needed for faxing its working flawless, I am just setting up an
SPA3102 as I needed an FXO port and that's 3/4 working. Those Linksys devices
were very cheap, ~80 CDN I paid for each.

>On the other side, are there any features of an analogue modem, I have 
>to pay attention for? Or will any Linux supported modem be okay to dial out?

Linux will support them fine, but that's not the only concern, IMHO you want
to avoid those, there is nothing that makes them appealing anymore from
my perspective...

jlc


From darklion11 at yahoo.com  Sun Jan 10 17:26:02 2010
From: darklion11 at yahoo.com (Edmar Cruz)
Date: Sun, 10 Jan 2010 17:26:02 -0800 (PST)
Subject: [Freeswitch-users]  Change Domain Freeswitch
Message-ID: <27104680.post@talk.nabble.com>


Dear All,

   How can i change the domain of my freeswitch 52.236.125.12 to
sip.grandminister.com to be able to detect the presence of the user whos
online or not...

Thanks
Edmar
-- 
View this message in context: http://old.nabble.com/Change-Domain-Freeswitch-tp27104680p27104680.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From larclap at yahoo.com  Sun Jan 10 18:25:00 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Sun, 10 Jan 2010 18:25:00 -0800
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <COL117-W6092C7A9A05EF45B305C54B96E0@phx.gbl>
References: <012901ca8fe6$e36b71c0$aa425540$@com>, ,
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>, ,
	<014801ca8fee$b75f8780$261e9680$@com>, ,
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>,
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>,
	<010001ca923e$4cd49bb0$e67dd310$@com>
	<COL117-W6092C7A9A05EF45B305C54B96E0@phx.gbl>
Message-ID: <014b01ca9265$46fa6e50$d4ef4af0$@com>

After realizing I need to build FreeSWITCH, which builds FreeSwitchCore.lib,
I then open fscomm and build it after setting the dependency. Now I get
these messages, which is really strange since the configuration for each
FreeSWITCH and fscomm is set to Debug. The missing files are in .\debug.
What am I missing?

 

Error  1      fatal error C1083: Cannot open source file:
'.\release\qrc_resources.cpp': No such file or directory     c1xx   FSComm

Error  2      fatal error C1083: Cannot open source file:
'.\release\moc_prefsofia.cpp': No such file or directory     c1xx   FSComm

Error  3      fatal error C1083: Cannot open source file:
'.\release\moc_prefportaudio.cpp': No such file or directory  c1xx   FSComm

Error  4      fatal error C1083: Cannot open source file:
'.\release\moc_prefdialog.cpp': No such file or directory     c1xx   FSComm

Error  5      fatal error C1083: Cannot open source file:
'.\release\moc_mainwindow.cpp': No such file or directory     c1xx   FSComm

Error  6      fatal error C1083: Cannot open source file:
'.\release\moc_fshost.cpp': No such file or directory     c1xx   FSComm

Error  7      fatal error C1083: Cannot open source file:
'.\release\moc_accountdialog.cpp': No such file or directory  c1xx   FSComm

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Sunday, January 10, 2010 2:16 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

 

you must also add FreeSwitchCoreLib as a dependency of FSComm
 
please update http://wiki.freeswitch.org/wiki/FSComm with any additional
information you found helpful

 

  _____  

From: larclap at yahoo.com
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 10 Jan 2010 13:45:59 -0800
Subject: Re: [Freeswitch-users] Compile error fscomm?

Jeff,

 

I downloaded and install QT as you suggested, and created an environmental
variable QTDIR as C:\Qt\4.6.0 (the installation directory).

 

I then downloaded FreeSWITCH source via svn (v16230) and built it
successfully. I then opened the FSCOMM project within the FreeSWITCH
directory and attempted to build it. It failed with the messages below. Any
ideas on what I might have done wrong? The environment is Windows 7/64.

 

Error  1      error LNK2019: unresolved external symbol
__imp__switch_core_set_globals at 0 referenced in function "public: __thiscall
FSHost::FSHost(class QObject *)" (??0FSHost@@QAE at PAVQObject@@@Z)
fshost.obj       FSComm

Error  2      error LNK2019: unresolved external symbol
__imp__switch_core_setrlimits at 0 referenced in function "public: __thiscall
FSHost::FSHost(class QObject *)" (??0FSHost@@QAE at PAVQObject@@@Z)
fshost.obj       FSComm

Error  3      error LNK2019: unresolved external symbol
__imp__switch_core_destroy at 0 referenced in function "protected: virtual void
__thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj
FSComm

Error  4      error LNK2019: unresolved external symbol
__imp__switch_event_unbind_callback at 4 referenced in function "protected:
virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)
fshost.obj       FSComm

Error  5      error LNK2019: unresolved external symbol
__imp__switch_core_runtime_loop at 4 referenced in function "protected: virtual
void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)    fshost.obj
FSComm

Error  6      error LNK2019: unresolved external symbol
__imp__switch_core_init_and_modload at 12 referenced in function "protected:
virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ) fshost.obj
FSComm

Error  7      error LNK2019: unresolved external symbol
__imp__switch_log_printf referenced in function "protected: virtual void
__thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj
FSComm

Error  8      error LNK2001: unresolved external symbol
__imp__switch_log_printf   mod_qsettings.obj       FSComm

Error  9      error LNK2001: unresolved external symbol
__imp__switch_log_printf   prefportaudio.obj       FSComm

Error  10     error LNK2019: unresolved external symbol
__imp__switch_event_bind at 20 referenced in function "protected: virtual void
__thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj
FSComm

Error  11     error LNK2019: unresolved external symbol
__imp__switch_core_init at 12 referenced in function "protected: virtual void
__thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj
FSComm

Error  12     error LNK2001: unresolved external symbol
__imp__SWITCH_GLOBAL_dirs  fshost.obj    FSComm

Error  13     error LNK2019: unresolved external symbol
__imp__switch_event_dup at 8 referenced in function "void __cdecl
eventHandlerCallback(struct switch_event *)"
(?eventHandlerCallback@@YAXPAUswitch_event@@@Z)       fshost.obj    FSComm

Error  14     error LNK2019: unresolved external symbol
__imp__switch_event_name at 4 referenced in function "private: enum
switch_status_t __thiscall FSHost::processAlegEvent(struct switch_event
*,class QString)"
(?processAlegEvent at FSHost@@AAE?AW4switch_status_t@@PAUswitch_event@@VQString
@@@Z) fshost.obj    FSComm

Error  15     error LNK2019: unresolved external symbol
__imp__switch_event_get_header at 8 referenced in function "private: enum
switch_status_t __thiscall FSHost::processAlegEvent(struct switch_event
*,class QString)"
(?processAlegEvent at FSHost@@AAE?AW4switch_status_t@@PAUswitch_event@@VQString
@@@Z)     fshost.obj       FSComm

Error  16     error LNK2019: unresolved external symbol
__imp__switch_api_execute at 16 referenced in function "public: enum
switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class
QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)
fshost.obj    FSComm

Error  17     error LNK2019: unresolved external symbol
__imp__switch_console_stream_raw_write referenced in function "public: enum
switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class
QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)
fshost.obj    FSComm

Error  18     error LNK2019: unresolved external symbol
__imp__switch_console_stream_write referenced in function "public: enum
switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class
QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)
fshost.obj    FSComm

Error  19     error LNK2019: unresolved external symbol
__imp__switch_xml_parse_str at 8 referenced in function "public: struct
switch_xml * __thiscall XMLBinding::getConfigXML(class QString)"
(?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)
mod_qsettings.obj    FSComm

Error  20     error LNK2019: unresolved external symbol
__imp__switch_event_expand_headers at 8 referenced in function "public: struct
switch_xml * __thiscall XMLBinding::getConfigXML(class QString)"
(?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)
mod_qsettings.obj    FSComm

Error  21     error LNK2019: unresolved external symbol
__imp__switch_event_add_header_string at 16 referenced in function "public:
struct switch_xml * __thiscall XMLBinding::getConfigXML(class QString)"
(?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)
mod_qsettings.obj    FSComm

Error  22     error LNK2019: unresolved external symbol
__imp__switch_event_create_subclass_detailed at 24 referenced in function
_switch_event_create_plain      mod_qsettings.obj    FSComm

Error  23     error LNK2019: unresolved external symbol
__imp__switch_xml_bind_search_function_ret at 16 referenced in function "enum
switch_status_t __cdecl do_config(void)"
(?do_config@@YA?AW4switch_status_t@@XZ)    mod_qsettings.obj    FSComm

Error  24     error LNK2019: unresolved external symbol
__imp__switch_xml_parse_section_string at 4 referenced in function "enum
switch_status_t __cdecl do_config(void)"
(?do_config@@YA?AW4switch_status_t@@XZ)       mod_qsettings.obj    FSComm

Error  25     error LNK2019: unresolved external symbol
__imp__switch_xml_attr_soft at 8 referenced in function "enum switch_status_t
__cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)
mod_qsettings.obj    FSComm

Error  26     error LNK2019: unresolved external symbol
__imp__switch_xml_free at 4 referenced in function "enum switch_status_t
__cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)
mod_qsettings.obj       FSComm

Error  27     error LNK2019: unresolved external symbol
__imp__switch_xml_child at 8 referenced in function "enum switch_status_t
__cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)
mod_qsettings.obj       FSComm

Error  28     error LNK2019: unresolved external symbol
__imp__switch_xml_open_cfg at 12 referenced in function "enum switch_status_t
__cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)
mod_qsettings.obj    FSComm

Error  29     error LNK2019: unresolved external symbol
__imp__switch_find_local_ip at 16 referenced in function "public: void
__thiscall PrefSofia::readConfig(void)" (?readConfig at PrefSofia@@QAEXXZ)
prefsofia.obj FSComm

Error  30     fatal error LNK1120: 27 unresolved externals
debug\FSComm.exe     FSComm

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Thursday, January 07, 2010 7:17 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

 

The windows support is very experimental at this time!
 
You must manually install
http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
 
Then set the environment variable QTDIR in the environment variables. This
can be set from the Computer/Properties/Advanced system settings/Environment
Variables/User Variables settings screen.
 
QTDIR=c:\qt\4.6.0 - or wherever you installed it
 
then restart VS
 

 

  _____  

Date: Thu, 7 Jan 2010 21:45:56 -0200
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

I don't have a Windows machine to test that. Maybe jlenk could give us a
hand since he is the one who has created the visual studio project? 

 

Regards,
Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927

On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:

No Qt installed. I just checked out from
http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into
VS2008. 

Do I need to get
http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

 

 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:

I just downloaded the fscomm project and loaded it into vs2008. I?ve never
programmed in C++ (or c), just C#, so I can?t make anything of the following
two messages:

 

Error      1              error PRJ0019: A tool returned an error code from
"RCC resources.qrc"  FSComm              FSComm

 

Warning               2              The following environment variables
were not found: $(QTDIR)                Project FSComm

 

Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.

 

The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)

 

Lars

Do you have Qt 4.6 installed?
-MC

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

 

  _____  

Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now.
<http://clk.atdmt.com/GBL/go/196390709/direct/01/> 

 

  _____  

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From andrew at hijacked.us  Sun Jan 10 18:52:48 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Sun, 10 Jan 2010 21:52:48 -0500
Subject: [Freeswitch-users] really no installer for w2k anywhere?
In-Reply-To: <4B46A5C4.9040809@gmail.com>
References: <4B46A5C4.9040809@gmail.com>
Message-ID: <20100111025248.GA10774@hijacked.us>

On Fri, Jan 08, 2010 at 04:25:56AM +0100, Pekka Kurki wrote:
> all installer versions fail with  missing  getnameinfo/getaddressinfo 
> support in w2k.
>

Sofia and some other FS bits use some XP and higher APIs, I never had
the time to properly try to backport FS to win2k.

Really you might want to consider upgrading from an OS that is now over
a decade old if you actually want to use it for a production purpose.
Alternately post a bounty for doing the backport (its a little more
complicated than I'd hoped, IIRC).

Andrew


From jeff at jefflenk.com  Sun Jan 10 20:18:18 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sun, 10 Jan 2010 22:18:18 -0600
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <014b01ca9265$46fa6e50$d4ef4af0$@com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>,
	,,<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>,
	,,<014801ca8fee$b75f8780$261e9680$@com>,
	, , <ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>, ,
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>, ,
	<010001ca923e$4cd49bb0$e67dd310$@com>,
	<COL117-W6092C7A9A05EF45B305C54B96E0@phx.gbl>,
	<014b01ca9265$46fa6e50$d4ef4af0$@com>
Message-ID: <COL117-W27A9215BD93159A12F8BEEB96D0@phx.gbl>


Please make sure you are on svn16231 or later
 


From: larclap at yahoo.com
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 10 Jan 2010 18:25:00 -0800
Subject: Re: [Freeswitch-users] Compile error fscomm?







After realizing I need to build FreeSWITCH, which builds FreeSwitchCore.lib, I then open fscomm and build it after setting the dependency. Now I get these messages, which is really strange since the configuration for each FreeSWITCH and fscomm is set to Debug. The missing files are in .\debug. What am I missing?
 
Error  1      fatal error C1083: Cannot open source file: '.\release\qrc_resources.cpp': No such file or directory     c1xx   FSComm
Error  2      fatal error C1083: Cannot open source file: '.\release\moc_prefsofia.cpp': No such file or directory     c1xx   FSComm
Error  3      fatal error C1083: Cannot open source file: '.\release\moc_prefportaudio.cpp': No such file or directory  c1xx   FSComm
Error  4      fatal error C1083: Cannot open source file: '.\release\moc_prefdialog.cpp': No such file or directory     c1xx   FSComm
Error  5      fatal error C1083: Cannot open source file: '.\release\moc_mainwindow.cpp': No such file or directory     c1xx   FSComm
Error  6      fatal error C1083: Cannot open source file: '.\release\moc_fshost.cpp': No such file or directory     c1xx   FSComm
Error  7      fatal error C1083: Cannot open source file: '.\release\moc_accountdialog.cpp': No such file or directory  c1xx   FSComm
 


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk
Sent: Sunday, January 10, 2010 2:16 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?
 
you must also add FreeSwitchCoreLib as a dependency of FSComm
 
please update http://wiki.freeswitch.org/wiki/FSComm with any additional information you found helpful

 



From: larclap at yahoo.com
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 10 Jan 2010 13:45:59 -0800
Subject: Re: [Freeswitch-users] Compile error fscomm?

Jeff,
 
I downloaded and install QT as you suggested, and created an environmental variable QTDIR as C:\Qt\4.6.0 (the installation directory).
 
I then downloaded FreeSWITCH source via svn (v16230) and built it successfully. I then opened the FSCOMM project within the FreeSWITCH directory and attempted to build it. It failed with the messages below. Any ideas on what I might have done wrong? The environment is Windows 7/64.
 
Error  1      error LNK2019: unresolved external symbol __imp__switch_core_set_globals at 0 referenced in function "public: __thiscall FSHost::FSHost(class QObject *)" (??0FSHost@@QAE at PAVQObject@@@Z)   fshost.obj       FSComm
Error  2      error LNK2019: unresolved external symbol __imp__switch_core_setrlimits at 0 referenced in function "public: __thiscall FSHost::FSHost(class QObject *)" (??0FSHost@@QAE at PAVQObject@@@Z)   fshost.obj       FSComm
Error  3      error LNK2019: unresolved external symbol __imp__switch_core_destroy at 0 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj    FSComm
Error  4      error LNK2019: unresolved external symbol __imp__switch_event_unbind_callback at 4 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)    fshost.obj       FSComm
Error  5      error LNK2019: unresolved external symbol __imp__switch_core_runtime_loop at 4 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)    fshost.obj       FSComm
Error  6      error LNK2019: unresolved external symbol __imp__switch_core_init_and_modload at 12 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ) fshost.obj       FSComm
Error  7      error LNK2019: unresolved external symbol __imp__switch_log_printf referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj    FSComm
Error  8      error LNK2001: unresolved external symbol __imp__switch_log_printf   mod_qsettings.obj       FSComm
Error  9      error LNK2001: unresolved external symbol __imp__switch_log_printf   prefportaudio.obj       FSComm
Error  10     error LNK2019: unresolved external symbol __imp__switch_event_bind at 20 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj    FSComm
Error  11     error LNK2019: unresolved external symbol __imp__switch_core_init at 12 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj    FSComm
Error  12     error LNK2001: unresolved external symbol __imp__SWITCH_GLOBAL_dirs  fshost.obj    FSComm
Error  13     error LNK2019: unresolved external symbol __imp__switch_event_dup at 8 referenced in function "void __cdecl eventHandlerCallback(struct switch_event *)" (?eventHandlerCallback@@YAXPAUswitch_event@@@Z)       fshost.obj    FSComm
Error  14     error LNK2019: unresolved external symbol __imp__switch_event_name at 4 referenced in function "private: enum switch_status_t __thiscall FSHost::processAlegEvent(struct switch_event *,class QString)" (?processAlegEvent at FSHost@@AAE?AW4switch_status_t@@PAUswitch_event@@VQString@@@Z) fshost.obj    FSComm
Error  15     error LNK2019: unresolved external symbol __imp__switch_event_get_header at 8 referenced in function "private: enum switch_status_t __thiscall FSHost::processAlegEvent(struct switch_event *,class QString)" (?processAlegEvent at FSHost@@AAE?AW4switch_status_t@@PAUswitch_event@@VQString@@@Z)     fshost.obj       FSComm
Error  16     error LNK2019: unresolved external symbol __imp__switch_api_execute at 16 referenced in function "public: enum switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)   fshost.obj    FSComm
Error  17     error LNK2019: unresolved external symbol __imp__switch_console_stream_raw_write referenced in function "public: enum switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)  fshost.obj    FSComm
Error  18     error LNK2019: unresolved external symbol __imp__switch_console_stream_write referenced in function "public: enum switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)    fshost.obj    FSComm
Error  19     error LNK2019: unresolved external symbol __imp__switch_xml_parse_str at 8 referenced in function "public: struct switch_xml * __thiscall XMLBinding::getConfigXML(class QString)" (?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)      mod_qsettings.obj    FSComm
Error  20     error LNK2019: unresolved external symbol __imp__switch_event_expand_headers at 8 referenced in function "public: struct switch_xml * __thiscall XMLBinding::getConfigXML(class QString)" (?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)      mod_qsettings.obj    FSComm
Error  21     error LNK2019: unresolved external symbol __imp__switch_event_add_header_string at 16 referenced in function "public: struct switch_xml * __thiscall XMLBinding::getConfigXML(class QString)" (?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)      mod_qsettings.obj    FSComm
Error  22     error LNK2019: unresolved external symbol __imp__switch_event_create_subclass_detailed at 24 referenced in function _switch_event_create_plain      mod_qsettings.obj    FSComm
Error  23     error LNK2019: unresolved external symbol __imp__switch_xml_bind_search_function_ret at 16 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)    mod_qsettings.obj    FSComm
Error  24     error LNK2019: unresolved external symbol __imp__switch_xml_parse_section_string at 4 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)       mod_qsettings.obj    FSComm
Error  25     error LNK2019: unresolved external symbol __imp__switch_xml_attr_soft at 8 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)       mod_qsettings.obj    FSComm
Error  26     error LNK2019: unresolved external symbol __imp__switch_xml_free at 4 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)   mod_qsettings.obj       FSComm
Error  27     error LNK2019: unresolved external symbol __imp__switch_xml_child at 8 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)   mod_qsettings.obj       FSComm
Error  28     error LNK2019: unresolved external symbol __imp__switch_xml_open_cfg at 12 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)       mod_qsettings.obj    FSComm
Error  29     error LNK2019: unresolved external symbol __imp__switch_find_local_ip at 16 referenced in function "public: void __thiscall PrefSofia::readConfig(void)" (?readConfig at PrefSofia@@QAEXXZ)       prefsofia.obj FSComm
Error  30     fatal error LNK1120: 27 unresolved externals    debug\FSComm.exe     FSComm
 


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk
Sent: Thursday, January 07, 2010 7:17 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?
 
The windows support is very experimental at this time!
 
You must manually install http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
 
Then set the environment variable QTDIR in the environment variables. This can be set from the Computer/Properties/Advanced system settings/Environment Variables/User Variables settings screen.
 
QTDIR=c:\qt\4.6.0 - or wherever you installed it
 
then restart VS
 

 



Date: Thu, 7 Jan 2010 21:45:56 -0200
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

I don't have a Windows machine to test that. Maybe jlenk could give us a hand since he is the one who has created the visual studio project? 

 

Regards,Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927

On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:


No Qt installed. I just checked out from http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into VS2008. 
Do I need to get http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?
 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?


 
 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:


I just downloaded the fscomm project and loaded it into vs2008. I?ve never programmed in C++ (or c), just C#, so I can?t make anything of the following two messages:
 
Error      1              error PRJ0019: A tool returned an error code from "RCC resources.qrc"  FSComm              FSComm
 
Warning               2              The following environment variables were not found: $(QTDIR)                Project FSComm
 
Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.
 
The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)
 
Lars

Do you have Qt 4.6 installed?
-MC
 

_______________________________________________
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From jmesquita at freeswitch.org  Sun Jan 10 20:46:18 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 11 Jan 2010 01:46:18 -0300
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <6b65470d1001081209v64d20f9ehdd7a7b4c6516c540@mail.gmail.com>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org>
	<23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com>
	<ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com>
	<23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com>
	<B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>
	<ebbc2d551001071227y22d6eaa8y43742fb4328a67b2@mail.gmail.com>
	<D03B492C-74B9-4B38-B67B-017690FF372C@freeswitch.org>
	<2d9149cd1001081103m4d2b6852l9848ce4c82005fd3@mail.gmail.com>
	<6b65470d1001081209v64d20f9ehdd7a7b4c6516c540@mail.gmail.com>
Message-ID: <ebbc2d551001102046o1dcf14cbkd9a639ccd16ef713@mail.gmail.com>

Thank you both for the support. This week I will make even more changes to
FSComm. Stay tuned and I hope you like it.

As for the sponsorship, Kristian, me and the project rely on that! ;-) Thank
you for even asking!

Regards,
Jo?o Mesquita
Paypal: jmesquita at gmail.com


On Fri, Jan 8, 2010 at 5:09 PM, William Suffill
<william.suffill at gmail.com>wrote:

> jmesquita at gmail.com is his Paypal account. Ya hope he gets better as well.
>
> -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mayamatakeshi at gmail.com  Sun Jan 10 21:00:57 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Mon, 11 Jan 2010 14:00:57 +0900
Subject: [Freeswitch-users] Avoiding restart of transfer_ringback file
	during successive bridge actions.
Message-ID: <15b9404e1001102100u157db096t636f06c0fa774291@mail.gmail.com>

Hi,
I'm implementing ACD functionality using a dialplan.
Basically, I prepare a list of members in a group and try each one of them
in successive bridge actions.
I cannot use a single bridge action with all members separated with pipes
because I am required to check some conditions like if the member is already
in another call. So what I do is to add a transfer action after the bridge
so that the call reenters the dialplan with the remaining of the list.
Up to this point things are fine.
However, if the call is answered and transferred to the group, I set
transfer_ringback so that I can play a file while the bridge is happening.
In case transfer_ringback is set to something like "local_stream://moh",
then we hear a small glitch every time a bridge action happens as the bridge
doesn't take into account that the ringback was already set up by a previous
operation and sets up the playfile operation again. But this is also fine.
The problem is in case transfer_ringback is set to an actual file and not
local_stream. In this case, every bridge action will cause restart of the
playback from the beginning.
So, is it possible to ask bridge to do not setup playback if it is already
running referencing the same file?

regards,
takeshi
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From gabe at gundy.org  Sun Jan 10 21:36:15 2010
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sun, 10 Jan 2010 22:36:15 -0700
Subject: [Freeswitch-users] fifo funk?
Message-ID: <903da5681001102136y6d51f960hd8c10dde4e74ac26@mail.gmail.com>

Anyone else notice that adding and deleting members of a queue works
better then reparsing with the exact same member info?

It seems to keep better track of where to send the next call.  Not a
big deal, just wondering if there is something going on that I don't
know about.

All in all we've really like fifos :) Thanks FS devs!


Gabe


From dujinfang at gmail.com  Sun Jan 10 21:49:52 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 11 Jan 2010 13:49:52 +0800
Subject: [Freeswitch-users] fifo funk?
In-Reply-To: <903da5681001102136y6d51f960hd8c10dde4e74ac26@mail.gmail.com>
References: <903da5681001102136y6d51f960hd8c10dde4e74ac26@mail.gmail.com>
Message-ID: <23f91031001102149x1f1913amefa97d02981261d@mail.gmail.com>

see fifo_member add/delete api

2010/1/11 Gabriel Gunderson <gabe at gundy.org>:
> Anyone else notice that adding and deleting members of a queue works
> better then reparsing with the exact same member info?
>
> It seems to keep better track of where to send the next call. ?Not a
> big deal, just wondering if there is something going on that I don't
> know about.
>
> All in all we've really like fifos :) Thanks FS devs!
>
>
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From sharad at coraltele.com  Sun Jan 10 23:30:55 2010
From: sharad at coraltele.com (Sharad)
Date: Sun, 10 Jan 2010 23:30:55 -0800 (PST)
Subject: [Freeswitch-users] Personal Greeting
In-Reply-To: <53FBFF55-D94D-4BCA-81EC-230F00DDC344@jerris.com>
References: <1262066443847-4226681.post@n2.nabble.com>
	<27073953.post@talk.nabble.com>
	<53FBFF55-D94D-4BCA-81EC-230F00DDC344@jerris.com>
Message-ID: <1263195055974-4284196.post@n2.nabble.com>



Michael Jerris wrote:
> 
> Huh?  What does this have to do with his question?
> 
> On Jan 8, 2010, at 5:25 AM, Edmar Cruz <darklion11 at yahoo.com> wrote:
> 
>>
>> You can set it in the dialplan
>>
>> <action application="playback" data="yourgreeting.wav" />
>>
>> For some cases softphones has its own greeting :working:
>>
>> Hope it can help you..
>>
>>
>> sharad-5 wrote:
>>>
>>>
>>>
>>> Hi
>>>
>>> I am new to Freeswitch so my question may be a stupid question.
>>>
>>> I just want to know how to disable the personal greeting to the  
>>> default
>>> one.
>>> One user has recorded his personal greeting now how can he make this
>>> default.
>>>
>>> I could not find any option for the same.
>>>
>>> Plz advice.
>>>
>>> Thanks & regards
>>> Sharad garg
>>> -- 
>>> View this message in context:
>>> http://n2.nabble.com/Personal-Greeting-tp4226681p4226681.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> -- 
>> View this message in context:
>> http://old.nabble.com/Personal-Greeting-tp26951471p27073953.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

Correct..I also think, dialplan wont help me.

sharad

-- 
View this message in context: http://n2.nabble.com/Personal-Greeting-tp4226681p4284196.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steve at justfone.com  Mon Jan 11 01:10:21 2010
From: steve at justfone.com (Steven Brown)
Date: Mon, 11 Jan 2010 09:10:21 +0000
Subject: [Freeswitch-users] RE Help with Portech <-> Freeswitch (Max
	Bridgewater)
Message-ID: <3e6d7b0c1001110110k474ba234lfc91210735e3a63d@mail.gmail.com>

Hi,

Two ways I've done it (not to say these are the 'correct' ways but
they do work for me after a lot of trial and error) are either to get
the Portech to register on FS as a regular endpoint then  bridge to it
as below

<extension name="gsm">
      <condition field="destination_number" expression="^9(.*)$">
	<action application="set" data="hangup_after_bridge=true"/>
        <action application="bridge" data="sofia/internal/1005^$1@${domain}">

where the Portech registers as 1005 and I prefix the number I wish to
dial with 9

or alternatively just bridge directly to it

<action application="bridge" data="sofia/default/$1 at 192.168.1.3:5060"/>

Where 192.168.1.3 is my Portech

In both cases you need to configure the Portech Lan to Mobile Table as below

Item  URL                 Call Num
0 	192.168.1.2 	#

where 192.168.1.2 is my FS box

This should get things going for one GSM channel, to get multiple
channels running you will have to tweak it differently depending on
the age of the Portech as I've discovered different firmware versions
seem to behave very differently in this respect.

Hope this helps

Steve




> Message: 1
> Date: Sat, 9 Jan 2010 12:00:54 -0500
> From: Max Bridgewater <max.bridgewater at gmail.com>
> Subject: [Freeswitch-users] Help with Portech <-> Freeswitch
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
> ? ? ? ?<c55d46951001090900p36df42e3ga91e61812d5b8d03 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Guys,
>
> It appears quite a few people in the list are using Portech. Can you please
> help me connect Freeswitch to ?it for termination puposes?
>
> Here is what I've done so far but without success.
>
> In Freeswitch I created a profile and stored it in under
> /usr/local/freeswitch/conf/sip_profiles/external/. Here is the content:
>
> <include>
> ?<gateway name="portech">
> ? ?<param name="username" value="noauthentication" />
> ? ?<param name="password" value="noauthentication" />
> ? ?<param name="proxy" value="74.24.22.59" />
> ? ?<param name="realm" value="74.24.22.59" />
> ? ?<param name="register" value="false" />
> ?</gateway>
> </include>
>
> Then, in the /usr/local/freeswitch/confi/dialplan/default.xml, I added the
> following extension:
>
> ? <extension name="portech">
> ? ? ?<condition field="destination_number" expression="^2801$">
> ? ? ? ?<action application="bridge"
> data="sofia/gateway/portech/5147237479"/>
> ? ? ?</condition>
> ? ?</extension>
>
> In Portech MV374, what I did is simply adding one entry in the Mobile/Lan to
> mobile table ?that consists of ?URL: 74.24.22.59 and call Num: #.
>
> Now, when I connect to Freeswitch with Xlite and try to dial extension 2801
> it simply says User Busy. i don't even see that attempts are being made to
> connect to the Portech gateway.
>
> Any idea?
> Thanks in advance.
>
> Max.


From oscav at hotmail.fr  Mon Jan 11 01:43:05 2010
From: oscav at hotmail.fr (Oscav)
Date: Mon, 11 Jan 2010 01:43:05 -0800 (PST)
Subject: [Freeswitch-users]  URGENT : DTMF during bridge
Message-ID: <27107895.post@talk.nabble.com>


Hi,

I need to handle DTMF during bridge in order to hangup the called party on
caller request. The DTMF sequence should be ##. Any idea on how to do that??

Thanks.
-- 
View this message in context: http://old.nabble.com/URGENT-%3A-DTMF-during-bridge-tp27107895p27107895.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From mcampbellsmith at gmail.com  Mon Jan 11 03:16:33 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 11 Jan 2010 22:16:33 +1100
Subject: [Freeswitch-users] Bypass_media mode
Message-ID: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>

Hi!
Hi!

I am calling from extension 2010 to extension 1000.  Both have ip
addesses 192.168.1.x.

In the 2000 series dialplan (a separate context) I have the following
to try to enable bypass_media.

            <extension name="Local_1000_9xxx">
                  <condition field="destination_number"
expression="^(10[01][0-9]|9\d{3})$">
                        <action application="set" data="dialed_extension=$1"/>
                        <action application="set" data="proxy_media=false"/>
                        <action application="set" data="bypass_media=true"/>
                        <action application="transfer"
data="${dialed_extension} XML default"/>
                  </condition>
            </extension>

Is this how bypass media should be enabled?

This fails fo me (the calls hang up and no audio).  The debug trace is
in http://pastebin.freeswitch.org/11737

What have I done wrong?

Thanks


From r.mokhtarpour at yahoo.com  Mon Jan 11 01:57:43 2010
From: r.mokhtarpour at yahoo.com (reza mokhtarpour)
Date: Mon, 11 Jan 2010 01:57:43 -0800 (PST)
Subject: [Freeswitch-users] gtalk and g723 codec
Message-ID: <769338.99062.qm@web33205.mail.mud.yahoo.com>

Hi there

I am begginer with FS , I use FS plus Gtalk?? everything is OK with PCMU codec but whenever? i replace it whit G723 codec i got "This codec is only usable in passthrough mode!" error.

SIP GW? >? FreeSwitch? >? Gtalk

these are my configuration files :

------- public.xml ----------------

<extension name="Gtalk">
??<condition field="source" expression="mod_sofia"/>
??<condition field="destination_number" expression="^1234$">
?<!-- <action application="set" data="proxy_media=true"/> -->
? <action application="set" data="bypass_media=true"/>
????<action application="bridge" data="dingaling/gmail.com/xxxxx at gmail.com"/>
??</condition>
</extension>

------------? dingaling.conf.xml ------------

<configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
? <settings>
??? <param name="debug" value="1"/>
??? <param name="codec-prefs" value="G723"/>
? </settings>

? <X-PRE-PROCESS cmd="include" data="../jingle_profiles/*.xml"/>

</configuration>

---------- sip_profiles/external.xml ---------------

<!-- 
?<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
?<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
-->
<param name="inbound-codec-prefs" value="G723"/>
? <param name="outbound-codec-prefs" value="G723"/>


I was googling for a while but I got nothing.
Thanks in advance.





      
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From a.alalousi at gmail.com  Mon Jan 11 04:18:11 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Mon, 11 Jan 2010 12:18:11 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
Message-ID: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>

Dear All,

I posted a thread re the subject but didn't get any joy, so perhaps second
time lucky.

I need to rewrite a couple of hangup causes to mean NORMAL_CONGESTION and no
matter which variables I set, this isn't happening. The idea is to control
what codes are returned to an end point after a successful bridge, as well
as deal with what codes are returned if the bridge is unsuccessful (e.g.
user_busy, originator_cancel ...etc).

I've had limited success by setting hangup_after_bridge=false then bridging
to error/<required_code>. This, however only works when the B-leg terminates
the call after a successful answer. Any other codes are not rewritten.

I've also tried playing with the bridge_hangup_code and hangup_code
variables prior and after bridging, still no joy. I have also set
sip_ignore_remote_cause=true prior to entering the bridge, as well
explicitly in vars.xml.

By the way, I'm running in proxy-media mode, but I did try it with
bypass-media as well. Same symptoms, same behaviour.

Any help with this would be highly appreciated.

Regards,

Ahmed.
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From john_re at fastmail.us  Mon Jan 11 04:21:57 2010
From: john_re at fastmail.us (giovanni_re)
Date: Mon, 11 Jan 2010 04:21:57 -0800
Subject: [Freeswitch-users] TONIGHT Join 5-6P Mon 11th - 1st Evening Meeting
 test IRC & VOIP online freeswitch at BerkeleyTIP-Global - for forwarding
Message-ID: <1263212517.5989.1354001243@webmail.messagingengine.com>

You're invited to the first test of the Global freeswitch bimonthly
evening meetings at BerkeleyTIP-Global.  :)

Join in tonight, Monday Jan 11, 5-6P Pacific, 8-9P Eastern, = Tues Jan
12 1A-2A UTC.
http://sites.google.com/site/berkeleytip/schedule

On #berkeleytip on irc.freenode.net,
& on voip - whatever is working - try btip server first.
http://sites.google.com/site/berkeleytip/remote-attendance

This will be an online only meeting - no in person meeting at UCB.

Hot topics:  Community Leadership Summit review of interesting sessions,
Spring 2010 efforts for UCB & all UC's & all college activities,
Upcoming KDE conference end of next week, for 1 week, in Los Angeles.

What do _you_ want to discuss?

==
Some people have asked for an evening meeting, because: a) they can't
make weekend meetings, b) they want more BTIP-Global. ;)

So, this will be a test, everyone invited, to see if we can make this
work.

==  BerkeleyTIP-Global is the Global All Free SW HW & Culture meeting
online via VOIP.
http://sites.google.com/site/berkeleytip/

Join the global mailing list, say "hi", & what you're interested in. :)
http://groups.google.com/group/BerkTIPGlobal

For Forwarding: You are invited to forward this announcement wherever it
might be appreciated.


From devel at thom.fr.eu.org  Mon Jan 11 04:53:45 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Mon, 11 Jan 2010 13:53:45 +0100
Subject: [Freeswitch-users] Sending call notification tone to a busy FXS port
Message-ID: <08abf4fa1e11a07bda4381fda8f93879@thom.fr.eu.org>

Hello,

I was just wondering if it is possible (and how) to send a call
notification tone to a phone connected to an FXS port and which is already
in communication.

Thanks

Fran?ois


From a.alalousi at gmail.com  Mon Jan 11 05:15:01 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Mon, 11 Jan 2010 13:15:01 +0000
Subject: [Freeswitch-users] Codecs and things
In-Reply-To: <b39273530912221645w8897121lf2938f59f541df8f@mail.gmail.com>
References: <cd93b3700912221255ve466667qb6e40961dbd05d99@mail.gmail.com>
	<b39273530912221645w8897121lf2938f59f541df8f@mail.gmail.com>
Message-ID: <cd93b3701001110515o47da76ffgc695222f3c4c6b97@mail.gmail.com>

Hi Rupa,

Thanks for your feedback.

One thing that caught my eye in your reply is you mentioning that proxy
media is a special hack for T38. The reason I'm using proxy mode is to fully
mask the identity of end points from each other. If there is another of
achieving this in FS then I am definitely very interested in hearing about
it and implementing it.

You also mention regular mode, and I couldn't find mention of this anywhere
in the Wiki. So far, my understanding was that you could run FS in either
proxy or bypass media modes. Can you elaborate a little bit and discuss this
a bit more ?

Thanks a lot in advance.

Regards,

Ahmed.


2009/12/23 Rupa Schomaker <rupa at rupa.com>

> On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:
> > Hello people,
> >
> > Can someone please clear the following ambiguities with codecs:
> >
> > Are we definitively able to run pass-through codecs (e.g. G.729) in Proxy
> > Media mode, or does FS need to be running in bypass-media ? the Wiki is
> not
> > clear in this regard
>
> Yes, you can use proxy media, bypass media, or even regular mode if
> you don't transcode (special for g729).  Proxy media is really a
> special hack that should only be used for T38 passthrough.  If you are
> using it for other purposes, think about it some more....
>
> > When an A-leg has negotiated a pass-through media codec, can the B-leg be
> > transcoded into a non-pass-through codec, and vice-versa ? think A-leg
> > incoming with a G.729 codec, and target for B-leg needs to be setup with
> a
> > GSM-codec, say
>
> That would require transcoding - which can't be done if the codec is
> pass-through.
>
> > Where in the developer's set of documentation are codecs discussed ? I
> would
> > like to start porting some code of mine for G.729a/b/ab form a ti DSP
> > platform to FS. FS lacking full G.729 support is proving quite a
> hindrance,
> > and there is no clear direction from the dev community as to when the
> same
> > will be available. Incidentally, any news on this effort ? where are we
> with
> > code, and what's an ETA for a Beta ?
>
> I'd say look at the broadvoice or other simple self-contained codecs
> are done.  Currently the only supported g729 solution is to use a
> digium board with mod_dahdi_codec.
>
> I don't have any info on a software based g729 solution.
>
> > On the same lines as (3) above, there is a codec dev template in the
> source
> > tree. Again, where can I find documentation relating to this ? the
> template
> > has hardly any docs at all.
> >
> > Best regards and warm wishes for a Merry Christmas and a great New Year
> to
> > one and all.
> >
> > Ahmed.
> >
> >
> > --
> > Ahmed A. Ibrahim-Naji Al-Alousi
> > Ph.D., MIEE, MBCS
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From rupa at rupa.com  Mon Jan 11 06:18:40 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 11 Jan 2010 08:18:40 -0600
Subject: [Freeswitch-users] Avoiding restart of transfer_ringback file
	during successive bridge actions.
In-Reply-To: <15b9404e1001102100u157db096t636f06c0fa774291@mail.gmail.com>
References: <15b9404e1001102100u157db096t636f06c0fa774291@mail.gmail.com>
Message-ID: <b39273531001110618r7d3df336i1cd9fb5d13a9ee7d@mail.gmail.com>

Maybe set a var that, when set, causes the dp to skip restarting the
ringback?

On Sun, Jan 10, 2010 at 11:00 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

> Hi,
> I'm implementing ACD functionality using a dialplan.
> Basically, I prepare a list of members in a group and try each one of them
> in successive bridge actions.
> I cannot use a single bridge action with all members separated with pipes
> because I am required to check some conditions like if the member is already
> in another call. So what I do is to add a transfer action after the bridge
> so that the call reenters the dialplan with the remaining of the list.
> Up to this point things are fine.
> However, if the call is answered and transferred to the group, I set
> transfer_ringback so that I can play a file while the bridge is happening.
> In case transfer_ringback is set to something like "local_stream://moh",
> then we hear a small glitch every time a bridge action happens as the bridge
> doesn't take into account that the ringback was already set up by a previous
> operation and sets up the playfile operation again. But this is also fine.
> The problem is in case transfer_ringback is set to an actual file and not
> local_stream. In this case, every bridge action will cause restart of the
> playback from the beginning.
> So, is it possible to ask bridge to do not setup playback if it is already
> running referencing the same file?
>
> regards,
> takeshi
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From rupa at rupa.com  Mon Jan 11 06:23:14 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 11 Jan 2010 08:23:14 -0600
Subject: [Freeswitch-users] Codecs and things
In-Reply-To: <cd93b3701001110515o47da76ffgc695222f3c4c6b97@mail.gmail.com>
References: <cd93b3700912221255ve466667qb6e40961dbd05d99@mail.gmail.com>
	<b39273530912221645w8897121lf2938f59f541df8f@mail.gmail.com>
	<cd93b3701001110515o47da76ffgc695222f3c4c6b97@mail.gmail.com>
Message-ID: <b39273531001110623q4795d6a9i2bc8bfa2760ba99a@mail.gmail.com>

On Mon, Jan 11, 2010 at 7:15 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:

> Hi Rupa,
>
> Thanks for your feedback.
>
> One thing that caught my eye in your reply is you mentioning that proxy
> media is a special hack for T38. The reason I'm using proxy mode is to fully
> mask the identity of end points from each other. If there is another of
> achieving this in FS then I am definitely very interested in hearing about
> it and implementing it.
>
>
The stock answer is that FS is not a proxy.  If you want a proxy use proxy
software (opensips/kamilio/whatever).  Proxy media mode is generally not the
right answer.


> You also mention regular mode, and I couldn't find mention of this anywhere
> in the Wiki. So far, my understanding was that you could run FS in either
> proxy or bypass media modes. Can you elaborate a little bit and discuss this
> a bit more ?
>
>
Regular mode is just the default mode FS runs in where all media passes
through it.


> Thanks a lot in advance.
>
> Regards,
>
> Ahmed.
>
>
> 2009/12/23 Rupa Schomaker <rupa at rupa.com>
>
> On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>> > Hello people,
>> >
>> > Can someone please clear the following ambiguities with codecs:
>> >
>> > Are we definitively able to run pass-through codecs (e.g. G.729) in
>> Proxy
>> > Media mode, or does FS need to be running in bypass-media ? the Wiki is
>> not
>> > clear in this regard
>>
>> Yes, you can use proxy media, bypass media, or even regular mode if
>> you don't transcode (special for g729).  Proxy media is really a
>> special hack that should only be used for T38 passthrough.  If you are
>> using it for other purposes, think about it some more....
>>
>> > When an A-leg has negotiated a pass-through media codec, can the B-leg
>> be
>> > transcoded into a non-pass-through codec, and vice-versa ? think A-leg
>> > incoming with a G.729 codec, and target for B-leg needs to be setup with
>> a
>> > GSM-codec, say
>>
>> That would require transcoding - which can't be done if the codec is
>> pass-through.
>>
>> > Where in the developer's set of documentation are codecs discussed ? I
>> would
>> > like to start porting some code of mine for G.729a/b/ab form a ti DSP
>> > platform to FS. FS lacking full G.729 support is proving quite a
>> hindrance,
>> > and there is no clear direction from the dev community as to when the
>> same
>> > will be available. Incidentally, any news on this effort ? where are we
>> with
>> > code, and what's an ETA for a Beta ?
>>
>> I'd say look at the broadvoice or other simple self-contained codecs
>> are done.  Currently the only supported g729 solution is to use a
>> digium board with mod_dahdi_codec.
>>
>> I don't have any info on a software based g729 solution.
>>
>> > On the same lines as (3) above, there is a codec dev template in the
>> source
>> > tree. Again, where can I find documentation relating to this ? the
>> template
>> > has hardly any docs at all.
>> >
>> > Best regards and warm wishes for a Merry Christmas and a great New Year
>> to
>> > one and all.
>> >
>> > Ahmed.
>> >
>> >
>> > --
>> > Ahmed A. Ibrahim-Naji Al-Alousi
>> > Ph.D., MIEE, MBCS
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From a.alalousi at gmail.com  Mon Jan 11 06:39:20 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Mon, 11 Jan 2010 14:39:20 +0000
Subject: [Freeswitch-users] Codecs and things
In-Reply-To: <b39273531001110623q4795d6a9i2bc8bfa2760ba99a@mail.gmail.com>
References: <cd93b3700912221255ve466667qb6e40961dbd05d99@mail.gmail.com>
	<b39273530912221645w8897121lf2938f59f541df8f@mail.gmail.com>
	<cd93b3701001110515o47da76ffgc695222f3c4c6b97@mail.gmail.com>
	<b39273531001110623q4795d6a9i2bc8bfa2760ba99a@mail.gmail.com>
Message-ID: <cd93b3701001110639t8b222d1i344ae445fb86f501@mail.gmail.com>

Hi Rupa,

Thanks for this.

Re: regular mode, are we saying to set both bypass-media and proxy-media to
false, and this would put it into regular mode ?

I'll look into alternatives re: proxy per your feedback.

Regards,

Ahmed.


2010/1/11 Rupa Schomaker <rupa at rupa.com>

>
>
> On Mon, Jan 11, 2010 at 7:15 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>
>> Hi Rupa,
>>
>> Thanks for your feedback.
>>
>> One thing that caught my eye in your reply is you mentioning that proxy
>> media is a special hack for T38. The reason I'm using proxy mode is to fully
>> mask the identity of end points from each other. If there is another of
>> achieving this in FS then I am definitely very interested in hearing about
>> it and implementing it.
>>
>>
> The stock answer is that FS is not a proxy.  If you want a proxy use proxy
> software (opensips/kamilio/whatever).  Proxy media mode is generally not the
> right answer.
>
>
>> You also mention regular mode, and I couldn't find mention of this
>> anywhere in the Wiki. So far, my understanding was that you could run FS in
>> either proxy or bypass media modes. Can you elaborate a little bit and
>> discuss this a bit more ?
>>
>>
> Regular mode is just the default mode FS runs in where all media passes
> through it.
>
>
>> Thanks a lot in advance.
>>
>> Regards,
>>
>> Ahmed.
>>
>>
>> 2009/12/23 Rupa Schomaker <rupa at rupa.com>
>>
>> On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>>> > Hello people,
>>> >
>>> > Can someone please clear the following ambiguities with codecs:
>>> >
>>> > Are we definitively able to run pass-through codecs (e.g. G.729) in
>>> Proxy
>>> > Media mode, or does FS need to be running in bypass-media ? the Wiki is
>>> not
>>> > clear in this regard
>>>
>>> Yes, you can use proxy media, bypass media, or even regular mode if
>>> you don't transcode (special for g729).  Proxy media is really a
>>> special hack that should only be used for T38 passthrough.  If you are
>>> using it for other purposes, think about it some more....
>>>
>>> > When an A-leg has negotiated a pass-through media codec, can the B-leg
>>> be
>>> > transcoded into a non-pass-through codec, and vice-versa ? think A-leg
>>> > incoming with a G.729 codec, and target for B-leg needs to be setup
>>> with a
>>> > GSM-codec, say
>>>
>>> That would require transcoding - which can't be done if the codec is
>>> pass-through.
>>>
>>> > Where in the developer's set of documentation are codecs discussed ? I
>>> would
>>> > like to start porting some code of mine for G.729a/b/ab form a ti DSP
>>> > platform to FS. FS lacking full G.729 support is proving quite a
>>> hindrance,
>>> > and there is no clear direction from the dev community as to when the
>>> same
>>> > will be available. Incidentally, any news on this effort ? where are we
>>> with
>>> > code, and what's an ETA for a Beta ?
>>>
>>> I'd say look at the broadvoice or other simple self-contained codecs
>>> are done.  Currently the only supported g729 solution is to use a
>>> digium board with mod_dahdi_codec.
>>>
>>> I don't have any info on a software based g729 solution.
>>>
>>> > On the same lines as (3) above, there is a codec dev template in the
>>> source
>>> > tree. Again, where can I find documentation relating to this ? the
>>> template
>>> > has hardly any docs at all.
>>> >
>>> > Best regards and warm wishes for a Merry Christmas and a great New Year
>>> to
>>> > one and all.
>>> >
>>> > Ahmed.
>>> >
>>> >
>>> > --
>>> > Ahmed A. Ibrahim-Naji Al-Alousi
>>> > Ph.D., MIEE, MBCS
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Ahmed Naji
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From sos at sokhapkin.dyndns.org  Mon Jan 11 06:48:49 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 11 Jan 2010 09:48:49 -0500
Subject: [Freeswitch-users] Codecs and things
In-Reply-To: <cd93b3701001110639t8b222d1i344ae445fb86f501@mail.gmail.com>
References: <cd93b3700912221255ve466667qb6e40961dbd05d99@mail.gmail.com>
	<b39273531001110623q4795d6a9i2bc8bfa2760ba99a@mail.gmail.com>
	<cd93b3701001110639t8b222d1i344ae445fb86f501@mail.gmail.com>
Message-ID: <201001110948.49710.sos@sokhapkin.dyndns.org>

Regular mode is when neither variable is set or set to false.

On Monday 11 January 2010, Ahmed Naji wrote:
> Hi Rupa,
>
> Thanks for this.
>
> Re: regular mode, are we saying to set both bypass-media and proxy-media to
> false, and this would put it into regular mode ?
>
> I'll look into alternatives re: proxy per your feedback.
>
> Regards,
>
> Ahmed.
>
>
> 2010/1/11 Rupa Schomaker <rupa at rupa.com>
>
> > On Mon, Jan 11, 2010 at 7:15 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:
> >> Hi Rupa,
> >>
> >> Thanks for your feedback.
> >>
> >> One thing that caught my eye in your reply is you mentioning that proxy
> >> media is a special hack for T38. The reason I'm using proxy mode is to
> >> fully mask the identity of end points from each other. If there is
> >> another of achieving this in FS then I am definitely very interested in
> >> hearing about it and implementing it.
> >
> > The stock answer is that FS is not a proxy.  If you want a proxy use
> > proxy software (opensips/kamilio/whatever).  Proxy media mode is
> > generally not the right answer.
> >
> >> You also mention regular mode, and I couldn't find mention of this
> >> anywhere in the Wiki. So far, my understanding was that you could run FS
> >> in either proxy or bypass media modes. Can you elaborate a little bit
> >> and discuss this a bit more ?
> >
> > Regular mode is just the default mode FS runs in where all media passes
> > through it.
> >
> >> Thanks a lot in advance.
> >>
> >> Regards,
> >>
> >> Ahmed.
> >>
> >>
> >> 2009/12/23 Rupa Schomaker <rupa at rupa.com>
> >>
> >> On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:
> >>> > Hello people,
> >>> >
> >>> > Can someone please clear the following ambiguities with codecs:
> >>> >
> >>> > Are we definitively able to run pass-through codecs (e.g. G.729) in
> >>>
> >>> Proxy
> >>>
> >>> > Media mode, or does FS need to be running in bypass-media ? the Wiki
> >>> > is
> >>>
> >>> not
> >>>
> >>> > clear in this regard
> >>>
> >>> Yes, you can use proxy media, bypass media, or even regular mode if
> >>> you don't transcode (special for g729).  Proxy media is really a
> >>> special hack that should only be used for T38 passthrough.  If you are
> >>> using it for other purposes, think about it some more....
> >>>
> >>> > When an A-leg has negotiated a pass-through media codec, can the
> >>> > B-leg
> >>>
> >>> be
> >>>
> >>> > transcoded into a non-pass-through codec, and vice-versa ? think
> >>> > A-leg incoming with a G.729 codec, and target for B-leg needs to be
> >>> > setup
> >>>
> >>> with a
> >>>
> >>> > GSM-codec, say
> >>>
> >>> That would require transcoding - which can't be done if the codec is
> >>> pass-through.
> >>>
> >>> > Where in the developer's set of documentation are codecs discussed ?
> >>> > I
> >>>
> >>> would
> >>>
> >>> > like to start porting some code of mine for G.729a/b/ab form a ti DSP
> >>> > platform to FS. FS lacking full G.729 support is proving quite a
> >>>
> >>> hindrance,
> >>>
> >>> > and there is no clear direction from the dev community as to when the
> >>>
> >>> same
> >>>
> >>> > will be available. Incidentally, any news on this effort ? where are
> >>> > we
> >>>
> >>> with
> >>>
> >>> > code, and what's an ETA for a Beta ?
> >>>
> >>> I'd say look at the broadvoice or other simple self-contained codecs
> >>> are done.  Currently the only supported g729 solution is to use a
> >>> digium board with mod_dahdi_codec.
> >>>
> >>> I don't have any info on a software based g729 solution.
> >>>
> >>> > On the same lines as (3) above, there is a codec dev template in the
> >>>
> >>> source
> >>>
> >>> > tree. Again, where can I find documentation relating to this ? the
> >>>
> >>> template
> >>>
> >>> > has hardly any docs at all.
> >>> >
> >>> > Best regards and warm wishes for a Merry Christmas and a great New
> >>> > Year
> >>>
> >>> to
> >>>
> >>> > one and all.
> >>> >
> >>> > Ahmed.
> >>> >
> >>> >
> >>> > --
> >>> > Ahmed A. Ibrahim-Naji Al-Alousi
> >>> > Ph.D., MIEE, MBCS
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> >>>
> >>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>
> >>> > http://www.freeswitch.org
> >>>
> >>> --
> >>> -Rupa
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> >>>s http://www.freeswitch.org
> >>
> >> --
> >> Ahmed Naji
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > -Rupa
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From rupa at rupa.com  Mon Jan 11 06:52:22 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 11 Jan 2010 08:52:22 -0600
Subject: [Freeswitch-users] Codecs and things
In-Reply-To: <cd93b3701001110639t8b222d1i344ae445fb86f501@mail.gmail.com>
References: <cd93b3700912221255ve466667qb6e40961dbd05d99@mail.gmail.com>
	<b39273530912221645w8897121lf2938f59f541df8f@mail.gmail.com>
	<cd93b3701001110515o47da76ffgc695222f3c4c6b97@mail.gmail.com>
	<b39273531001110623q4795d6a9i2bc8bfa2760ba99a@mail.gmail.com>
	<cd93b3701001110639t8b222d1i344ae445fb86f501@mail.gmail.com>
Message-ID: <b39273531001110652k6920cdb4k48f9729f5fa5f13e@mail.gmail.com>

On Mon, Jan 11, 2010 at 8:39 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:

> Hi Rupa,
>
> Thanks for this.
>
> Re: regular mode, are we saying to set both bypass-media and proxy-media to
> false, and this would put it into regular mode ?
>
>
Or just don't set either.  Default is regular mode (why I called it regular
mode).  I'm not sure what the proper term is, "media mode" maybe ?


> I'll look into alternatives re: proxy per your feedback.
>
> Regards,
>
> Ahmed.
>
>
> 2010/1/11 Rupa Schomaker <rupa at rupa.com>
>
>
>>
>> On Mon, Jan 11, 2010 at 7:15 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>>
>>> Hi Rupa,
>>>
>>> Thanks for your feedback.
>>>
>>> One thing that caught my eye in your reply is you mentioning that proxy
>>> media is a special hack for T38. The reason I'm using proxy mode is to fully
>>> mask the identity of end points from each other. If there is another of
>>> achieving this in FS then I am definitely very interested in hearing about
>>> it and implementing it.
>>>
>>>
>> The stock answer is that FS is not a proxy.  If you want a proxy use proxy
>> software (opensips/kamilio/whatever).  Proxy media mode is generally not the
>> right answer.
>>
>>
>>> You also mention regular mode, and I couldn't find mention of this
>>> anywhere in the Wiki. So far, my understanding was that you could run FS in
>>> either proxy or bypass media modes. Can you elaborate a little bit and
>>> discuss this a bit more ?
>>>
>>>
>> Regular mode is just the default mode FS runs in where all media passes
>> through it.
>>
>>
>>> Thanks a lot in advance.
>>>
>>> Regards,
>>>
>>> Ahmed.
>>>
>>>
>>> 2009/12/23 Rupa Schomaker <rupa at rupa.com>
>>>
>>> On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji <a.alalousi at gmail.com>
>>>> wrote:
>>>> > Hello people,
>>>> >
>>>> > Can someone please clear the following ambiguities with codecs:
>>>> >
>>>> > Are we definitively able to run pass-through codecs (e.g. G.729) in
>>>> Proxy
>>>> > Media mode, or does FS need to be running in bypass-media ? the Wiki
>>>> is not
>>>> > clear in this regard
>>>>
>>>> Yes, you can use proxy media, bypass media, or even regular mode if
>>>> you don't transcode (special for g729).  Proxy media is really a
>>>> special hack that should only be used for T38 passthrough.  If you are
>>>> using it for other purposes, think about it some more....
>>>>
>>>> > When an A-leg has negotiated a pass-through media codec, can the B-leg
>>>> be
>>>> > transcoded into a non-pass-through codec, and vice-versa ? think A-leg
>>>> > incoming with a G.729 codec, and target for B-leg needs to be setup
>>>> with a
>>>> > GSM-codec, say
>>>>
>>>> That would require transcoding - which can't be done if the codec is
>>>> pass-through.
>>>>
>>>> > Where in the developer's set of documentation are codecs discussed ? I
>>>> would
>>>> > like to start porting some code of mine for G.729a/b/ab form a ti DSP
>>>> > platform to FS. FS lacking full G.729 support is proving quite a
>>>> hindrance,
>>>> > and there is no clear direction from the dev community as to when the
>>>> same
>>>> > will be available. Incidentally, any news on this effort ? where are
>>>> we with
>>>> > code, and what's an ETA for a Beta ?
>>>>
>>>> I'd say look at the broadvoice or other simple self-contained codecs
>>>> are done.  Currently the only supported g729 solution is to use a
>>>> digium board with mod_dahdi_codec.
>>>>
>>>> I don't have any info on a software based g729 solution.
>>>>
>>>> > On the same lines as (3) above, there is a codec dev template in the
>>>> source
>>>> > tree. Again, where can I find documentation relating to this ? the
>>>> template
>>>> > has hardly any docs at all.
>>>> >
>>>> > Best regards and warm wishes for a Merry Christmas and a great New
>>>> Year to
>>>> > one and all.
>>>> >
>>>> > Ahmed.
>>>> >
>>>> >
>>>> > --
>>>> > Ahmed A. Ibrahim-Naji Al-Alousi
>>>> > Ph.D., MIEE, MBCS
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Ahmed Naji
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From brian at freeswitch.org  Mon Jan 11 07:01:18 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Jan 2010 09:01:18 -0600
Subject: [Freeswitch-users] gtalk and g723 codec
In-Reply-To: <769338.99062.qm@web33205.mail.mud.yahoo.com>
References: <769338.99062.qm@web33205.mail.mud.yahoo.com>
Message-ID: <A8CA30A1-555D-4145-AC23-D468E21CE54A@freeswitch.org>

You can't use G723, Its only a passthru codec.

/b

On Jan 11, 2010, at 3:57 AM, reza mokhtarpour wrote:

> Hi there
> 
> I am begginer with FS , I use FS plus Gtalk   everything is OK with PCMU codec but whenever  i replace it whit G723 codec i got "This codec is only usable in passthrough mode!" error.
> 
> SIP GW  >  FreeSwitch  >  Gtalk
> 
> these are my configuration files :

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From ibrahimtunali at gmail.com  Mon Jan 11 07:43:59 2010
From: ibrahimtunali at gmail.com (itunali)
Date: Mon, 11 Jan 2010 07:43:59 -0800 (PST)
Subject: [Freeswitch-users]  Performance issue on 1.0.4 vs 1.0.5
Message-ID: <27112490.post@talk.nabble.com>


Hi,

I did performance tests to measure that freeswitch limits. The test just
dial echo extension 9996 at default context and wait 6 sec then hangup.

I used sipp test tool and set all variables/environment as described on
(http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations)

I reached 80 cps on 1.0.4 (install from freeswitch-drivers ppa) but not get
similar rate on SVN trunk build (1.5 cps) and get many retransmission SIP
packets.

I build SVN trunk code with default ./bootstrap.sh && ./configure && make &&
make install process.

My server specs;
Ubuntu 9.10 Karmic 
Linux 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 17:01:44 UTC 2009 x86_64
GNU/Linux

Is there any .deb packets for 1.0.5?

Regards,
Ibrahim
-- 
View this message in context: http://old.nabble.com/Performance-issue-on-1.0.4-vs-1.0.5-tp27112490p27112490.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From jcasale at activenetwerx.com  Mon Jan 11 08:15:48 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Mon, 11 Jan 2010 16:15:48 +0000
Subject: [Freeswitch-users] Outbound call problem
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9076F42@prato.activenetwerx.local>

Likely an issue with my SPA3102, but when I route a call
to its FXO port, I can almost faintly hear the operator
if its misdialed, but otherwise the connection is loaded
with feedback and static.

Anyone have a suggestion on where to start looking? Inbound
from that FXO port is flawless.

Thanks,
jlc


From jerry.richards at teotech.com  Mon Jan 11 08:48:48 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 11 Jan 2010 08:48:48 -0800
Subject: [Freeswitch-users] http://jira.freeswitch.org/browse/FSCORE-262
In-Reply-To: <5817BC3F-C091-42E3-AAAE-2CF4A21A9531@freeswitch.org>
References: <39ED1D5111554D33AF502370BC34D288@greyhawk.tonecommander.com>
	<5817BC3F-C091-42E3-AAAE-2CF4A21A9531@freeswitch.org>
Message-ID: <D266691DAA854ABD9B266AC8187802AE@greyhawk.tonecommander.com>

No I don't have a patch, but I suspect it might be a sofia SIP stack issue.

Jerry
 

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Friday, January 08, 2010 11:44 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] http://jira.freeswitch.org/browse/FSCORE-262

Do you happen to have a patch for that?

/b

On Jan 8, 2010, at 1:38 PM, Jerry Richards wrote:

> Is there a plan to fix this JIRA issue:
> http://jira.freeswitch.org/browse/FSCORE-262
> 
> This is causing a problem in sharing presence data between FS and 
> another gateway.
> 
> Thanks,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org






From mike at jerris.com  Mon Jan 11 09:03:56 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 11 Jan 2010 12:03:56 -0500
Subject: [Freeswitch-users] http://jira.freeswitch.org/browse/FSCORE-262
In-Reply-To: <D266691DAA854ABD9B266AC8187802AE@greyhawk.tonecommander.com>
References: <39ED1D5111554D33AF502370BC34D288@greyhawk.tonecommander.com>
	<5817BC3F-C091-42E3-AAAE-2CF4A21A9531@freeswitch.org>
	<D266691DAA854ABD9B266AC8187802AE@greyhawk.tonecommander.com>
Message-ID: <4B646557-1ED8-4767-A87F-846EC071D64E@jerris.com>

as noted on the bug, please try r16218 

Mike

On Jan 11, 2010, at 11:48 AM, Jerry Richards wrote:

> No I don't have a patch, but I suspect it might be a sofia SIP stack issue.
> 
> Jerry
> 
> 
> -----Original Message-----
> From: Brian West [mailto:brian at freeswitch.org] 
> Sent: Friday, January 08, 2010 11:44 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] http://jira.freeswitch.org/browse/FSCORE-262
> 
> Do you happen to have a patch for that?
> 
> /b
> 
> On Jan 8, 2010, at 1:38 PM, Jerry Richards wrote:
> 
>> Is there a plan to fix this JIRA issue:
>> http://jira.freeswitch.org/browse/FSCORE-262
>> 
>> This is causing a problem in sharing presence data between FS and 
>> another gateway.
>> 
>> Thanks,
>> Jerry
>> 


From talk2ram at gmail.com  Mon Jan 11 09:17:10 2010
From: talk2ram at gmail.com (ram)
Date: Mon, 11 Jan 2010 22:47:10 +0530
Subject: [Freeswitch-users] Outbound call problem
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9076F42@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E9076F42@prato.activenetwerx.local>
Message-ID: <b74751491001110917n6709a93cua5dcdce67232ce7a@mail.gmail.com>

post the logs

ram

On Mon, Jan 11, 2010 at 9:45 PM, Joseph L. Casale <jcasale at activenetwerx.com
> wrote:

> Likely an issue with my SPA3102, but when I route a call
> to its FXO port, I can almost faintly hear the operator
> if its misdialed, but otherwise the connection is loaded
> with feedback and static.
>
> Anyone have a suggestion on where to start looking? Inbound
> from that FXO port is flawless.
>
> Thanks,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tayeb.meftah at gmail.com  Mon Jan 11 09:04:44 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Mon, 11 Jan 2010 18:04:44 +0100
Subject: [Freeswitch-users] Performance issue on 1.0.4 vs 1.0.5
In-Reply-To: <27112490.post@talk.nabble.com>
References: <27112490.post@talk.nabble.com>
Message-ID: <E712CC50028F4DA0B5A53289032A839A@PCdeuser>

    hi ibrahim,
please try  to apt-get update
and apt-get upgrade and return fidback
thanks
----- Original Message ----- 
From: "itunali" <ibrahimtunali at gmail.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, January 11, 2010 4:43 PM
Subject: [Freeswitch-users] Performance issue on 1.0.4 vs 1.0.5


>
> Hi,
>
> I did performance tests to measure that freeswitch limits. The test just
> dial echo extension 9996 at default context and wait 6 sec then hangup.
>
> I used sipp test tool and set all variables/environment as described on
> (http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations)
>
> I reached 80 cps on 1.0.4 (install from freeswitch-drivers ppa) but not 
> get
> similar rate on SVN trunk build (1.5 cps) and get many retransmission SIP
> packets.
>
> I build SVN trunk code with default ./bootstrap.sh && ./configure && make 
> &&
> make install process.
>
> My server specs;
> Ubuntu 9.10 Karmic
> Linux 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 17:01:44 UTC 2009 x86_64
> GNU/Linux
>
> Is there any .deb packets for 1.0.5?
>
> Regards,
> Ibrahim
> -- 
> View this message in context: 
> http://old.nabble.com/Performance-issue-on-1.0.4-vs-1.0.5-tp27112490p27112490.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From anatoliy at kounitskiy.com  Mon Jan 11 09:18:49 2010
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Mon, 11 Jan 2010 19:18:49 +0200
Subject: [Freeswitch-users] Problem between revision 16223 and 16242 on
	Debian (stable) 64bit
Message-ID: <1263230329.2504.33.camel@lenovor400-laptop>

Hello,
I just made a checkout of the svn and tried to configure it, but there
is an error in the arp-util lib, after the ./bootstrap.sh


Freeswitch revision: 16242
OS: Debian 64b (stable)
Command used: ./configure --prefix=/usr/local/freeswitch
--enable-optimization --enable-64
Error:
checking for Expat in xml/expat... yes
configuring package in xml/expat now
configure: error: expected an absolute directory name for --bindir:
NONE/bin
configure failed for xml/expat
configure: error: ./configure.gnu failed for libs/apr-util

If I execute "sh configure.gnu"/"sh configure.gnu --enable-64" in
libs/apr-util it goes without an error.

On the same server with Freeswitch revision: 16223 (with the same
configure command), it goes as planned - without errors.

Regards,

-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540



From brian at freeswitch.org  Mon Jan 11 09:20:48 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Jan 2010 11:20:48 -0600
Subject: [Freeswitch-users] Performance issue on 1.0.4 vs 1.0.5
In-Reply-To: <E712CC50028F4DA0B5A53289032A839A@PCdeuser>
References: <27112490.post@talk.nabble.com>
	<E712CC50028F4DA0B5A53289032A839A@PCdeuser>
Message-ID: <B6E8FC8D-EB6F-4D70-B606-439CC8DFB8B9@freeswitch.org>

How about we compile from src?  

/b

On Jan 11, 2010, at 11:04 AM, Meftah Tayeb wrote:

>    hi ibrahim,
> please try  to apt-get update
> and apt-get upgrade and return fidback
> thanks



From andrew at hijacked.us  Mon Jan 11 10:12:03 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Mon, 11 Jan 2010 13:12:03 -0500
Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS
	based callcenter)
In-Reply-To: <855e4dcf1001041842u46c6d36q9c2e4ece2ced7380@mail.gmail.com>
References: <20091219014359.GA21798@hijacked.us>
	<855e4dcf1001041842u46c6d36q9c2e4ece2ced7380@mail.gmail.com>
Message-ID: <20100111181203.GD10774@hijacked.us>

On Tue, Jan 05, 2010 at 03:42:57PM +1300, Tim Uckun wrote:
> > http://opencsm.org/downloads/spice-telephony-0.9.6.tar.gz
> >
> 
> There seems to be something wrong with both opencsm.org and wiki.opencsm.org.
> 
> Just thought I'd let you know.
>

Yeah, my company wanted me to move it, so I re-hosted it as a github
project (with a new name):

http://github.com/Vagabond/OpenACD

I hadn't announced the change yet because I've been away for the last
week and didn't have time. The wiki contents haven't been moved over
yet, but they needed some cleanup anyway.

On the other hand attended transfer support finally materialized.

Andrew


From anatoliy at kounitskiy.com  Mon Jan 11 10:30:49 2010
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Mon, 11 Jan 2010 20:30:49 +0200
Subject: [Freeswitch-users] Problem between revision 16223 and 16242 on
 Debian (stable) 64bit
In-Reply-To: <1263230329.2504.33.camel@lenovor400-laptop>
References: <1263230329.2504.33.camel@lenovor400-laptop>
Message-ID: <1263234649.2504.35.camel@lenovor400-laptop>

Ok, i found in which revision it is broken.

Until revision 16237 - it works as charm
In Revision 16238 - it doesn't work

svn log --revision 16237:16238
------------------------------------------------------------------------
r16238 | mikej | 2010-01-11 16:36:29 +0200 (Mon, 11 Jan 2010) | 1 line

wip move towards adding directory layout control to configure
------------------------------------------------------------------------

Regards,

On Mon, 2010-01-11 at 19:18 +0200, Anatoliy Kounitskiy wrote:
> Hello,
> I just made a checkout of the svn and tried to configure it, but there
> is an error in the arp-util lib, after the ./bootstrap.sh
> 
> 
> Freeswitch revision: 16242
> OS: Debian 64b (stable)
> Command used: ./configure --prefix=/usr/local/freeswitch
> --enable-optimization --enable-64
> Error:
> checking for Expat in xml/expat... yes
> configuring package in xml/expat now
> configure: error: expected an absolute directory name for --bindir:
> NONE/bin
> configure failed for xml/expat
> configure: error: ./configure.gnu failed for libs/apr-util
> 
> If I execute "sh configure.gnu"/"sh configure.gnu --enable-64" in
> libs/apr-util it goes without an error.
> 
> On the same server with Freeswitch revision: 16223 (with the same
> configure command), it goes as planned - without errors.
> 
> Regards,
> 


-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540



From gabe at gundy.org  Mon Jan 11 11:05:29 2010
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 11 Jan 2010 12:05:29 -0700
Subject: [Freeswitch-users] fifo funk?
In-Reply-To: <23f91031001102149x1f1913amefa97d02981261d@mail.gmail.com>
References: <903da5681001102136y6d51f960hd8c10dde4e74ac26@mail.gmail.com>
	<23f91031001102149x1f1913amefa97d02981261d@mail.gmail.com>
Message-ID: <903da5681001111105j25e46e98n8e6cf095651c010f@mail.gmail.com>

On Sun, Jan 10, 2010 at 10:49 PM, Seven Du <dujinfang at gmail.com> wrote:
> see fifo_member add/delete api

I must not have been clear.  We *do* use the add and del api and find
that it works well (better than reparsing).

Gabe


From anthony.minessale at gmail.com  Mon Jan 11 11:13:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Jan 2010 13:13:07 -0600
Subject: [Freeswitch-users] fifo funk?
In-Reply-To: <903da5681001111105j25e46e98n8e6cf095651c010f@mail.gmail.com>
References: <903da5681001102136y6d51f960hd8c10dde4e74ac26@mail.gmail.com>
	<23f91031001102149x1f1913amefa97d02981261d@mail.gmail.com>
	<903da5681001111105j25e46e98n8e6cf095651c010f@mail.gmail.com>
Message-ID: <191c3a031001111113t4f61ff8t21c6c80bce35f53f@mail.gmail.com>

that is the best way.

The re-parse is a much more harsh operation designed for configuration
changes.


On Mon, Jan 11, 2010 at 1:05 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Sun, Jan 10, 2010 at 10:49 PM, Seven Du <dujinfang at gmail.com> wrote:
> > see fifo_member add/delete api
>
> I must not have been clear.  We *do* use the add and del api and find
> that it works well (better than reparsing).
>
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From r.mokhtarpour at yahoo.com  Mon Jan 11 12:04:02 2010
From: r.mokhtarpour at yahoo.com (reza mokhtarpour)
Date: Mon, 11 Jan 2010 12:04:02 -0800 (PST)
Subject: [Freeswitch-users] gtalk and g723 codec
In-Reply-To: <A8CA30A1-555D-4145-AC23-D468E21CE54A@freeswitch.org>
Message-ID: <702240.99335.qm@web33207.mail.mud.yahoo.com>

Sorry ,? I don't understand.

i want use it in this mode .

my SIP gateway originate call with g723 codec and gtalk support this codec? so these endpoints should can communicate with each other in pass through mode.

if it is not true please correct me.

--- On Mon, 1/11/10, Brian West <brian at freeswitch.org> wrote:

From: Brian West <brian at freeswitch.org>
Subject: Re: [Freeswitch-users] gtalk and g723 codec
To: freeswitch-users at lists.freeswitch.org
Date: Monday, January 11, 2010, 7:01 AM

You can't use G723, Its only a passthru codec.
/b
On Jan 11, 2010, at 3:57 AM, reza mokhtarpour wrote:
Hi there

I am begginer with FS , I use FS plus Gtalk?? everything is OK with PCMU codec but whenever? i replace it whit G723 codec i got "This codec is only usable in passthrough mode!" error.

SIP GW? >? FreeSwitch? >? Gtalk

these are my configuration files :

-----Inline Attachment Follows-----

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      
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From brian at freeswitch.org  Mon Jan 11 12:12:21 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Jan 2010 14:12:21 -0600
Subject: [Freeswitch-users] gtalk and g723 codec
In-Reply-To: <702240.99335.qm@web33207.mail.mud.yahoo.com>
References: <702240.99335.qm@web33207.mail.mud.yahoo.com>
Message-ID: <C76820EA-8DCD-4034-9CFC-D3FD4D6D07E8@freeswitch.org>

It might not work correctly in that configuration.

/b

On Jan 11, 2010, at 2:04 PM, reza mokhtarpour wrote:

> Sorry ,  I don't understand.
> 
> i want use it in this mode .
> 
> my SIP gateway originate call with g723 codec and gtalk support this codec  so these endpoints should can communicate with each other in pass through mode.
> 
> if it is not true please correct me.

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From anthony.minessale at gmail.com  Mon Jan 11 12:18:08 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Jan 2010 14:18:08 -0600
Subject: [Freeswitch-users] Performance issue on 1.0.4 vs 1.0.5
In-Reply-To: <27112490.post@talk.nabble.com>
References: <27112490.post@talk.nabble.com>
Message-ID: <191c3a031001111218l645342d8k5ba8b1304e303025@mail.gmail.com>

We have a policy against getting involved in load testing.
Many people are getting well beyond 1.5 cps with trunk so you are probably
doing something wrong.
That's about all I have to offer on the topic.



On Mon, Jan 11, 2010 at 9:43 AM, itunali <ibrahimtunali at gmail.com> wrote:

>
> Hi,
>
> I did performance tests to measure that freeswitch limits. The test just
> dial echo extension 9996 at default context and wait 6 sec then hangup.
>
> I used sipp test tool and set all variables/environment as described on
> (http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations)
>
> I reached 80 cps on 1.0.4 (install from freeswitch-drivers ppa) but not get
> similar rate on SVN trunk build (1.5 cps) and get many retransmission SIP
> packets.
>
> I build SVN trunk code with default ./bootstrap.sh && ./configure && make
> &&
> make install process.
>
> My server specs;
> Ubuntu 9.10 Karmic
> Linux 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 17:01:44 UTC 2009 x86_64
> GNU/Linux
>
> Is there any .deb packets for 1.0.5?
>
> Regards,
> Ibrahim
> --
> View this message in context:
> http://old.nabble.com/Performance-issue-on-1.0.4-vs-1.0.5-tp27112490p27112490.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at van.lammeren.net  Mon Jan 11 12:49:35 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Mon, 11 Jan 2010 15:49:35 -0500
Subject: [Freeswitch-users] lua+sqlite example?
In-Reply-To: <06ca01ca777a$a04125e0$e0c371a0$@com>
References: <04a201ca7623$2c0b2020$84216060$@com>
	<191c3a030912070935u183ff728j8b2c99576da1f5b8@mail.gmail.com>
	<06ca01ca777a$a04125e0$e0c371a0$@com>
Message-ID: <5d2828f1001111249t29331786t6dc9705897b77442@mail.gmail.com>

LuaSQL supports both sqlite and sqlite3 natively.
You can find more info here:
http://www.keplerproject.org/luasql/index.html

And you can download the source from here:
http://luaforge.net/frs/?group_id=12

Mike van Lammeren
<http://luaforge.net/frs/?group_id=12>

On Mon, Dec 7, 2009 at 3:19 PM, Steve Klein <sklein at singular.com> wrote:

>  Thanks. We?ll look at that.
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 07, 2009 9:35 AM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] lua+sqlite example?
>
>
>
> yes if you use the lua odbc sql plugin you should be able to use that for
> sqlite, they may also have a native one.
>
>  On Sat, Dec 5, 2009 at 9:21 PM, Steve Klein <sklein at singular.com> wrote:
>
> Greetings. We are attempting to add sqlite access to an IVR application we
> are prototyping. We are using lua for the scripts. Is there an example
> anywhere of a lua + sqlite script? Do we need to install luasql? Any
> help/pointers greatly appreciated.
>
>
>
> --Steve Klein
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> No virus found in this incoming message.
> Checked by AVG - www.avg.com
> Version: 8.5.426 / Virus Database: 270.14.83/2529 - Release Date: 12/07/09
> 07:33:00
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jcasale at activenetwerx.com  Mon Jan 11 13:34:02 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Mon, 11 Jan 2010 21:34:02 +0000
Subject: [Freeswitch-users] Outbound call problem
In-Reply-To: <b74751491001110917n6709a93cua5dcdce67232ce7a@mail.gmail.com>
References: <CA5A491E9DEFBE4CB777DE97E21575E9076F42@prato.activenetwerx.local>
	<b74751491001110917n6709a93cua5dcdce67232ce7a@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9077570@prato.activenetwerx.local>

>post the logs
?
Ram,
I don't know what to say:) I spent two days on this without results.
Someone onsite finally called me back with a chance to dial out and
it worked? I had to review the log three times because I didn't believe
it...

Can't say I mind, but I wish I knew why, heh.

Thanks!
jlc



From a.alalousi at gmail.com  Mon Jan 11 14:10:39 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Mon, 11 Jan 2010 22:10:39 +0000
Subject: [Freeswitch-users] Revision 16238,
	16237 compiling and call handling issue
Message-ID: <cd93b3701001111410labccf1eme3e9c680cb13c249@mail.gmail.com>

People,

It seems there are  a few things that are broken in this release. So far,
I've come across two issues: first of, configure fails, per what others
reported, complaining about NONE/bin not being an absolute directory path;
1637 has no problems with that.

There is also what appears to be a serious call handling issue that was not
present in 1.0.4 trunks which is the following:

Calls are initiated correctly, Leg-B is set-up and remote end rings,
ring-back is reported to Leg-A, remote end picks up, but Leg-A still gets
the ringing tone. This is pretty much the case with both internal and
external profiles, irrespective of whether or not a gateway is used to route
the call, irrespective of which media mode FS is running in and irrespective
of codec.

I know dev and support groups don't like to get involved in performance and
load testing loops, but for he who cares, performance on 1.0.5 trunks is
miserable compared to 1.0.4. I've bombarded 1.0.4 with over 150 cps with
hardly a glitch on quad-xeon machines with 6Gb RAM and GigE controllers.
1.0.5 falls far, far short of that and the figure is a fraction at just
under 20 for the same spec. machine. I run CentOS 5.0.4, and have had the
exact same experience with Debian stable and Ubuntu Karmic on the same
hardware.

I'm tracing the call handling issue now, and will report back if I find
anything useful. Meantime, it would be great if someone from support can
open a Jira for this or let me know how to do that - sorry, my own ignorance
of how the dev/support system works .. only converted to FS 6 weeks ago :).
Happy to post traces and findings.

Regards,

Ahmed.
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From anthony.minessale at gmail.com  Mon Jan 11 14:32:22 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Jan 2010 16:32:22 -0600
Subject: [Freeswitch-users] Revision 16238,
	16237 compiling and call 	handling issue
In-Reply-To: <cd93b3701001111410labccf1eme3e9c680cb13c249@mail.gmail.com>
References: <cd93b3701001111410labccf1eme3e9c680cb13c249@mail.gmail.com>
Message-ID: <191c3a031001111432j5734c6begc52d5398b4844c0e@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Reporting_Bugs


On Mon, Jan 11, 2010 at 4:10 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:

> People,
>
> It seems there are  a few things that are broken in this release. So far,
> I've come across two issues: first of, configure fails, per what others
> reported, complaining about NONE/bin not being an absolute directory path;
> 1637 has no problems with that.
>
> There is also what appears to be a serious call handling issue that was not
> present in 1.0.4 trunks which is the following:
>
> Calls are initiated correctly, Leg-B is set-up and remote end rings,
> ring-back is reported to Leg-A, remote end picks up, but Leg-A still gets
> the ringing tone. This is pretty much the case with both internal and
> external profiles, irrespective of whether or not a gateway is used to route
> the call, irrespective of which media mode FS is running in and irrespective
> of codec.
>
> I know dev and support groups don't like to get involved in performance and
> load testing loops, but for he who cares, performance on 1.0.5 trunks is
> miserable compared to 1.0.4. I've bombarded 1.0.4 with over 150 cps with
> hardly a glitch on quad-xeon machines with 6Gb RAM and GigE controllers.
> 1.0.5 falls far, far short of that and the figure is a fraction at just
> under 20 for the same spec. machine. I run CentOS 5.0.4, and have had the
> exact same experience with Debian stable and Ubuntu Karmic on the same
> hardware.
>
> I'm tracing the call handling issue now, and will report back if I find
> anything useful. Meantime, it would be great if someone from support can
> open a Jira for this or let me know how to do that - sorry, my own ignorance
> of how the dev/support system works .. only converted to FS 6 weeks ago :).
> Happy to post traces and findings.
>
> Regards,
>
> Ahmed.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From djbinter at yahoo.com  Mon Jan 11 14:41:14 2010
From: djbinter at yahoo.com (DJB)
Date: Mon, 11 Jan 2010 14:41:14 -0800 (PST)
Subject: [Freeswitch-users] Bypass Media mode seems to be broken
Message-ID: <78283.656.qm@web37506.mail.mud.yahoo.com>

I wonder whether anyone experienced this problem.

SVN Version: 16249

Trace log:  http://pastebin.freeswitch.org/11754

Dialplan:
   <extension name="tollfree">
     <condition field="destination_number" expression="^(\+1|1)?(8(00|88|77|66)[2-9]\d{6})$">
       <action application="set" data="bypass_media=true"/>
       <action application="bridge" data="sofia/external/1$2 at sip.tollfreegateway.com"/>
     </condition>
   </extension>

Problem:
FS is not passing 200 OK to inbound leg when it received from outbound leg


      
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From mike at van.lammeren.net  Mon Jan 11 14:53:24 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Mon, 11 Jan 2010 17:53:24 -0500
Subject: [Freeswitch-users] How to provide dynamic directory information?
Message-ID: <5d2828f1001111453t8044774w3cc882988e18ced@mail.gmail.com>

Hello!

I'd like to be able to have FreeSWITCH check a database for authorization,
every time a user registers. There are some great examples on the wiki,
which use either MOD_XML_CURL or Lua to dynamically provide a dialplan, but
I cannot find an example for providing a directory.

I am developing an application that will have thousands of users, and will
run on multiple FreeSWITCH servers behind a load balancer. Ideally,
FreeSWITCH would only look-up directory information, specifically, username
and password, whenever a user attempts to connect. The directory information
will be changing regularly, as users are added or removed from the system.

Is this possible with FreeSWITCH? Or can only dialplan information be
provided dynamically?

I've written a script in Lua that provides the XML data, such as that found
in the example /freeswitch/conf/directory/default/ folders, and I try to
call it with this bit of XML in /freeswitch/conf/directory/default.xml:

<groups>
<group name="default">
<users>
<param name="xml-handler-script" value="/directory.lua"/>
<param name="xml-handler-bindings" value="users"/>
</users>
</groups>

Is this the right approach? Am I going about this the right way?

I would appreciate any tips that anyone can provide!

Thanks!

Mike van Lammeren
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From mcampbellsmith at gmail.com  Mon Jan 11 14:58:05 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 12 Jan 2010 09:58:05 +1100
Subject: [Freeswitch-users] Bypass Media mode seems to be broken
In-Reply-To: <78283.656.qm@web37506.mail.mud.yahoo.com>
References: <78283.656.qm@web37506.mail.mud.yahoo.com>
Message-ID: <33c87fa31001111458s595c0164l7bd64c9031b38ed7@mail.gmail.com>

I sent this yesterday (my FS version is FreeSWITCH Version 1.0.trunk (16131) )

Not sure if this is related ...

Hi!

I am calling from extension 2010 to extension 1000.  Both have ip
addesses 192.168.1.x.

In the 2000 series dialplan (a separate context) I have the following
to try to enable bypass_media.

           <extension name="Local_1000_9xxx">
                 <condition field="destination_number"
expression="^(10[01][0-9]|9\d{3})$">
                       <action application="set" data="dialed_extension=$1"/>
                       <action application="set" data="proxy_media=false"/>
                       <action application="set" data="bypass_media=true"/>
                       <action application="transfer"
data="${dialed_extension} XML default"/>
                 </condition>
           </extension>

Is this how bypass media should be enabled?

This fails fo me (the calls hang up and no audio).  The debug trace is
in http://pastebin.freeswitch.org/11737

What have I done wrong?

Thanks


On Tue, Jan 12, 2010 at 9:41 AM, DJB <djbinter at yahoo.com> wrote:
> I wonder whether anyone experienced this problem.
> SVN Version: 16249
> Trace log: ?http://pastebin.freeswitch.org/11754
> Dialplan:
> ?? <extension name="tollfree">
> ?? ? <condition field="destination_number"
> expression="^(\+1|1)?(8(00|88|77|66)[2-9]\d{6})$">
> ?? ? ? <action application="set" data="bypass_media=true"/>
> ?? ? ? <action application="bridge"
> data="sofia/external/1$2 at sip.tollfreegateway.com"/>
> ?? ? </condition>
> ?? </extension>
> Problem:
> FS is not passing 200 OK to inbound leg when it received from outbound leg
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From egable+freeswitch at gmail.com  Mon Jan 11 15:02:37 2010
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Mon, 11 Jan 2010 18:02:37 -0500
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
Message-ID: <bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>

Assuming you're using SVN trunk, there is a bug that is likely causing
your issue. Anthony is fixing it right now.

On Mon, Jan 11, 2010 at 6:16 AM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Hi!
> Hi!
>
> I am calling from extension 2010 to extension 1000. ?Both have ip
> addesses 192.168.1.x.
>
> In the 2000 series dialplan (a separate context) I have the following
> to try to enable bypass_media.
>
> ? ? ? ? ? ?<extension name="Local_1000_9xxx">
> ? ? ? ? ? ? ? ? ?<condition field="destination_number"
> expression="^(10[01][0-9]|9\d{3})$">
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="dialed_extension=$1"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="proxy_media=false"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="bypass_media=true"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="transfer"
> data="${dialed_extension} XML default"/>
> ? ? ? ? ? ? ? ? ?</condition>
> ? ? ? ? ? ?</extension>
>
> Is this how bypass media should be enabled?
>
> This fails fo me (the calls hang up and no audio). ?The debug trace is
> in http://pastebin.freeswitch.org/11737
>
> What have I done wrong?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero


From egable+freeswitch at gmail.com  Mon Jan 11 15:06:30 2010
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Mon, 11 Jan 2010 18:06:30 -0500
Subject: [Freeswitch-users] Bypass Media mode seems to be broken
In-Reply-To: <33c87fa31001111458s595c0164l7bd64c9031b38ed7@mail.gmail.com>
References: <78283.656.qm@web37506.mail.mud.yahoo.com>
	<33c87fa31001111458s595c0164l7bd64c9031b38ed7@mail.gmail.com>
Message-ID: <bf6923ed1001111506q471ac349md187f4df91ae9280@mail.gmail.com>

Anthony reproduced it and is fixing it right now.


On Mon, Jan 11, 2010 at 5:58 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> I sent this yesterday (my FS version is FreeSWITCH Version 1.0.trunk (16131) )
>
> Not sure if this is related ...
>
> Hi!
>
> I am calling from extension 2010 to extension 1000. ?Both have ip
> addesses 192.168.1.x.
>
> In the 2000 series dialplan (a separate context) I have the following
> to try to enable bypass_media.
>
> ? ? ? ? ? <extension name="Local_1000_9xxx">
> ? ? ? ? ? ? ? ? <condition field="destination_number"
> expression="^(10[01][0-9]|9\d{3})$">
> ? ? ? ? ? ? ? ? ? ? ? <action application="set" data="dialed_extension=$1"/>
> ? ? ? ? ? ? ? ? ? ? ? <action application="set" data="proxy_media=false"/>
> ? ? ? ? ? ? ? ? ? ? ? <action application="set" data="bypass_media=true"/>
> ? ? ? ? ? ? ? ? ? ? ? <action application="transfer"
> data="${dialed_extension} XML default"/>
> ? ? ? ? ? ? ? ? </condition>
> ? ? ? ? ? </extension>
>
> Is this how bypass media should be enabled?
>
> This fails fo me (the calls hang up and no audio). ?The debug trace is
> in http://pastebin.freeswitch.org/11737
>
> What have I done wrong?
>
> Thanks
>
>
> On Tue, Jan 12, 2010 at 9:41 AM, DJB <djbinter at yahoo.com> wrote:
>> I wonder whether anyone experienced this problem.
>> SVN Version: 16249
>> Trace log: ?http://pastebin.freeswitch.org/11754
>> Dialplan:
>> ?? <extension name="tollfree">
>> ?? ? <condition field="destination_number"
>> expression="^(\+1|1)?(8(00|88|77|66)[2-9]\d{6})$">
>> ?? ? ? <action application="set" data="bypass_media=true"/>
>> ?? ? ? <action application="bridge"
>> data="sofia/external/1$2 at sip.tollfreegateway.com"/>
>> ?? ? </condition>
>> ?? </extension>
>> Problem:
>> FS is not passing 200 OK to inbound leg when it received from outbound leg
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero


From mcampbellsmith at gmail.com  Mon Jan 11 15:09:40 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 12 Jan 2010 10:09:40 +1100
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
Message-ID: <33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>

Thanks Eliot.  I'm using FreeSWITCH Version 1.0.trunk (16131).   Do
you kno wif the fault was present in 16131?

Cheers

On Tue, Jan 12, 2010 at 10:02 AM, Eliot Gable
<egable+freeswitch at gmail.com> wrote:
> Assuming you're using SVN trunk, there is a bug that is likely causing
> your issue. Anthony is fixing it right now.
>
> On Mon, Jan 11, 2010 at 6:16 AM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>> Hi!
>> Hi!
>>
>> I am calling from extension 2010 to extension 1000. ?Both have ip
>> addesses 192.168.1.x.
>>
>> In the 2000 series dialplan (a separate context) I have the following
>> to try to enable bypass_media.
>>
>> ? ? ? ? ? ?<extension name="Local_1000_9xxx">
>> ? ? ? ? ? ? ? ? ?<condition field="destination_number"
>> expression="^(10[01][0-9]|9\d{3})$">
>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="dialed_extension=$1"/>
>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="proxy_media=false"/>
>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="bypass_media=true"/>
>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="transfer"
>> data="${dialed_extension} XML default"/>
>> ? ? ? ? ? ? ? ? ?</condition>
>> ? ? ? ? ? ?</extension>
>>
>> Is this how bypass media should be enabled?
>>
>> This fails fo me (the calls hang up and no audio). ?The debug trace is
>> in http://pastebin.freeswitch.org/11737
>>
>> What have I done wrong?
>>
>> Thanks
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Mon Jan 11 15:10:09 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Jan 2010 17:10:09 -0600
Subject: [Freeswitch-users] Bypass Media mode seems to be broken
In-Reply-To: <bf6923ed1001111506q471ac349md187f4df91ae9280@mail.gmail.com>
References: <78283.656.qm@web37506.mail.mud.yahoo.com>
	<33c87fa31001111458s595c0164l7bd64c9031b38ed7@mail.gmail.com>
	<bf6923ed1001111506q471ac349md187f4df91ae9280@mail.gmail.com>
Message-ID: <EBB7B2A4-295A-4E3D-B26B-855601858D27@freeswitch.org>

YEP YEP!

/b

On Jan 11, 2010, at 5:06 PM, Eliot Gable wrote:

> Anthony reproduced it and is fixing it right now.
> 



From mike at jerris.com  Mon Jan 11 15:18:12 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 11 Jan 2010 18:18:12 -0500
Subject: [Freeswitch-users] Revision 16238,
	16237 compiling and call handling issue
In-Reply-To: <cd93b3701001111410labccf1eme3e9c680cb13c249@mail.gmail.com>
References: <cd93b3701001111410labccf1eme3e9c680cb13c249@mail.gmail.com>
Message-ID: <FF3124B2-775D-4308-B405-DC9D14DC3396@jerris.com>


On Jan 11, 2010, at 5:10 PM, Ahmed Naji wrote:

> People,
> 
> It seems there are  a few things that are broken in this release. So far, I've come across two issues: first of, configure fails, per what others reported, complaining about NONE/bin not being an absolute directory path; 1637 has no problems with that.

fixed in tree already.



From egable+freeswitch at gmail.com  Mon Jan 11 15:22:07 2010
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Mon, 11 Jan 2010 18:22:07 -0500
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
Message-ID: <bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>

I can't say for certain. I know it's not in 16016. Wait for Anthony to
fix the issue in the current version, then update and try again.

On Mon, Jan 11, 2010 at 6:09 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Thanks Eliot. ?I'm using FreeSWITCH Version 1.0.trunk (16131). ? Do
> you kno wif the fault was present in 16131?
>
> Cheers
>
> On Tue, Jan 12, 2010 at 10:02 AM, Eliot Gable
> <egable+freeswitch at gmail.com> wrote:
>> Assuming you're using SVN trunk, there is a bug that is likely causing
>> your issue. Anthony is fixing it right now.
>>
>> On Mon, Jan 11, 2010 at 6:16 AM, Mark Campbell-Smith
>> <mcampbellsmith at gmail.com> wrote:
>>> Hi!
>>> Hi!
>>>
>>> I am calling from extension 2010 to extension 1000. ?Both have ip
>>> addesses 192.168.1.x.
>>>
>>> In the 2000 series dialplan (a separate context) I have the following
>>> to try to enable bypass_media.
>>>
>>> ? ? ? ? ? ?<extension name="Local_1000_9xxx">
>>> ? ? ? ? ? ? ? ? ?<condition field="destination_number"
>>> expression="^(10[01][0-9]|9\d{3})$">
>>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="dialed_extension=$1"/>
>>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="proxy_media=false"/>
>>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="bypass_media=true"/>
>>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="transfer"
>>> data="${dialed_extension} XML default"/>
>>> ? ? ? ? ? ? ? ? ?</condition>
>>> ? ? ? ? ? ?</extension>
>>>
>>> Is this how bypass media should be enabled?
>>>
>>> This fails fo me (the calls hang up and no audio). ?The debug trace is
>>> in http://pastebin.freeswitch.org/11737
>>>
>>> What have I done wrong?
>>>
>>> Thanks
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Eliot Gable
>>
>> "We do not inherit the Earth from our ancestors: we borrow it from our
>> children." ~David Brower
>>
>> "I decided the words were too conservative for me. We're not borrowing
>> from our children, we're stealing from them--and it's not even
>> considered to be a crime." ~David Brower
>>
>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>> live; not live to eat.) ~Marcus Tullius Cicero
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero


From a.alalousi at gmail.com  Mon Jan 11 15:32:33 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Mon, 11 Jan 2010 23:32:33 +0000
Subject: [Freeswitch-users] Revision 16238,
	16237 compiling and call 	handling issue
In-Reply-To: <191c3a031001111432j5734c6begc52d5398b4844c0e@mail.gmail.com>
References: <cd93b3701001111410labccf1eme3e9c680cb13c249@mail.gmail.com>
	<191c3a031001111432j5734c6begc52d5398b4844c0e@mail.gmail.com>
Message-ID: <cd93b3701001111532y6a9f6f74j18c85fc9b02553ba@mail.gmail.com>

Thanks for this.

2010/1/11 Anthony Minessale <anthony.minessale at gmail.com>

>
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>
>
> On Mon, Jan 11, 2010 at 4:10 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>
>> People,
>>
>> It seems there are  a few things that are broken in this release. So far,
>> I've come across two issues: first of, configure fails, per what others
>> reported, complaining about NONE/bin not being an absolute directory path;
>> 1637 has no problems with that.
>>
>> There is also what appears to be a serious call handling issue that was
>> not present in 1.0.4 trunks which is the following:
>>
>> Calls are initiated correctly, Leg-B is set-up and remote end rings,
>> ring-back is reported to Leg-A, remote end picks up, but Leg-A still gets
>> the ringing tone. This is pretty much the case with both internal and
>> external profiles, irrespective of whether or not a gateway is used to route
>> the call, irrespective of which media mode FS is running in and irrespective
>> of codec.
>>
>> I know dev and support groups don't like to get involved in performance
>> and load testing loops, but for he who cares, performance on 1.0.5 trunks is
>> miserable compared to 1.0.4. I've bombarded 1.0.4 with over 150 cps with
>> hardly a glitch on quad-xeon machines with 6Gb RAM and GigE controllers.
>> 1.0.5 falls far, far short of that and the figure is a fraction at just
>> under 20 for the same spec. machine. I run CentOS 5.0.4, and have had the
>> exact same experience with Debian stable and Ubuntu Karmic on the same
>> hardware.
>>
>> I'm tracing the call handling issue now, and will report back if I find
>> anything useful. Meantime, it would be great if someone from support can
>> open a Jira for this or let me know how to do that - sorry, my own ignorance
>> of how the dev/support system works .. only converted to FS 6 weeks ago :).
>> Happy to post traces and findings.
>>
>> Regards,
>>
>> Ahmed.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From nicolas at medularis.com  Mon Jan 11 15:42:58 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 11 Jan 2010 20:42:58 -0300
Subject: [Freeswitch-users] Calls getting queued?
In-Reply-To: <191c3a031001071022j1e5796fdvf63900f5968bc01b@mail.gmail.com>
References: <1b46b4e81001070743v5389ef3ewb75dbeb06b402c9a@mail.gmail.com>
	<87f2f3b91001070926j60139f2cxe0ae9687e25a302e@mail.gmail.com>
	<1b46b4e81001071012y237fb8c0jd68232137b7d8e96@mail.gmail.com>
	<191c3a031001071022j1e5796fdvf63900f5968bc01b@mail.gmail.com>
Message-ID: <1b46b4e81001111542t51629dd0ic22f5cc908283778@mail.gmail.com>

Thanks. I actually got rid of all the JS callbacks and left only the main JS
script which originates 2 calls and then bridges them together. I moved all
event detection to an Event Sockets daemon. I thought I was off the hook,
but today the issue started happening again, and there was no curl involved.

Without looking at sip traces, what do you think could create a situation
like this?

I have no idea how to reproduce this issue, except wait for a few hours or
maybe even a few days, so I'm not sure recording all sip traffic would be
such a good idea. How would I go about getting traces for this?

Thanks.


On Thu, Jan 7, 2010 at 3:22 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try setting the timeout in curl
>
> conf/autoload_configs/xml_curl.conf.xml:
> <param name="timeout" value="5"/>
>
>
> On Thu, Jan 7, 2010 at 12:12 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> Michael,
>>
>> Thanks for your help. Yes, if I restart FS things go back to normal
>> for a while and then the same thing starts happening again.
>>
>> The weird thing is, it started only 2 days ago, and happened only once
>> or twice. Before that I had no trouble, and I only made 1 change,
>> which I reverted, but it wasn't that. Today it's happening all the
>> time, if I restart FS things will work for maybe an hour and then it
>> will start doing the same thing.
>>
>> I'm guessing it might be something external to FS, like curl calls not
>> finishing properly because of the url they are requesting or something
>> like that.
>>
>> What kind of info should I collect? I don't think it has to do with
>> sofia or any sip-related problems. I'm also using the default
>> dialplan, no changes at all, I'm doing everything through JS, well and
>> one really small lua script.
>>
>> This is the main JS file:
>> It originates 2 calls and bridges them.
>>
>> - http://pastebin.freeswitch.org/11706
>>
>>
>> This is another JS script which gets called when each call is hanged up:
>> It gets some info and then requests a url using curl to update call
>> status on an external db.
>>
>> - http://pastebin.freeswitch.org/11707
>>
>>
>> This lua script calls a ruby script to do some other stuff when a call
>> is answered:
>>
>> - http://pastebin.freeswitch.org/11708
>>
>>
>> Thanks!
>>
>>
>> Nico
>>
>>
>>
>> On Thu, Jan 7, 2010 at 2:26 PM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> >
>> >
>> > On Thu, Jan 7, 2010 at 7:43 AM, Nicolas Brenner <nicolas at medularis.com>
>> > wrote:
>> >>
>> >> Hi, I'm having a strange problem with FS. I'm using a few JS scripts
>> >> to generate calls and bridge them together. Usually everything works
>> >> just fine, but them at some point it's like if FS choked, calls for
>> >> the first leg of the bridges are apparently made, but the second leg
>> >> is never called. The call is not hanged up for several minutes and the
>> >> system keeps opening new channels but never connecting a call.
>> >>
>> >> For example, right now, doing 'show channels' on the console, I get a
>> >> list of 72 open channels (it's adding up, it was 40 a couple minutes
>> >> ago), but doing a 'show calls' gives me 0 active calls. The usual
>> >> behavior, when everything's working fine, is to get twice as many
>> >> channels as there are active calls and no channels at all when there
>> >> are no calls, unless they haven't been bridged yet.
>> >>
>> >> The originate string is something like this:
>> >>
>> >> var stUsRing = "%(2000,4000,440,480)";
>> >> var timeout = 45;
>> >> originate_str1 = "{api_hangup_hook=jsapi::callback.js
>> >> l1,execute_on_answer=lua answered.lua 1
>> >>
>> >>
>> c2c_call,ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,ringback='"+stUsRing+"',medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]"+dialstr1;
>> >>
>> >> Where diasltr1 has the phonenumber and and gateway info. The
>> >> callback.js has a curl request to update some call info on an external
>> >> database and answered.lua calls a ruby script through the os.execute()
>> >> function (I know, I should be doing all this through the event socket,
>> >> I was doing that but had trouble and had to come up with a quick
>> >> solution).
>> >>
>> >> The system is not loaded at all, at least not for what I think and
>> >> read that FS can handle. We are having at most 10 concurrent calls (20
>> >> channels), with maybe 5 to 10 calls per minute.
>> >>
>> >> What worries me is not only that I don't know where the problem is,
>> >> but that I have no clue how to debug it or send you guys more
>> >> "lowlevel" and detailed information to give you an insight about
>> >> what's going on. Any help would be greatly appreciated!
>> >>
>> >> Thanks!
>> >>
>> >> Nico
>> >>
>> > First off you'll want to get familiar with the resources mentioned here:
>> > http://wiki.freeswitch.org/wiki/Reporting_Bugs
>> >
>> > It has good tips on how to collect and report information.
>> >
>> > Second, I recommend that you pastebin your relevant portion of the
>> dialplan
>> > and the whole javascript program that you are using so that others can
>> take
>> > a look.
>> >
>> > Last thing: if you restart FreeSWITCH does everything work fine for a
>> while
>> > but then eventually it breaks down and exhibits the behavior that you
>> are
>> > reporting?
>> >
>> > -MC
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Jan 11 16:07:59 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Jan 2010 18:07:59 -0600
Subject: [Freeswitch-users] Revision 16238,
	16237 compiling and call 	handling issue
In-Reply-To: <cd93b3701001111532y6a9f6f74j18c85fc9b02553ba@mail.gmail.com>
References: <cd93b3701001111410labccf1eme3e9c680cb13c249@mail.gmail.com>
	<191c3a031001111432j5734c6begc52d5398b4844c0e@mail.gmail.com>
	<cd93b3701001111532y6a9f6f74j18c85fc9b02553ba@mail.gmail.com>
Message-ID: <191c3a031001111607j63a72fdft6f47084820b81bf4@mail.gmail.com>

too late,
its fixed in rev 16250


On Mon, Jan 11, 2010 at 5:32 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:

> Thanks for this.
>
> 2010/1/11 Anthony Minessale <anthony.minessale at gmail.com>
>
>
>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>
>>
>> On Mon, Jan 11, 2010 at 4:10 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>>
>>> People,
>>>
>>> It seems there are  a few things that are broken in this release. So far,
>>> I've come across two issues: first of, configure fails, per what others
>>> reported, complaining about NONE/bin not being an absolute directory path;
>>> 1637 has no problems with that.
>>>
>>> There is also what appears to be a serious call handling issue that was
>>> not present in 1.0.4 trunks which is the following:
>>>
>>> Calls are initiated correctly, Leg-B is set-up and remote end rings,
>>> ring-back is reported to Leg-A, remote end picks up, but Leg-A still gets
>>> the ringing tone. This is pretty much the case with both internal and
>>> external profiles, irrespective of whether or not a gateway is used to route
>>> the call, irrespective of which media mode FS is running in and irrespective
>>> of codec.
>>>
>>> I know dev and support groups don't like to get involved in performance
>>> and load testing loops, but for he who cares, performance on 1.0.5 trunks is
>>> miserable compared to 1.0.4. I've bombarded 1.0.4 with over 150 cps with
>>> hardly a glitch on quad-xeon machines with 6Gb RAM and GigE controllers.
>>> 1.0.5 falls far, far short of that and the figure is a fraction at just
>>> under 20 for the same spec. machine. I run CentOS 5.0.4, and have had the
>>> exact same experience with Debian stable and Ubuntu Karmic on the same
>>> hardware.
>>>
>>> I'm tracing the call handling issue now, and will report back if I find
>>> anything useful. Meantime, it would be great if someone from support can
>>> open a Jira for this or let me know how to do that - sorry, my own ignorance
>>> of how the dev/support system works .. only converted to FS 6 weeks ago :).
>>> Happy to post traces and findings.
>>>
>>> Regards,
>>>
>>> Ahmed.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Mon Jan 11 16:08:40 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Jan 2010 18:08:40 -0600
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
Message-ID: <191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>

fixed in 16250

On Mon, Jan 11, 2010 at 5:22 PM, Eliot Gable
<egable+freeswitch at gmail.com<egable%2Bfreeswitch at gmail.com>
> wrote:

> I can't say for certain. I know it's not in 16016. Wait for Anthony to
> fix the issue in the current version, then update and try again.
>
> On Mon, Jan 11, 2010 at 6:09 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
> > Thanks Eliot.  I'm using FreeSWITCH Version 1.0.trunk (16131).   Do
> > you kno wif the fault was present in 16131?
> >
> > Cheers
> >
> > On Tue, Jan 12, 2010 at 10:02 AM, Eliot Gable
> > <egable+freeswitch at gmail.com <egable%2Bfreeswitch at gmail.com>> wrote:
> >> Assuming you're using SVN trunk, there is a bug that is likely causing
> >> your issue. Anthony is fixing it right now.
> >>
> >> On Mon, Jan 11, 2010 at 6:16 AM, Mark Campbell-Smith
> >> <mcampbellsmith at gmail.com> wrote:
> >>> Hi!
> >>> Hi!
> >>>
> >>> I am calling from extension 2010 to extension 1000.  Both have ip
> >>> addesses 192.168.1.x.
> >>>
> >>> In the 2000 series dialplan (a separate context) I have the following
> >>> to try to enable bypass_media.
> >>>
> >>>            <extension name="Local_1000_9xxx">
> >>>                  <condition field="destination_number"
> >>> expression="^(10[01][0-9]|9\d{3})$">
> >>>                        <action application="set"
> data="dialed_extension=$1"/>
> >>>                        <action application="set"
> data="proxy_media=false"/>
> >>>                        <action application="set"
> data="bypass_media=true"/>
> >>>                        <action application="transfer"
> >>> data="${dialed_extension} XML default"/>
> >>>                  </condition>
> >>>            </extension>
> >>>
> >>> Is this how bypass media should be enabled?
> >>>
> >>> This fails fo me (the calls hang up and no audio).  The debug trace is
> >>> in http://pastebin.freeswitch.org/11737
> >>>
> >>> What have I done wrong?
> >>>
> >>> Thanks
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> >> --
> >> Eliot Gable
> >>
> >> "We do not inherit the Earth from our ancestors: we borrow it from our
> >> children." ~David Brower
> >>
> >> "I decided the words were too conservative for me. We're not borrowing
> >> from our children, we're stealing from them--and it's not even
> >> considered to be a crime." ~David Brower
> >>
> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> >> live; not live to eat.) ~Marcus Tullius Cicero
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Mon Jan 11 16:08:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Jan 2010 18:08:24 -0600
Subject: [Freeswitch-users] Bypass Media mode seems to be broken
In-Reply-To: <EBB7B2A4-295A-4E3D-B26B-855601858D27@freeswitch.org>
References: <78283.656.qm@web37506.mail.mud.yahoo.com>
	<33c87fa31001111458s595c0164l7bd64c9031b38ed7@mail.gmail.com>
	<bf6923ed1001111506q471ac349md187f4df91ae9280@mail.gmail.com>
	<EBB7B2A4-295A-4E3D-B26B-855601858D27@freeswitch.org>
Message-ID: <191c3a031001111608o37eef419i7fcc556979b3e2ff@mail.gmail.com>

fixed in 16250


On Mon, Jan 11, 2010 at 5:10 PM, Brian West <brian at freeswitch.org> wrote:

> YEP YEP!
>
> /b
>
> On Jan 11, 2010, at 5:06 PM, Eliot Gable wrote:
>
> > Anthony reproduced it and is fixing it right now.
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Mon Jan 11 16:10:01 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Jan 2010 18:10:01 -0600
Subject: [Freeswitch-users] Calls getting queued?
In-Reply-To: <1b46b4e81001111542t51629dd0ic22f5cc908283778@mail.gmail.com>
References: <1b46b4e81001070743v5389ef3ewb75dbeb06b402c9a@mail.gmail.com>
	<87f2f3b91001070926j60139f2cxe0ae9687e25a302e@mail.gmail.com>
	<1b46b4e81001071012y237fb8c0jd68232137b7d8e96@mail.gmail.com>
	<191c3a031001071022j1e5796fdvf63900f5968bc01b@mail.gmail.com>
	<1b46b4e81001111542t51629dd0ic22f5cc908283778@mail.gmail.com>
Message-ID: <191c3a031001111610o66bbc21fxbceabd037dddcf76@mail.gmail.com>

js is notorious for garbage collection issues.
you would be wise to just build a dial string and use the bridge application
to bridge them rather than bridge them manually in JS

On Mon, Jan 11, 2010 at 5:42 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Thanks. I actually got rid of all the JS callbacks and left only the main
> JS script which originates 2 calls and then bridges them together. I moved
> all event detection to an Event Sockets daemon. I thought I was off the
> hook, but today the issue started happening again, and there was no curl
> involved.
>
> Without looking at sip traces, what do you think could create a situation
> like this?
>
> I have no idea how to reproduce this issue, except wait for a few hours or
> maybe even a few days, so I'm not sure recording all sip traffic would be
> such a good idea. How would I go about getting traces for this?
>
> Thanks.
>
>
> On Thu, Jan 7, 2010 at 3:22 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> try setting the timeout in curl
>>
>> conf/autoload_configs/xml_curl.conf.xml:
>> <param name="timeout" value="5"/>
>>
>>
>> On Thu, Jan 7, 2010 at 12:12 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>
>>> Michael,
>>>
>>> Thanks for your help. Yes, if I restart FS things go back to normal
>>> for a while and then the same thing starts happening again.
>>>
>>> The weird thing is, it started only 2 days ago, and happened only once
>>> or twice. Before that I had no trouble, and I only made 1 change,
>>> which I reverted, but it wasn't that. Today it's happening all the
>>> time, if I restart FS things will work for maybe an hour and then it
>>> will start doing the same thing.
>>>
>>> I'm guessing it might be something external to FS, like curl calls not
>>> finishing properly because of the url they are requesting or something
>>> like that.
>>>
>>> What kind of info should I collect? I don't think it has to do with
>>> sofia or any sip-related problems. I'm also using the default
>>> dialplan, no changes at all, I'm doing everything through JS, well and
>>> one really small lua script.
>>>
>>> This is the main JS file:
>>> It originates 2 calls and bridges them.
>>>
>>> - http://pastebin.freeswitch.org/11706
>>>
>>>
>>> This is another JS script which gets called when each call is hanged up:
>>> It gets some info and then requests a url using curl to update call
>>> status on an external db.
>>>
>>> - http://pastebin.freeswitch.org/11707
>>>
>>>
>>> This lua script calls a ruby script to do some other stuff when a call
>>> is answered:
>>>
>>> - http://pastebin.freeswitch.org/11708
>>>
>>>
>>> Thanks!
>>>
>>>
>>> Nico
>>>
>>>
>>>
>>> On Thu, Jan 7, 2010 at 2:26 PM, Michael Collins <msc at freeswitch.org>
>>> wrote:
>>> >
>>> >
>>> > On Thu, Jan 7, 2010 at 7:43 AM, Nicolas Brenner <nicolas at medularis.com
>>> >
>>> > wrote:
>>> >>
>>> >> Hi, I'm having a strange problem with FS. I'm using a few JS scripts
>>> >> to generate calls and bridge them together. Usually everything works
>>> >> just fine, but them at some point it's like if FS choked, calls for
>>> >> the first leg of the bridges are apparently made, but the second leg
>>> >> is never called. The call is not hanged up for several minutes and the
>>> >> system keeps opening new channels but never connecting a call.
>>> >>
>>> >> For example, right now, doing 'show channels' on the console, I get a
>>> >> list of 72 open channels (it's adding up, it was 40 a couple minutes
>>> >> ago), but doing a 'show calls' gives me 0 active calls. The usual
>>> >> behavior, when everything's working fine, is to get twice as many
>>> >> channels as there are active calls and no channels at all when there
>>> >> are no calls, unless they haven't been bridged yet.
>>> >>
>>> >> The originate string is something like this:
>>> >>
>>> >> var stUsRing = "%(2000,4000,440,480)";
>>> >> var timeout = 45;
>>> >> originate_str1 = "{api_hangup_hook=jsapi::callback.js
>>> >> l1,execute_on_answer=lua answered.lua 1
>>> >>
>>> >>
>>> c2c_call,ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,ringback='"+stUsRing+"',medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]"+dialstr1;
>>> >>
>>> >> Where diasltr1 has the phonenumber and and gateway info. The
>>> >> callback.js has a curl request to update some call info on an external
>>> >> database and answered.lua calls a ruby script through the os.execute()
>>> >> function (I know, I should be doing all this through the event socket,
>>> >> I was doing that but had trouble and had to come up with a quick
>>> >> solution).
>>> >>
>>> >> The system is not loaded at all, at least not for what I think and
>>> >> read that FS can handle. We are having at most 10 concurrent calls (20
>>> >> channels), with maybe 5 to 10 calls per minute.
>>> >>
>>> >> What worries me is not only that I don't know where the problem is,
>>> >> but that I have no clue how to debug it or send you guys more
>>> >> "lowlevel" and detailed information to give you an insight about
>>> >> what's going on. Any help would be greatly appreciated!
>>> >>
>>> >> Thanks!
>>> >>
>>> >> Nico
>>> >>
>>> > First off you'll want to get familiar with the resources mentioned
>>> here:
>>> > http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>> >
>>> > It has good tips on how to collect and report information.
>>> >
>>> > Second, I recommend that you pastebin your relevant portion of the
>>> dialplan
>>> > and the whole javascript program that you are using so that others can
>>> take
>>> > a look.
>>> >
>>> > Last thing: if you restart FreeSWITCH does everything work fine for a
>>> while
>>> > but then eventually it breaks down and exhibits the behavior that you
>>> are
>>> > reporting?
>>> >
>>> > -MC
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mcampbellsmith at gmail.com  Mon Jan 11 16:14:47 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 12 Jan 2010 11:14:47 +1100
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
Message-ID: <33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>

Beauty!  Thanks.. will update now.

On Tue, Jan 12, 2010 at 11:08 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> fixed in 16250
>
> On Mon, Jan 11, 2010 at 5:22 PM, Eliot Gable <egable+freeswitch at gmail.com>
> wrote:
>>
>> I can't say for certain. I know it's not in 16016. Wait for Anthony to
>> fix the issue in the current version, then update and try again.
>>
>> On Mon, Jan 11, 2010 at 6:09 PM, Mark Campbell-Smith
>> <mcampbellsmith at gmail.com> wrote:
>> > Thanks Eliot. ?I'm using FreeSWITCH Version 1.0.trunk (16131). ? Do
>> > you kno wif the fault was present in 16131?
>> >
>> > Cheers
>> >
>> > On Tue, Jan 12, 2010 at 10:02 AM, Eliot Gable
>> > <egable+freeswitch at gmail.com> wrote:
>> >> Assuming you're using SVN trunk, there is a bug that is likely causing
>> >> your issue. Anthony is fixing it right now.
>> >>
>> >> On Mon, Jan 11, 2010 at 6:16 AM, Mark Campbell-Smith
>> >> <mcampbellsmith at gmail.com> wrote:
>> >>> Hi!
>> >>> Hi!
>> >>>
>> >>> I am calling from extension 2010 to extension 1000. ?Both have ip
>> >>> addesses 192.168.1.x.
>> >>>
>> >>> In the 2000 series dialplan (a separate context) I have the following
>> >>> to try to enable bypass_media.
>> >>>
>> >>> ? ? ? ? ? ?<extension name="Local_1000_9xxx">
>> >>> ? ? ? ? ? ? ? ? ?<condition field="destination_number"
>> >>> expression="^(10[01][0-9]|9\d{3})$">
>> >>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set"
>> >>> data="dialed_extension=$1"/>
>> >>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set"
>> >>> data="proxy_media=false"/>
>> >>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set"
>> >>> data="bypass_media=true"/>
>> >>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="transfer"
>> >>> data="${dialed_extension} XML default"/>
>> >>> ? ? ? ? ? ? ? ? ?</condition>
>> >>> ? ? ? ? ? ?</extension>
>> >>>
>> >>> Is this how bypass media should be enabled?
>> >>>
>> >>> This fails fo me (the calls hang up and no audio). ?The debug trace is
>> >>> in http://pastebin.freeswitch.org/11737
>> >>>
>> >>> What have I done wrong?
>> >>>
>> >>> Thanks
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >>
>> >> --
>> >> Eliot Gable
>> >>
>> >> "We do not inherit the Earth from our ancestors: we borrow it from our
>> >> children." ~David Brower
>> >>
>> >> "I decided the words were too conservative for me. We're not borrowing
>> >> from our children, we're stealing from them--and it's not even
>> >> considered to be a crime." ~David Brower
>> >>
>> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>> >> live; not live to eat.) ~Marcus Tullius Cicero
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Eliot Gable
>>
>> "We do not inherit the Earth from our ancestors: we borrow it from our
>> children." ~David Brower
>>
>> "I decided the words were too conservative for me. We're not borrowing
>> from our children, we're stealing from them--and it's not even
>> considered to be a crime." ~David Brower
>>
>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>> live; not live to eat.) ~Marcus Tullius Cicero
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From a.alalousi at gmail.com  Mon Jan 11 16:21:26 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Tue, 12 Jan 2010 00:21:26 +0000
Subject: [Freeswitch-users] Revision 16238,
	16237 compiling and call 	handling issue
In-Reply-To: <191c3a031001111607j63a72fdft6f47084820b81bf4@mail.gmail.com>
References: <cd93b3701001111410labccf1eme3e9c680cb13c249@mail.gmail.com>
	<191c3a031001111432j5734c6begc52d5398b4844c0e@mail.gmail.com>
	<cd93b3701001111532y6a9f6f74j18c85fc9b02553ba@mail.gmail.com>
	<191c3a031001111607j63a72fdft6f47084820b81bf4@mail.gmail.com>
Message-ID: <cd93b3701001111621l771c2333r56dd1a097fbf7a58@mail.gmail.com>

we are not worthy :)

well done.

2010/1/12 Anthony Minessale <anthony.minessale at gmail.com>

> too late,
> its fixed in rev 16250
>
>
>
> On Mon, Jan 11, 2010 at 5:32 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>
>> Thanks for this.
>>
>> 2010/1/11 Anthony Minessale <anthony.minessale at gmail.com>
>>
>>
>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>
>>>
>>> On Mon, Jan 11, 2010 at 4:10 PM, Ahmed Naji <a.alalousi at gmail.com>wrote:
>>>
>>>> People,
>>>>
>>>> It seems there are  a few things that are broken in this release. So
>>>> far, I've come across two issues: first of, configure fails, per what others
>>>> reported, complaining about NONE/bin not being an absolute directory path;
>>>> 1637 has no problems with that.
>>>>
>>>> There is also what appears to be a serious call handling issue that was
>>>> not present in 1.0.4 trunks which is the following:
>>>>
>>>> Calls are initiated correctly, Leg-B is set-up and remote end rings,
>>>> ring-back is reported to Leg-A, remote end picks up, but Leg-A still gets
>>>> the ringing tone. This is pretty much the case with both internal and
>>>> external profiles, irrespective of whether or not a gateway is used to route
>>>> the call, irrespective of which media mode FS is running in and irrespective
>>>> of codec.
>>>>
>>>> I know dev and support groups don't like to get involved in performance
>>>> and load testing loops, but for he who cares, performance on 1.0.5 trunks is
>>>> miserable compared to 1.0.4. I've bombarded 1.0.4 with over 150 cps with
>>>> hardly a glitch on quad-xeon machines with 6Gb RAM and GigE controllers.
>>>> 1.0.5 falls far, far short of that and the figure is a fraction at just
>>>> under 20 for the same spec. machine. I run CentOS 5.0.4, and have had the
>>>> exact same experience with Debian stable and Ubuntu Karmic on the same
>>>> hardware.
>>>>
>>>> I'm tracing the call handling issue now, and will report back if I find
>>>> anything useful. Meantime, it would be great if someone from support can
>>>> open a Jira for this or let me know how to do that - sorry, my own ignorance
>>>> of how the dev/support system works .. only converted to FS 6 weeks ago :).
>>>> Happy to post traces and findings.
>>>>
>>>> Regards,
>>>>
>>>> Ahmed.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Ahmed Naji
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From msc at freeswitch.org  Mon Jan 11 16:25:42 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 16:25:42 -0800
Subject: [Freeswitch-users] URGENT : DTMF during bridge
In-Reply-To: <27107895.post@talk.nabble.com>
References: <27107895.post@talk.nabble.com>
Message-ID: <87f2f3b91001111625t2e959dc0g6fd2fb1f58aa0da3@mail.gmail.com>

On Mon, Jan 11, 2010 at 1:43 AM, Oscav <oscav at hotmail.fr> wrote:

>
> Hi,
>
> I need to handle DTMF during bridge in order to hangup the called party on
> caller request. The DTMF sequence should be ##. Any idea on how to do
> that??
>
> I think you need bind_meta_app. Look at the wiki as well as the
Local_Extension example to see how bind_meta_app works. The catch for you,
though, is that you can't use ## as your digit sequence, instead you'll need
to use *# or ** or something like that.

-MC
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From msc at freeswitch.org  Mon Jan 11 16:36:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 16:36:03 -0800
Subject: [Freeswitch-users] How to provide dynamic directory information?
In-Reply-To: <5d2828f1001111453t8044774w3cc882988e18ced@mail.gmail.com>
References: <5d2828f1001111453t8044774w3cc882988e18ced@mail.gmail.com>
Message-ID: <87f2f3b91001111636j1613b6ak5b5de86804f2125e@mail.gmail.com>

On Mon, Jan 11, 2010 at 2:53 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Hello!
>
> I'd like to be able to have FreeSWITCH check a database for authorization,
> every time a user registers. There are some great examples on the wiki,
> which use either MOD_XML_CURL or Lua to dynamically provide a dialplan, but
> I cannot find an example for providing a directory.
>
> I am developing an application that will have thousands of users, and will
> run on multiple FreeSWITCH servers behind a load balancer. Ideally,
> FreeSWITCH would only look-up directory information, specifically, username
> and password, whenever a user attempts to connect. The directory information
> will be changing regularly, as users are added or removed from the system.
>
> Is this possible with FreeSWITCH? Or can only dialplan information be
> provided dynamically?
>
> I've written a script in Lua that provides the XML data, such as that found
> in the example /freeswitch/conf/directory/default/ folders, and I try to
> call it with this bit of XML in /freeswitch/conf/directory/default.xml:
>
> <groups>
> <group name="default">
> <users>
>  <param name="xml-handler-script" value="/directory.lua"/>
> <param name="xml-handler-bindings" value="users"/>
>  </users>
> </groups>
>
> Is this the right approach? Am I going about this the right way?
>

You can bind "directory" as well as "dialplan" and a few others. I
personally don't use xml_curl in production but for kicks I tried to learn
it and I documented some of my journey on my personal blog. (
http://telecommusings.blogspot.com/)

xml_curl was designed to scale and be applied in your type of scenario.
Raymond (intralanman on IRC) has played with it quite a bit as have a number
of others.
-MC
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From msc at freeswitch.org  Mon Jan 11 16:44:11 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 16:44:11 -0800
Subject: [Freeswitch-users] Gateway Configuration
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9074E14@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E9074E14@prato.activenetwerx.local>
Message-ID: <87f2f3b91001111644o766fccdap5ccf4fcde3a7536b@mail.gmail.com>

On Sat, Jan 9, 2010 at 9:40 AM, Joseph L. Casale
<jcasale at activenetwerx.com>wrote:

> It seems there are two ways to configure an spa3102's fxo port w/
> pbx's, you can set the dial plan to <phone_#_of_pstn>@<freeswitch_ip>
> or <ext_to_dial>@<freeswitch_ip>.
>
> >From fs's perspective, what exactly is the difference here?
> Are there any significant differences between the two methods?
> Are there any best practices that should be considered?
>

I've only got a PAP2T (2 FXS) but from what you describe I'd say that it
makes sense for the FXO port to be "phone_#_of_pstn" and FXS to be
"ext_to_dial." I believe FS tries to be endpoint-type agnostic in this
scenario. It's a SIP call in and gets routed in the dialplan.

>
> Incoming sip did's and a zap line I had all were configured so that
> they entered the public context filtered by <phone_#>.
>
> If a call is coming in from an actual PSTN line then hitting the public
context makes a lot of sense.

> Most of the examples I see for setting up the spa don't function like this
> but a couple do?
>
As usual "it depends." However, the simplest rule of thumb would be that FXS
ports are analogous to SIP users and FXO ports are analogous to SIP
gateways.

Out of curiosity, what is your application?
-MC
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From anatoliy at kounitskiy.com  Mon Jan 11 16:44:48 2010
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Tue, 12 Jan 2010 02:44:48 +0200
Subject: [Freeswitch-users] Problem between revision 16223 and 16242 on
 Debian (stable) 64bit
In-Reply-To: <1263234649.2504.35.camel@lenovor400-laptop>
References: <1263230329.2504.33.camel@lenovor400-laptop>
	<1263234649.2504.35.camel@lenovor400-laptop>
Message-ID: <1263257088.5515.6.camel@lenovor400-laptop>

Thank you for fixing it so fast :) (tested with 16250)

Regards,
Anatoliy


On Mon, 2010-01-11 at 20:30 +0200, Anatoliy Kounitskiy wrote:
> Ok, i found in which revision it is broken.
> 
> Until revision 16237 - it works as charm
> In Revision 16238 - it doesn't work
> 
> svn log --revision 16237:16238
> ------------------------------------------------------------------------
> r16238 | mikej | 2010-01-11 16:36:29 +0200 (Mon, 11 Jan 2010) | 1 line
> 
> wip move towards adding directory layout control to configure
> ------------------------------------------------------------------------
> 
> Regards,
> 
> On Mon, 2010-01-11 at 19:18 +0200, Anatoliy Kounitskiy wrote:
> > Hello,
> > I just made a checkout of the svn and tried to configure it, but there
> > is an error in the arp-util lib, after the ./bootstrap.sh
> > 
> > 
> > Freeswitch revision: 16242
> > OS: Debian 64b (stable)
> > Command used: ./configure --prefix=/usr/local/freeswitch
> > --enable-optimization --enable-64
> > Error:
> > checking for Expat in xml/expat... yes
> > configuring package in xml/expat now
> > configure: error: expected an absolute directory name for --bindir:
> > NONE/bin
> > configure failed for xml/expat
> > configure: error: ./configure.gnu failed for libs/apr-util
> > 
> > If I execute "sh configure.gnu"/"sh configure.gnu --enable-64" in
> > libs/apr-util it goes without an error.
> > 
> > On the same server with Freeswitch revision: 16223 (with the same
> > configure command), it goes as planned - without errors.
> > 
> > Regards,
> > 
> 
> 


-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540



From rupa at rupa.com  Mon Jan 11 16:47:48 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 11 Jan 2010 18:47:48 -0600
Subject: [Freeswitch-users] URGENT : DTMF during bridge
In-Reply-To: <27107895.post@talk.nabble.com>
References: <27107895.post@talk.nabble.com>
Message-ID: <b39273531001111647n2867f859rfa08cd506a5964cb@mail.gmail.com>

Is it really necessary to say "URGENT"?  I doubt anyone will respond any
faster/sooner.

Anyway:

bind_meta_app will do * + a char to do an action.  So *# could be hangup.

You could listen to DTMF events over ESL and do whatever action you want.
 Keep in mind you should have a maximum time duration between the first a
second char.  That way, if I hit # 1s into the call and then again 5min into
the call you don't hangup on me.

On Mon, Jan 11, 2010 at 3:43 AM, Oscav <oscav at hotmail.fr> wrote:

>
> Hi,
>
> I need to handle DTMF during bridge in order to hangup the called party on
> caller request. The DTMF sequence should be ##. Any idea on how to do
> that??
>
> Thanks.
> --
> View this message in context:
> http://old.nabble.com/URGENT-%3A-DTMF-during-bridge-tp27107895p27107895.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From msc at freeswitch.org  Mon Jan 11 16:50:56 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 16:50:56 -0800
Subject: [Freeswitch-users] Help with Portech <-> Freeswitch
In-Reply-To: <c55d46951001090900p36df42e3ga91e61812d5b8d03@mail.gmail.com>
References: <c55d46951001090900p36df42e3ga91e61812d5b8d03@mail.gmail.com>
Message-ID: <87f2f3b91001111650n6d635e8dk9de30a4249a6dd73@mail.gmail.com>

On Sat, Jan 9, 2010 at 9:00 AM, Max Bridgewater
<max.bridgewater at gmail.com>wrote:

> Hi Guys,
>
> It appears quite a few people in the list are using Portech. Can you please
> help me connect Freeswitch to  it for termination puposes?
>
> Here is what I've done so far but without success.
>
> In Freeswitch I created a profile and stored it in under
> /usr/local/freeswitch/conf/sip_profiles/external/. Here is the content:
>
> <include>
>   <gateway name="portech">
>     <param name="username" value="noauthentication" />
>     <param name="password" value="noauthentication" />
>     <param name="proxy" value="74.24.22.59" />
>     <param name="realm" value="74.24.22.59" />
>     <param name="register" value="false" />
>   </gateway>
> </include>
>
> Then, in the /usr/local/freeswitch/confi/dialplan/default.xml, I added the
> following extension:
>
>    <extension name="portech">
>       <condition field="destination_number" expression="^2801$">
>         <action application="bridge"
> data="sofia/gateway/portech/5147237479"/>
>       </condition>
>     </extension>
>
> In Portech MV374, what I did is simply adding one entry in the Mobile/Lan
> to mobile table  that consists of  URL: 74.24.22.59 and call Num: #.
>
> Now, when I connect to Freeswitch with Xlite and try to dial extension 2801
> it simply says User Busy. i don't even see that attempts are being made to
> connect to the Portech gateway.
>
> Any idea?
> Thanks in advance.
>

Turn on debug level console output and turn on SIP trace and make a test
call capturing the output. Put into a pastebin and reply to this thread with
the link. Hopefully the log will tell you (and us) what is going on.

FYI, this page has lots have handy tips for how to gather information and
report it:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

That page is your friend because it gives you lots of tools and skills for
gathering information. :)
-MC
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From msc at freeswitch.org  Mon Jan 11 16:53:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 16:53:03 -0800
Subject: [Freeswitch-users] URGENT : DTMF during bridge
In-Reply-To: <b39273531001111647n2867f859rfa08cd506a5964cb@mail.gmail.com>
References: <27107895.post@talk.nabble.com>
	<b39273531001111647n2867f859rfa08cd506a5964cb@mail.gmail.com>
Message-ID: <87f2f3b91001111653y5c78bca3r47170eaa064ee9d1@mail.gmail.com>

On Mon, Jan 11, 2010 at 4:47 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Is it really necessary to say "URGENT"?  I doubt anyone will respond any
> faster/sooner.
>
> Anyway:
>
> bind_meta_app will do * + a char to do an action.  So *# could be hangup.
>
> You could listen to DTMF events over ESL and do whatever action you want.
>  Keep in mind you should have a maximum time duration between the first a
> second char.  That way, if I hit # 1s into the call and then again 5min into
> the call you don't hangup on me.
>

Rupa is wise... :)
-MC
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From brian at freeswitch.org  Mon Jan 11 16:53:52 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Jan 2010 18:53:52 -0600
Subject: [Freeswitch-users] Problem between revision 16223 and 16242 on
	Debian (stable) 64bit
In-Reply-To: <1263257088.5515.6.camel@lenovor400-laptop>
References: <1263230329.2504.33.camel@lenovor400-laptop>
	<1263234649.2504.35.camel@lenovor400-laptop>
	<1263257088.5515.6.camel@lenovor400-laptop>
Message-ID: <5246A6FF-3A35-4153-B3E6-3BEB501A947D@freeswitch.org>

Tony has a wish list on the FAQ ;)  First question.

/b

On Jan 11, 2010, at 6:44 PM, Anatoliy Kounitskiy wrote:

> Thank you for fixing it so fast :) (tested with 16250)
> 
> Regards,
> Anatoliy



From msc at freeswitch.org  Mon Jan 11 16:55:08 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 16:55:08 -0800
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
Message-ID: <87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>

On Mon, Jan 11, 2010 at 4:18 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:

> Dear All,
>
> I posted a thread re the subject but didn't get any joy, so perhaps second
> time lucky.
>
> I need to rewrite a couple of hangup causes to mean NORMAL_CONGESTION and
> no matter which variables I set, this isn't happening. The idea is to
> control what codes are returned to an end point after a successful bridge,
> as well as deal with what codes are returned if the bridge is unsuccessful
> (e.g. user_busy, originator_cancel ...etc).
>
> I've had limited success by setting hangup_after_bridge=false then bridging
> to error/<required_code>. This, however only works when the B-leg terminates
> the call after a successful answer. Any other codes are not rewritten.
>
> I've also tried playing with the bridge_hangup_code and hangup_code
> variables prior and after bridging, still no joy. I have also set
> sip_ignore_remote_cause=true prior to entering the bridge, as well
> explicitly in vars.xml.
>
> By the way, I'm running in proxy-media mode, but I did try it with
> bypass-media as well. Same symptoms, same behaviour.
>
> Any help with this would be highly appreciated.
>
> Well, I do know that when you do a hangup in the dialplan you can pass an
optional cause as well:
<action application="hangup" data="USER_BUSY"/>
If you are doing the hanging up then you have a fair amount of control...
-MC
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From msc at freeswitch.org  Mon Jan 11 16:55:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 16:55:53 -0800
Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS
	based callcenter)
In-Reply-To: <20100111181203.GD10774@hijacked.us>
References: <20091219014359.GA21798@hijacked.us>
	<855e4dcf1001041842u46c6d36q9c2e4ece2ced7380@mail.gmail.com>
	<20100111181203.GD10774@hijacked.us>
Message-ID: <87f2f3b91001111655x3c785b6au3a279a32f3eec6da@mail.gmail.com>

On Mon, Jan 11, 2010 at 10:12 AM, Andrew Thompson <andrew at hijacked.us>wrote:

> On Tue, Jan 05, 2010 at 03:42:57PM +1300, Tim Uckun wrote:
> > > http://opencsm.org/downloads/spice-telephony-0.9.6.tar.gz
> > >
> >
> > There seems to be something wrong with both opencsm.org and
> wiki.opencsm.org.
> >
> > Just thought I'd let you know.
> >
>
> Yeah, my company wanted me to move it, so I re-hosted it as a github
> project (with a new name):
>
> http://github.com/Vagabond/OpenACD
>
> I hadn't announced the change yet because I've been away for the last
> week and didn't have time. The wiki contents haven't been moved over
> yet, but they needed some cleanup anyway.
>
> On the other hand attended transfer support finally materialized.
>
Thanks Andrew!
-MC
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From msc at freeswitch.org  Mon Jan 11 16:57:02 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 16:57:02 -0800
Subject: [Freeswitch-users] Sending call notification tone to a busy FXS
	port
In-Reply-To: <08abf4fa1e11a07bda4381fda8f93879@thom.fr.eu.org>
References: <08abf4fa1e11a07bda4381fda8f93879@thom.fr.eu.org>
Message-ID: <87f2f3b91001111657g51f63d55yce889283fee655f6@mail.gmail.com>

On Mon, Jan 11, 2010 at 4:53 AM, Fran?ois Legal <devel at thom.fr.eu.org>wrote:

> Hello,
>
> I was just wondering if it is possible (and how) to send a call
> notification tone to a phone connected to an FXS port and which is already
> in communication.
>
> What is the interface? ATA or a TDM card? What model? Need to make sure
this feature is supported by your specific device.
-MC
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From carlos.talbot at gmail.com  Mon Jan 11 17:22:12 2010
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Mon, 11 Jan 2010 19:22:12 -0600
Subject: [Freeswitch-users] FSComm Windows build
Message-ID: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>

FYI,

there's a Windows pre-compiled binary of FSComm now available for those who
want to check it.

http://files.freeswitch.org/windows_installer/FSComm.exe

regards,

Carlos
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From msc at freeswitch.org  Mon Jan 11 17:42:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 17:42:00 -0800
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
Message-ID: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>

We want to let everyone know that FreeSWITCH now supports the Broadsoft SCA
method of doing shared lines. The story is here:

http://www.freeswitch.org/node/227

Tony and Brian spent many hours laboring over this, so please be sure to
show your appreciation to them for this new feature and all of the great
things they do for the FreeSWITCH community and VoIP in general!

-Michael
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From jcasale at activenetwerx.com  Mon Jan 11 18:24:50 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 12 Jan 2010 02:24:50 +0000
Subject: [Freeswitch-users] Gateway Configuration
In-Reply-To: <87f2f3b91001111644o766fccdap5ccf4fcde3a7536b@mail.gmail.com>
References: <CA5A491E9DEFBE4CB777DE97E21575E9074E14@prato.activenetwerx.local>
	<87f2f3b91001111644o766fccdap5ccf4fcde3a7536b@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9077A6F@prato.activenetwerx.local>

>I've only got a PAP2T (2 FXS) but from what you describe I'd say that it makes
>sense for the FXO port to be "phone_#_of_pstn" and FXS to be "ext_to_dial."
>I believe FS tries to be endpoint-type agnostic in this scenario. It's a SIP
>call in and gets routed in the dialplan. 

This is what I thought as well, where I am unclear is how you register this device
as a gateway then as it isn't like a regular sip provider by means of a gateway
definition? The gateway definition requires a username and password for example
so how do you create the basic definition to allow the spa3102 to push the call into
the public context?

I can make an outbound call as I just construct a dialplan like:
"sofia/internal/$1 at spa3102.domain.local:5060"

>>Incoming sip did's and a zap line I had all were configured so that
>>they entered the public context filtered by <phone_#>.
>If a call is coming in from an actual PSTN line then hitting the public context makes a lot of sense.? 

>Out of curiosity, what is your application?

We use this one pstn line as backup when voip is down for whatever reason. The
company # given out is the single pstn line, its routed on busy to the sip provider
for a series of other lines. As we expect the pstn to always be working, if the
voip is up and more people call we get the calls. If voip is down, at least we have
1 line always working w/o any manual intervention. Outgoing always tries sip first
and then routes to the pstn line if required.

Thanks,
jlc


From Mailings at kh-dev.de  Mon Jan 11 19:01:27 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Tue, 12 Jan 2010 04:01:27 +0100
Subject: [Freeswitch-users] proxy_media seems to be broken
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403B145DC1941@srv02.khdev.corp>

Hi,

I just tested with the latest tarball (11. Jan).

And now my T.38 Fax config with proxy_media doesn't work anymore.

Here's the config:
Fax <-> Cisco SPA2102 <-> FS <-> Lancom SIP/ISDN-Gateway <-> ISDN
proxy_media and late negotiation is turned on.

Here's what happens:
- Fax makes call
- Lancom routes the call to ISDN
- Remote fax takes the call
- Lancom sends reinvite with T.38
- The Cisco just keeps on ringing without noticing that the call was answered

This config worked in older FS versions without problems.

Might this already be fixed in rev 16250 (bypass_media problem)?
Or did the config of proxy_media change?

Thanks, Klaus
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From msc at freeswitch.org  Mon Jan 11 19:26:34 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 19:26:34 -0800
Subject: [Freeswitch-users] Gateway Configuration
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9077A6F@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E9074E14@prato.activenetwerx.local>
	<87f2f3b91001111644o766fccdap5ccf4fcde3a7536b@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E9077A6F@prato.activenetwerx.local>
Message-ID: <87f2f3b91001111926xcb72feepf147c45697b38c8f@mail.gmail.com>

On Mon, Jan 11, 2010 at 6:24 PM, Joseph L. Casale <jcasale at activenetwerx.com
> wrote:

> >I've only got a PAP2T (2 FXS) but from what you describe I'd say that it
> makes
> >sense for the FXO port to be "phone_#_of_pstn" and FXS to be
> "ext_to_dial."
> >I believe FS tries to be endpoint-type agnostic in this scenario. It's a
> SIP
> >call in and gets routed in the dialplan.
>
> This is what I thought as well, where I am unclear is how you register this
> device
> as a gateway then as it isn't like a regular sip provider by means of a
> gateway
> definition? The gateway definition requires a username and password for
> example
> so how do you create the basic definition to allow the spa3102 to push the
> call into
> the public context?
>

Please email me off list and we'll work out the particulars and then create
a more complete wiki entry.
-MC
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From djbinter at yahoo.com  Mon Jan 11 19:34:51 2010
From: djbinter at yahoo.com (DJB)
Date: Mon, 11 Jan 2010 19:34:51 -0800 (PST)
Subject: [Freeswitch-users] proxy_media seems to be broken
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F0403B145DC1941@srv02.khdev.corp>
References: <A2F1C0E212BBBA4E8C0E7BE568361F0403B145DC1941@srv02.khdev.corp>
Message-ID: <608345.82340.qm@web37502.mail.mud.yahoo.com>

Yes, it has been fixed in 16250.



________________________________
From: Klaus Hochlehnert <Mailings at kh-dev.de>
To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
Sent: Mon, January 11, 2010 7:01:27 PM
Subject: [Freeswitch-users] proxy_media seems to be broken

 
Hi,
 
I just tested with the latest tarball (11. Jan).
 
And now my T.38 Fax config with proxy_media
doesn?t work anymore.
 
Here?s the config:
Fax <-> Cisco SPA2102 <-> FS
<-> Lancom SIP/ISDN-Gateway <-> ISDN
proxy_media and late negotiation is turned
on.
 
Here?s what happens:
- Fax makes call
- Lancom routes the call to ISDN
- Remote fax takes the call
- Lancom sends reinvite with T.38
- The Cisco just keeps on ringing without
noticing that the call was answered
 
This config worked in older FS versions without
problems.
 
Might this already be fixed in rev 16250 (bypass_media
problem)?
Or did the config of proxy_media change?
 
Thanks, Klaus


      
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From a.alalousi at gmail.com  Mon Jan 11 23:56:36 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Tue, 12 Jan 2010 07:56:36 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
	<87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
Message-ID: <cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>

Hi Michael,

This is exactly what I'm doing, but it's just not happening.

Thanks,

Ahmed.


2010/1/12 Michael Collins <msc at freeswitch.org>

>
>
> On Mon, Jan 11, 2010 at 4:18 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>
>> Dear All,
>>
>> I posted a thread re the subject but didn't get any joy, so perhaps second
>> time lucky.
>>
>> I need to rewrite a couple of hangup causes to mean NORMAL_CONGESTION and
>> no matter which variables I set, this isn't happening. The idea is to
>> control what codes are returned to an end point after a successful bridge,
>> as well as deal with what codes are returned if the bridge is unsuccessful
>> (e.g. user_busy, originator_cancel ...etc).
>>
>> I've had limited success by setting hangup_after_bridge=false then
>> bridging to error/<required_code>. This, however only works when the B-leg
>> terminates the call after a successful answer. Any other codes are not
>> rewritten.
>>
>> I've also tried playing with the bridge_hangup_code and hangup_code
>> variables prior and after bridging, still no joy. I have also set
>> sip_ignore_remote_cause=true prior to entering the bridge, as well
>> explicitly in vars.xml.
>>
>> By the way, I'm running in proxy-media mode, but I did try it with
>> bypass-media as well. Same symptoms, same behaviour.
>>
>> Any help with this would be highly appreciated.
>>
>> Well, I do know that when you do a hangup in the dialplan you can pass an
> optional cause as well:
> <action application="hangup" data="USER_BUSY"/>
> If you are doing the hanging up then you have a fair amount of control...
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From steveayre at gmail.com  Tue Jan 12 00:46:27 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 12 Jan 2010 08:46:27 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
	<87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
	<cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>
Message-ID: <eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>

Can you show us the dialplan extension you're trying?

Thanks,
-Steve

2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
> Hi Michael,
>
> This is exactly what I'm doing, but it's just not happening.
>
> Thanks,
>
> Ahmed.
>
>
> 2010/1/12 Michael Collins <msc at freeswitch.org>
>>
>>
>> On Mon, Jan 11, 2010 at 4:18 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>>>
>>> Dear All,
>>>
>>> I posted a thread re the subject but didn't get any joy, so perhaps
>>> second time lucky.
>>>
>>> I need to rewrite a couple of hangup causes to mean NORMAL_CONGESTION and
>>> no matter which variables I set, this isn't happening. The idea is to
>>> control what codes are returned to an end point after a successful bridge,
>>> as well as deal with what codes are returned if the bridge is unsuccessful
>>> (e.g. user_busy, originator_cancel ...etc).
>>>
>>> I've had limited success by setting hangup_after_bridge=false then
>>> bridging to error/<required_code>. This, however only works when the B-leg
>>> terminates the call after a successful answer. Any other codes are not
>>> rewritten.
>>>
>>> I've also tried playing with the bridge_hangup_code and hangup_code
>>> variables prior and after bridging, still no joy. I have also set
>>> sip_ignore_remote_cause=true prior to entering the bridge, as well
>>> explicitly in vars.xml.
>>>
>>> By the way, I'm running in proxy-media mode, but I did try it with
>>> bypass-media as well. Same symptoms, same behaviour.
>>>
>>> Any help with this would be highly appreciated.
>>>
>> Well, I do know that when you do a hangup in the dialplan you can pass an
>> optional cause as well:
>> <action application="hangup" data="USER_BUSY"/>
>> If you are doing the hanging up then you have a fair amount of control...
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From devel at thom.fr.eu.org  Tue Jan 12 01:25:11 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Tue, 12 Jan 2010 10:25:11 +0100
Subject: [Freeswitch-users] Sending call notification tone to a busy FXS
	port
In-Reply-To: <87f2f3b91001111657g51f63d55yce889283fee655f6@mail.gmail.com>
References: <08abf4fa1e11a07bda4381fda8f93879@thom.fr.eu.org>
	<87f2f3b91001111657g51f63d55yce889283fee655f6@mail.gmail.com>
Message-ID: <1b556d9a7c2b1d64fd745170e5b172c9@thom.fr.eu.org>



This is a TDM Sangoma A400. 

I will check with Sangoma. If it is
supported by HW, is it supported by openzap ? 

Fran?ois 

On Mon, 11 Jan
2010 16:57:02 -0800, Michael Collins  wrote: 

 On Mon, Jan 11, 2010 at
4:53 AM, Fran?ois Legal  wrote:
 Hello,

 I was just wondering if it is
possible (and how) to send a call
 notification tone to a phone connected
to an FXS port and which is already
 in communication.

 What is the
interface? ATA or a TDM card? What model? Need to make sure this feature is
supported by your specific device.
-MC  

 

Links:
------
[1]
mailto:devel at thom.fr.eu.org
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From mcampbellsmith at gmail.com  Tue Jan 12 01:32:28 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 12 Jan 2010 20:32:28 +1100
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
	<33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
Message-ID: <33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>

I've updated and tested bypass_media.

It works if I remove this line from the B leg dialplan (ie 2010 calls
1000 - this is in the 1000 section of the dialplan):
                        <action application="tone_detect" data="fax
1100 r +5000 transfer 'receivefax XML features' 1" />

Does bypass_media work with tone_detect?

Thanks!

On Tue, Jan 12, 2010 at 11:14 AM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Beauty! ?Thanks.. will update now.
>
> On Tue, Jan 12, 2010 at 11:08 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> fixed in 16250
>>
>> On Mon, Jan 11, 2010 at 5:22 PM, Eliot Gable <egable+freeswitch at gmail.com>
>> wrote:
>>>
>>> I can't say for certain. I know it's not in 16016. Wait for Anthony to
>>> fix the issue in the current version, then update and try again.
>>>
>>> On Mon, Jan 11, 2010 at 6:09 PM, Mark Campbell-Smith
>>> <mcampbellsmith at gmail.com> wrote:
>>> > Thanks Eliot. ?I'm using FreeSWITCH Version 1.0.trunk (16131). ? Do
>>> > you kno wif the fault was present in 16131?
>>> >
>>> > Cheers
>>> >
>>> > On Tue, Jan 12, 2010 at 10:02 AM, Eliot Gable
>>> > <egable+freeswitch at gmail.com> wrote:
>>> >> Assuming you're using SVN trunk, there is a bug that is likely causing
>>> >> your issue. Anthony is fixing it right now.
>>> >>
>>> >> On Mon, Jan 11, 2010 at 6:16 AM, Mark Campbell-Smith
>>> >> <mcampbellsmith at gmail.com> wrote:
>>> >>> Hi!
>>> >>> Hi!
>>> >>>
>>> >>> I am calling from extension 2010 to extension 1000. ?Both have ip
>>> >>> addesses 192.168.1.x.
>>> >>>
>>> >>> In the 2000 series dialplan (a separate context) I have the following
>>> >>> to try to enable bypass_media.
>>> >>>
>>> >>> ? ? ? ? ? ?<extension name="Local_1000_9xxx">
>>> >>> ? ? ? ? ? ? ? ? ?<condition field="destination_number"
>>> >>> expression="^(10[01][0-9]|9\d{3})$">
>>> >>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set"
>>> >>> data="dialed_extension=$1"/>
>>> >>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set"
>>> >>> data="proxy_media=false"/>
>>> >>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set"
>>> >>> data="bypass_media=true"/>
>>> >>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="transfer"
>>> >>> data="${dialed_extension} XML default"/>
>>> >>> ? ? ? ? ? ? ? ? ?</condition>
>>> >>> ? ? ? ? ? ?</extension>
>>> >>>
>>> >>> Is this how bypass media should be enabled?
>>> >>>
>>> >>> This fails fo me (the calls hang up and no audio). ?The debug trace is
>>> >>> in http://pastebin.freeswitch.org/11737
>>> >>>
>>> >>> What have I done wrong?
>>> >>>
>>> >>> Thanks
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>
>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>>
>>> >>
>>> >>
>>> >>
>>> >> --
>>> >> Eliot Gable
>>> >>
>>> >> "We do not inherit the Earth from our ancestors: we borrow it from our
>>> >> children." ~David Brower
>>> >>
>>> >> "I decided the words were too conservative for me. We're not borrowing
>>> >> from our children, we're stealing from them--and it's not even
>>> >> considered to be a crime." ~David Brower
>>> >>
>>> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>>> >> live; not live to eat.) ~Marcus Tullius Cicero
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>>
>>>
>>> --
>>> Eliot Gable
>>>
>>> "We do not inherit the Earth from our ancestors: we borrow it from our
>>> children." ~David Brower
>>>
>>> "I decided the words were too conservative for me. We're not borrowing
>>> from our children, we're stealing from them--and it's not even
>>> considered to be a crime." ~David Brower
>>>
>>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>>> live; not live to eat.) ~Marcus Tullius Cicero
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From Claudio.Cavalera at italtel.it  Tue Jan 12 01:43:34 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Tue, 12 Jan 2010 10:43:34 +0100
Subject: [Freeswitch-users] playing with sessions in lua
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F362D9@BESONE.corp.dom>

Hello,
this should be simple in theory therefore I'm probably missing the right
way to do it.
I want to play with sessions in lua, bridge them, park them, etc...

example1: Consider this simple lua script in which i create two
sessions:

api = freeswitch.API();
api:execute("originate", "sofia/internal/1001%192.168.1.1 &park");
api:execute("originate", "sofia/internal/1002%192.168.1.1 &park");

now if i want to bridge them i suppose i should use something like

api:execute("uuid_bridge", "uuid_1 uuid_2");

but how do i get uuid_1 and uuid_2, i.e. the uuids of the two sessions?


example2: I could create sessions with

local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
local session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");

but then there is NOT a bridge API to bridge the sessions like:
bridge(session1, session2);

I admit I have not yet understood why such bridge possibility exist in
javascript but does not exist in lua.
http://wiki.freeswitch.org/wiki/Javascript_Misc_bridge
I guess there is a reason for this but I can't figure it out.


example3: yet another possibility
local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
session1:originate(session2, "sofia/internal/1002%192.168.1.1", 1000);

but it does not work either.

Besides with this third example something strange happen:

freeswitch at internal> show channels
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr
,dest,application,application_data,dialplan,context,read_codec,read_rate
,write_codec,write_rate,secure
1c5db2df-14ce-4516-94f2-bb7c087e0802,outbound,2010-01-12
10:50:59,1263289859,sofia/internal/1004,CS_REPORTING,FreeSWITCH,00000000
00,,1004,,,,default,PCMA,8000,PCMA,8000,

1 total.

freeswitch at internal> uuid_kill 1c5db2df-14ce-4516-94f2-bb7c087e0802
-ERR No Such Channel!

freeswitch at internal> show channels
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr
,dest,application,application_data,dialplan,context,read_codec,read_rate
,write_codec,write_rate,secure
1c5db2df-14ce-4516-94f2-bb7c087e0802,outbound,2010-01-12
10:50:59,1263289859,sofia/internal/1004,CS_REPORTING,FreeSWITCH,00000000
00,,1004,,,,default,PCMA,8000,PCMA,8000,

1 total.

freeswitch at internal>

If you are interested the full log is here:
http://pastebin.freeswitch.org/11757
but I admit i'm not on latest trunk yet!

Thanks.
Ciao,
Claudio


PS: Is there a reason why there is a uuid_park command but not
uuid_valet_park ?


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From a.alalousi at gmail.com  Tue Jan 12 01:52:08 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Tue, 12 Jan 2010 09:52:08 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
	<87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
	<cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>
	<eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>
Message-ID: <cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>

Here you go:

<condition field="destination_number" expression="^(?:5824)(44(1|2)\d+)$"
break="on-true">
  <action inline="true" application="set" data="bypass_media=false"/>
  <action inline="true" application="set" data="prpoxy_media=false"/>
  <action inline="true" application="set" data="disable_q850_reason=true"/>
  <action inline="true" application="set" data="hangup_after_bridge=false"/>
  <action inline="true" application="set"
data="sip_ignore_remote_cause=true"/>
  <action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1"/>
  <action inline="true" application="set"
data="bridge_hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
  <action inline="true" application="set"
data="hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
  <action application="bridge" data="error/normal_circuit_congestion"/>
  <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
</condition>

As you can see, I am trying to rewrite the hangup codes in a multitude of
ways and places, but still exhibit the same behaviour.

Any help appreciated.

Regards,

Ahmed.

2010/1/12 Steven Ayre <steveayre at gmail.com>

> Can you show us the dialplan extension you're trying?
>
> Thanks,
> -Steve
>
> 2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
> > Hi Michael,
> >
> > This is exactly what I'm doing, but it's just not happening.
> >
> > Thanks,
> >
> > Ahmed.
> >
> >
> > 2010/1/12 Michael Collins <msc at freeswitch.org>
> >>
> >>
> >> On Mon, Jan 11, 2010 at 4:18 AM, Ahmed Naji <a.alalousi at gmail.com>
> wrote:
> >>>
> >>> Dear All,
> >>>
> >>> I posted a thread re the subject but didn't get any joy, so perhaps
> >>> second time lucky.
> >>>
> >>> I need to rewrite a couple of hangup causes to mean NORMAL_CONGESTION
> and
> >>> no matter which variables I set, this isn't happening. The idea is to
> >>> control what codes are returned to an end point after a successful
> bridge,
> >>> as well as deal with what codes are returned if the bridge is
> unsuccessful
> >>> (e.g. user_busy, originator_cancel ...etc).
> >>>
> >>> I've had limited success by setting hangup_after_bridge=false then
> >>> bridging to error/<required_code>. This, however only works when the
> B-leg
> >>> terminates the call after a successful answer. Any other codes are not
> >>> rewritten.
> >>>
> >>> I've also tried playing with the bridge_hangup_code and hangup_code
> >>> variables prior and after bridging, still no joy. I have also set
> >>> sip_ignore_remote_cause=true prior to entering the bridge, as well
> >>> explicitly in vars.xml.
> >>>
> >>> By the way, I'm running in proxy-media mode, but I did try it with
> >>> bypass-media as well. Same symptoms, same behaviour.
> >>>
> >>> Any help with this would be highly appreciated.
> >>>
> >> Well, I do know that when you do a hangup in the dialplan you can pass
> an
> >> optional cause as well:
> >> <action application="hangup" data="USER_BUSY"/>
> >> If you are doing the hanging up then you have a fair amount of
> control...
> >> -MC
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Ahmed Naji
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From jason at jasonjgw.net  Tue Jan 12 01:53:32 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 12 Jan 2010 20:53:32 +1100
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
	<33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
	<33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>
Message-ID: <20100112095332.GA32294@jdc.jasonjgw.net>

Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> I've updated and tested bypass_media.
> 
> It works if I remove this line from the B leg dialplan (ie 2010 calls
> 1000 - this is in the 1000 section of the dialplan):
>                         <action application="tone_detect" data="fax
> 1100 r +5000 transfer 'receivefax XML features' 1" />
> 
> Does bypass_media work with tone_detect?

As I understand it, tone_detect detects tones in the RTP stream (i.e., in the
audio). For this to be possible, FreeSWITCH has to be in the audio path, hence
bypass media cannot be used

If this reasoning isn't obvious to you, then you've misunderstood what
tone_detect does or what bypass media is (the audio traffic flows directly
between the two endpoints without passing through the FreeSWITCH system that
establishes the connection, therefore FreeSWITCH can't process it to detect
tones and consequently bypass media and tone detection are inherently
incompatible.)



From mcampbellsmith at gmail.com  Tue Jan 12 02:23:33 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 12 Jan 2010 21:23:33 +1100
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <20100112095332.GA32294@jdc.jasonjgw.net>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
	<33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
	<33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>
	<20100112095332.GA32294@jdc.jasonjgw.net>
Message-ID: <33c87fa31001120223t13efc07agcc595e48405fd8bb@mail.gmail.com>

Hi Jason,

I have understood that.  Its not that a difficult concept to understand!

In the log I see:
2010-01-12 21:03:17.585598 [DEBUG] mod_dptools.c:818
sofia/internal/1000 at 192.168.1.120 SET
[ringback]=[v=-7;%(400,200,413,438);%(400,2000,413,438)]
2010-01-12 21:03:17.605591 [DEBUG] switch_core_session.c:1509
Application tone_detect Requires media! pre_answering channel
sofia/internal/1000 at 192.168.1.120

I thought the SIP re-Invite message can be used to update media
parameters, including IP address endpoints.  Does FS try too do this
in the case that tone_detect is used?

On Tue, Jan 12, 2010 at 8:53 PM, Jason White <jason at jasonjgw.net> wrote:
> Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>> I've updated and tested bypass_media.
>>
>> It works if I remove this line from the B leg dialplan (ie 2010 calls
>> 1000 - this is in the 1000 section of the dialplan):
>> ? ? ? ? ? ? ? ? ? ? ? ? <action application="tone_detect" data="fax
>> 1100 r +5000 transfer 'receivefax XML features' 1" />
>>
>> Does bypass_media work with tone_detect?
>
> As I understand it, tone_detect detects tones in the RTP stream (i.e., in the
> audio). For this to be possible, FreeSWITCH has to be in the audio path, hence
> bypass media cannot be used
>
> If this reasoning isn't obvious to you, then you've misunderstood what
> tone_detect does or what bypass media is (the audio traffic flows directly
> between the two endpoints without passing through the FreeSWITCH system that
> establishes the connection, therefore FreeSWITCH can't process it to detect
> tones and consequently bypass media and tone detection are inherently
> incompatible.)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From dome at tel.co.th  Tue Jan 12 03:13:12 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 12 Jan 2010 18:13:12 +0700
Subject: [Freeswitch-users] CDR and reporting.
Message-ID: <8ccbff061001120313u78701883lf9e212947af0528@mail.gmail.com>

Dear All,
           I have 200k cdr record daily. what's good solution to
record and report ?
Now I'm thinking about CDRTOOL from ag project and areski stat (now
part of a2billing).
My idea use mod_xml_cdr -> http (may be php, or perlembded on nginx) -> mysql

i want to report ASR, ACD ,
any comment ?

Dome C.


From sos at sokhapkin.dyndns.org  Tue Jan 12 03:39:37 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 12 Jan 2010 06:39:37 -0500
Subject: [Freeswitch-users] CDR and reporting.
In-Reply-To: <8ccbff061001120313u78701883lf9e212947af0528@mail.gmail.com>
References: <8ccbff061001120313u78701883lf9e212947af0528@mail.gmail.com>
Message-ID: <201001120639.38098.sos@sokhapkin.dyndns.org>

Look at default config for mod_cdr_csv, sql template. It can produce a 
sequence of SQL INSERT statements which can be later fed (after daily log 
rotation) to mysql client program.

I see no reason to employ xml for this task, the language hard to read and 
parse for both human and computer.

On Tuesday 12 January 2010, Dome Charoenyost wrote:
> Dear All,
>            I have 200k cdr record daily. what's good solution to
> record and report ?
> Now I'm thinking about CDRTOOL from ag project and areski stat (now
> part of a2billing).
> My idea use mod_xml_cdr -> http (may be php, or perlembded on nginx) ->
> mysql
>
> i want to report ASR, ACD ,
> any comment ?
>
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From Russell.Mosemann at cune.org  Tue Jan 12 04:25:47 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Tue, 12 Jan 2010 06:25:47 -0600
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com><87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com><cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com><eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>
	<cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>
Message-ID: <FDE78FE432FE49D091FA62A0BE7A3CCD@cune.pri>

>   <action inline="true" application="set" data="prpoxy_media=false"/>

prpoxy? ;-)

--
Russell Mosemann



From a.alalousi at gmail.com  Tue Jan 12 05:07:26 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Tue, 12 Jan 2010 13:07:26 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <FDE78FE432FE49D091FA62A0BE7A3CCD@cune.pri>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
	<87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
	<cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>
	<eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>
	<cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>
	<FDE78FE432FE49D091FA62A0BE7A3CCD@cune.pri>
Message-ID: <cd93b3701001120507j941367etc70d56cd92ccd398@mail.gmail.com>

;)


2010/1/12 Russell Mosemann <Russell.Mosemann at cune.org>

> >   <action inline="true" application="set" data="prpoxy_media=false"/>
>
> prpoxy? ;-)
>
> --
> Russell Mosemann
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From steveayre at gmail.com  Tue Jan 12 05:20:45 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 12 Jan 2010 13:20:45 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
	<87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
	<cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>
	<eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>
	<cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>
Message-ID: <eafa772f1001120520i719751b9hcbc22a969a89a585@mail.gmail.com>

- prpoxy_media should be proxy_media
- bypass_media and proxy_media shouldn't need setting to false -
that's their default (unless you're set one of them to true on the sip
profile?)
- why do you need to disable q850 reason?

I do something very similar - try this...

<condition field="destination_number" expression="^(?:5824)(44(1|2)\d+)$">
  <action application="set" data="continue_on_fail=true"/>
  <action application="set" data="hangup_after_bridge=true"/>
  <action application="set" data="sip_ignore_remote_cause=true"/>
  <action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1"/>
  <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
</condition>

By the way, you'll be unable to rewrite the hangup cause for a bridge
that gets a 180 or 183 packet from the gateway before getting a 4xx,
5xx or 6xx packet (because those bridges don't 'fail') - I've seen
this on a SIP provider before who gives 183 Session Progress before a
404 Not Found if the PSTN number dialled doesn't exist.

Regards,
-Steve


2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
> Here you go:
>
> <condition field="destination_number" expression="^(?:5824)(44(1|2)\d+)$"
> break="on-true">
> ? <action inline="true" application="set" data="bypass_media=false"/>
> ? <action inline="true" application="set" data="prpoxy_media=false"/>
> ? <action inline="true" application="set" data="disable_q850_reason=true"/>
> ? <action inline="true" application="set" data="hangup_after_bridge=false"/>
> ? <action inline="true" application="set"
> data="sip_ignore_remote_cause=true"/>
> ? <action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1"/>
> ? <action inline="true" application="set"
> data="bridge_hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
> ? <action inline="true" application="set"
> data="hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
> ? <action application="bridge" data="error/normal_circuit_congestion"/>
> ? <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
> </condition>
>
> As you can see, I am trying to rewrite the hangup codes in a multitude of
> ways and places, but still exhibit the same behaviour.
>
> Any help appreciated.
>
> Regards,
>
> Ahmed.
>
> 2010/1/12 Steven Ayre <steveayre at gmail.com>
>>
>> Can you show us the dialplan extension you're trying?
>>
>> Thanks,
>> -Steve
>>
>> 2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
>> > Hi Michael,
>> >
>> > This is exactly what I'm doing, but it's just not happening.
>> >
>> > Thanks,
>> >
>> > Ahmed.
>> >
>> >
>> > 2010/1/12 Michael Collins <msc at freeswitch.org>
>> >>
>> >>
>> >> On Mon, Jan 11, 2010 at 4:18 AM, Ahmed Naji <a.alalousi at gmail.com>
>> >> wrote:
>> >>>
>> >>> Dear All,
>> >>>
>> >>> I posted a thread re the subject but didn't get any joy, so perhaps
>> >>> second time lucky.
>> >>>
>> >>> I need to rewrite a couple of hangup causes to mean NORMAL_CONGESTION
>> >>> and
>> >>> no matter which variables I set, this isn't happening. The idea is to
>> >>> control what codes are returned to an end point after a successful
>> >>> bridge,
>> >>> as well as deal with what codes are returned if the bridge is
>> >>> unsuccessful
>> >>> (e.g. user_busy, originator_cancel ...etc).
>> >>>
>> >>> I've had limited success by setting hangup_after_bridge=false then
>> >>> bridging to error/<required_code>. This, however only works when the
>> >>> B-leg
>> >>> terminates the call after a successful answer. Any other codes are not
>> >>> rewritten.
>> >>>
>> >>> I've also tried playing with the bridge_hangup_code and hangup_code
>> >>> variables prior and after bridging, still no joy. I have also set
>> >>> sip_ignore_remote_cause=true prior to entering the bridge, as well
>> >>> explicitly in vars.xml.
>> >>>
>> >>> By the way, I'm running in proxy-media mode, but I did try it with
>> >>> bypass-media as well. Same symptoms, same behaviour.
>> >>>
>> >>> Any help with this would be highly appreciated.
>> >>>
>> >> Well, I do know that when you do a hangup in the dialplan you can pass
>> >> an
>> >> optional cause as well:
>> >> <action application="hangup" data="USER_BUSY"/>
>> >> If you are doing the hanging up then you have a fair amount of
>> >> control...
>> >> -MC
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > Ahmed Naji
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From pmhshz at gmail.com  Tue Jan 12 07:22:18 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Tue, 12 Jan 2010 20:52:18 +0530
Subject: [Freeswitch-users] Defunct process in ESL testserver example
Message-ID: <a792304b1001120722o2d4fbf91vb0456ef4f0436a5e@mail.gmail.com>

Hello everybody,

I am creating a C program of ESL outbound for call processing.
I am using testserver.c example, and till now it seems fine.

But i noticed that every call testserver process, a new process is being
created which I can see in Linux system with below command:
For example, when I make two calls and even after hangup, I saw three
process like below:
ps -A | grep testserver
 9345 pts/2    00:00:00 testserver
 9350 pts/2    00:00:00 testserver
 9357 pts/2    00:00:00 testserver

This get increased for every call i make.

I did some workout and placed below two lines (close & exit) at the end of
mycallback function, (as I found them on ivrd.c file):

    esl_disconnect(&handle);
    close(client_sock);
    exit(0);
}
But after that the process becomes defunct/zombie

 9440 pts/2    00:00:00 conflisten
 9442 pts/2    00:00:00 conflisten <defunct>
 9452 pts/2    00:00:00 conflisten <defunct>


Will anybody please suggest me how can I eliminate this process, which
remains in memory even after call hangup?

Thanks for any response.
MohammedShehzad
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From brian at freeswitch.org  Tue Jan 12 07:53:03 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 09:53:03 -0600
Subject: [Freeswitch-users] proxy_media seems to be broken
In-Reply-To: <608345.82340.qm@web37502.mail.mud.yahoo.com>
References: <A2F1C0E212BBBA4E8C0E7BE568361F0403B145DC1941@srv02.khdev.corp>
	<608345.82340.qm@web37502.mail.mud.yahoo.com>
Message-ID: <194CB7A1-D382-4F98-AB9A-21AFFABFFD6B@freeswitch.org>

And the tarball is updated already automatically too.  Please update to the latest FreeSWITCH... report any issues to jira if you have them in the future.

In the future please read thru the mailing list as this was discussed in two different threads yesterday with the details and the rev where it was fixed. 

Thanks,
/b

On Jan 11, 2010, at 9:34 PM, DJB wrote:

> Yes, it has been fixed in 16250.
> 

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From mailinglist at fribert.dk  Tue Jan 12 08:04:59 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Tue, 12 Jan 2010 17:04:59 +0100
Subject: [Freeswitch-users] Multi-Homed setup,
	starting over - still not working
Message-ID: <4B4CABBB020000E10000038E@mail.fribert.dk>

Hi Guys
 
I really would like to have this up and running, but I'm constantly running into things that doesn't work, and I have no idea where the problem is...
 
I've started from scratch with pfSense and Freeswitch.
I've followed the Multi Home tutorial here: http://wiki.freeswitch.org/wiki/Multi_home_tutorial
 
I've deleted the 'lan' profile, and altered the 'internal' profile as recommended, and I've gotten to the paragraph:
APPLYING YOUR CHANGES AND CHECKING YOUR WORK
When I start up my x-lite program I get this error:
2010-01-12 16:38:54.172731 [WARNING] sofia_reg.c:1755 Can't find user [1000 at 83.89.x.x]
You must define a domain called '83.89.x.xin your directory and add a user with the id="1000" attribute
and you must configure your device to use the proper domain in it's authentication credentials.
and you must configure your device to use the proper domain in it's authentication credentials.
 
83.89.x.x is my external IP, and not my internal IP???
 
Any help on this?
 
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From anthony.minessale at gmail.com  Tue Jan 12 08:34:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Jan 2010 10:34:23 -0600
Subject: [Freeswitch-users] Defunct process in ESL testserver example
In-Reply-To: <a792304b1001120722o2d4fbf91vb0456ef4f0436a5e@mail.gmail.com>
References: <a792304b1001120722o2d4fbf91vb0456ef4f0436a5e@mail.gmail.com>
Message-ID: <191c3a031001120834q36f72791s811b6352ab7c12eb@mail.gmail.com>

It's forking process code.
you need to ignore sigchld or create a sig handler for sigchld and use the
wait syscall to reap the process.


On Tue, Jan 12, 2010 at 9:22 AM, MohammedShehzad <pmhshz at gmail.com> wrote:

> Hello everybody,
>
> I am creating a C program of ESL outbound for call processing.
> I am using testserver.c example, and till now it seems fine.
>
> But i noticed that every call testserver process, a new process is being
> created which I can see in Linux system with below command:
> For example, when I make two calls and even after hangup, I saw three
> process like below:
> ps -A | grep testserver
>  9345 pts/2    00:00:00 testserver
>  9350 pts/2    00:00:00 testserver
>  9357 pts/2    00:00:00 testserver
>
> This get increased for every call i make.
>
> I did some workout and placed below two lines (close & exit) at the end of
> mycallback function, (as I found them on ivrd.c file):
>
>     esl_disconnect(&handle);
>     close(client_sock);
>     exit(0);
> }
> But after that the process becomes defunct/zombie
>
>  9440 pts/2    00:00:00 conflisten
>  9442 pts/2    00:00:00 conflisten <defunct>
>  9452 pts/2    00:00:00 conflisten <defunct>
>
>
> Will anybody please suggest me how can I eliminate this process, which
> remains in memory even after call hangup?
>
> Thanks for any response.
> MohammedShehzad
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at van.lammeren.net  Tue Jan 12 08:50:07 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Tue, 12 Jan 2010 11:50:07 -0500
Subject: [Freeswitch-users] How to provide dynamic directory information?
In-Reply-To: <87f2f3b91001111636j1613b6ak5b5de86804f2125e@mail.gmail.com>
References: <5d2828f1001111453t8044774w3cc882988e18ced@mail.gmail.com>
	<87f2f3b91001111636j1613b6ak5b5de86804f2125e@mail.gmail.com>
Message-ID: <5d2828f1001120850qb676df6wfecbb30c05e65f23@mail.gmail.com>

Hi Michael!

I've had a look at your blog. Good stuff!

Thanks!

On Mon, Jan 11, 2010 at 7:36 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Jan 11, 2010 at 2:53 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:
>
>> Hello!
>>
>> I'd like to be able to have FreeSWITCH check a database for authorization,
>> every time a user registers. There are some great examples on the wiki,
>> which use either MOD_XML_CURL or Lua to dynamically provide a dialplan, but
>> I cannot find an example for providing a directory.
>>
>> I am developing an application that will have thousands of users, and will
>> run on multiple FreeSWITCH servers behind a load balancer. Ideally,
>> FreeSWITCH would only look-up directory information, specifically, username
>> and password, whenever a user attempts to connect. The directory information
>> will be changing regularly, as users are added or removed from the system.
>>
>> Is this possible with FreeSWITCH? Or can only dialplan information be
>> provided dynamically?
>>
>> I've written a script in Lua that provides the XML data, such as that
>> found in the example /freeswitch/conf/directory/default/ folders, and I try
>> to call it with this bit of XML in /freeswitch/conf/directory/default.xml:
>>
>> <groups>
>> <group name="default">
>> <users>
>>  <param name="xml-handler-script" value="/directory.lua"/>
>> <param name="xml-handler-bindings" value="users"/>
>>  </users>
>> </groups>
>>
>> Is this the right approach? Am I going about this the right way?
>>
>
> You can bind "directory" as well as "dialplan" and a few others. I
> personally don't use xml_curl in production but for kicks I tried to learn
> it and I documented some of my journey on my personal blog. (
> http://telecommusings.blogspot.com/)
>
> xml_curl was designed to scale and be applied in your type of scenario.
> Raymond (intralanman on IRC) has played with it quite a bit as have a number
> of others.
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jcasale at activenetwerx.com  Tue Jan 12 09:29:18 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 12 Jan 2010 17:29:18 +0000
Subject: [Freeswitch-users] Multi-Homed setup,
 starting over - still not working
In-Reply-To: <4B4CABBB020000E10000038E@mail.fribert.dk>
References: <4B4CABBB020000E10000038E@mail.fribert.dk>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E907873D@prato.activenetwerx.local>

>Hi Guys
>?
>I really would like to have this up and running, but I'm constantly running into things that doesn't work, and I have no idea where the problem is...
>?
>I've started from scratch with pfSense and Freeswitch.
>I've followed the Multi Home tutorial here: http://wiki.freeswitch.org/wiki/Multi_home_tutorial
>?
>I've deleted the 'lan' profile, and altered the 'internal' profile as recommended, and I've gotten to the paragraph:
>
>APPLYING YOUR CHANGES AND CHECKING YOUR WORK
>
>When I start up my x-lite program I get this error:
>2010-01-12 16:38:54.172731 [WARNING] sofia_reg.c:1755 Can't find user [1000 at 83.89.x.x]
>You must define a domain called '83.89.x.xin your directory and add a user with the id="1000" attribute
>and you must configure your device to use the proper domain in it's authentication credentials.
>and you must configure your device to use the proper domain in it's authentication credentials.
>?
>83.89.x.x is my external IP, and not my internal IP???
>
>Any help on this?

This is because you haven't set your domain in vars.xml. The behavior is that
$${local_ip_v4} evals to your wan ip. This is the first step in that tutorial:)
http://wiki.freeswitch.org/wiki/Multi_home_tutorial#INTERNAL_LAN

Open vars.xml, make the line:
  <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
  <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>

Match your lan ip:
  <X-PRE-PROCESS cmd="set" data="domain=192.168.13.1"/>
  <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
?
restart fs, then goto the fs_cli and type `eval ${domain}` it should
come back with "your" lan ip.


From xanlich at gmail.com  Tue Jan 12 09:08:08 2010
From: xanlich at gmail.com (Chia-Yen Wu)
Date: Wed, 13 Jan 2010 01:08:08 +0800
Subject: [Freeswitch-users] Hello , about calculate numbers
Message-ID: <314dc3f81001120908l5772b13fqd676a3be367bb662@mail.gmail.com>

Hello everyone,
Is there anyway to calculate number in dialplan.xml?

like example:
<action application="set" data="var=1"/>
and with an action , I can do : var=var+1

I have tried mod_expr, but failed to catch the variable, like below:
<action application="set" data="var=${expr($var+1)}"/>
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From sos at sokhapkin.dyndns.org  Tue Jan 12 09:39:06 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 12 Jan 2010 12:39:06 -0500
Subject: [Freeswitch-users] Hello , about calculate numbers
In-Reply-To: <314dc3f81001120908l5772b13fqd676a3be367bb662@mail.gmail.com>
References: <314dc3f81001120908l5772b13fqd676a3be367bb662@mail.gmail.com>
Message-ID: <201001121239.06756.sos@sokhapkin.dyndns.org>

<action application="set" data="var=${expr(${var}+1)}"/>

On Tuesday 12 January 2010, Chia-Yen Wu wrote:
> Hello everyone,
> Is there anyway to calculate number in dialplan.xml?
>
> like example:
> <action application="set" data="var=1"/>
> and with an action , I can do : var=var+1
>
> I have tried mod_expr, but failed to catch the variable, like below:
> <action application="set" data="var=${expr($var+1)}"/>




From jcasale at activenetwerx.com  Tue Jan 12 10:20:21 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 12 Jan 2010 18:20:21 +0000
Subject: [Freeswitch-users] Fax Codecs
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E907880A@prato.activenetwerx.local>

While faxing is working, I am seeing these in the log when I
attempt to use rxfax:

switch_ivr_play_say.c:1154 Codec Activated L16 at 8000hz 1 channels 30ms
switch_core_io.c:652 sofia/internal/1002 at 192.168.13.1 receive message [TRANSCODING_NECESSARY]

So later I see:
switch_core_codec.c:128 sofia/internal/1002 at 192.168.13.1 Restore previous codec PCMU:0

So I am actually not clear on what's happening? I thought I had the ua set to PCMU but I
guess fs is seeing the incoming sip session as L16 which I read is bad for fax? What's
the significance of the last line where it states its restoring the old codec?

Can anyone shed some light on what's actually happening?

Thanks!
jlc



From steveu at coppice.org  Tue Jan 12 10:32:34 2010
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 13 Jan 2010 02:32:34 +0800
Subject: [Freeswitch-users] Fax Codecs
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E907880A@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E907880A@prato.activenetwerx.local>
Message-ID: <4B4CC042.3090404@coppice.org>

On 01/13/2010 02:20 AM, Joseph L. Casale wrote:
> While faxing is working, I am seeing these in the log when I
> attempt to use rxfax:
>
> switch_ivr_play_say.c:1154 Codec Activated L16 at 8000hz 1 channels 30ms
> switch_core_io.c:652 sofia/internal/1002 at 192.168.13.1 receive message [TRANSCODING_NECESSARY]
>
> So later I see:
> switch_core_codec.c:128 sofia/internal/1002 at 192.168.13.1 Restore previous codec PCMU:0
>
> So I am actually not clear on what's happening? I thought I had the ua set to PCMU but I
> guess fs is seeing the incoming sip session as L16 which I read is bad for fax? What's
> the significance of the last line where it states its restoring the old codec?
>
> Can anyone shed some light on what's actually happening?
>    
A-law and u-law are the right things for the transmission of FAX 
signals, but you can't actually do signal processing on an A-law or 
u-law signal. It needs to be converted to a 16 bit linear signal (i.e. 
L16) for that. What you see is the correct action.

Steve



From jmesquita at freeswitch.org  Tue Jan 12 11:51:12 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 12 Jan 2010 17:51:12 -0200
Subject: [Freeswitch-users] FSComm Windows build
In-Reply-To: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
References: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
Message-ID: <ebbc2d551001121151v5593ee83m96aabea917ea12f5@mail.gmail.com>

Thank you very much Carlos for your support.

All FSComm testers. Please, beware that FSComm has released its project and
not the real thing! We are still on the early ages of development and don't
expect everything to work! Nonetheless, what is not working has to be
reported. :-)

Thank you,

Jo?o Mesquita


On Mon, Jan 11, 2010 at 11:22 PM, Carlos Talbot <carlos.talbot at gmail.com>wrote:

> FYI,
>
> there's a Windows pre-compiled binary of FSComm now available for those who
> want to check it.
>
> http://files.freeswitch.org/windows_installer/FSComm.exe
>
> regards,
>
> Carlos
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From john at acsol.net  Tue Jan 12 12:08:53 2010
From: john at acsol.net (John)
Date: Tue, 12 Jan 2010 13:08:53 -0700
Subject: [Freeswitch-users] ShoreTel to FS connection
Message-ID: <4B4CD6D5.1010104@acsol.net>

I am trying to utilize a freeswitch system as a SIP trunk for ShoreTel. 
I have had to change the port from 5080 to 5060. When I dial through, I 
see on the FS console show the call; however I can't seem to change the 
public.xml dialplan to effect any changes in the behaviour. How can I 
tell which dialplan the external call is using?

  2010-01-12 13:04:17.705548 [NOTICE] switch_channel.c:613 New Channel 
sofia/external/+19705551212 at 192.168.155.13:5060 
[aa2f2fde-ffb5-11de-9590-015cd8b8c273]
2010-01-12 13:04:17.705548 [INFO] mod_dialplan_xml.c:408 Processing 
Extension 1002->5551212 in context public
2010-01-12 13:04:17.705548 [NOTICE] switch_core_state_machine.c:187 
Hangup sofia/external/+19705551212 at 192.168.155.13:5060 [CS_EXECUTE] 
[NORMAL_CLEARING

Thanks John


From peder at networkoblivion.com  Tue Jan 12 12:11:36 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 12 Jan 2010 14:11:36 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
Message-ID: <064e01ca93c3$71b259a0$55170ce0$@com>

Anybody know the settings on the SPA5xx to make SCA work?  I've tried all
sorts of combinations of shared/private on the ext and phone and set the
Server type to Broadsoft, etc and  it just never seems to work.  Do I need
the BLF info?  Do I set shared/private on the phone, or ext, or both?  Do I
need to enable SCA Line ID Mapping?  Or SCA Barge-In Enable?  Or both?

Peder




From brian at freeswitch.org  Tue Jan 12 12:17:07 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 14:17:07 -0600
Subject: [Freeswitch-users] ShoreTel to FS connection
In-Reply-To: <4B4CD6D5.1010104@acsol.net>
References: <4B4CD6D5.1010104@acsol.net>
Message-ID: <38D0A853-889C-4F53-B768-53C1283C8B5C@freeswitch.org>

Its looking for 5551212 in context public.  you can ONLY communicate with shoretel on port 5060

/b

On Jan 12, 2010, at 2:08 PM, John wrote:

> I am trying to utilize a freeswitch system as a SIP trunk for ShoreTel. 
> I have had to change the port from 5080 to 5060. When I dial through, I 
> see on the FS console show the call; however I can't seem to change the 
> public.xml dialplan to effect any changes in the behaviour. How can I 
> tell which dialplan the external call is using?
> 
>  2010-01-12 13:04:17.705548 [NOTICE] switch_channel.c:613 New Channel 
> sofia/external/+19705551212 at 192.168.155.13:5060 
> [aa2f2fde-ffb5-11de-9590-015cd8b8c273]
> 2010-01-12 13:04:17.705548 [INFO] mod_dialplan_xml.c:408 Processing 
> Extension 1002->5551212 in context public
> 2010-01-12 13:04:17.705548 [NOTICE] switch_core_state_machine.c:187 
> Hangup sofia/external/+19705551212 at 192.168.155.13:5060 [CS_EXECUTE] 
> [NORMAL_CLEARING
> 
> Thanks John
> 
> _____________



From brian at freeswitch.org  Tue Jan 12 12:19:13 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 14:19:13 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <064e01ca93c3$71b259a0$55170ce0$@com>
References: <064e01ca93c3$71b259a0$55170ce0$@com>
Message-ID: <DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>

Enable line.
Set to share.

On phone tab.. mark it shared and select the line.

Attendant console set broadsoft.

save.

reboot
try

/b






On Jan 12, 2010, at 2:11 PM, Peder wrote:

> Anybody know the settings on the SPA5xx to make SCA work?  I've tried all
> sorts of combinations of shared/private on the ext and phone and set the
> Server type to Broadsoft, etc and  it just never seems to work.  Do I need
> the BLF info?  Do I set shared/private on the phone, or ext, or both?  Do I
> need to enable SCA Line ID Mapping?  Or SCA Barge-In Enable?  Or both?
> 
> Peder



From peder at networkoblivion.com  Tue Jan 12 12:32:49 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 12 Jan 2010 14:32:49 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>
References: <064e01ca93c3$71b259a0$55170ce0$@com>
	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>
Message-ID: <066301ca93c6$6882a300$3987e900$@com>

We can answer and put it on hold and pickup from the other phone, but we
can't barge in.  Should that work?  It doesn't seem to.  Also, private hold
doesn't seem to work.  Whether I use private hold or regular hold, the other
phone can pickup the line.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, January 12, 2010 2:19 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cisco SPA5xx SCA

Enable line.
Set to share.

On phone tab.. mark it shared and select the line.

Attendant console set broadsoft.

save.

reboot
try

/b






On Jan 12, 2010, at 2:11 PM, Peder wrote:

> Anybody know the settings on the SPA5xx to make SCA work?  I've tried all
> sorts of combinations of shared/private on the ext and phone and set the
> Server type to Broadsoft, etc and  it just never seems to work.  Do I need
> the BLF info?  Do I set shared/private on the phone, or ext, or both?  Do
I
> need to enable SCA Line ID Mapping?  Or SCA Barge-In Enable?  Or both?
> 
> Peder


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From mike at van.lammeren.net  Tue Jan 12 12:33:08 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Tue, 12 Jan 2010 15:33:08 -0500
Subject: [Freeswitch-users] How to provide dynamic directory information?
In-Reply-To: <5d2828f1001120850qb676df6wfecbb30c05e65f23@mail.gmail.com>
References: <5d2828f1001111453t8044774w3cc882988e18ced@mail.gmail.com>
	<87f2f3b91001111636j1613b6ak5b5de86804f2125e@mail.gmail.com>
	<5d2828f1001120850qb676df6wfecbb30c05e65f23@mail.gmail.com>
Message-ID: <5d2828f1001121233h7a374f46hbf6964f2e1887b95@mail.gmail.com>

Hello!

I am now successfully pulling directory information from the database with a
Lua script.

I based my work on this section of the wiki:
http://wiki.freeswitch.org/wiki/Mod_lua#For_serving_configuration

There were a few odd things in the wiki that initially threw me for a loop,
and I have since improved the wiki entry linked above. I hope that it will
be easier for the next person!

There is definitely a learning curve to FreeSWITCH, but in the end,
FreeSWITCH always does the trick! Keep with it, and you will be rewarded!

Mike van Lammeren
<http://wiki.freeswitch.org/wiki/Mod_lua#For_serving_configuration>

On Tue, Jan 12, 2010 at 11:50 AM, Mike van Lammeren
<mike at van.lammeren.net>wrote:

> Hi Michael!
>
> I've had a look at your blog. Good stuff!
>
> Thanks!
>
> On Mon, Jan 11, 2010 at 7:36 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Mon, Jan 11, 2010 at 2:53 PM, Mike van Lammeren <mike at van.lammeren.net
>> > wrote:
>>
>>> Hello!
>>>
>>> I'd like to be able to have FreeSWITCH check a database for
>>> authorization, every time a user registers. There are some great examples on
>>> the wiki, which use either MOD_XML_CURL or Lua to dynamically provide a
>>> dialplan, but I cannot find an example for providing a directory.
>>>
>>> I am developing an application that will have thousands of users, and
>>> will run on multiple FreeSWITCH servers behind a load balancer. Ideally,
>>> FreeSWITCH would only look-up directory information, specifically, username
>>> and password, whenever a user attempts to connect. The directory information
>>> will be changing regularly, as users are added or removed from the system.
>>>
>>> Is this possible with FreeSWITCH? Or can only dialplan information be
>>> provided dynamically?
>>>
>>> I've written a script in Lua that provides the XML data, such as that
>>> found in the example /freeswitch/conf/directory/default/ folders, and I try
>>> to call it with this bit of XML in /freeswitch/conf/directory/default.xml:
>>>
>>> <groups>
>>> <group name="default">
>>> <users>
>>>  <param name="xml-handler-script" value="/directory.lua"/>
>>> <param name="xml-handler-bindings" value="users"/>
>>>  </users>
>>> </groups>
>>>
>>> Is this the right approach? Am I going about this the right way?
>>>
>>
>> You can bind "directory" as well as "dialplan" and a few others. I
>> personally don't use xml_curl in production but for kicks I tried to learn
>> it and I documented some of my journey on my personal blog. (
>> http://telecommusings.blogspot.com/)
>>
>> xml_curl was designed to scale and be applied in your type of scenario.
>> Raymond (intralanman on IRC) has played with it quite a bit as have a number
>> of others.
>> -MC
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From brian at freeswitch.org  Tue Jan 12 12:40:58 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 14:40:58 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <066301ca93c6$6882a300$3987e900$@com>
References: <064e01ca93c3$71b259a0$55170ce0$@com>
	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>
	<066301ca93c6$6882a300$3987e900$@com>
Message-ID: <5DE42506-5400-4B40-A522-D1FE5C5AC1EB@freeswitch.org>

Those phones do not support barge in, If they do I couldn't find it.  As for private hold how did you mark it private hold?

/b

On Jan 12, 2010, at 2:32 PM, Peder wrote:

> We can answer and put it on hold and pickup from the other phone, but we
> can't barge in.  Should that work?  It doesn't seem to.  Also, private hold
> doesn't seem to work.  Whether I use private hold or regular hold, the other
> phone can pickup the line.



From peder at networkoblivion.com  Tue Jan 12 12:50:40 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 12 Jan 2010 14:50:40 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <5DE42506-5400-4B40-A522-D1FE5C5AC1EB@freeswitch.org>
References: <064e01ca93c3$71b259a0$55170ce0$@com>	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>	<066301ca93c6$6882a300$3987e900$@com>
	<5DE42506-5400-4B40-A522-D1FE5C5AC1EB@freeswitch.org>
Message-ID: <066c01ca93c8$e67c2b80$b3748280$@com>

Barge:
Under Phone there is an "SCA Barge-In Enable:" line that defaults to no.  I
set it to yes, but when I hit it, I got "unauthorized".

Private Hold:
The soft keys say "PrvHld" when you are on a call.  It functioned the same
as the regular hold button.  From reading the Polycom SCA docs, it says that
private hold should block anybody else from grabbing it.



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, January 12, 2010 2:41 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cisco SPA5xx SCA

Those phones do not support barge in, If they do I couldn't find it.  As for
private hold how did you mark it private hold?

/b

On Jan 12, 2010, at 2:32 PM, Peder wrote:

> We can answer and put it on hold and pickup from the other phone, but we
> can't barge in.  Should that work?  It doesn't seem to.  Also, private
hold
> doesn't seem to work.  Whether I use private hold or regular hold, the
other
> phone can pickup the line.


_______________________________________________
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From brian at freeswitch.org  Tue Jan 12 12:55:41 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 14:55:41 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <066c01ca93c8$e67c2b80$b3748280$@com>
References: <064e01ca93c3$71b259a0$55170ce0$@com>	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>	<066301ca93c6$6882a300$3987e900$@com>
	<5DE42506-5400-4B40-A522-D1FE5C5AC1EB@freeswitch.org>
	<066c01ca93c8$e67c2b80$b3748280$@com>
Message-ID: <D0EA8A36-6AA9-41C6-A9AF-3F374D24902A@freeswitch.org>

Can you post me a sip trace of this.. I only have a 501G without a display.  So I can't see this.

/b

On Jan 12, 2010, at 2:50 PM, Peder wrote:

> Barge:
> Under Phone there is an "SCA Barge-In Enable:" line that defaults to no.  I
> set it to yes, but when I hit it, I got "unauthorized".
> 
> Private Hold:
> The soft keys say "PrvHld" when you are on a call.  It functioned the same
> as the regular hold button.  From reading the Polycom SCA docs, it says that
> private hold should block anybody else from grabbing it.



From peder at networkoblivion.com  Tue Jan 12 13:16:13 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 12 Jan 2010 15:16:13 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <D0EA8A36-6AA9-41C6-A9AF-3F374D24902A@freeswitch.org>
References: <064e01ca93c3$71b259a0$55170ce0$@com>	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>	<066301ca93c6$6882a300$3987e900$@com>	<5DE42506-5400-4B40-A522-D1FE5C5AC1EB@freeswitch.org>	<066c01ca93c8$e67c2b80$b3748280$@com>
	<D0EA8A36-6AA9-41C6-A9AF-3F374D24902A@freeswitch.org>
Message-ID: <067201ca93cc$789d8100$69d88300$@com>

Will do.  Interesting note.  The Linksys SPA922 sort of supports SCA.  You
can do the shared line reg and it looks ok.  The Linksys can answer and then
put on hold and the other phone can pick it up.  The only issue is that the
Linksys can't pickup a call that the other phone puts on hold.  

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, January 12, 2010 2:56 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cisco SPA5xx SCA

Can you post me a sip trace of this.. I only have a 501G without a display.
So I can't see this.

/b

On Jan 12, 2010, at 2:50 PM, Peder wrote:

> Barge:
> Under Phone there is an "SCA Barge-In Enable:" line that defaults to no.
I
> set it to yes, but when I hit it, I got "unauthorized".
> 
> Private Hold:
> The soft keys say "PrvHld" when you are on a call.  It functioned the same
> as the regular hold button.  From reading the Polycom SCA docs, it says
that
> private hold should block anybody else from grabbing it.


_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Tue Jan 12 13:19:56 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 15:19:56 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <067201ca93cc$789d8100$69d88300$@com>
References: <064e01ca93c3$71b259a0$55170ce0$@com>	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>	<066301ca93c6$6882a300$3987e900$@com>	<5DE42506-5400-4B40-A522-D1FE5C5AC1EB@freeswitch.org>	<066c01ca93c8$e67c2b80$b3748280$@com>
	<D0EA8A36-6AA9-41C6-A9AF-3F374D24902A@freeswitch.org>
	<067201ca93cc$789d8100$69d88300$@com>
Message-ID: <5579C5DD-ED93-445A-AF56-61A9D3A3705B@freeswitch.org>

Yah I never said it would work with those :P

/b

On Jan 12, 2010, at 3:16 PM, Peder wrote:

> Will do.  Interesting note.  The Linksys SPA922 sort of supports SCA.  You
> can do the shared line reg and it looks ok.  The Linksys can answer and then
> put on hold and the other phone can pick it up.  The only issue is that the
> Linksys can't pickup a call that the other phone puts on hold.  



From anthony.minessale at gmail.com  Tue Jan 12 13:20:08 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Jan 2010 15:20:08 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <066301ca93c6$6882a300$3987e900$@com>
References: <064e01ca93c3$71b259a0$55170ce0$@com>
	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>
	<066301ca93c6$6882a300$3987e900$@com>
Message-ID: <191c3a031001121320s1840280aq2f99385b4b6102f7@mail.gmail.com>

Maybe you want to let us on your box to work on it or buy us your phones to
have in our lab.
Our phones did not support those options.





On Tue, Jan 12, 2010 at 2:32 PM, Peder <peder at networkoblivion.com> wrote:

> We can answer and put it on hold and pickup from the other phone, but we
> can't barge in.  Should that work?  It doesn't seem to.  Also, private hold
> doesn't seem to work.  Whether I use private hold or regular hold, the
> other
> phone can pickup the line.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
> Sent: Tuesday, January 12, 2010 2:19 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Cisco SPA5xx SCA
>
> Enable line.
> Set to share.
>
> On phone tab.. mark it shared and select the line.
>
> Attendant console set broadsoft.
>
> save.
>
> reboot
> try
>
> /b
>
>
>
>
>
>
> On Jan 12, 2010, at 2:11 PM, Peder wrote:
>
> > Anybody know the settings on the SPA5xx to make SCA work?  I've tried all
> > sorts of combinations of shared/private on the ext and phone and set the
> > Server type to Broadsoft, etc and  it just never seems to work.  Do I
> need
> > the BLF info?  Do I set shared/private on the phone, or ext, or both?  Do
> I
> > need to enable SCA Line ID Mapping?  Or SCA Barge-In Enable?  Or both?
> >
> > Peder
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From msc at freeswitch.org  Tue Jan 12 13:30:05 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Jan 2010 13:30:05 -0800
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <067201ca93cc$789d8100$69d88300$@com>
References: <064e01ca93c3$71b259a0$55170ce0$@com>
	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>
	<066301ca93c6$6882a300$3987e900$@com>
	<5DE42506-5400-4B40-A522-D1FE5C5AC1EB@freeswitch.org>
	<066c01ca93c8$e67c2b80$b3748280$@com>
	<D0EA8A36-6AA9-41C6-A9AF-3F374D24902A@freeswitch.org>
	<067201ca93cc$789d8100$69d88300$@com>
Message-ID: <87f2f3b91001121330t55d81787k3c7030e429a606b6@mail.gmail.com>

Peder,

Thanks for trying all this out in the real world. When you get it working
please let me know so that we can get proper documentation for these devices
in the wiki. I'd like to see the documentation for specific devices written
by people who are physically using them.

Thanks!
-MC

On Tue, Jan 12, 2010 at 1:16 PM, Peder <peder at networkoblivion.com> wrote:

> Will do.  Interesting note.  The Linksys SPA922 sort of supports SCA.  You
> can do the shared line reg and it looks ok.  The Linksys can answer and
> then
> put on hold and the other phone can pick it up.  The only issue is that the
> Linksys can't pickup a call that the other phone puts on hold.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
> Sent: Tuesday, January 12, 2010 2:56 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Cisco SPA5xx SCA
>
> Can you post me a sip trace of this.. I only have a 501G without a display.
> So I can't see this.
>
> /b
>
> On Jan 12, 2010, at 2:50 PM, Peder wrote:
>
> > Barge:
> > Under Phone there is an "SCA Barge-In Enable:" line that defaults to no.
> I
> > set it to yes, but when I hit it, I got "unauthorized".
> >
> > Private Hold:
> > The soft keys say "PrvHld" when you are on a call.  It functioned the
> same
> > as the regular hold button.  From reading the Polycom SCA docs, it says
> that
> > private hold should block anybody else from grabbing it.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Jan 12 13:34:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Jan 2010 15:34:49 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <067201ca93cc$789d8100$69d88300$@com>
References: <064e01ca93c3$71b259a0$55170ce0$@com>
	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>
	<066301ca93c6$6882a300$3987e900$@com>
	<5DE42506-5400-4B40-A522-D1FE5C5AC1EB@freeswitch.org>
	<066c01ca93c8$e67c2b80$b3748280$@com>
	<D0EA8A36-6AA9-41C6-A9AF-3F374D24902A@freeswitch.org>
	<067201ca93cc$789d8100$69d88300$@com>
Message-ID: <191c3a031001121334j6a0725aq74d64ca5cc1ea363@mail.gmail.com>

an isolated trace of that pickup not working with debug log and debug_sla=10
set in sofia.conf.xml under settings would be nice.


On Tue, Jan 12, 2010 at 3:16 PM, Peder <peder at networkoblivion.com> wrote:

> Will do.  Interesting note.  The Linksys SPA922 sort of supports SCA.  You
> can do the shared line reg and it looks ok.  The Linksys can answer and
> then
> put on hold and the other phone can pick it up.  The only issue is that the
> Linksys can't pickup a call that the other phone puts on hold.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
> Sent: Tuesday, January 12, 2010 2:56 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Cisco SPA5xx SCA
>
> Can you post me a sip trace of this.. I only have a 501G without a display.
> So I can't see this.
>
> /b
>
> On Jan 12, 2010, at 2:50 PM, Peder wrote:
>
> > Barge:
> > Under Phone there is an "SCA Barge-In Enable:" line that defaults to no.
> I
> > set it to yes, but when I hit it, I got "unauthorized".
> >
> > Private Hold:
> > The soft keys say "PrvHld" when you are on a call.  It functioned the
> same
> > as the regular hold button.  From reading the Polycom SCA docs, it says
> that
> > private hold should block anybody else from grabbing it.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jerry.richards at teotech.com  Tue Jan 12 13:47:23 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 12 Jan 2010 13:47:23 -0800
Subject: [Freeswitch-users] PRI Goes Down Upon Calling Cell Phone
Message-ID: <24C642A4E3CB4C9F9865199154120E72@greyhawk.tonecommander.com>

 
I am having a problem where my PRI goes down after attempting to call a cell
phone from an internal phone.  I am running Freeswitch
(freeswitch-1.0.5-20100112-0400.tar.gz) with an A101D Sangoma Card and
wanpipe driver: wanpipe-3.5.10.tgz.  I am not using libpri.
 
The scenario is as follows:
 
1) Call desktop phone 3491006
2) Call completes normally
3) Call cell phone 4181432 (see http://pastebin.freeswitch.org/11767)
    ***** call fails; cell phone rings, but does not complete *****
4) Call desktop phone 3491006
    ***** call fails; far-end does not ring *****
5) Any subsequent calls through PRI fail
   
I notice the call to my cell phone receives a Q.931 PROGRESS message instead
of Q.931 ALERTING.  Has this issue already been identified as a bug?  Do you
know if this is a FS issue?  Or Wanpipe driver issue?
 
Best Regards,
Jerry



From msc at freeswitch.org  Tue Jan 12 14:09:38 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Jan 2010 14:09:38 -0800
Subject: [Freeswitch-users] PRI Goes Down Upon Calling Cell Phone
In-Reply-To: <24C642A4E3CB4C9F9865199154120E72@greyhawk.tonecommander.com>
References: <24C642A4E3CB4C9F9865199154120E72@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91001121409i141ac5fco8a2c50fed523dce1@mail.gmail.com>

Can you get a Q931 trace on this and put it up where we can download it? The
procedure is documented on Sangoma's website; see openzap wiki page under
debugging sangoma boost.

Thanks,
MC

On Tue, Jan 12, 2010 at 1:47 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> I am having a problem where my PRI goes down after attempting to call a
> cell
> phone from an internal phone.  I am running Freeswitch
> (freeswitch-1.0.5-20100112-0400.tar.gz) with an A101D Sangoma Card and
> wanpipe driver: wanpipe-3.5.10.tgz.  I am not using libpri.
>
> The scenario is as follows:
>
> 1) Call desktop phone 3491006
> 2) Call completes normally
> 3) Call cell phone 4181432 (see http://pastebin.freeswitch.org/11767)
>    ***** call fails; cell phone rings, but does not complete *****
> 4) Call desktop phone 3491006
>    ***** call fails; far-end does not ring *****
> 5) Any subsequent calls through PRI fail
>
> I notice the call to my cell phone receives a Q.931 PROGRESS message
> instead
> of Q.931 ALERTING.  Has this issue already been identified as a bug?  Do
> you
> know if this is a FS issue?  Or Wanpipe driver issue?
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jerry.richards at teotech.com  Tue Jan 12 15:10:25 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 12 Jan 2010 15:10:25 -0800
Subject: [Freeswitch-users] Inbound DTMF Not Recognized In Latest Version
Message-ID: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com>


I am having a problem where in inbound call from the PSTN going to the demo
IVR does not recognize DTMF digits.  I posted a log at
http://pastebin.freeswitch.org/11769.  I am running Freeswitch
(freeswitch-1.0.5-20100112-0400.tar.gz) with an A101D Sangoma Card and
wanpipe driver: wanpipe-3.5.10.tgz.  I am not using libpri.

It appears that each DTMF digit might be getting detected twice.  I tried to
dial "5401" and it thought I dialed "5544".

I am following the instructions at
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf, as shown
below.  Also, I tried moving the "start_dtmf" statement in between all the
statements within the condition tag, but it didn't make any difference.

<extension name="ivr_demo">
   <condition field="destination_number" expression="5000">
      <action application="answer"/>
      <action application="start_dtmf" />
      <action application="sleep" data="2000"/>
      <action application="ivr" data="demo_ivr"/>
   </condition>
</extension>

Also, I noticed at line #293, the log is saying:

[ERR] mod_sndfile.c:194 Error Opening File
[/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System
error : No such file or directory.]

Has anyone else encountered this issue?

Best Regards,
Jerry



From lart2150 at gmail.com  Tue Jan 12 15:08:27 2010
From: lart2150 at gmail.com (Brian Engert)
Date: Tue, 12 Jan 2010 17:08:27 -0600
Subject: [Freeswitch-users] Channel Variables with spaces
Message-ID: <a88e84c21001121508s191b89bdr6d67079cba62be34@mail.gmail.com>

I'm trying to set a channel variable with a space in it when I run the
originate command.  The command only seems to work when I escape the
space with a \ however that backspace get's passed on.  Some examples
are

originate {fax_ident=1231231234,fax_header=testing\
spaces}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)
originate {fax_ident=1231231234,fax_header=testing
spaces}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)

The first one will go through just fine but the fax at the other end
gets "testing\ spaces" the second one gives me
"2010-01-12 16:35:18.41325 [ERR] switch_ivr_originate.c:990 Parse Error!"

I think this could also be done as a profile but I would rather do it
from the call.  I've looked around the wiki and have not seen any
examples that use a space when doing the originate call.

- Brian


From brian at freeswitch.org  Tue Jan 12 15:25:58 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 17:25:58 -0600
Subject: [Freeswitch-users] Inbound DTMF Not Recognized In Latest Version
In-Reply-To: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com>
References: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com>
Message-ID: <44468DAE-9201-40AF-ACA3-E634E6682F04@freeswitch.org>

looks like you don't have the latest sound files installed.

/b

On Jan 12, 2010, at 5:10 PM, Jerry Richards wrote:

> [ERR] mod_sndfile.c:194 Error Opening File
> [/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System
> error : No such file or directory.]



From mike at jerris.com  Tue Jan 12 15:35:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 12 Jan 2010 18:35:20 -0500
Subject: [Freeswitch-users] Hello , about calculate numbers
In-Reply-To: <201001121239.06756.sos@sokhapkin.dyndns.org>
References: <314dc3f81001120908l5772b13fqd676a3be367bb662@mail.gmail.com>
	<201001121239.06756.sos@sokhapkin.dyndns.org>
Message-ID: <EDF668FE-A76D-4A5F-96D1-756A7ACD13A6@jerris.com>

Remember you can't do conditions on these vars being set unless you transfer back into the dialplan as these actions are not run immediately, but rather after dialplan parse, unless you use http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions

Mie

On Jan 12, 2010, at 12:39 PM, Sergey Okhapkin wrote:

> <action application="set" data="var=${expr(${var}+1)}"/>
> 
> On Tuesday 12 January 2010, Chia-Yen Wu wrote:
>> Hello everyone,
>> Is there anyway to calculate number in dialplan.xml?
>> 
>> like example:
>> <action application="set" data="var=1"/>
>> and with an action , I can do : var=var+1
>> 
>> I have tried mod_expr, but failed to catch the variable, like below:
>> <action application="set" data="var=${expr($var+1)}"/>
> 



From anthony.minessale at gmail.com  Tue Jan 12 15:35:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Jan 2010 17:35:38 -0600
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0F362D9@BESONE.corp.dom>
References: <A2167AE7633A5C4DAE56EEBD1428ECC0F362D9@BESONE.corp.dom>
Message-ID: <191c3a031001121535t4819ee44p74cc9f7cd4fe972f@mail.gmail.com>

On Tue, Jan 12, 2010 at 3:43 AM, Cavalera Claudio Luigi <
Claudio.Cavalera at italtel.it> wrote:

> Hello,
> this should be simple in theory therefore I'm probably missing the right
> way to do it.
> I want to play with sessions in lua, bridge them, park them, etc...
>
> example1: Consider this simple lua script in which i create two
> sessions:
>
> api = freeswitch.API();
> api:execute("originate", "sofia/internal/1001%192.168.1.1 &park");
> api:execute("originate", "sofia/internal/1002%192.168.1.1 &park");
>
>
capture the output from api:execute the uuid is in there



> now if i want to bridge them i suppose i should use something like
>
> api:execute("uuid_bridge", "uuid_1 uuid_2");
>
> but how do i get uuid_1 and uuid_2, i.e. the uuids of the two sessions?
>
>
> example2: I could create sessions with
>
> local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
> local session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");
>
> but then there is NOT a bridge API to bridge the sessions like:
> bridge(session1, session2);
>
> I admit I have not yet understood why such bridge possibility exist in
> javascript but does not exist in lua.
> http://wiki.freeswitch.org/wiki/Javascript_Misc_bridge
> I guess there is a reason for this but I can't figure it out.
>
>
>
because lua calls it freeswitch.bridge

session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1", session1);
freeswitch.bridge(session1, session2);





> example3: yet another possibility
> local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
> session1:originate(session2, "sofia/internal/1002%192.168.1.1", 1000);
>
> but it does not work either.
>
>
The above is gibberish try:

local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
session1:execute("bridge", "sofia/internal/1002%192.168.1.1");




> Besides with this third example something strange happen:
>
> freeswitch at internal> show channels
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr
> ,dest,application,application_data,dialplan,context,read_codec,read_rate
> ,write_codec,write_rate,secure
> 1c5db2df-14ce-4516-94f2-bb7c087e0802,outbound,2010-01-12
> 10:50:59,1263289859,sofia/internal/1004,CS_REPORTING,FreeSWITCH,00000000
> 00,,1004,,,,default,PCMA,8000,PCMA,8000,
>
> 1 total.
>
> freeswitch at internal> uuid_kill 1c5db2df-14ce-4516-94f2-bb7c087e0802
> -ERR No Such Channel!
>
> freeswitch at internal> show channels
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr
> ,dest,application,application_data,dialplan,context,read_codec,read_rate
> ,write_codec,write_rate,secure
> 1c5db2df-14ce-4516-94f2-bb7c087e0802,outbound,2010-01-12
> 10:50:59,1263289859,sofia/internal/1004,CS_REPORTING,FreeSWITCH,00000000
> 00,,1004,,,,default,PCMA,8000,PCMA,8000,
>
> 1 total.
>
> freeswitch at internal>
>
> If you are interested the full log is here:
> http://pastebin.freeswitch.org/11757
> but I admit i'm not on latest trunk yet!
>
> Thanks.
> Ciao,
> Claudio
>
>
> PS: Is there a reason why there is a uuid_park command but not
> uuid_valet_park ?
>
>
> Internet Email Confidentiality Footer
>
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>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From brian at freeswitch.org  Tue Jan 12 15:37:42 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 17:37:42 -0600
Subject: [Freeswitch-users] Channel Variables with spaces
In-Reply-To: <a88e84c21001121508s191b89bdr6d67079cba62be34@mail.gmail.com>
References: <a88e84c21001121508s191b89bdr6d67079cba62be34@mail.gmail.com>
Message-ID: <0F330D84-2C15-47D3-986B-704370A6F745@freeswitch.org>

Have you tried to properly quote them?

/b

On Jan 12, 2010, at 5:08 PM, Brian Engert wrote:

> The first one will go through just fine but the fax at the other end
> gets "testing\ spaces" the second one gives me
> "2010-01-12 16:35:18.41325 [ERR] switch_ivr_originate.c:990 Parse Error!"



From msc at freeswitch.org  Tue Jan 12 15:40:43 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Jan 2010 15:40:43 -0800
Subject: [Freeswitch-users] Channel Variables with spaces
In-Reply-To: <a88e84c21001121508s191b89bdr6d67079cba62be34@mail.gmail.com>
References: <a88e84c21001121508s191b89bdr6d67079cba62be34@mail.gmail.com>
Message-ID: <87f2f3b91001121540j70367d55n1aa02aed36faccbf@mail.gmail.com>

On Tue, Jan 12, 2010 at 3:08 PM, Brian Engert <lart2150 at gmail.com> wrote:

> I'm trying to set a channel variable with a space in it when I run the
> originate command.  The command only seems to work when I escape the
> space with a \ however that backspace get's passed on.  Some examples
> are
>
> originate {fax_ident=1231231234,fax_header=testing\
> spaces}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)
> originate {fax_ident=1231231234,fax_header=testing
> spaces}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)
>
> The first one will go through just fine but the fax at the other end
> gets "testing\ spaces" the second one gives me
> "2010-01-12 16:35:18.41325 [ERR] switch_ivr_originate.c:990 Parse Error!"
>
> I think this could also be done as a profile but I would rather do it
> from the call.  I've looked around the wiki and have not seen any
> examples that use a space when doing the originate call.
>
> I'll give you the answer if you promise to put it into the wiki. ;)

Okay, use single quotes like this:

originate {fax_ident=1231231234,fax_header='testing
spaces'}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)

-MC
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From lart2150 at gmail.com  Tue Jan 12 16:18:14 2010
From: lart2150 at gmail.com (Brian Engert)
Date: Tue, 12 Jan 2010 18:18:14 -0600
Subject: [Freeswitch-users] Channel Variables with spaces
In-Reply-To: <87f2f3b91001121540j70367d55n1aa02aed36faccbf@mail.gmail.com>
References: <a88e84c21001121508s191b89bdr6d67079cba62be34@mail.gmail.com>
	<87f2f3b91001121540j70367d55n1aa02aed36faccbf@mail.gmail.com>
Message-ID: <a88e84c21001121618l59c196f0qe9655c7057d57f64@mail.gmail.com>

I'll try that tomorrow and update the wiki!  is it only single quotes
or should double quotes work as well?



On Tue, Jan 12, 2010 at 5:40 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Tue, Jan 12, 2010 at 3:08 PM, Brian Engert <lart2150 at gmail.com> wrote:
>>
>> I'm trying to set a channel variable with a space in it when I run the
>> originate command. ?The command only seems to work when I escape the
>> space with a \ however that backspace get's passed on. ?Some examples
>> are
>>
>> originate {fax_ident=1231231234,fax_header=testing\
>> spaces}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)
>> originate {fax_ident=1231231234,fax_header=testing
>> spaces}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)
>>
>> The first one will go through just fine but the fax at the other end
>> gets "testing\ spaces" the second one gives me
>> "2010-01-12 16:35:18.41325 [ERR] switch_ivr_originate.c:990 Parse Error!"
>>
>> I think this could also be done as a profile but I would rather do it
>> from the call. ?I've looked around the wiki and have not seen any
>> examples that use a space when doing the originate call.
>>
> I'll give you the answer if you promise to put it into the wiki. ;)
>
> Okay, use single quotes like this:
>
> originate {fax_ident=1231231234,fax_header='testing
> spaces'}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From john at acsol.net  Tue Jan 12 16:20:25 2010
From: john at acsol.net (John)
Date: Tue, 12 Jan 2010 17:20:25 -0700
Subject: [Freeswitch-users] ShoreTel to FS connection
In-Reply-To: <38D0A853-889C-4F53-B768-53C1283C8B5C@freeswitch.org>
References: <4B4CD6D5.1010104@acsol.net>
	<38D0A853-889C-4F53-B768-53C1283C8B5C@freeswitch.org>
Message-ID: <4B4D11C9.1060007@acsol.net>

5551212 is the number that I am dialing with the ShoreTel system. It is 
passing it to FS. Some how I need it to pass that number to the gateway 
defined on the FS.



On 1/12/2010 1:17 PM, Brian West wrote:
> Its looking for 5551212 in context public.  you can ONLY communicate with shoretel on port 5060
>
> /b
>
> On Jan 12, 2010, at 2:08 PM, John wrote:
>
>    
>> I am trying to utilize a freeswitch system as a SIP trunk for ShoreTel.
>> I have had to change the port from 5080 to 5060. When I dial through, I
>> see on the FS console show the call; however I can't seem to change the
>> public.xml dialplan to effect any changes in the behaviour. How can I
>> tell which dialplan the external call is using?
>>
>>   2010-01-12 13:04:17.705548 [NOTICE] switch_channel.c:613 New Channel
>> sofia/external/+19705551212 at 192.168.155.13:5060
>> [aa2f2fde-ffb5-11de-9590-015cd8b8c273]
>> 2010-01-12 13:04:17.705548 [INFO] mod_dialplan_xml.c:408 Processing
>> Extension 1002->5551212 in context public
>> 2010-01-12 13:04:17.705548 [NOTICE] switch_core_state_machine.c:187
>> Hangup sofia/external/+19705551212 at 192.168.155.13:5060 [CS_EXECUTE]
>> [NORMAL_CLEARING
>>
>> Thanks John
>>
>> _____________
>>      
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    



From brian at freeswitch.org  Tue Jan 12 16:25:34 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 18:25:34 -0600
Subject: [Freeswitch-users] Channel Variables with spaces
In-Reply-To: <a88e84c21001121618l59c196f0qe9655c7057d57f64@mail.gmail.com>
References: <a88e84c21001121508s191b89bdr6d67079cba62be34@mail.gmail.com>
	<87f2f3b91001121540j70367d55n1aa02aed36faccbf@mail.gmail.com>
	<a88e84c21001121618l59c196f0qe9655c7057d57f64@mail.gmail.com>
Message-ID: <77CA31CE-3238-45E7-826E-04BA9E7D8357@freeswitch.org>

Single
On Jan 12, 2010, at 6:18 PM, Brian Engert wrote:

> I'll try that tomorrow and update the wiki!  is it only single quotes
> or should double quotes work as well?



From brian at freeswitch.org  Tue Jan 12 16:26:11 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 18:26:11 -0600
Subject: [Freeswitch-users] ShoreTel to FS connection
In-Reply-To: <4B4D11C9.1060007@acsol.net>
References: <4B4CD6D5.1010104@acsol.net>
	<38D0A853-889C-4F53-B768-53C1283C8B5C@freeswitch.org>
	<4B4D11C9.1060007@acsol.net>
Message-ID: <4F42AEBC-7F10-4D33-B79A-F560F598637E@freeswitch.org>

The put a dialplan entry in public context to catch and redirect the number elsewhere or use bridge to send it out a gateway.

/b

On Jan 12, 2010, at 6:20 PM, John wrote:

> 5551212 is the number that I am dialing with the ShoreTel system. It is 
> passing it to FS. Some how I need it to pass that number to the gateway 
> defined on the FS.



From jeff at jefflenk.com  Tue Jan 12 18:35:25 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 12 Jan 2010 18:35:25 -0800 (PST)
Subject: [Freeswitch-users] FSComm Windows build
In-Reply-To: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
References: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
Message-ID: <1263350125191-4295967.post@n2.nabble.com>


Very Cool!

Just curious Carlos do you have any experience with x64 with regard to QT on
Win?

Thanks
Jeff


Carlos Talbot wrote:
> 
> FYI,
> 
> there's a Windows pre-compiled binary of FSComm now available for those
> who
> want to check it.
> 
> http://files.freeswitch.org/windows_installer/FSComm.exe
> 
> regards,
> 
> Carlos
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/FSComm-Windows-build-tp4289256p4295967.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From carlos.talbot at gmail.com  Tue Jan 12 19:18:26 2010
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Tue, 12 Jan 2010 21:18:26 -0600
Subject: [Freeswitch-users] FSComm Windows build
In-Reply-To: <1263350125191-4295967.post@n2.nabble.com>
References: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
	<1263350125191-4295967.post@n2.nabble.com>
Message-ID: <5800526b1001121918r15a336f5jbc5b724e66ac2825@mail.gmail.com>

Jeff,

not yet. I'm still linking with a 32bit QT library of 4.5.2 I compiled for
VS2008 sometime last year. I followed a guide similar to this one:
http://dcsoft.com/community_server/blogs/dcsoft/archive/2009/03/06/how-to-setup-qt-4-5-visual-studio-integration.aspx

At the time I didn't see a need for a 64bit version of FsGui. Since FSComm
has FreeSWITCH in the back end this kind of changes things.

Looks like someone put up a wiki on compiling QT for Win x64:
http://en.wikibooks.org/wiki/Opticks_Developer_Guide/Getting_Started/Building_Qt_From_Source

Now that QT 4.6 is out I might have to revisit a new library build with 64
bit in mind.

Carlos


On Tue, Jan 12, 2010 at 8:35 PM, Jeff Lenk <jeff at jefflenk.com> wrote:

>
> Very Cool!
>
> Just curious Carlos do you have any experience with x64 with regard to QT
> on
> Win?
>
> Thanks
> Jeff
>
>
> Carlos Talbot wrote:
> >
> > FYI,
> >
> > there's a Windows pre-compiled binary of FSComm now available for those
> > who
> > want to check it.
> >
> > http://files.freeswitch.org/windows_installer/FSComm.exe
> >
> > regards,
> >
> > Carlos
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://n2.nabble.com/FSComm-Windows-build-tp4289256p4295967.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From xanlich at gmail.com  Tue Jan 12 19:37:00 2010
From: xanlich at gmail.com (Chia-Yen Wu)
Date: Wed, 13 Jan 2010 11:37:00 +0800
Subject: [Freeswitch-users] Hello , about calculate numbers
In-Reply-To: <EDF668FE-A76D-4A5F-96D1-756A7ACD13A6@jerris.com>
References: <314dc3f81001120908l5772b13fqd676a3be367bb662@mail.gmail.com>
	<201001121239.06756.sos@sokhapkin.dyndns.org>
	<EDF668FE-A76D-4A5F-96D1-756A7ACD13A6@jerris.com>
Message-ID: <314dc3f81001121937l1b415c3amb715d50017850ebb@mail.gmail.com>

Thanks for reply, I have tried the new one and did transfer back
but it still dont work, when I was posting this question again with my test
commands.
suddenly I found out what the problem it is.

the variable cannot with number in it, like:

work:
<action application="set" data="var=1"/>
<action application="set" data="var=${expr(${var}+1)}"/>
fail:
<action application="set" data="var1=1"/>
<action application="set" data="var1=${expr(${var1}+1)}"/>


2010/1/13 Michael Jerris <mike at jerris.com>

> Remember you can't do conditions on these vars being set unless you
> transfer back into the dialplan as these actions are not run immediately,
> but rather after dialplan parse, unless you use
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>
> Mie
>
> On Jan 12, 2010, at 12:39 PM, Sergey Okhapkin wrote:
>
> > <action application="set" data="var=${expr(${var}+1)}"/>
> >
> > On Tuesday 12 January 2010, Chia-Yen Wu wrote:
> >> Hello everyone,
> >> Is there anyway to calculate number in dialplan.xml?
> >>
> >> like example:
> >> <action application="set" data="var=1"/>
> >> and with an action , I can do : var=var+1
> >>
> >> I have tried mod_expr, but failed to catch the variable, like below:
> >> <action application="set" data="var=${expr($var+1)}"/>
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mcampbellsmith at gmail.com  Tue Jan 12 20:04:24 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 13 Jan 2010 15:04:24 +1100
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <33c87fa31001120223t13efc07agcc595e48405fd8bb@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
	<33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
	<33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>
	<20100112095332.GA32294@jdc.jasonjgw.net>
	<33c87fa31001120223t13efc07agcc595e48405fd8bb@mail.gmail.com>
Message-ID: <33c87fa31001122004g3ad8fe11w701b14a2fcd03442@mail.gmail.com>

Hi All,

Does anyone know if tone_detect can be used with bypass_media?

I thought the SIP re-Invite message can be used to update media
parameters, including IP address endpoints.  Does FS try to do this
in the case that tone_detect is used?

In my case, the calls are dropped.

On Tue, Jan 12, 2010 at 9:23 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Hi Jason,
>
> I have understood that. ?Its not that a difficult concept to understand!
>
> In the log I see:
> 2010-01-12 21:03:17.585598 [DEBUG] mod_dptools.c:818
> sofia/internal/1000 at 192.168.1.120 SET
> [ringback]=[v=-7;%(400,200,413,438);%(400,2000,413,438)]
> 2010-01-12 21:03:17.605591 [DEBUG] switch_core_session.c:1509
> Application tone_detect Requires media! pre_answering channel
> sofia/internal/1000 at 192.168.1.120
>
> I thought the SIP re-Invite message can be used to update media
> parameters, including IP address endpoints. ?Does FS try too do this
> in the case that tone_detect is used?
>
> On Tue, Jan 12, 2010 at 8:53 PM, Jason White <jason at jasonjgw.net> wrote:
>> Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>>> I've updated and tested bypass_media.
>>>
>>> It works if I remove this line from the B leg dialplan (ie 2010 calls
>>> 1000 - this is in the 1000 section of the dialplan):
>>> ? ? ? ? ? ? ? ? ? ? ? ? <action application="tone_detect" data="fax
>>> 1100 r +5000 transfer 'receivefax XML features' 1" />
>>>
>>> Does bypass_media work with tone_detect?
>>
>> As I understand it, tone_detect detects tones in the RTP stream (i.e., in the
>> audio). For this to be possible, FreeSWITCH has to be in the audio path, hence
>> bypass media cannot be used
>>
>> If this reasoning isn't obvious to you, then you've misunderstood what
>> tone_detect does or what bypass media is (the audio traffic flows directly
>> between the two endpoints without passing through the FreeSWITCH system that
>> establishes the connection, therefore FreeSWITCH can't process it to detect
>> tones and consequently bypass media and tone detection are inherently
>> incompatible.)
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From mike at jerris.com  Tue Jan 12 20:09:29 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 12 Jan 2010 23:09:29 -0500
Subject: [Freeswitch-users] Hello , about calculate numbers
In-Reply-To: <314dc3f81001121937l1b415c3amb715d50017850ebb@mail.gmail.com>
References: <314dc3f81001120908l5772b13fqd676a3be367bb662@mail.gmail.com>
	<201001121239.06756.sos@sokhapkin.dyndns.org>
	<EDF668FE-A76D-4A5F-96D1-756A7ACD13A6@jerris.com>
	<314dc3f81001121937l1b415c3amb715d50017850ebb@mail.gmail.com>
Message-ID: <2C92D649-BFC9-4FF9-AA1F-E6D0FDB5CA1A@jerris.com>

Strange, could you post the debug log of this so we can see how it's  
expanded?

Mike

On Jan 12, 2010, at 10:37 PM, Chia-Yen Wu <xanlich at gmail.com> wrote:

> Thanks for reply, I have tried the new one and did transfer back
> but it still dont work, when I was posting this question again with  
> my test commands.
> suddenly I found out what the problem it is.
>
> the variable cannot with number in it, like:
>
> work:
> <action application="set" data="var=1"/>
> <action application="set" data="var=${expr(${var}+1)}"/>
> fail:
> <action application="set" data="var1=1"/>
> <action application="set" data="var1=${expr(${var1}+1)}"/>
>
>
> 2010/1/13 Michael Jerris <mike at jerris.com>
> Remember you can't do conditions on these vars being set unless you  
> transfer back into the dialplan as these actions are not run  
> immediately, but rather after dialplan parse, unless you use http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>
> Mie
>
> On Jan 12, 2010, at 12:39 PM, Sergey Okhapkin wrote:
>
> > <action application="set" data="var=${expr(${var}+1)}"/>
> >
> > On Tuesday 12 January 2010, Chia-Yen Wu wrote:
> >> Hello everyone,
> >> Is there anyway to calculate number in dialplan.xml?
> >>
> >> like example:
> >> <action application="set" data="var=1"/>
> >> and with an action , I can do : var=var+1
> >>
> >> I have tried mod_expr, but failed to catch the variable, like  
> below:
> >> <action application="set" data="var=${expr($var+1)}"/>
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From magesh.freeswitch at gmail.com  Tue Jan 12 20:21:20 2010
From: magesh.freeswitch at gmail.com (Magesh R)
Date: Tue, 12 Jan 2010 23:21:20 -0500
Subject: [Freeswitch-users] Sangoma PRI installation for FreeSWITCH
Message-ID: <369c72d81001122021o6885913cl618965791aec4621@mail.gmail.com>

Dear All,
         I have installed Sangoma PRI card in machine with following steps,
                 * wget
ftp://ftp.sangoma.com/linux/custom/3.5/wanpipe-3.5.8.7.tgz
                 * tar -xvfz wanpipe-3.5.8.7.tgz
                 * cd wanpipe-3.5.8.7
                 * make openzap
                 * make install
                 * make install_pri

           I have executed "wanrouter hwprobe" command it prints the
following details,

           1 . AFT-A102-SH : SLOT=4 : BUS=3 : IRQ=11 : CPU=A : PORT=1 :
HWEC=0 : V=36
           2 . AFT-A102-SH : SLOT=4 : BUS=3 : IRQ=11 : CPU=A : PORT=2 :
HWEC=0 : V=36

        Card Cnt: A101-2=1

       Next I have executed wancfg_fs script to configure the sangoma for
freeswitch.
It creates the following configuration files
                     * wanpipe1.conf
                     * wanpipe2.conf
                     * smg_prid.conf
                     * openzap.conf
                     * openzap.conf.xml
       I have attached those files.
       I have started the wanrouter and printed the wanrouter status,
              Wanrouter Status:

Device name | Protocol | Station | Status        |
wanpipe1    | AFT TE1  | N/A     | Connected     |
wanpipe2    | AFT TE1  | N/A     | Disconnected  |

      Next I have started the smg_ctrl, but it failed to start. It prints
the following things,
smg_ctrl start

Starting processes...
Loading SCTP...OK
Starting sangoma_prid...OK
sangoma_prid failed to start
check /var/log/sangoma_mgd.log for errors

Stopping running processes...
safe_sangoma not running...
sangoma_prid is stopped
Removing PID files...done

I have checked /var/log/sangoma_mgd.log file. But nothing was there.

Could any please tell me where I made mistake?

Thanks,
Mag
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From brian at freeswitch.org  Tue Jan 12 20:41:35 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 22:41:35 -0600
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <33c87fa31001122004g3ad8fe11w701b14a2fcd03442@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
	<33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
	<33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>
	<20100112095332.GA32294@jdc.jasonjgw.net>
	<33c87fa31001120223t13efc07agcc595e48405fd8bb@mail.gmail.com>
	<33c87fa31001122004g3ad8fe11w701b14a2fcd03442@mail.gmail.com>
Message-ID: <12C239EE-DAB3-4B52-9317-32399B5290E3@freeswitch.org>

NO.

/b

On Jan 12, 2010, at 10:04 PM, Mark Campbell-Smith wrote:

> 
> Does anyone know if tone_detect can be used with bypass_media?



From vgoget at yahoo.com  Tue Jan 12 16:54:54 2010
From: vgoget at yahoo.com (VG Oget)
Date: Tue, 12 Jan 2010 16:54:54 -0800 (PST)
Subject: [Freeswitch-users] Fw: SIP client authorization issues (Globarange
	phone)
Message-ID: <271101.94727.qm@web53107.mail.re2.yahoo.com>

Hi, I am trying to register a GlobaRange SIP phone to FreeSwitch (was able to do it with Asterisk some time ago). It is a locked VOIP phone to Joip.com. On Ubuntu with latest version as of a few days ago?

I redirected the traffic to Freeswitch (basically proxy.joip.com points and to the Freeswitch host) and also changed the SIP port to 23768 (what the phone wants). Very close to this: http://darkskiez.co.uk/index.php?page=Use_The_Panasonic_Globarange_With_Asterisk


I have Xlite successfully registering (SIP port 23768 and using proxy.joip.com not the IP address) so I?m confident the freeswitch config is close to being okay?

Now I have intercepted the traffic during registration. The issue (I think) is that the SIP authentication phases use a different username and I don?t know how to configure it.
I looked into this page extensively: http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide
And also tried to understand sip_suth_username but I cannot find the solution.

If someone can point me in the right direction, that would be perfect.

Thank you,
G.


--------------

Phone: 192.168.1.11
Freeswitch: 192.168.1.5 (=proxy.joip.com on local dns)
SIP port: 5060
Extension(changed): 1234567890 (joip username)
Authentication username (changed): 98765432

------------


REGISTER sip:proxy.joip.com:23768 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:6060;branch=z9hG4bK991d765d
Max-Forwards: 70
To: sip:1234567890 at proxy.joip.com
From: sip:1234567890 at proxy.joip.com;tag=3957008896
Call-ID: yyyy-xxxxxxxxxxxx at 192.168.1.11
CSeq: 1 REGISTER
Contact: sip:1234567890 at 192.168.1.11:6060;nat=3;deviceid=2
Expires: 304
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY
User-Agent: Panasonic GT1500/a13.32/xxxxxxxxxxxx
Content-Length: 0

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.11:6060;branch=z9hG4bK991d765d
From: <sip:1234567890 at proxy.joip.com>;tag=3957008896
To: <sip:1234567890 at proxy.joip.com>;tag=96BeFKm5U4jFH
Call-ID: yyyy-xxxxxxxxxxxx at 192.168.1.11
CSeq: 1 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.0.5-20100106-0400-hacked
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
WWW-Authenticate: Digest realm="proxy.joip.com", nonce="67132386-fe25-11de-97c3-c7963d15da66", algorithm=MD5, qop="auth"
Content-Length: 0

REGISTER sip:proxy.joip.com:23768 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:6060;branch=z9hG4bKe585be9f
Max-Forwards: 70
To: sip:1234567890 at proxy.joip.com
From: sip:1234567890 at proxy.joip.com;tag=3957008896
Call-ID: yyyy-xxxxxxxxxxxx at 192.168.1.11
CSeq: 2 REGISTER
Contact: sip:1234567890 at 192.168.1.11:6060;nat=3;deviceid=2
Expires: 304
Authorization: Digest realm="proxy.joip.com", nonce="67132386-fe25-11de-97c3-c7963d15da66", algorithm=MD5, qop=auth, cnonce="7FDF680B", nc=00000001, uri="sip:proxy.joip.com:23768", username="98765432", 

response="7bd8c1898e9c01ee1817146ddc86c8b4"
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY
User-Agent: Panasonic GT1500/a13.32/xxxxxxxxxxxx
Content-Length: 0

SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.11:6060;branch=z9hG4bKe585be9f
From: <sip:1234567890 at proxy.joip.com>;tag=3957008896
To: <sip:1234567890 at proxy.joip.com>;tag=ag56ge58rD91c
Call-ID: yyyy-xxxxxxxxxxxx at 192.168.1.11
CSeq: 2 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.0.5-20100106-0400-hacked
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Length: 0


----




      Get your preferred Email name!
Now you can @ymail.com and @rocketmail.com. 
http://mail.promotions.yahoo.com/newdomains/aa/


From mike at jerris.com  Tue Jan 12 20:46:00 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 12 Jan 2010 23:46:00 -0500
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <33c87fa31001122004g3ad8fe11w701b14a2fcd03442@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
	<33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
	<33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>
	<20100112095332.GA32294@jdc.jasonjgw.net>
	<33c87fa31001120223t13efc07agcc595e48405fd8bb@mail.gmail.com>
	<33c87fa31001122004g3ad8fe11w701b14a2fcd03442@mail.gmail.com>
Message-ID: <03E1D9E8-ADB2-4D3D-BCDC-3F54ADD8879A@jerris.com>

As soon as it pulls in the media (in this case, to do the tone detect), its going to give up on bypass media unless you adjust settings to behave otherwise. (such as http://wiki.freeswitch.org/wiki/Variable_bypass_media_after_bridge)

Mike

On Jan 12, 2010, at 11:04 PM, Mark Campbell-Smith wrote:

> Hi All,
> 
> Does anyone know if tone_detect can be used with bypass_media?
> 
> I thought the SIP re-Invite message can be used to update media
> parameters, including IP address endpoints.  Does FS try to do this
> in the case that tone_detect is used?
> 
> In my case, the calls are dropped.
> 
> On Tue, Jan 12, 2010 at 9:23 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>> Hi Jason,
>> 
>> I have understood that.  Its not that a difficult concept to understand!
>> 
>> In the log I see:
>> 2010-01-12 21:03:17.585598 [DEBUG] mod_dptools.c:818
>> sofia/internal/1000 at 192.168.1.120 SET
>> [ringback]=[v=-7;%(400,200,413,438);%(400,2000,413,438)]
>> 2010-01-12 21:03:17.605591 [DEBUG] switch_core_session.c:1509
>> Application tone_detect Requires media! pre_answering channel
>> sofia/internal/1000 at 192.168.1.120
>> 
>> I thought the SIP re-Invite message can be used to update media
>> parameters, including IP address endpoints.  Does FS try too do this
>> in the case that tone_detect is used?
>> 
>> On Tue, Jan 12, 2010 at 8:53 PM, Jason White <jason at jasonjgw.net> wrote:
>>> Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>>>> I've updated and tested bypass_media.
>>>> 
>>>> It works if I remove this line from the B leg dialplan (ie 2010 calls
>>>> 1000 - this is in the 1000 section of the dialplan):
>>>>                         <action application="tone_detect" data="fax
>>>> 1100 r +5000 transfer 'receivefax XML features' 1" />
>>>> 
>>>> Does bypass_media work with tone_detect?
>>> 
>>> As I understand it, tone_detect detects tones in the RTP stream (i.e., in the
>>> audio). For this to be possible, FreeSWITCH has to be in the audio path, hence
>>> bypass media cannot be used
>>> 
>>> If this reasoning isn't obvious to you, then you've misunderstood what
>>> tone_detect does or what bypass media is (the audio traffic flows directly
>>> between the two endpoints without passing through the FreeSWITCH system that
>>> establishes the connection, therefore FreeSWITCH can't process it to detect
>>> tones and consequently bypass media and tone detection are inherently
>>> incompatible.)
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jeff at jefflenk.com  Tue Jan 12 20:51:14 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 12 Jan 2010 20:51:14 -0800 (PST)
Subject: [Freeswitch-users] FSComm Windows build
In-Reply-To: <5800526b1001121918r15a336f5jbc5b724e66ac2825@mail.gmail.com>
References: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
	<1263350125191-4295967.post@n2.nabble.com>
	<5800526b1001121918r15a336f5jbc5b724e66ac2825@mail.gmail.com>
Message-ID: <1263358274180-4311097.post@n2.nabble.com>


Carlos,

Thanks for those links!

I am currenltly using prebuilt 4.6 32 bit libs and was hoping Nokia prebuilt
the x64 too. I guess I might have to look into building myself as well.

-Jeff


Jeff,

not yet. I'm still linking with a 32bit QT library of 4.5.2 I compiled for
VS2008 sometime last year. I followed a guide similar to this one:
http://dcsoft.com/community_server/blogs/dcsoft/archive/2009/03/06/how-to-setup-qt-4-5-visual-studio-integration.aspx

At the time I didn't see a need for a 64bit version of FsGui. Since FSComm
has FreeSWITCH in the back end this kind of changes things.

Looks like someone put up a wiki on compiling QT for Win x64:
http://en.wikibooks.org/wiki/Opticks_Developer_Guide/Getting_Started/Building_Qt_From_Source

Now that QT 4.6 is out I might have to revisit a new library build with 64
bit in mind.

Carlos

-- 
View this message in context: http://n2.nabble.com/FSComm-Windows-build-tp4289256p4311097.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From pmhshz at gmail.com  Tue Jan 12 20:56:44 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Wed, 13 Jan 2010 10:26:44 +0530
Subject: [Freeswitch-users] Defunct process in ESL testserver example
In-Reply-To: <191c3a031001120834q36f72791s811b6352ab7c12eb@mail.gmail.com>
References: <a792304b1001120722o2d4fbf91vb0456ef4f0436a5e@mail.gmail.com>
	<191c3a031001120834q36f72791s811b6352ab7c12eb@mail.gmail.com>
Message-ID: <a792304b1001122056m1885bf01ka5e2d64c9c183710@mail.gmail.com>

Thanks Anthony,

I added the below line before esl_listen to ignore the sigchild, and it
resolved the problem:
signal(SIGCHLD, SIG_IGN);
esl_listen(ip, port, mycallback);

-MohammedShehzad

On Tue, Jan 12, 2010 at 10:04 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> It's forking process code.
> you need to ignore sigchld or create a sig handler for sigchld and use the
> wait syscall to reap the process.
>
>
> On Tue, Jan 12, 2010 at 9:22 AM, MohammedShehzad <pmhshz at gmail.com> wrote:
>
>> Hello everybody,
>>
>> I am creating a C program of ESL outbound for call processing.
>> I am using testserver.c example, and till now it seems fine.
>>
>> But i noticed that every call testserver process, a new process is being
>> created which I can see in Linux system with below command:
>> For example, when I make two calls and even after hangup, I saw three
>> process like below:
>> ps -A | grep testserver
>>  9345 pts/2    00:00:00 testserver
>>  9350 pts/2    00:00:00 testserver
>>  9357 pts/2    00:00:00 testserver
>>
>> This get increased for every call i make.
>>
>> I did some workout and placed below two lines (close & exit) at the end of
>> mycallback function, (as I found them on ivrd.c file):
>>
>>     esl_disconnect(&handle);
>>     close(client_sock);
>>     exit(0);
>> }
>> But after that the process becomes defunct/zombie
>>
>>  9440 pts/2    00:00:00 conflisten
>>  9442 pts/2    00:00:00 conflisten <defunct>
>>  9452 pts/2    00:00:00 conflisten <defunct>
>>
>>
>> Will anybody please suggest me how can I eliminate this process, which
>> remains in memory even after call hangup?
>>
>> Thanks for any response.
>> MohammedShehzad
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From mcampbellsmith at gmail.com  Tue Jan 12 21:02:01 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 13 Jan 2010 16:02:01 +1100
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <03E1D9E8-ADB2-4D3D-BCDC-3F54ADD8879A@jerris.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
	<33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
	<33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>
	<20100112095332.GA32294@jdc.jasonjgw.net>
	<33c87fa31001120223t13efc07agcc595e48405fd8bb@mail.gmail.com>
	<33c87fa31001122004g3ad8fe11w701b14a2fcd03442@mail.gmail.com>
	<03E1D9E8-ADB2-4D3D-BCDC-3F54ADD8879A@jerris.com>
Message-ID: <33c87fa31001122102v573ebe3aj365915e850d5b926@mail.gmail.com>

ahha... thanks Mike.

So I could do something like this, if 1000 is the b-leg?:

                  <condition field="destination_number"
expression="^(10[01][0-9])$">
                        <action application="set" data="dialed_extension=$1"/>
                        <action application="set"
data="bypass_media_after_bridge=true"/>
                        <action application="tone_detect" data="fax
1100 r +5000 transfer 'receivefax XML features' 1" />
                        <action application="bridge"
data="user/${dialed_extension}@${domain}"/>
                  </condition>

Looks interesting - I'll have to give it a whirl later ....

On Wed, Jan 13, 2010 at 3:46 PM, Michael Jerris <mike at jerris.com> wrote:
> As soon as it pulls in the media (in this case, to do the tone detect), its going to give up on bypass media unless you adjust settings to behave otherwise. (such as http://wiki.freeswitch.org/wiki/Variable_bypass_media_after_bridge)
>
> Mike
>
> On Jan 12, 2010, at 11:04 PM, Mark Campbell-Smith wrote:
>
>> Hi All,
>>
>> Does anyone know if tone_detect can be used with bypass_media?
>>
>> I thought the SIP re-Invite message can be used to update media
>> parameters, including IP address endpoints. ?Does FS try to do this
>> in the case that tone_detect is used?
>>
>> In my case, the calls are dropped.
>>
>> On Tue, Jan 12, 2010 at 9:23 PM, Mark Campbell-Smith
>> <mcampbellsmith at gmail.com> wrote:
>>> Hi Jason,
>>>
>>> I have understood that. ?Its not that a difficult concept to understand!
>>>
>>> In the log I see:
>>> 2010-01-12 21:03:17.585598 [DEBUG] mod_dptools.c:818
>>> sofia/internal/1000 at 192.168.1.120 SET
>>> [ringback]=[v=-7;%(400,200,413,438);%(400,2000,413,438)]
>>> 2010-01-12 21:03:17.605591 [DEBUG] switch_core_session.c:1509
>>> Application tone_detect Requires media! pre_answering channel
>>> sofia/internal/1000 at 192.168.1.120
>>>
>>> I thought the SIP re-Invite message can be used to update media
>>> parameters, including IP address endpoints. ?Does FS try too do this
>>> in the case that tone_detect is used?
>>>
>>> On Tue, Jan 12, 2010 at 8:53 PM, Jason White <jason at jasonjgw.net> wrote:
>>>> Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>>>>> I've updated and tested bypass_media.
>>>>>
>>>>> It works if I remove this line from the B leg dialplan (ie 2010 calls
>>>>> 1000 - this is in the 1000 section of the dialplan):
>>>>> ? ? ? ? ? ? ? ? ? ? ? ? <action application="tone_detect" data="fax
>>>>> 1100 r +5000 transfer 'receivefax XML features' 1" />
>>>>>
>>>>> Does bypass_media work with tone_detect?
>>>>
>>>> As I understand it, tone_detect detects tones in the RTP stream (i.e., in the
>>>> audio). For this to be possible, FreeSWITCH has to be in the audio path, hence
>>>> bypass media cannot be used
>>>>
>>>> If this reasoning isn't obvious to you, then you've misunderstood what
>>>> tone_detect does or what bypass media is (the audio traffic flows directly
>>>> between the two endpoints without passing through the FreeSWITCH system that
>>>> establishes the connection, therefore FreeSWITCH can't process it to detect
>>>> tones and consequently bypass media and tone detection are inherently
>>>> incompatible.)
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mike at jerris.com  Tue Jan 12 21:14:55 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 00:14:55 -0500
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <33c87fa31001122102v573ebe3aj365915e850d5b926@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
	<33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
	<33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>
	<20100112095332.GA32294@jdc.jasonjgw.net>
	<33c87fa31001120223t13efc07agcc595e48405fd8bb@mail.gmail.com>
	<33c87fa31001122004g3ad8fe11w701b14a2fcd03442@mail.gmail.com>
	<03E1D9E8-ADB2-4D3D-BCDC-3F54ADD8879A@jerris.com>
	<33c87fa31001122102v573ebe3aj365915e850d5b926@mail.gmail.com>
Message-ID: <ABAE5818-5254-4DCA-8AAB-36E0D065EA5B@jerris.com>

I doubt that works as you will still need media after it has been bridged.  I was just giving an example of something that could be used.  You might be able to set that var, set tone detect, answer, play fake ringback for a short time (as long as you have you tone detect checking for tone), and then bridge.

Mike

On Jan 13, 2010, at 12:02 AM, Mark Campbell-Smith wrote:

> ahha... thanks Mike.
> 
> So I could do something like this, if 1000 is the b-leg?:
> 
>                  <condition field="destination_number"
> expression="^(10[01][0-9])$">
>                        <action application="set" data="dialed_extension=$1"/>
>                        <action application="set"
> data="bypass_media_after_bridge=true"/>
>                        <action application="tone_detect" data="fax
> 1100 r +5000 transfer 'receivefax XML features' 1" />
>                        <action application="bridge"
> data="user/${dialed_extension}@${domain}"/>
>                  </condition>
> 
> Looks interesting - I'll have to give it a whirl later ....



From xanlich at gmail.com  Wed Jan 13 00:48:40 2010
From: xanlich at gmail.com (Chia-Yen Wu)
Date: Wed, 13 Jan 2010 16:48:40 +0800
Subject: [Freeswitch-users] Hello , about calculate numbers
In-Reply-To: <2C92D649-BFC9-4FF9-AA1F-E6D0FDB5CA1A@jerris.com>
References: <314dc3f81001120908l5772b13fqd676a3be367bb662@mail.gmail.com>
	<201001121239.06756.sos@sokhapkin.dyndns.org>
	<EDF668FE-A76D-4A5F-96D1-756A7ACD13A6@jerris.com>
	<314dc3f81001121937l1b415c3amb715d50017850ebb@mail.gmail.com>
	<2C92D649-BFC9-4FF9-AA1F-E6D0FDB5CA1A@jerris.com>
Message-ID: <314dc3f81001130048k3fec1deaocf444a30aeb5a9ce@mail.gmail.com>

Sorry, my bad, I finally found out that I got an incorrect condition and the
stage goes wrong.
Everything is working fine now, thanks for help!


2010/1/13 Michael Jerris <mike at jerris.com>

> Strange, could you post the debug log of this so we can see how it's
> expanded?
>
> Mike
>
>
> On Jan 12, 2010, at 10:37 PM, Chia-Yen Wu <xanlich at gmail.com> wrote:
>
> Thanks for reply, I have tried the new one and did transfer back
> but it still dont work, when I was posting this question again with my test
> commands.
> suddenly I found out what the problem it is.
>
> the variable cannot with number in it, like:
>
> work:
> <action application="set" data="var=1"/>
> <action application="set" data="var=${expr(${var}+1)}"/>
> fail:
> <action application="set" data="var1=1"/>
> <action application="set" data="var1=${expr(${var1}+1)}"/>
>
>
> 2010/1/13 Michael Jerris < <mike at jerris.com>mike at jerris.com>
>
>> Remember you can't do conditions on these vars being set unless you
>> transfer back into the dialplan as these actions are not run immediately,
>> but rather after dialplan parse, unless you use
>> <http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions>
>> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>>
>> Mie
>>
>> On Jan 12, 2010, at 12:39 PM, Sergey Okhapkin wrote:
>>
>> > <action application="set" data="var=${expr(${var}+1)}"/>
>> >
>> > On Tuesday 12 January 2010, Chia-Yen Wu wrote:
>> >> Hello everyone,
>> >> Is there anyway to calculate number in dialplan.xml?
>> >>
>> >> like example:
>> >> <action application="set" data="var=1"/>
>> >> and with an action , I can do : var=var+1
>> >>
>> >> I have tried mod_expr, but failed to catch the variable, like below:
>> >> <action application="set" data="var=${expr($var+1)}"/>
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Wed Jan 13 01:13:03 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 13 Jan 2010 14:43:03 +0530
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
Message-ID: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>

Hi all,

I've done a sample program (In perl ESL) , which play a file to the caller
and then it will call recvEvent() and print the event name. I've handled
signals also.

When I send SIGINT to my program (Perl), the signal handler is called and I
can see the print output. But in the same time,  I received
SERVER_DISCONNECTED from freeswitch as event.

I don't know why I received SERVER_DISCONNECTED from freeswitch. Is it
because, the recvEvent() from perl internally  calls the recvevent function
in the Esl.c and when it waits to receive the information from socket, the
signal occurred???

Please clarify me!!

Here is my program
require ESL;
use IO::Socket::INET;
use Data::Dumper;

my $ip = "192.168.1.222";
my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
Proto => 'tcp',  Listen => 1,  Reuse => 1 );
die "Could not create socket: $!\n" unless $sock;
&register_Signals();

for(;;) {
    my $new_sock = $sock->accept();
    next if (not defined ($new_sock));
    my $pid = fork();
    if ($pid) {
        close($new_sock);
        next;
    }
    print "CHILD PID: $$\n";
    my $host = $new_sock->sockhost();
    my $fd = fileno($new_sock);

    my $con = new ESL::ESLconnection($fd);
    my $info = $con->getInfo();

    my $uuid = $info->getHeader("unique-id");

    printf "Connected call %s, from %s\n", $uuid,
$info->getHeader("caller-caller-id-number");
    my $r=$con->execute("answer");
    print Dumper $r;
    $con->events("plain","all");
    my
$re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");

    while($con->connected()) {
        my $e = $con->recvEvent();

        if ($e) {
            my $name = $e->getHeader("event-name");
            print "EVENT [$name]\n";
            if ($name eq "DTMF") {
                my $digit = $e->getHeader("dtmf-digit");
                print "$digit\n";
            }
        }
    }
    close($new_sock);
}
sub register_Signals() {
    foreach ( keys %SIG ) {
        $SIG{$_} = 'sig_Handler';
    }
}

sub sig_Handler() {
    my $handle=$_[0];
    if($handle eq "INT") {
        print "$$: SIGNAL SIG$handle is generated\n";
    }
}
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From tayeb.meftah at gmail.com  Wed Jan 13 01:56:30 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Wed, 13 Jan 2010 10:56:30 +0100
Subject: [Freeswitch-users] unable to dial out troug siemens Hie9200
	softswitch
Message-ID: <4B4D98CE.1060808@gmail.com>

hi dear all,
we have a siemens HIE 9200 softswitch that support SIP, SIP-T and H.248
we are trying to interconnect freeswitch with it
if HIE9200 dial out troug freeswitch, is passing the call, but without RTP
but if the freeswitch dial out troug the HIE9200, the call is unable to 
pass with error 500
the trace is here:

http://siplabs.net/tracebin/fs-siemens-500.pcap
http://siplabs.net/tracebin/fs-siemens-rtp.pcap
unfortunatly the freeswitch don't support H.248 otherwise i will control 
the media gateway (HIG 1100)
thanks for any help


From Claudio.Cavalera at italtel.it  Wed Jan 13 02:22:29 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Wed, 13 Jan 2010 11:22:29 +0100
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <191c3a031001121535t4819ee44p74cc9f7cd4fe972f@mail.gmail.com>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F362E0@BESONE.corp.dom>

Thanks a lot Anthony,
some comments inline (and please forgive me for my broken email client).
 
>>  example1: Consider this simple lua script in which i create two
sessions:

 >>  api = freeswitch.API();
 >>  api:execute("originate", "sofia/internal/1001%192.168.1.1 &park");
 >>  api:execute("originate", "sofia/internal/1002%192.168.1.1 &park");

 >  capture the output from api:execute the uuid is in there

Thx a lot,
this was one piece i was missing although it's already on the wiki here:
http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls 


 > because lua calls it freeswitch.bridge

 > session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
 > session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1",
session1);
 > freeswitch.bridge(session1, session2); 
 
good to now, there isn't any example of freeswitch.bridge in the wiki
and i would like to add one.
Where I could find the full api of
freeswitch.Session( ) ?
because I've seen this working also without "session1" in the second
line:
session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");
freeswitch.bridge(session1, session2); 

 also is there any difference between  freeswitch.bridge and
freeswitch.execute(uuid_bridge ...) ?
 
 >> example3: yet another possibility
 >> local session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
 >> session1:originate(session2, "sofia/internal/1002%192.168.1.1",
1000);
 >> but it does not work either.

 > The above is gibberish try:
 > local session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
 > session1:execute("bridge", "sofia/internal/1002%192.168.1.1"); 
 
Okay i will report also this bridge example on the wiki which was
missing.
But does session:originate make sense in some cases or not? Otherwise
i'm going to remove this line on the wiki
 http://wiki.freeswitch.org/wiki/Mod_lua#session:originate
<http://wiki.freeswitch.org/wiki/Mod_lua#session:originate> 
 
Thanks,
Claudio


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From david.villasmil.work at gmail.com  Wed Jan 13 03:11:34 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 13 Jan 2010 12:11:34 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
Message-ID: <5F707113-F78F-44C0-96A4-2C211F1C4791@gmail.com>

is it possible? can i bridge to multiple b-sides and m?ltiple




From david.villasmil.work at gmail.com  Wed Jan 13 03:15:14 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 13 Jan 2010 12:15:14 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
Message-ID: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>

hello

is it possible to bridge multiple b-legs and provide all audio  
(progress) until there is an answer on one channel?

thanks guys

David


From Prometheus001 at gmx.net  Wed Jan 13 03:26:13 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 13 Jan 2010 12:26:13 +0100
Subject: [Freeswitch-users] Multiple registrations: Only calling the first
 element in the list in this mode
Message-ID: <4B4DADD5.3010507@gmx.net>

Hello,

I habe the following behaviour

when I call a user which is registered twice with 2 phones via
    bridge user/100 at domain
both phones are ringing. This is correct as I allow multiple
registrations in a profile

However when I call multiple endpoints via
    bridge user/100 at domain,user/101 at domain,user/102 at domain
only one phone with number100 is ringing.

Console log shows "Only calling the first element in the list in this
mode.":
2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734 variable
string 0 = [presence_id=100 at domain]
2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734 variable
string 1 = [transfer_fallback_extension=100]
2010-01-12 19:52:18.236361 [WARNING] switch_ivr_originate.c:2048 Only
calling the first element in the list in this mode.
2010-01-12 19:52:18.236361 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip:100 at 10.11.12.203:2048
[9b95fcdc-ffab-11de-9ba2-13b2daa7ce61]

Is there any way to work around this? I need all phones to be ringing in
this scenario.

Best regards
Peter


From mailinglist at fribert.dk  Wed Jan 13 03:44:37 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Wed, 13 Jan 2010 12:44:37 +0100
Subject: [Freeswitch-users] Svar: Re:  Multi-Homed setup,
	starting over	 - still not working
Message-ID: <4B4DC035020000E1000003A2@mail.fribert.dk>

Hi Joseph

Oh yes I have :-)
But are you sayng that "eval ${domain}" will do more than is done when I restart the fs?
I don't see that anywhere in the guide :-D

I'll try it out asap!

Thanks
Fribse


>>> "Joseph L. Casale"  12-01-10 18:41 >>>
>Hi Guys
> 
>I really would like to have this up and running, but I'm constantly running into things that doesn't work, and I have no idea where the problem is...
> 
>I've started from scratch with pfSense and Freeswitch.
>I've followed the Multi Home tutorial here: http://wiki.freeswitch.org/wiki/Multi_home_tutorial
> 
>I've deleted the 'lan' profile, and altered the 'internal' profile as recommended, and I've gotten to the paragraph:
>
>APPLYING YOUR CHANGES AND CHECKING YOUR WORK
>
>When I start up my x-lite program I get this error:
>2010-01-12 16:38:54.172731 [WARNING] sofia_reg.c:1755 Can't find user [1000 at 83.89.x.x]
>You must define a domain called '83.89.x.xin your directory and add a user with the id="1000" attribute
>and you must configure your device to use the proper domain in it's authentication credentials.
>and you must configure your device to use the proper domain in it's authentication credentials.
> 
>83.89.x.x is my external IP, and not my internal IP???
>
>Any help on this?

This is because you haven't set your domain in vars.xml. The behavior is that
$${local_ip_v4} evals to your wan ip. This is the first step in that tutorial:)
http://wiki.freeswitch.org/wiki/Multi_home_tutorial#INTERNAL_LAN

Open vars.xml, make the line:
  
  

Match your lan ip:
  
  
 
restart fs, then goto the fs_cli and type `eval ${domain}` it should
come back with "your" lan ip.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From sos at sokhapkin.dyndns.org  Wed Jan 13 04:46:47 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 07:46:47 -0500
Subject: [Freeswitch-users] 10 seconds delay on fallback
Message-ID: <201001130746.47968.sos@sokhapkin.dyndns.org>

To implement fallback to backup PSTN routes and do LCR my dialplan executes 
the following commands:

set execute_on_answer=set hangup_after_bridge=true
set some custom channel variables
bridge sofia/gateway1/number
set some custom channel variables
bridge sofia/gateway2/number
....

This generally works fine if gateway1 returns SIP error, dialplan sets channel 
variables to another values and calls gateway2 immediately. But if GW1 
responds with 183 early media and SIP error after that, bad thing happens - 
dialplan continues immediately, sets channel variables, executes bridge 
application, but INVITE to GW2 is sent after 10 seconds delay. Caller gets 10 
extra seconds of post dial delay. I suspect the delay happens because of 
attempt to read audio frame, switch_ivr_originate() has the following lines:

if (switch_channel_media_ready(caller_channel)) {
	tstatus = switch_core_session_read_frame()
      ...

Any advice how to avoid extra delay in this situation?


From rupa at rupa.com  Wed Jan 13 04:50:21 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 13 Jan 2010 06:50:21 -0600
Subject: [Freeswitch-users] Multiple registrations: Only calling the
	first element in the list in this mode
In-Reply-To: <4B4DADD5.3010507@gmx.net>
References: <4B4DADD5.3010507@gmx.net>
Message-ID: <b39273531001130450k16af27d2v209a300f441d39d8@mail.gmail.com>

Try:

bridge user/100 at domain:_:user/101 at domain:_:user/102 at domain

Then document it up if it works.

On Wed, Jan 13, 2010 at 5:26 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello,
>
> I habe the following behaviour
>
> when I call a user which is registered twice with 2 phones via
>    bridge user/100 at domain
> both phones are ringing. This is correct as I allow multiple
> registrations in a profile
>
> However when I call multiple endpoints via
>    bridge user/100 at domain,user/101 at domain,user/102 at domain
> only one phone with number100 is ringing.
>
> Console log shows "Only calling the first element in the list in this
> mode.":
> 2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734 variable
> string 0 = [presence_id=100 at domain]
> 2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734 variable
> string 1 = [transfer_fallback_extension=100]
> 2010-01-12 19:52:18.236361 [WARNING] switch_ivr_originate.c:2048 Only
> calling the first element in the list in this mode.
> 2010-01-12 19:52:18.236361 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/sip:100 at 10.11.12.203:2048
> [9b95fcdc-ffab-11de-9ba2-13b2daa7ce61]
>
> Is there any way to work around this? I need all phones to be ringing in
> this scenario.
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From a.alalousi at gmail.com  Wed Jan 13 05:51:54 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Wed, 13 Jan 2010 13:51:54 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <eafa772f1001120520i719751b9hcbc22a969a89a585@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
	<87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
	<cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>
	<eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>
	<cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>
	<eafa772f1001120520i719751b9hcbc22a969a89a585@mail.gmail.com>
Message-ID: <cd93b3701001130551v26fe6355w77f532a86ba4e08a@mail.gmail.com>

Steve/All,

Thanks for all your feedback, this thread can be closed.

Here is some feedback:

Re: Steve's queries:

- "pproxy" was a typo on my part, but it should not affect anything. I set
them both to false in vars.xml, and don't over-ride them anywhere.
- true re: default values for proxy_media & bypass_media being false. I am
explicitly setting them here out of 1) paranoia and 2) I like to make sure I
know my variable values and not leave them to defaults - relics of being a
developer
- I was disabling the q850 code as part of my attempts to crack this nut.

Re: the solution, I've managed to rewrite some of the codes with a call to
the hangup app, which is what Steve is also using, and his findings re:
bridges getting 183 and 180 before 4xx.

I wonder if it's possible to rewrite causes from such bridges by executing a
JS or similar app attached to the bridge. I'll report on this as and when.

Regards,

Ahmed.




2010/1/12 Steven Ayre <steveayre at gmail.com>

> - prpoxy_media should be proxy_media
> - bypass_media and proxy_media shouldn't need setting to false -
> that's their default (unless you're set one of them to true on the sip
> profile?)
> - why do you need to disable q850 reason?
>
> I do something very similar - try this...
>
> <condition field="destination_number" expression="^(?:5824)(44(1|2)\d+)$">
>   <action application="set" data="continue_on_fail=true"/>
>  <action application="set" data="hangup_after_bridge=true"/>
>  <action application="set" data="sip_ignore_remote_cause=true"/>
>   <action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1"/>
>   <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
> </condition>
>
> By the way, you'll be unable to rewrite the hangup cause for a bridge
> that gets a 180 or 183 packet from the gateway before getting a 4xx,
> 5xx or 6xx packet (because those bridges don't 'fail') - I've seen
> this on a SIP provider before who gives 183 Session Progress before a
> 404 Not Found if the PSTN number dialled doesn't exist.
>
> Regards,
> -Steve
>
>
> 2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
> > Here you go:
> >
> > <condition field="destination_number" expression="^(?:5824)(44(1|2)\d+)$"
> > break="on-true">
> >   <action inline="true" application="set" data="bypass_media=false"/>
> >   <action inline="true" application="set" data="prpoxy_media=false"/>
> >   <action inline="true" application="set"
> data="disable_q850_reason=true"/>
> >   <action inline="true" application="set"
> data="hangup_after_bridge=false"/>
> >   <action inline="true" application="set"
> > data="sip_ignore_remote_cause=true"/>
> >   <action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1"/>
> >   <action inline="true" application="set"
> > data="bridge_hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
> >   <action inline="true" application="set"
> > data="hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
> >   <action application="bridge" data="error/normal_circuit_congestion"/>
> >   <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
> > </condition>
> >
> > As you can see, I am trying to rewrite the hangup codes in a multitude of
> > ways and places, but still exhibit the same behaviour.
> >
> > Any help appreciated.
> >
> > Regards,
> >
> > Ahmed.
> >
> > 2010/1/12 Steven Ayre <steveayre at gmail.com>
> >>
> >> Can you show us the dialplan extension you're trying?
> >>
> >> Thanks,
> >> -Steve
> >>
> >> 2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
> >> > Hi Michael,
> >> >
> >> > This is exactly what I'm doing, but it's just not happening.
> >> >
> >> > Thanks,
> >> >
> >> > Ahmed.
> >> >
> >> >
> >> > 2010/1/12 Michael Collins <msc at freeswitch.org>
> >> >>
> >> >>
> >> >> On Mon, Jan 11, 2010 at 4:18 AM, Ahmed Naji <a.alalousi at gmail.com>
> >> >> wrote:
> >> >>>
> >> >>> Dear All,
> >> >>>
> >> >>> I posted a thread re the subject but didn't get any joy, so perhaps
> >> >>> second time lucky.
> >> >>>
> >> >>> I need to rewrite a couple of hangup causes to mean
> NORMAL_CONGESTION
> >> >>> and
> >> >>> no matter which variables I set, this isn't happening. The idea is
> to
> >> >>> control what codes are returned to an end point after a successful
> >> >>> bridge,
> >> >>> as well as deal with what codes are returned if the bridge is
> >> >>> unsuccessful
> >> >>> (e.g. user_busy, originator_cancel ...etc).
> >> >>>
> >> >>> I've had limited success by setting hangup_after_bridge=false then
> >> >>> bridging to error/<required_code>. This, however only works when the
> >> >>> B-leg
> >> >>> terminates the call after a successful answer. Any other codes are
> not
> >> >>> rewritten.
> >> >>>
> >> >>> I've also tried playing with the bridge_hangup_code and hangup_code
> >> >>> variables prior and after bridging, still no joy. I have also set
> >> >>> sip_ignore_remote_cause=true prior to entering the bridge, as well
> >> >>> explicitly in vars.xml.
> >> >>>
> >> >>> By the way, I'm running in proxy-media mode, but I did try it with
> >> >>> bypass-media as well. Same symptoms, same behaviour.
> >> >>>
> >> >>> Any help with this would be highly appreciated.
> >> >>>
> >> >> Well, I do know that when you do a hangup in the dialplan you can
> pass
> >> >> an
> >> >> optional cause as well:
> >> >> <action application="hangup" data="USER_BUSY"/>
> >> >> If you are doing the hanging up then you have a fair amount of
> >> >> control...
> >> >> -MC
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> >
> >> > --
> >> > Ahmed Naji
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Ahmed Naji
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From kond at nstel.ru  Wed Jan 13 06:10:02 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Wed, 13 Jan 2010 17:10:02 +0300
Subject: [Freeswitch-users] sip trunk question: why call through external
	profile is challenged?
Message-ID: <20100113141003.970B311F32@mail.nstel.ru>

Hi all!

 

I'm brand new to FreeSwitch, but have some experience with SipX.

Our company is evaluating FS.

For test purposes I set up FS on a virtual machine (vmware esxi). I use
CentOS.

The FS version I use is 1.0.5-20100110-0400.

 

I have a question regarding sip trunk between FS and SipX.

I created the following GW in external profile: 

[freeswitch at freeswitch external]$ cat sipx-lab.xml | grep -v '<!--'

<include>

  <gateway name="sipx4.lab.nstel.ru">

  <param name="username" value="zxcv"/>

  <param name="password" value="2007"/>

  <param name="register" value="false"/>

  </gateway>

</include>

 

External.xml file is not modified after installation.

 

I see this gateway via fs_cli:

freeswitch at internal> sofia status gateway sipx4.lab.nstel.ru

============================================================================
=====================

Name            sipx4.lab.nstel.ru

Profile         external

Scheme          Digest

Realm           sipx4.lab.nstel.ru

Username        zxcv

Password        yes

>From            <sip:zxcv at sipx4.lab.nstel.ru;transport=udp>

Contact
<sip:gw+sipx4.lab.nstel.ru at 172.23.22.49:5080;transport=udp;gw=sipx4.lab.nste
l.ru>

Exten           zxcv

To              sip:zxcv at sipx4.lab.nstel.ru

Proxy           sip:sipx4.lab.nstel.ru

Context         public

Expires         3600

Freq            3600

Ping            0

PingFreq        0

PingState       0/0/0

State           NOREG

Status          UP

CallsIN         0

CallsOUT        0

============================================================================
=====================

 

I created new FS extension 2853. I registers (xlite) and I can call it.

 

Now I want to call FS user from sipx.

 

>From the sipx side one can configure link to FS differently. There are two
options:

1. Call FS directly trough sipxproxy (the core part of sipx works as sip
proxy, not as b2bua)

2. Call trough embedded b2bua, named sipxbridge.

 

When a call is going trough sipxbridge, it is successfully landed at FS
extension 2853.

But when a call is going from the sipxproxy, it is challenged with Status
407 "proxy authentication required', and then call fails.

(I'm not sure if sipx should handle this challenge, but this is separate
question for the sipx forum).

At the default log level I see nothing in the freeswitch.log.

 

So the question is why one call through external profile is being challenged
while the other is not?

I suspect that the reason is in the difference in the two Invite messages:

 

Here is challenged Invite:

INVITE sip:2853 at fs.lab.nstel.ru:5080;transport=udp SIP/2.0

Record-Route:
<sip:172.23.12.104:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EOTRiZjI3MjgtYWMxN
zBjZTktMTNjNC0yZjZkZWMtNjNkNmUwN2UtMmY2ZGVj.900_ntap%2Aid%7EMjIyNjAtMQ%60%60
%213ba6d85f6946c4c6001bee1d3b54474f>

From:
"testphone3857"<sip:3857 at lab.nstel.ru>;tag=94bf2728-ac170ce9-13c4-2f6dec-63d
6e07e-2f6dec

To: <sip:2853 at lab.nstel.ru>

Call-Id: 94bed2a0-ac170ce9-13c4-2f6dec-68ffdb81-2f6dec at lab.nstel.ru

Cseq: 2 INVITE

Via: SIP/2.0/UDP
172.23.12.104;branch=z9hG4bK-sipXecs-000eaf48a36fe4029c7cde004a6f44424847

Via: SIP/2.0/TCP
172.23.12.104;branch=z9hG4bK-sipXecs-000bf3008cfa3f3e70470789c75232ba9499~ac
7fd729330fd563f83cacd941311e75;id=22260-1

Via: SIP/2.0/UDP 172.23.12.233:5060;branch=z9hG4bK-2f6dec-b9456412-7ad07784

Max-Forwards: 18

Supported: replaces

User-Agent: LG-Nortel LIP 6812 v1.2.38sp SN/00405A18B634

Contact: <sip:3857 at 172.23.12.233:5060;x-sipX-nonat>

Proxy-Authorization: Digest
username="3857",realm="lab.nstel.ru",nonce="e2152722611af1bbe59f3a8eb31b8eb8
4b4db2ca",uri="sip:2853 at lab.nstel.ru",response="a09c34fbd897783c9b9af50bd044
ccaa",algorithm=MD5

Content-Type: application/sdp

Content-Length: 301

Date: Wed, 13 Jan 2010 11:47:22 GMT

Expires: 60

X-Sipx-Handled: X172.23.12.104-81.211.30.104

 

v=0

o=LGEIPP 16246 16247 IN IP4 172.23.12.233

s=SIP Call

c=IN IP4 172.23.12.233

t=0 0

m=audio 23008 RTP/AVP 0 8 18 4 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:18 annexb=no

a=fmtp:101 0-11

a=sendrecv

 

 

And here is successful Invite:

INVITE sip:2853 at fs.lab.nstel.ru;user=phone SIP/2.0

Call-ID: 94bec3b8-ac170ce9-13c4-2f6d47-2fee09d0-2f6d47 at lab.nstel.ru.0

CSeq: 1 INVITE

From: "testphone3857" <sip:3857 at 172.23.12.104>;tag=816640414244159961

To: <sip:2853 at fs.lab.nstel.ru;user=phone>

Via: SIP/2.0/UDP
172.23.12.104:5080;branch=z9hG4bK885dde2c8680f5315845cd3350b8b605373534

Max-Forwards: 70

User-Agent: sipXecs/4.0.2 sipXecs/sipxbridge (Linux)

P-Asserted-Identity: <sip:1004 at 172.23.12.104>

Contact: <sip:1004 at 172.23.12.104:5080;transport=udp>

Route: <sip:172.23.22.49:5080;transport=udp;lr>

Session-Expires: 1800;refresher=uac

Allow: INVITE,BYE,ACK,CANCEL,OPTIONS

Content-Type: application/sdp

Content-Length: 315

 

v=0

o=sipxbridge 6640787141824452741 1 IN IP4 172.23.12.104

s=SIP Call

c=IN IP4 172.23.12.104

t=0 0

m=audio 30248 RTP/AVP 0 8 18 4 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:18 annexb=no

a=fmtp:101 0-11

a=sendrecv

 

Can somebody please if it is a FS configuration problem or a software
problem or is it a problem on the sipx side?

 

Thanks in advance,

Nikolay.

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From juanbackson at gmail.com  Wed Jan 13 06:19:33 2010
From: juanbackson at gmail.com (Juan Backson)
Date: Wed, 13 Jan 2010 22:19:33 +0800
Subject: [Freeswitch-users] need help with switch_ivr_originate
Message-ID: <27c25bc41001130619g3213e37bo8f79e6ee1984701a@mail.gmail.com>

Hi,

I tried to use the function switch_ivr_originate to originate a call instead
of using the "bridge" app.

However, I am getting "ORIGINATOR CANCEL" error after switch_ivr_originate
is executed.

Does anyone know why would freeswitch cancel the call ?

I have checked all sample usage of switch_ivr_originate in the code but no
luck.

thx,
jb
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From steveayre at gmail.com  Wed Jan 13 06:25:32 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 13 Jan 2010 14:25:32 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <cd93b3701001130551v26fe6355w77f532a86ba4e08a@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
	<87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
	<cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>
	<eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>
	<cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>
	<eafa772f1001120520i719751b9hcbc22a969a89a585@mail.gmail.com>
	<cd93b3701001130551v26fe6355w77f532a86ba4e08a@mail.gmail.com>
Message-ID: <eafa772f1001130625r14c40cb7q584f5f84a175e21b@mail.gmail.com>

"I wonder if it's possible to rewrite causes from such bridges by
executing a JS or similar app attached to the bridge. I'll report on
this as and when."

If you run a LUA (or JS, etc) script where the hangup app currently
is, then you'll be able to check the current hangup cause at that
point within the script and conditionally run the hangup app from your
script with the cause you want to rewrite to.

-Steve

2010/1/13 Ahmed Naji <a.alalousi at gmail.com>:
> Steve/All,
>
> Thanks for all your feedback, this thread can be closed.
>
> Here is some feedback:
>
> Re: Steve's queries:
>
> - "pproxy" was a typo on my part, but it should not affect anything. I set
> them both to false in vars.xml, and don't over-ride them anywhere.
> - true re: default values for proxy_media & bypass_media being false. I am
> explicitly setting them here out of 1) paranoia and 2) I like to make sure I
> know my variable values and not leave them to defaults - relics of being a
> developer
> - I was disabling the q850 code as part of my attempts to crack this nut.
>
> Re: the solution, I've managed to rewrite some of the codes with a call to
> the hangup app, which is what Steve is also using, and his findings re:
> bridges getting 183 and 180 before 4xx.
>
> I wonder if it's possible to rewrite causes from such bridges by executing a
> JS or similar app attached to the bridge. I'll report on this as and when.
>
> Regards,
>
> Ahmed.
>
>
>
>
> 2010/1/12 Steven Ayre <steveayre at gmail.com>
>>
>> - prpoxy_media should be proxy_media
>> - bypass_media and proxy_media shouldn't need setting to false -
>> that's their default (unless you're set one of them to true on the sip
>> profile?)
>> - why do you need to disable q850 reason?
>>
>> I do something very similar - try this...
>>
>> <condition field="destination_number" expression="^(?:5824)(44(1|2)\d+)$">
>> ?<action application="set" data="continue_on_fail=true"/>
>> ?<action application="set" data="hangup_after_bridge=true"/>
>> ?<action application="set" data="sip_ignore_remote_cause=true"/>
>> ?<action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1"/>
>> ?<action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
>> </condition>
>>
>> By the way, you'll be unable to rewrite the hangup cause for a bridge
>> that gets a 180 or 183 packet from the gateway before getting a 4xx,
>> 5xx or 6xx packet (because those bridges don't 'fail') - I've seen
>> this on a SIP provider before who gives 183 Session Progress before a
>> 404 Not Found if the PSTN number dialled doesn't exist.
>>
>> Regards,
>> -Steve
>>
>>
>> 2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
>> > Here you go:
>> >
>> > <condition field="destination_number"
>> > expression="^(?:5824)(44(1|2)\d+)$"
>> > break="on-true">
>> > ? <action inline="true" application="set" data="bypass_media=false"/>
>> > ? <action inline="true" application="set" data="prpoxy_media=false"/>
>> > ? <action inline="true" application="set"
>> > data="disable_q850_reason=true"/>
>> > ? <action inline="true" application="set"
>> > data="hangup_after_bridge=false"/>
>> > ? <action inline="true" application="set"
>> > data="sip_ignore_remote_cause=true"/>
>> > ? <action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1"/>
>> > ? <action inline="true" application="set"
>> > data="bridge_hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
>> > ? <action inline="true" application="set"
>> > data="hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
>> > ? <action application="bridge" data="error/normal_circuit_congestion"/>
>> > ? <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
>> > </condition>
>> >
>> > As you can see, I am trying to rewrite the hangup codes in a multitude
>> > of
>> > ways and places, but still exhibit the same behaviour.
>> >
>> > Any help appreciated.
>> >
>> > Regards,
>> >
>> > Ahmed.
>> >
>> > 2010/1/12 Steven Ayre <steveayre at gmail.com>
>> >>
>> >> Can you show us the dialplan extension you're trying?
>> >>
>> >> Thanks,
>> >> -Steve
>> >>
>> >> 2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
>> >> > Hi Michael,
>> >> >
>> >> > This is exactly what I'm doing, but it's just not happening.
>> >> >
>> >> > Thanks,
>> >> >
>> >> > Ahmed.
>> >> >
>> >> >
>> >> > 2010/1/12 Michael Collins <msc at freeswitch.org>
>> >> >>
>> >> >>
>> >> >> On Mon, Jan 11, 2010 at 4:18 AM, Ahmed Naji <a.alalousi at gmail.com>
>> >> >> wrote:
>> >> >>>
>> >> >>> Dear All,
>> >> >>>
>> >> >>> I posted a thread re the subject but didn't get any joy, so perhaps
>> >> >>> second time lucky.
>> >> >>>
>> >> >>> I need to rewrite a couple of hangup causes to mean
>> >> >>> NORMAL_CONGESTION
>> >> >>> and
>> >> >>> no matter which variables I set, this isn't happening. The idea is
>> >> >>> to
>> >> >>> control what codes are returned to an end point after a successful
>> >> >>> bridge,
>> >> >>> as well as deal with what codes are returned if the bridge is
>> >> >>> unsuccessful
>> >> >>> (e.g. user_busy, originator_cancel ...etc).
>> >> >>>
>> >> >>> I've had limited success by setting hangup_after_bridge=false then
>> >> >>> bridging to error/<required_code>. This, however only works when
>> >> >>> the
>> >> >>> B-leg
>> >> >>> terminates the call after a successful answer. Any other codes are
>> >> >>> not
>> >> >>> rewritten.
>> >> >>>
>> >> >>> I've also tried playing with the bridge_hangup_code and hangup_code
>> >> >>> variables prior and after bridging, still no joy. I have also set
>> >> >>> sip_ignore_remote_cause=true prior to entering the bridge, as well
>> >> >>> explicitly in vars.xml.
>> >> >>>
>> >> >>> By the way, I'm running in proxy-media mode, but I did try it with
>> >> >>> bypass-media as well. Same symptoms, same behaviour.
>> >> >>>
>> >> >>> Any help with this would be highly appreciated.
>> >> >>>
>> >> >> Well, I do know that when you do a hangup in the dialplan you can
>> >> >> pass
>> >> >> an
>> >> >> optional cause as well:
>> >> >> <action application="hangup" data="USER_BUSY"/>
>> >> >> If you are doing the hanging up then you have a fair amount of
>> >> >> control...
>> >> >> -MC
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> > Ahmed Naji
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Ahmed Naji
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From dftoro at yahoo.com  Wed Jan 13 06:44:13 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Wed, 13 Jan 2010 06:44:13 -0800 (PST)
Subject: [Freeswitch-users] sangoma echo cancellation software
In-Reply-To: <4B05F78E.3080007@coppice.org>
Message-ID: <796826.47173.qm@web33506.mail.mud.yahoo.com>

greetings,
?
I have sangoma analog card without echo cancellation hardware, I have problems with echo on calls, does FreeSwitch support echo cancellation software on windows ?
?
Thanks

Diego Toro
http://lacarretade.blogspot.com


      
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From a.alalousi at gmail.com  Wed Jan 13 07:13:09 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Wed, 13 Jan 2010 15:13:09 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <eafa772f1001130625r14c40cb7q584f5f84a175e21b@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
	<87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
	<cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>
	<eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>
	<cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>
	<eafa772f1001120520i719751b9hcbc22a969a89a585@mail.gmail.com>
	<cd93b3701001130551v26fe6355w77f532a86ba4e08a@mail.gmail.com>
	<eafa772f1001130625r14c40cb7q584f5f84a175e21b@mail.gmail.com>
Message-ID: <cd93b3701001130713x5078457fpbf56b5ee45238e6e@mail.gmail.com>

Precisely my line of thought Steve. I'm almost done with a LUA to do the
call setup/tare-down and manipulate hangup causes as needed.

While I'm at it, I think I may as well handle codec prefs ...etc. within the
same script.

Will let you and the others how things progress.

BR,

Ahmed.

2010/1/13 Steven Ayre <steveayre at gmail.com>

> "I wonder if it's possible to rewrite causes from such bridges by
> executing a JS or similar app attached to the bridge. I'll report on
> this as and when."
>
> If you run a LUA (or JS, etc) script where the hangup app currently
> is, then you'll be able to check the current hangup cause at that
> point within the script and conditionally run the hangup app from your
> script with the cause you want to rewrite to.
>
> -Steve
>
> 2010/1/13 Ahmed Naji <a.alalousi at gmail.com>:
> > Steve/All,
> >
> > Thanks for all your feedback, this thread can be closed.
> >
> > Here is some feedback:
> >
> > Re: Steve's queries:
> >
> > - "pproxy" was a typo on my part, but it should not affect anything. I
> set
> > them both to false in vars.xml, and don't over-ride them anywhere.
> > - true re: default values for proxy_media & bypass_media being false. I
> am
> > explicitly setting them here out of 1) paranoia and 2) I like to make
> sure I
> > know my variable values and not leave them to defaults - relics of being
> a
> > developer
> > - I was disabling the q850 code as part of my attempts to crack this nut.
> >
> > Re: the solution, I've managed to rewrite some of the codes with a call
> to
> > the hangup app, which is what Steve is also using, and his findings re:
> > bridges getting 183 and 180 before 4xx.
> >
> > I wonder if it's possible to rewrite causes from such bridges by
> executing a
> > JS or similar app attached to the bridge. I'll report on this as and
> when.
> >
> > Regards,
> >
> > Ahmed.
> >
> >
> >
> >
> > 2010/1/12 Steven Ayre <steveayre at gmail.com>
> >>
> >> - prpoxy_media should be proxy_media
> >> - bypass_media and proxy_media shouldn't need setting to false -
> >> that's their default (unless you're set one of them to true on the sip
> >> profile?)
> >> - why do you need to disable q850 reason?
> >>
> >> I do something very similar - try this...
> >>
> >> <condition field="destination_number"
> expression="^(?:5824)(44(1|2)\d+)$">
> >>  <action application="set" data="continue_on_fail=true"/>
> >>  <action application="set" data="hangup_after_bridge=true"/>
> >>  <action application="set" data="sip_ignore_remote_cause=true"/>
> >>  <action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1"/>
> >>  <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
> >> </condition>
> >>
> >> By the way, you'll be unable to rewrite the hangup cause for a bridge
> >> that gets a 180 or 183 packet from the gateway before getting a 4xx,
> >> 5xx or 6xx packet (because those bridges don't 'fail') - I've seen
> >> this on a SIP provider before who gives 183 Session Progress before a
> >> 404 Not Found if the PSTN number dialled doesn't exist.
> >>
> >> Regards,
> >> -Steve
> >>
> >>
> >> 2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
> >> > Here you go:
> >> >
> >> > <condition field="destination_number"
> >> > expression="^(?:5824)(44(1|2)\d+)$"
> >> > break="on-true">
> >> >   <action inline="true" application="set" data="bypass_media=false"/>
> >> >   <action inline="true" application="set" data="prpoxy_media=false"/>
> >> >   <action inline="true" application="set"
> >> > data="disable_q850_reason=true"/>
> >> >   <action inline="true" application="set"
> >> > data="hangup_after_bridge=false"/>
> >> >   <action inline="true" application="set"
> >> > data="sip_ignore_remote_cause=true"/>
> >> >   <action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1
> "/>
> >> >   <action inline="true" application="set"
> >> > data="bridge_hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
> >> >   <action inline="true" application="set"
> >> > data="hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
> >> >   <action application="bridge"
> data="error/normal_circuit_congestion"/>
> >> >   <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
> >> > </condition>
> >> >
> >> > As you can see, I am trying to rewrite the hangup codes in a multitude
> >> > of
> >> > ways and places, but still exhibit the same behaviour.
> >> >
> >> > Any help appreciated.
> >> >
> >> > Regards,
> >> >
> >> > Ahmed.
> >> >
> >> > 2010/1/12 Steven Ayre <steveayre at gmail.com>
> >> >>
> >> >> Can you show us the dialplan extension you're trying?
> >> >>
> >> >> Thanks,
> >> >> -Steve
> >> >>
> >> >> 2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
> >> >> > Hi Michael,
> >> >> >
> >> >> > This is exactly what I'm doing, but it's just not happening.
> >> >> >
> >> >> > Thanks,
> >> >> >
> >> >> > Ahmed.
> >> >> >
> >> >> >
> >> >> > 2010/1/12 Michael Collins <msc at freeswitch.org>
> >> >> >>
> >> >> >>
> >> >> >> On Mon, Jan 11, 2010 at 4:18 AM, Ahmed Naji <a.alalousi at gmail.com
> >
> >> >> >> wrote:
> >> >> >>>
> >> >> >>> Dear All,
> >> >> >>>
> >> >> >>> I posted a thread re the subject but didn't get any joy, so
> perhaps
> >> >> >>> second time lucky.
> >> >> >>>
> >> >> >>> I need to rewrite a couple of hangup causes to mean
> >> >> >>> NORMAL_CONGESTION
> >> >> >>> and
> >> >> >>> no matter which variables I set, this isn't happening. The idea
> is
> >> >> >>> to
> >> >> >>> control what codes are returned to an end point after a
> successful
> >> >> >>> bridge,
> >> >> >>> as well as deal with what codes are returned if the bridge is
> >> >> >>> unsuccessful
> >> >> >>> (e.g. user_busy, originator_cancel ...etc).
> >> >> >>>
> >> >> >>> I've had limited success by setting hangup_after_bridge=false
> then
> >> >> >>> bridging to error/<required_code>. This, however only works when
> >> >> >>> the
> >> >> >>> B-leg
> >> >> >>> terminates the call after a successful answer. Any other codes
> are
> >> >> >>> not
> >> >> >>> rewritten.
> >> >> >>>
> >> >> >>> I've also tried playing with the bridge_hangup_code and
> hangup_code
> >> >> >>> variables prior and after bridging, still no joy. I have also set
> >> >> >>> sip_ignore_remote_cause=true prior to entering the bridge, as
> well
> >> >> >>> explicitly in vars.xml.
> >> >> >>>
> >> >> >>> By the way, I'm running in proxy-media mode, but I did try it
> with
> >> >> >>> bypass-media as well. Same symptoms, same behaviour.
> >> >> >>>
> >> >> >>> Any help with this would be highly appreciated.
> >> >> >>>
> >> >> >> Well, I do know that when you do a hangup in the dialplan you can
> >> >> >> pass
> >> >> >> an
> >> >> >> optional cause as well:
> >> >> >> <action application="hangup" data="USER_BUSY"/>
> >> >> >> If you are doing the hanging up then you have a fair amount of
> >> >> >> control...
> >> >> >> -MC
> >> >> >>
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >>
> >> >> >
> >> >> >
> >> >> >
> >> >> > --
> >> >> > Ahmed Naji
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > --
> >> > Ahmed Naji
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Ahmed Naji
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From jeff at jefflenk.com  Wed Jan 13 07:20:12 2010
From: jeff at jefflenk.com (Jeff Lenk )
Date: Wed, 13 Jan 2010 15:20:12 +0000
Subject: [Freeswitch-users] sangoma echo cancellation software
Message-ID: <COL117-DS9F4F51BA87712328B80E3B96B0@phx.gbl>

No such luck - I dont know any solution other than get the other card


-----Original Message-----
From: Diego Toro
Sent: 1/13/2010 2:44:13 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] sangoma echo cancellation software
greetings,

I have sangoma analog card without echo cancellation hardware, I have problems with echo on calls, does FreeSwitch support echo cancellation software on windows ?

Thanks

Diego Toro
http://lacarretade.blogspot.com


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From jcasale at activenetwerx.com  Wed Jan 13 07:30:10 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 13 Jan 2010 15:30:10 +0000
Subject: [Freeswitch-users] Svar: Re:  Multi-Homed setup,
	starting over	 - still not working
In-Reply-To: <4B4DC035020000E1000003A2@mail.fribert.dk>
References: <4B4DC035020000E1000003A2@mail.fribert.dk>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9079C65@prato.activenetwerx.local>

>But are you sayng that "eval ${domain}" will do more than is done when I restart the fs?
>I don't see that anywhere in the guide :-D
>
>I'll try it out asap!

What I am saying is it "evaluates" that placeholder and shows
its value. You can use it to show the current value to verify
it matches what you intend, which is for it to be the same as
your LAN ip.


From devel at thom.fr.eu.org  Wed Jan 13 07:32:16 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Wed, 13 Jan 2010 16:32:16 +0100
Subject: [Freeswitch-users] =?utf-8?q?playback_and_play=5Fand=5Fget=5Fdigi?=
 =?utf-8?q?ts_strange_misunderstanding?=
Message-ID: <cdeeb122e98434d6303df4b05f8cbada@thom.fr.eu.org>



Hello, 

trying to make so dialplan extensions that use the playbvack
and play_and_get_digits applications, but I'm having trouble with the file
name specification. 

The files I want to play are in the french language
(fr/fr/julie as configured in lang/fr/fr.xml) 

My extension is as follows
: 

The channel is using a bit rate of 8000 Hz, so by the set
default_language=fr I would expect freeswitch to playback the file at
${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/8000/ringing_disabled.wav
whereas it tries to playback the file at
${FREESWITCH_PATH}/sounds/en/us/callie/misc/ringing_disabled.wav 

I have
the same with play_and_get_digits application. 

What am I doing wrong ?


Fran?ois
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From Claudio.Cavalera at italtel.it  Wed Jan 13 07:51:10 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Wed, 13 Jan 2010 16:51:10 +0100
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0F362E0@BESONE.corp.dom>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F362E4@BESONE.corp.dom>

Also after
api = freeswitch.API();
what is the difference between
api:execute()
 
and
api:executeString() ?
 
I've found example of both uses in
http://wiki.freeswitch.org/wiki/Mod_lua
and in 
http://wiki.freeswitch.org/wiki/Make_API_calls_directly_from_Lua_code
 
Thanks,
Claudio
 
 


________________________________

	From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Cavalera Claudio Luigi
	Sent: Wednesday, January 13, 2010 11:22 AM
	To: freeswitch-users at lists.freeswitch.org
	Subject: Re: [Freeswitch-users] playing with sessions in lua
	
	
	Thanks a lot Anthony,
	some comments inline (and please forgive me for my broken email
client).
	 
	>>  example1: Consider this simple lua script in which i create
two sessions:
	
	 >>  api = freeswitch.API();
	 >>  api:execute("originate", "sofia/internal/1001%192.168.1.1
&park");
	 >>  api:execute("originate", "sofia/internal/1002%192.168.1.1
&park");
	
	 >  capture the output from api:execute the uuid is in there
	
	Thx a lot,
	this was one piece i was missing although it's already on the
wiki here:
	http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls 
	


	 > because lua calls it freeswitch.bridge
	
	 > session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
	 > session2 =
freeswitch.Session("sofia/internal/1002%192.168.1.1", session1);
	 > freeswitch.bridge(session1, session2); 
	 
	good to now, there isn't any example of freeswitch.bridge in the
wiki and i would like to add one.
	Where I could find the full api of
	freeswitch.Session( ) ?
	because I've seen this working also without "session1" in the
second line:
	session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
	session2 =
freeswitch.Session("sofia/internal/1002%192.168.1.1");
	freeswitch.bridge(session1, session2); 
	
	 also is there any difference between  freeswitch.bridge and
freeswitch.execute(uuid_bridge ...) ?
	 
	 >> example3: yet another possibility
	 >> local session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
	 >> session1:originate(session2,
"sofia/internal/1002%192.168.1.1", 1000);
	 >> but it does not work either.
	
	 > The above is gibberish try:
	 > local session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
	 > session1:execute("bridge",
"sofia/internal/1002%192.168.1.1"); 
	 
	Okay i will report also this bridge example on the wiki which
was missing.
	But does session:originate make sense in some cases or not?
Otherwise i'm going to remove this line on the wiki
	 http://wiki.freeswitch.org/wiki/Mod_lua#session:originate
<http://wiki.freeswitch.org/wiki/Mod_lua#session:originate> 
	 
	Thanks,
	Claudio

	 

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From scott.torr.fs at letterboxes.org  Wed Jan 13 07:56:03 2010
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Thu, 14 Jan 2010 02:56:03 +1100
Subject: [Freeswitch-users] Inbound DTMF Not Recognized In Latest Version
In-Reply-To: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com>
References: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com>
Message-ID: <1263398163.29661.1354446823@webmail.messagingengine.com>

Hi Jerrey,

>From your debug information:
2010-01-12 14:50:27.778790 [DEBUG] switch_ivr_async.c:1532 DTMF
DETECTED: [5]

This is the FS software 'inband' detection of the digit 5 DTMF tone as a
result of issuing 'start_dtmf'.


2010-01-12 14:50:28.038716 [DEBUG] mod_openzap.c:672 queue DTMF [5]

mod_openzap.c -- OPENZAP Endpoint Module
This is the A101D Sangoma Card detection of the digit 5 DTMF tone. I
assume your card has the 'DTMF decoding and tone recognition' optional
DSP daughterboard installed.


So there is no need to use <action application="start_dtmf" /> which
will start the FS software 'inband' DTMF tone detection algorithm.


If someone would like to confirm this, please do.


regards,
Scott Torr


On Tue, 12 Jan 2010 15:10 -0800, "Jerry Richards"
<jerry.richards at teotech.com> wrote:
> 
> I am having a problem where in inbound call from the PSTN going to the
> demo
> IVR does not recognize DTMF digits.  I posted a log at
> http://pastebin.freeswitch.org/11769.  I am running Freeswitch
> (freeswitch-1.0.5-20100112-0400.tar.gz) with an A101D Sangoma Card and
> wanpipe driver: wanpipe-3.5.10.tgz.  I am not using libpri.
> 
> It appears that each DTMF digit might be getting detected twice.  I tried
> to
> dial "5401" and it thought I dialed "5544".
> 
> I am following the instructions at
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf, as shown
> below.  Also, I tried moving the "start_dtmf" statement in between all
> the
> statements within the condition tag, but it didn't make any difference.
> 
> <extension name="ivr_demo">
>    <condition field="destination_number" expression="5000">
>       <action application="answer"/>
>       <action application="start_dtmf" />
>       <action application="sleep" data="2000"/>
>       <action application="ivr" data="demo_ivr"/>
>    </condition>
> </extension>
> 
> Also, I noticed at line #293, the log is saying:
> 
> [ERR] mod_sndfile.c:194 Error Opening File
> [/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav]
> [System
> error : No such file or directory.]
> 
> Has anyone else encountered this issue?
> 
> Best Regards,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From jerry.richards at teotech.com  Wed Jan 13 08:00:29 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 13 Jan 2010 08:00:29 -0800
Subject: [Freeswitch-users] Inbound DTMF Not Recognized In Latest Version
In-Reply-To: <44468DAE-9201-40AF-ACA3-E634E6682F04@freeswitch.org>
References: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com>
	<44468DAE-9201-40AF-ACA3-E634E6682F04@freeswitch.org>
Message-ID: <46044FFE91874AA79BA08DC8EE31159E@greyhawk.tonecommander.com>

These are my install steps.  

1) ./bootstrap.sh
2) ./configure
3) uncomment ../../libs/openzap/mod_openzap in ./modules.conf
4) make
5) make install sounds-install moh-install hd-sounds-install hd-moh-install
uhd-sounds-install uhd-moh-install
6) contrib/trixter/makemodconf.pl modules.conf >
/usr/local/freeswitch/conf/autoload_configs/modules.conf.xml 

Are these correct?

Thanks,
Jerry


-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Tuesday, January 12, 2010 3:26 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Inbound DTMF Not Recognized In Latest
Version

looks like you don't have the latest sound files installed.

/b

On Jan 12, 2010, at 5:10 PM, Jerry Richards wrote:

> [ERR] mod_sndfile.c:194 Error Opening File 
> [/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] 
> [System error : No such file or directory.]






From brian at freeswitch.org  Wed Jan 13 08:00:55 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 10:00:55 -0600
Subject: [Freeswitch-users] Inbound DTMF Not Recognized In Latest Version
In-Reply-To: <1263398163.29661.1354446823@webmail.messagingengine.com>
References: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com>
	<1263398163.29661.1354446823@webmail.messagingengine.com>
Message-ID: <2A012832-A0B1-4DBC-ABB3-85FCBB71810A@freeswitch.org>

If you're using OpenZAP you don't need start_dtmf thats mostly for RTP streams that are inband and OpenZAP does the DTMF bits for you if you start that you'll end up with double digits most likely.

/b

On Jan 13, 2010, at 9:56 AM, Scott Torr wrote:

> If someone would like to confirm this, please do.



From brian at freeswitch.org  Wed Jan 13 08:04:06 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 10:04:06 -0600
Subject: [Freeswitch-users] Inbound DTMF Not Recognized In Latest Version
In-Reply-To: <46044FFE91874AA79BA08DC8EE31159E@greyhawk.tonecommander.com>
References: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com>
	<44468DAE-9201-40AF-ACA3-E634E6682F04@freeswitch.org>
	<46044FFE91874AA79BA08DC8EE31159E@greyhawk.tonecommander.com>
Message-ID: <5553C06E-B1B2-428B-B140-CF3044169470@freeswitch.org>

Yes but you can just do uhd-sounds-install uhd-moh-install they include everything below it.

/b

On Jan 13, 2010, at 10:00 AM, Jerry Richards wrote:

> These are my install steps.  
> 
> 1) ./bootstrap.sh
> 2) ./configure
> 3) uncomment ../../libs/openzap/mod_openzap in ./modules.conf
> 4) make
> 5) make install sounds-install moh-install hd-sounds-install hd-moh-install
> uhd-sounds-install uhd-moh-install
> 6) contrib/trixter/makemodconf.pl modules.conf >
> /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml 
> 
> Are these correct?
> 
> Thanks,
> Jerry



From moises.silva at gmail.com  Wed Jan 13 08:13:14 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 13 Jan 2010 11:13:14 -0500
Subject: [Freeswitch-users] Sangoma PRI installation for FreeSWITCH
In-Reply-To: <369c72d81001122021o6885913cl618965791aec4621@mail.gmail.com>
References: <369c72d81001122021o6885913cl618965791aec4621@mail.gmail.com>
Message-ID: <c4d05cbe1001130813q26ba6fc9r846aaad7cf01c5f5@mail.gmail.com>

Hi Magesh,

On Tue, Jan 12, 2010 at 11:21 PM, Magesh R <magesh.freeswitch at gmail.com>wrote:

>       Next I have started the smg_ctrl, but it failed to start. It prints
> the following things,
> smg_ctrl start
>
> Starting processes...
> Loading SCTP...OK
> Starting sangoma_prid...OK
> sangoma_prid failed to start
> check /var/log/sangoma_mgd.log for errors
>
> Stopping running processes...
> safe_sangoma not running...
> sangoma_prid is stopped
> Removing PID files...done
>
> I have checked /var/log/sangoma_mgd.log file. But nothing was there.
>
> Could any please tell me where I made mistake?
>

Check that you have local2.* /var/log/sangoma_mgd.log in your syslog
configuration (which Linux distribution are you using?), and then restart
the syslog service, otherwise nothing will be logged.

If you still can't get the logs after doing that, you can try to start
manually sangoma_prid by doing this:

# export LD_LIBRARY_PATH=/usr/lib/sangoma_prid
# /usr/sbin/sangoma_prid &> /tmp/sangoma_prid.log

This will try to start sangoma_prid and save all the output to
/tmp/sangoma_prid.log, try doing that and then paste (
http://pastebin.freeswitch.org/) the resulting log.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From jerry.richards at teotech.com  Wed Jan 13 08:15:58 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 13 Jan 2010 08:15:58 -0800
Subject: [Freeswitch-users] Sangoma PRI installation for FreeSWITCH
In-Reply-To: <369c72d81001122021o6885913cl618965791aec4621@mail.gmail.com>
References: <369c72d81001122021o6885913cl618965791aec4621@mail.gmail.com>
Message-ID: <B4A27408A33F4E74A4DBEA492B906C20@greyhawk.tonecommander.com>

Here are my instructions to install the wanpipe driver and I have not had
your problem:
 
1)       Download the following Wanpipe driver tarball wanpipe-<version>.tgz

2)       Store the wanpipe-<version>.tgz file in the /opt folder.

3)       tar xvfz wanpipe-<version>.tgz

4)       cd wanpipe-<version>

5)       make openzap

6)       make install

7)       make install_pri

8)       wanrouter hwprobe     // confirms successful wanpipe installation

9)       /usr/sbin/wancfg_fs   // starts wanpipe configuration utility

10)   1=NO            // Change Freeswitch Configuration Directory

11)   1=T1            // Select Media Type

12)   1=YES (keep)    // Configure Port 1 T1, B8ZS, ESF

13)   1=NORMAL        // Select Clock

14)   1=NATIONAL      // Select Switchtype

15)   1=CPE           // Select Signaling Type

16)   1               // Input Group #

17)   1=YES           // Enable H/W DTMF Detect

18)   1=YES           // Enable FAX Detect

19)   1=YES=CONTINUE  // Configuration Complete

20)   1=YES           // Save Configuration

21)   1=YES           // Wanrouter start on reboot

22)   1=YES           // smg_ctrl start/stop on wanrouter start

23)   Done.

 
Jerry


  _____  

From: Magesh R [mailto:magesh.freeswitch at gmail.com] 
Sent: Tuesday, January 12, 2010 8:21 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Sangoma PRI installation for FreeSWITCH


Dear All,
         I have installed Sangoma PRI card in machine with following steps,
                 * wget
ftp://ftp.sangoma.com/linux/custom/3.5/wanpipe-3.5.8.7.tgz
                 * tar -xvfz wanpipe-3.5.8.7.tgz
                 * cd wanpipe-3.5.8.7
                 * make openzap
                 * make install
                 * make install_pri
      
           I have executed "wanrouter hwprobe" command it prints the
following details,
    
           1 . AFT-A102-SH : SLOT=4 : BUS=3 : IRQ=11 : CPU=A : PORT=1 :
HWEC=0 : V=36
           2 . AFT-A102-SH : SLOT=4 : BUS=3 : IRQ=11 : CPU=A : PORT=2 :
HWEC=0 : V=36
        
        Card Cnt: A101-2=1

       Next I have executed wancfg_fs script to configure the sangoma for
freeswitch. 
It creates the following configuration files
                     * wanpipe1.conf 
                     * wanpipe2.conf
                     * smg_prid.conf
                     * openzap.conf
                     * openzap.conf.xml 
       I have attached those files.
       I have started the wanrouter and printed the wanrouter status,
              Wanrouter Status:

Device name | Protocol | Station | Status        |
wanpipe1    | AFT TE1  | N/A     | Connected     |
wanpipe2    | AFT TE1  | N/A     | Disconnected  |

      Next I have started the smg_ctrl, but it failed to start. It prints
the following things,
smg_ctrl start

Starting processes...
Loading SCTP...OK
Starting sangoma_prid...OK
sangoma_prid failed to start
check /var/log/sangoma_mgd.log for errors

Stopping running processes...
safe_sangoma not running...
sangoma_prid is stopped
Removing PID files...done

I have checked /var/log/sangoma_mgd.log file. But nothing was there.

Could any please tell me where I made mistake?

Thanks,
Mag


              



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From anthony.minessale at gmail.com  Wed Jan 13 08:18:42 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Jan 2010 10:18:42 -0600
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
References: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
Message-ID: <191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>

you would have to register signals in your child process too

On Wed, Jan 13, 2010 at 3:13 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi all,
>
> I've done a sample program (In perl ESL) , which play a file to the caller
> and then it will call recvEvent() and print the event name. I've handled
> signals also.
>
> When I send SIGINT to my program (Perl), the signal handler is called and I
> can see the print output. But in the same time,  I received
> SERVER_DISCONNECTED from freeswitch as event.
>
> I don't know why I received SERVER_DISCONNECTED from freeswitch. Is it
> because, the recvEvent() from perl internally  calls the recvevent function
> in the Esl.c and when it waits to receive the information from socket, the
> signal occurred???
>
> Please clarify me!!
>
> Here is my program
> require ESL;
> use IO::Socket::INET;
> use Data::Dumper;
>
> my $ip = "192.168.1.222";
> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
> Proto => 'tcp',  Listen => 1,  Reuse => 1 );
> die "Could not create socket: $!\n" unless $sock;
> &register_Signals();
>
> for(;;) {
>     my $new_sock = $sock->accept();
>     next if (not defined ($new_sock));
>     my $pid = fork();
>     if ($pid) {
>         close($new_sock);
>         next;
>     }
>     print "CHILD PID: $$\n";
>     my $host = $new_sock->sockhost();
>     my $fd = fileno($new_sock);
>
>     my $con = new ESL::ESLconnection($fd);
>     my $info = $con->getInfo();
>
>     my $uuid = $info->getHeader("unique-id");
>
>     printf "Connected call %s, from %s\n", $uuid,
> $info->getHeader("caller-caller-id-number");
>     my $r=$con->execute("answer");
>     print Dumper $r;
>     $con->events("plain","all");
>     my
> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>
>     while($con->connected()) {
>         my $e = $con->recvEvent();
>
>         if ($e) {
>             my $name = $e->getHeader("event-name");
>             print "EVENT [$name]\n";
>             if ($name eq "DTMF") {
>                 my $digit = $e->getHeader("dtmf-digit");
>                 print "$digit\n";
>             }
>         }
>     }
>     close($new_sock);
> }
> sub register_Signals() {
>     foreach ( keys %SIG ) {
>         $SIG{$_} = 'sig_Handler';
>     }
> }
>
> sub sig_Handler() {
>     my $handle=$_[0];
>     if($handle eq "INT") {
>         print "$$: SIGNAL SIG$handle is generated\n";
>     }
> }
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From john at acsol.net  Wed Jan 13 08:38:38 2010
From: john at acsol.net (John)
Date: Wed, 13 Jan 2010 09:38:38 -0700
Subject: [Freeswitch-users] Sip provider Using IP authenication
Message-ID: <4B4DF70E.50007@acsol.net>

I'm am attempting to use Voipinnovations.com (M2Tel) as a SIP provider. 
They do not utilize username/password; how would I go about creating a 
gateway entry for them without username/password? Thanks


From gmaruzz at celliax.org  Wed Jan 13 08:40:27 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 13 Jan 2010 17:40:27 +0100
Subject: [Freeswitch-users] GSM modem for Skype
In-Reply-To: <SNT134-w3842E3232C8D3C80F105EAF47E0@phx.gbl>
References: <mailman.4361.1261716035.2495.freeswitch-users@lists.freeswitch.org>
	<SNT134-w3842E3232C8D3C80F105EAF47E0@phx.gbl>
Message-ID: <7b197bef1001130840k6a2aac9bk9e173c8a79fa3376@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#


2009/12/25 ??? <qinglan_zeng at hotmail.com>:
> Hello All,
>
> I had a GSM VoIP Gateway which can be connected to service providers who use
> SIP protocal. I remember somebody in Freeswitch community mentioned there is
> a software which can connect GSM network with Skype network. I'm not sure if
> this software can work with my GSM VOIP gateway or not.  If somebody can
> send me such software that would be much appriciated and even license fee
> required would be accepted. I hope I can have a testing on this or I can
> send this gateway to those guys who have this kind of software for testing.
>
> Thanks and Merry Christmas.
>
> Daniel Zeng
>
>
>
> ________________________________
> ??????????MSN??? ?????
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmartinr82 at gmail.com  Wed Jan 13 08:41:36 2010
From: gmartinr82 at gmail.com (Martin Rodriguez)
Date: Wed, 13 Jan 2010 13:41:36 -0300
Subject: [Freeswitch-users] FSComm Windows build
In-Reply-To: <1263358274180-4311097.post@n2.nabble.com>
References: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
	<1263350125191-4295967.post@n2.nabble.com>
	<5800526b1001121918r15a336f5jbc5b724e66ac2825@mail.gmail.com>
	<1263358274180-4311097.post@n2.nabble.com>
Message-ID: <fc0ed37b1001130841k11b3ae8ekaeefb227fb4a227f@mail.gmail.com>

Hi List:

Test Installation of this compilation but wanting to run it generates
the following error. I have installed Windows XP SP3.

Attached screenshot.

Bye
Martin Rodriguez


2010/1/13 Jeff Lenk <jeff at jefflenk.com>:
>
> Carlos,
>
> Thanks for those links!
>
> I am currenltly using prebuilt 4.6 32 bit libs and was hoping Nokia prebuilt
> the x64 too. I guess I might have to look into building myself as well.
>
> -Jeff
>
>
> Jeff,
>
> not yet. I'm still linking with a 32bit QT library of 4.5.2 I compiled for
> VS2008 sometime last year. I followed a guide similar to this one:
> http://dcsoft.com/community_server/blogs/dcsoft/archive/2009/03/06/how-to-setup-qt-4-5-visual-studio-integration.aspx
>
> At the time I didn't see a need for a 64bit version of FsGui. Since FSComm
> has FreeSWITCH in the back end this kind of changes things.
>
> Looks like someone put up a wiki on compiling QT for Win x64:
> http://en.wikibooks.org/wiki/Opticks_Developer_Guide/Getting_Started/Building_Qt_From_Source
>
> Now that QT 4.6 is out I might have to revisit a new library build with 64
> bit in mind.
>
> Carlos
>
> --
> View this message in context: http://n2.nabble.com/FSComm-Windows-build-tp4289256p4311097.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Jan 13 08:45:15 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 10:45:15 -0600
Subject: [Freeswitch-users] Sip provider Using IP authenication
In-Reply-To: <4B4DF70E.50007@acsol.net>
References: <4B4DF70E.50007@acsol.net>
Message-ID: <103A262C-672A-445E-832D-386EF944FA0E@freeswitch.org>

You don't.  You just send an invite in the form of sofia/profile/number at ip

/b

On Jan 13, 2010, at 10:38 AM, John wrote:

> I'm am attempting to use Voipinnovations.com (M2Tel) as a SIP provider. 
> They do not utilize username/password; how would I go about creating a 
> gateway entry for them without username/password? Thanks

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From rupa at rupa.com  Wed Jan 13 08:56:44 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 13 Jan 2010 10:56:44 -0600
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0F362E4@BESONE.corp.dom>
References: <A2167AE7633A5C4DAE56EEBD1428ECC0F362E0@BESONE.corp.dom>
	<A2167AE7633A5C4DAE56EEBD1428ECC0F362E4@BESONE.corp.dom>
Message-ID: <b39273531001130856l6f72fe4bmcc61c41b04b2a7c1@mail.gmail.com>

execute("api", "args")
executeString("api args")

one needs the api name as a separate arg

On Wed, Jan 13, 2010 at 9:51 AM, Cavalera Claudio Luigi <
Claudio.Cavalera at italtel.it> wrote:

>  Also after
> api = freeswitch.API();
> what is the difference between
> api:execute()
>
> and
> api:executeString() ?
>
> I've found example of both uses in http://wiki.freeswitch.org/wiki/Mod_lua
> and in
> http://wiki.freeswitch.org/wiki/Make_API_calls_directly_from_Lua_code
>
> Thanks,
> Claudio
>
>
>
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cavalera
> Claudio Luigi
> *Sent:* Wednesday, January 13, 2010 11:22 AM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] playing with sessions in lua
>
>  Thanks a lot Anthony,
> some comments inline (and please forgive me for my broken email client).
>
> >>  example1: Consider this simple lua script in which i create two
> sessions:
>
>  >>  api = freeswitch.API();
>  >>  api:execute("originate", "sofia/internal/1001%192.168.1.1 &park");
>  >>  api:execute("originate", "sofia/internal/1002%192.168.1.1 &park");
>
>  >  capture the output from api:execute the uuid is in there
>
> Thx a lot,
> this was one piece i was missing although it's already on the wiki here:
> http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls
>
>
>  > because lua calls it freeswitch.bridge
>
>  > session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
>  > session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1",
> session1);
>  > freeswitch.bridge(session1, session2);
>
> good to now, there isn't any example of freeswitch.bridge in the wiki and i
> would like to add one.
> Where I could find the full api of
> freeswitch.Session( ) ?
> because I've seen this working also without "session1" in the second line:
> session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
> session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");
> freeswitch.bridge(session1, session2);
>
>  also is there any difference between  freeswitch.bridge
> and freeswitch.execute(uuid_bridge ...) ?
>
>  >> example3: yet another possibility
>  >> local session1 =
> freeswitch.Session("sofia/internal/1001%192.168.1.1");
>  >> session1:originate(session2, "sofia/internal/1002%192.168.1.1", 1000);
>  >> but it does not work either.
>
>  > The above is gibberish try:
>  > local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
>  > session1:execute("bridge", "sofia/internal/1002%192.168.1.1");
>
> Okay i will report also this bridge example on the wiki which was missing.
> But does session:originate make sense in some cases or not? Otherwise i'm
> going to remove this line on the wiki
>  http://wiki.freeswitch.org/wiki/Mod_lua#session:originate
>
> Thanks,
> Claudio
>
>
>
> Internet Email Confidentiality Footer
>
>
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>
> La presente comunicazione, con le informazioni in essa contenute e ogni
> documento o file allegato, e' rivolta unicamente alla/e persona/e cui e'
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> ********************************************************************************************************************************************
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From john at acsol.net  Wed Jan 13 09:02:36 2010
From: john at acsol.net (John)
Date: Wed, 13 Jan 2010 10:02:36 -0700
Subject: [Freeswitch-users] Sip provider Using IP authenication
In-Reply-To: <103A262C-672A-445E-832D-386EF944FA0E@freeswitch.org>
References: <4B4DF70E.50007@acsol.net>
	<103A262C-672A-445E-832D-386EF944FA0E@freeswitch.org>
Message-ID: <4B4DFCAC.6070804@acsol.net>

So you don't need a gateway.xml file in the external defining it? You 
would just need a bridge statement in the dialplan?

Thanks John

On 1/13/2010 9:45 AM, Brian West wrote:
> You don't.  You just send an invite in the form of 
> sofia/profile/number at ip
>
> /b
>
> On Jan 13, 2010, at 10:38 AM, John wrote:
>
>> I'm am attempting to use Voipinnovations.com 
>> <http://Voipinnovations.com/> (M2Tel) as a SIP provider.
>> They do not utilize username/password; how would I go about creating a
>> gateway entry for them without username/password? Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From brian at freeswitch.org  Wed Jan 13 09:06:13 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 11:06:13 -0600
Subject: [Freeswitch-users] Sip provider Using IP authenication
In-Reply-To: <4B4DFCAC.6070804@acsol.net>
References: <4B4DF70E.50007@acsol.net>
	<103A262C-672A-445E-832D-386EF944FA0E@freeswitch.org>
	<4B4DFCAC.6070804@acsol.net>
Message-ID: <632BEFBA-74BA-4479-8C31-029AF7A0A4AB@freeswitch.org>


On Jan 13, 2010, at 11:02 AM, John wrote:

> So you don't need a gateway.xml file in the external defining it?

Nope.

> You would just need a bridge statement in the dialplan?

Yep.

> 
> Thanks John 



From mailinglist at fribert.dk  Wed Jan 13 09:08:36 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Wed, 13 Jan 2010 18:08:36 +0100
Subject: [Freeswitch-users] Svar: Re:  Multi-Homed setup,
	starting	 over	 - still not working
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9079C65@prato.activenetwerx.local>
References: <4B4DC035020000E1000003A2@mail.fribert.dk>
	<CA5A491E9DEFBE4CB777DE97E21575E9079C65@prato.activenetwerx.local>
Message-ID: <4B4E0C24020000E1000003A7@mail.fribert.dk>

Hi Joseph
 
Hmm, you were right, I hadn't set it there, as the guide explains me that I should set it in internal.xml.
After setting it in vars.xml it authenticates as well.
 
So I'm contiuing the quest :-)
 
Thankyou very much for your assistance!
 
/Fribse


>>> 13-01-2010 kl. 16:30 skrev "Joseph L. Casale" <jcasale at activenetwerx.com> i meddelelsen <CA5A491E9DEFBE4CB777DE97E21575E9079C65 at prato.activenetwerx.local>:

>But are you sayng that "eval ${domain}" will do more than is done when I restart the fs?
>I don't see that anywhere in the guide :-D
>
>I'll try it out asap!

What I am saying is it "evaluates" that placeholder and shows
its value. You can use it to show the current value to verify
it matches what you intend, which is for it to be the same as
your LAN ip.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From jerry.richards at teotech.com  Wed Jan 13 09:15:53 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 13 Jan 2010 09:15:53 -0800
Subject: [Freeswitch-users] Inbound DTMF Not Recognized In Latest Version
In-Reply-To: <2A012832-A0B1-4DBC-ABB3-85FCBB71810A@freeswitch.org>
References: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com><1263398163.29661.1354446823@webmail.messagingengine.com>
	<2A012832-A0B1-4DBC-ABB3-85FCBB71810A@freeswitch.org>
Message-ID: <E3ACF9CD600448E789F5D7178E4AE3DB@greyhawk.tonecommander.com>


Okay, cool.  Now it's working.  It is possible I screwed up the XML files
when I upgraded (since there were several new paths relating to accessing
sound files), so I rebuilt/reinstalled it from scratch and then more
carefully edited the XML files to preserve the new sound paths.  I also took
out the "start_dtmf".  So this my issue resolved.

Thanks,
Jerry


-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Wednesday, January 13, 2010 8:01 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Inbound DTMF Not Recognized In Latest
Version

If you're using OpenZAP you don't need start_dtmf thats mostly for RTP
streams that are inband and OpenZAP does the DTMF bits for you if you start
that you'll end up with double digits most likely.

/b

On Jan 13, 2010, at 9:56 AM, Scott Torr wrote:

> If someone would like to confirm this, please do.






From mike at jerris.com  Wed Jan 13 09:21:01 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 12:21:01 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
Message-ID: <EA8724C8-84A8-44DB-9F76-EFA517E76402@jerris.com>

Are you asking if you can mux all of the progress audio from multiple b-legs?  if so, no, and why would you want to?

Mike

On Jan 13, 2010, at 6:15 AM, David Villasmil wrote:

> hello
> 
> is it possible to bridge multiple b-legs and provide all audio  
> (progress) until there is an answer on one channel?



From mike at jerris.com  Wed Jan 13 09:29:35 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 12:29:35 -0500
Subject: [Freeswitch-users] sip trunk question: why call through
	external profile is challenged?
In-Reply-To: <20100113141003.970B311F32@mail.nstel.ru>
References: <20100113141003.970B311F32@mail.nstel.ru>
Message-ID: <98955F81-E9C1-4926-A648-AD49FB9D38A4@jerris.com>

Look at how sipx sets up the users when they build the extensions and such for conferences, there was something special here, but I can't recall what.

Mike

On Jan 13, 2010, at 9:10 AM, Nikolay Kondratyev wrote:

> Hi all!
>  
> I?m brand new to FreeSwitch, but have some experience with SipX.
> Our company is evaluating FS.
> For test purposes I set up FS on a virtual machine (vmware esxi). I use CentOS.
> The FS version I use is 1.0.5-20100110-0400.
>  
> I have a question regarding sip trunk between FS and SipX.
> I created the following GW in external profile:
> [freeswitch at freeswitch external]$ cat sipx-lab.xml | grep -v '<!--'
> <include>
>   <gateway name="sipx4.lab.nstel.ru">
>   <param name="username" value="zxcv"/>
>   <param name="password" value="2007"/>
>   <param name="register" value="false"/>
>   </gateway>
> </include>
>  
> External.xml file is not modified after installation.
>  
> I see this gateway via fs_cli:
> freeswitch at internal> sofia status gateway sipx4.lab.nstel.ru
> =================================================================================================
> Name            sipx4.lab.nstel.ru
> Profile         external
> Scheme          Digest
> Realm           sipx4.lab.nstel.ru
> Username        zxcv
> Password        yes
> From            <sip:zxcv at sipx4.lab.nstel.ru;transport=udp>
> Contact         <sip:gw+sipx4.lab.nstel.ru at 172.23.22.49:5080;transport=udp;gw=sipx4.lab.nstel.ru>
> Exten           zxcv
> To              sip:zxcv at sipx4.lab.nstel.ru
> Proxy           sip:sipx4.lab.nstel.ru
> Context         public
> Expires         3600
> Freq            3600
> Ping            0
> PingFreq        0
> PingState       0/0/0
> State           NOREG
> Status          UP
> CallsIN         0
> CallsOUT        0
> =================================================================================================
>  
> I created new FS extension 2853. I registers (xlite) and I can call it.
>  
> Now I want to call FS user from sipx.
>  
> From the sipx side one can configure link to FS differently. There are two options:
> 1. Call FS directly trough sipxproxy (the core part of sipx works as sip proxy, not as b2bua)
> 2. Call trough embedded b2bua, named sipxbridge.
>  
> When a call is going trough sipxbridge, it is successfully landed at FS extension 2853.
> But when a call is going from the sipxproxy, it is challenged with Status 407 ?proxy authentication required?, and then call fails.
> (I?m not sure if sipx should handle this challenge, but this is separate question for the sipx forum).
> At the default log level I see nothing in the freeswitch.log.
>  
> So the question is why one call through external profile is being challenged while the other is not?
> I suspect that the reason is in the difference in the two Invite messages:
>  
> Here is challenged Invite:
> INVITE sip:2853 at fs.lab.nstel.ru:5080;transport=udp SIP/2.0
> Record-Route: <sip:172.23.12.104:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EOTRiZjI3MjgtYWMxNzBjZTktMTNjNC0yZjZkZWMtNjNkNmUwN2UtMmY2ZGVj.900_ntap%2Aid%7EMjIyNjAtMQ%60%60%213ba6d85f6946c4c6001bee1d3b54474f>
> From: "testphone3857"<sip:3857 at lab.nstel.ru>;tag=94bf2728-ac170ce9-13c4-2f6dec-63d6e07e-2f6dec
> To: <sip:2853 at lab.nstel.ru>
> Call-Id: 94bed2a0-ac170ce9-13c4-2f6dec-68ffdb81-2f6dec at lab.nstel.ru
> Cseq: 2 INVITE
> Via: SIP/2.0/UDP 172.23.12.104;branch=z9hG4bK-sipXecs-000eaf48a36fe4029c7cde004a6f44424847
> Via: SIP/2.0/TCP 172.23.12.104;branch=z9hG4bK-sipXecs-000bf3008cfa3f3e70470789c75232ba9499~ac7fd729330fd563f83cacd941311e75;id=22260-1
> Via: SIP/2.0/UDP 172.23.12.233:5060;branch=z9hG4bK-2f6dec-b9456412-7ad07784
> Max-Forwards: 18
> Supported: replaces
> User-Agent: LG-Nortel LIP 6812 v1.2.38sp SN/00405A18B634
> Contact: <sip:3857 at 172.23.12.233:5060;x-sipX-nonat>
> Proxy-Authorization: Digest username="3857",realm="lab.nstel.ru",nonce="e2152722611af1bbe59f3a8eb31b8eb84b4db2ca",uri="sip:2853 at lab.nstel.ru",response="a09c34fbd897783c9b9af50bd044ccaa",algorithm=MD5
> Content-Type: application/sdp
> Content-Length: 301
> Date: Wed, 13 Jan 2010 11:47:22 GMT
> Expires: 60
> X-Sipx-Handled: X172.23.12.104-81.211.30.104
>  
> v=0
> o=LGEIPP 16246 16247 IN IP4 172.23.12.233
> s=SIP Call
> c=IN IP4 172.23.12.233
> t=0 0
> m=audio 23008 RTP/AVP 0 8 18 4 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:18 annexb=no
> a=fmtp:101 0-11
> a=sendrecv
>  
>  
> And here is successful Invite:
> INVITE sip:2853 at fs.lab.nstel.ru;user=phone SIP/2.0
> Call-ID: 94bec3b8-ac170ce9-13c4-2f6d47-2fee09d0-2f6d47 at lab.nstel.ru.0
> CSeq: 1 INVITE
> From: "testphone3857" <sip:3857 at 172.23.12.104>;tag=816640414244159961
> To: <sip:2853 at fs.lab.nstel.ru;user=phone>
> Via: SIP/2.0/UDP 172.23.12.104:5080;branch=z9hG4bK885dde2c8680f5315845cd3350b8b605373534
> Max-Forwards: 70
> User-Agent: sipXecs/4.0.2 sipXecs/sipxbridge (Linux)
> P-Asserted-Identity: <sip:1004 at 172.23.12.104>
> Contact: <sip:1004 at 172.23.12.104:5080;transport=udp>
> Route: <sip:172.23.22.49:5080;transport=udp;lr>
> Session-Expires: 1800;refresher=uac
> Allow: INVITE,BYE,ACK,CANCEL,OPTIONS
> Content-Type: application/sdp
> Content-Length: 315
>  
> v=0
> o=sipxbridge 6640787141824452741 1 IN IP4 172.23.12.104
> s=SIP Call
> c=IN IP4 172.23.12.104
> t=0 0
> m=audio 30248 RTP/AVP 0 8 18 4 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:18 annexb=no
> a=fmtp:101 0-11
> a=sendrecv
>  
> Can somebody please if it is a FS configuration problem or a software problem or is it a problem on the sipx side?
>  
> Thanks in advance,
> Nikolay.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Wed Jan 13 09:33:42 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 12:33:42 -0500
Subject: [Freeswitch-users] Inbound DTMF Not Recognized In Latest Version
In-Reply-To: <E3ACF9CD600448E789F5D7178E4AE3DB@greyhawk.tonecommander.com>
References: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com><1263398163.29661.1354446823@webmail.messagingengine.com>
	<2A012832-A0B1-4DBC-ABB3-85FCBB71810A@freeswitch.org>
	<E3ACF9CD600448E789F5D7178E4AE3DB@greyhawk.tonecommander.com>
Message-ID: <7F7949F8-EE9A-4D9B-8594-7C85607E2112@jerris.com>

The way I changed the configs, they should all still work fine with old configs, just not if sounds directory is moved out of prefix, which there is not a way to do yet.

Mike

On Jan 13, 2010, at 12:15 PM, Jerry Richards wrote:

> 
> Okay, cool.  Now it's working.  It is possible I screwed up the XML files
> when I upgraded (since there were several new paths relating to accessing
> sound files), so I rebuilt/reinstalled it from scratch and then more
> carefully edited the XML files to preserve the new sound paths.  I also took
> out the "start_dtmf".  So this my issue resolved.
> 
> Thanks,
> Jerry
> 
> 
> -----Original Message-----
> From: Brian West [mailto:brian at freeswitch.org] 
> Sent: Wednesday, January 13, 2010 8:01 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Inbound DTMF Not Recognized In Latest
> Version
> 
> If you're using OpenZAP you don't need start_dtmf thats mostly for RTP
> streams that are inband and OpenZAP does the DTMF bits for you if you start
> that you'll end up with double digits most likely.
> 
> /b
> 
> On Jan 13, 2010, at 9:56 AM, Scott Torr wrote:
> 
>> If someone would like to confirm this, please do.
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jeff at jefflenk.com  Wed Jan 13 09:38:13 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 13 Jan 2010 11:38:13 -0600
Subject: [Freeswitch-users] FSComm Windows build
In-Reply-To: <fc0ed37b1001130841k11b3ae8ekaeefb227fb4a227f@mail.gmail.com>
References: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>,
	<1263350125191-4295967.post@n2.nabble.com>,
	<5800526b1001121918r15a336f5jbc5b724e66ac2825@mail.gmail.com>,
	<1263358274180-4311097.post@n2.nabble.com>,
	<fc0ed37b1001130841k11b3ae8ekaeefb227fb4a227f@mail.gmail.com>
Message-ID: <COL117-W3855379D4087BF8CA339A8B96B0@phx.gbl>


Use the prebuilt install from Carlos

 

http://files.freeswitch.org/windows_installer/

 

This will list all the dependents that you need

 


 
> Date: Wed, 13 Jan 2010 13:41:36 -0300
> From: gmartinr82 at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FSComm Windows build
> 
> Hi List:
> 
> Test Installation of this compilation but wanting to run it generates
> the following error. I have installed Windows XP SP3.
> 
> Attached screenshot.
> 
> Bye
> Martin Rodriguez
> 
> 
> 2010/1/13 Jeff Lenk <jeff at jefflenk.com>:
> >
> > Carlos,
> >
> > Thanks for those links!
> >
> > I am currenltly using prebuilt 4.6 32 bit libs and was hoping Nokia prebuilt
> > the x64 too. I guess I might have to look into building myself as well.
> >
> > -Jeff
> >
> >
> > Jeff,
> >
> > not yet. I'm still linking with a 32bit QT library of 4.5.2 I compiled for
> > VS2008 sometime last year. I followed a guide similar to this one:
> > http://dcsoft.com/community_server/blogs/dcsoft/archive/2009/03/06/how-to-setup-qt-4-5-visual-studio-integration.aspx
> >
> > At the time I didn't see a need for a 64bit version of FsGui. Since FSComm
> > has FreeSWITCH in the back end this kind of changes things.
> >
> > Looks like someone put up a wiki on compiling QT for Win x64:
> > http://en.wikibooks.org/wiki/Opticks_Developer_Guide/Getting_Started/Building_Qt_From_Source
> >
> > Now that QT 4.6 is out I might have to revisit a new library build with 64
> > bit in mind.
> >
> > Carlos
> >
> > --
> > View this message in context: http://n2.nabble.com/FSComm-Windows-build-tp4289256p4311097.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
http://clk.atdmt.com/GBL/go/196390706/direct/01/
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From carlos.talbot at gmail.com  Wed Jan 13 09:56:03 2010
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Wed, 13 Jan 2010 11:56:03 -0600
Subject: [Freeswitch-users] FSComm Windows build
In-Reply-To: <fc0ed37b1001130841k11b3ae8ekaeefb227fb4a227f@mail.gmail.com>
References: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
	<1263350125191-4295967.post@n2.nabble.com>
	<5800526b1001121918r15a336f5jbc5b724e66ac2825@mail.gmail.com>
	<1263358274180-4311097.post@n2.nabble.com>
	<fc0ed37b1001130841k11b3ae8ekaeefb227fb4a227f@mail.gmail.com>
Message-ID: <5800526b1001130956g9f00a5ai2020c9d00d89992e@mail.gmail.com>

D'oh! I forgot to include the VS 2008 redistributable package like I do with
FreeSWITCH. You'll have to manually install one of the following below until
I've updated the install.

32bit version
http://www.microsoft.com/downloads/details.aspx?FamilyID=9b2da534-3e03-4391-8a4d-074b9f2bc1bf&displaylang=en


regards,

Carlos

On Wed, Jan 13, 2010 at 10:41 AM, Martin Rodriguez <gmartinr82 at gmail.com>wrote:

> Hi List:
>
> Test Installation of this compilation but wanting to run it generates
> the following error. I have installed Windows XP SP3.
>
> Attached screenshot.
>
> Bye
> Martin Rodriguez
>
>
> 2010/1/13 Jeff Lenk <jeff at jefflenk.com>:
> >
> > Carlos,
> >
> > Thanks for those links!
> >
> > I am currenltly using prebuilt 4.6 32 bit libs and was hoping Nokia
> prebuilt
> > the x64 too. I guess I might have to look into building myself as well.
> >
> > -Jeff
> >
> >
> > Jeff,
> >
> > not yet. I'm still linking with a 32bit QT library of 4.5.2 I compiled
> for
> > VS2008 sometime last year. I followed a guide similar to this one:
> >
> http://dcsoft.com/community_server/blogs/dcsoft/archive/2009/03/06/how-to-setup-qt-4-5-visual-studio-integration.aspx
> >
> > At the time I didn't see a need for a 64bit version of FsGui. Since
> FSComm
> > has FreeSWITCH in the back end this kind of changes things.
> >
> > Looks like someone put up a wiki on compiling QT for Win x64:
> >
> http://en.wikibooks.org/wiki/Opticks_Developer_Guide/Getting_Started/Building_Qt_From_Source
> >
> > Now that QT 4.6 is out I might have to revisit a new library build with
> 64
> > bit in mind.
> >
> > Carlos
> >
> > --
> > View this message in context:
> http://n2.nabble.com/FSComm-Windows-build-tp4289256p4311097.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch-users at digitaldan.com  Wed Jan 13 10:01:50 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Wed, 13 Jan 2010 11:01:50 -0700 (MST)
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <21466640.91263405615737.JavaMail.root@zimbra>
Message-ID: <10842021.111263405710193.JavaMail.root@zimbra>

Hi guys, I'm having an issue with DTMF detection from our cisco 5300 gateways. DTMF detection has been working great when someone was manually inputing the digits (ie there are small pauses between the DTMF digits), but when the other end is sending those digits from say a speed dial or programmed dial string and can send the DTMF digits one right after the other we start to drop one or more digits about 75% of the time. 


I upgraded to trunk this morning and have tried everything that was listed on http://wiki.freeswitch.org/wiki/RTP_Issues . I have a pcap output of the SIP and RTP data, it looks fine to me, I can see the DTMF payloads coming as they should be. Any ideas? I would be fine sending or posting the pcap file to someone privately. 


Thanks 
Dan- 
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From mailinglist at fribert.dk  Wed Jan 13 10:06:59 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Wed, 13 Jan 2010 19:06:59 +0100
Subject: [Freeswitch-users] Svar: Re:  Multi-Homed setup,
	starting		 over	 - still not working
In-Reply-To: <4B4E0C24020000E1000003A7@mail.fribert.dk>
References: <4B4DC035020000E1000003A2@mail.fribert.dk>
	<CA5A491E9DEFBE4CB777DE97E21575E9079C65@prato.activenetwerx.local>
	<4B4E0C24020000E1000003A7@mail.fribert.dk>
Message-ID: <4B4E19D3020000E1000003AB@mail.fribert.dk>

Hi again
 
Ok, got a bit further, just had a success, I called out, weeheee :-D
 
Now I want to call in, and it seems the SIP provider I'm using has multiple IP's, I'm not sure if that's the issue here?
I've created the dialplan part explained in the guide, and placed it in dialplan/public as explained in IN-COMING CALLS, RINGING A GROUP

<include>
  <extension name="Privat">
    <condition field="destination_number" expression="^(4692xxxx)$">  (my phonenumber)
      <action application="set" data="domain_name=10.11.12.25"/>
      <action application="transfer" data="*8201 XML default"/>
    </condition>
  </extension>
</include>
I've joined the two soft phones I have to the group.
 
When I call in I get this displayed:
2010-01-13 18:57:51.396370 [NOTICE] switch_channel.c:602 New Channel sofia/external/2680xxxx at 87.54.25.116 [fedccd2a-6d00-df11-aee0-000c29b7b4cb]
2010-01-13 18:57:51.396370 [NOTICE] sofia.c:3419 Hangup sofia/external/26802233 at 87.54.25.116 [CS_NEW] [INCOMPATIBLE_DESTINATION]
2010-01-13 18:57:51.399569 [NOTICE] switch_core_session.c:1085 Session 19 (sofia/external/2680xxxx at 87.54.25.116) Ended
2010-01-13 18:57:51.399569 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/2680xxxx at 87.54.25.116 [CS_DESTROY]

The 87.54.25.116 is from musimi.dk, but outgoing is handled by 87.54.25.114 (which musimi.dk translates to).

2680xxxx is my cellphones number.

Hmm, help :-)

>>> 13-01-2010 kl. 18:08 skrev "mailinglist" <mailinglist at fribert.dk> i meddelelsen <4B4E0C24020000E1000003A7 at mail.fribert.dk>:

Hi Joseph
 
Hmm, you were right, I hadn't set it there, as the guide explains me that I should set it in internal.xml.
After setting it in vars.xml it authenticates as well.
 
So I'm contiuing the quest :-)
 
Thankyou very much for your assistance!
 
/Fribse


>>> 13-01-2010 kl. 16:30 skrev "Joseph L. Casale" <jcasale at activenetwerx.com> i meddelelsen <CA5A491E9DEFBE4CB777DE97E21575E9079C65 at prato.activenetwerx.local>:

>But are you sayng that "eval ${domain}" will do more than is done when I restart the fs?
>I don't see that anywhere in the guide :-D
>
>I'll try it out asap!

What I am saying is it "evaluates" that placeholder and shows
its value. You can use it to show the current value to verify
it matches what you intend, which is for it to be the same as
your LAN ip.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From msc at freeswitch.org  Wed Jan 13 10:14:25 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Jan 2010 10:14:25 -0800
Subject: [Freeswitch-users] CDR and reporting.
In-Reply-To: <201001120639.38098.sos@sokhapkin.dyndns.org>
References: <8ccbff061001120313u78701883lf9e212947af0528@mail.gmail.com>
	<201001120639.38098.sos@sokhapkin.dyndns.org>
Message-ID: <87f2f3b91001131014w4d19027bl4e90895954a40bbd@mail.gmail.com>

Also, you can have a cron job collect these more frequently than once per
day. There are folks out there we grab them several times per minute and
process them so that it is almost real-time. Of course, using mod_xml_cdr
requires more effort up front but it *IS* real-time and more scalable.

-MC

On Tue, Jan 12, 2010 at 3:39 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Look at default config for mod_cdr_csv, sql template. It can produce a
> sequence of SQL INSERT statements which can be later fed (after daily log
> rotation) to mysql client program.
>
> I see no reason to employ xml for this task, the language hard to read and
> parse for both human and computer.
>
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From devel at thom.fr.eu.org  Wed Jan 13 10:25:20 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Wed, 13 Jan 2010 19:25:20 +0100
Subject: [Freeswitch-users] No hangup tone after zap channel closed,
	tones in general
Message-ID: <8b18ffe229516c90285de8a54a00e845@thom.fr.eu.org>



Hello, 

How shall I do to get a hangup tone on an FXS port after the
channel is closed ? 

Is there something specific to configure in openzap ?


For instance, if I call the following extension from an FXS port : 

I
get no tone after the hangup application is called. 

I also wonder if
there is some documentation on the tones.conf file format, and about the
variables uk-ring, us-ring, bong-ring, sit in vars.xml (when are they used,
what is the syntax). I could not find any info on wiki. 

Thank you for
helping 

Fran?ois
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From msc at freeswitch.org  Wed Jan 13 10:41:40 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Jan 2010 10:41:40 -0800
Subject: [Freeswitch-users] Fw: SIP client authorization issues
	(Globarange phone)
In-Reply-To: <271101.94727.qm@web53107.mail.re2.yahoo.com>
References: <271101.94727.qm@web53107.mail.re2.yahoo.com>
Message-ID: <87f2f3b91001131041t1aea3a89n68c6f6948db1253d@mail.gmail.com>

On Tue, Jan 12, 2010 at 4:54 PM, VG Oget <vgoget at yahoo.com> wrote:

> Hi, I am trying to register a GlobaRange SIP phone to FreeSwitch (was able
> to do it with Asterisk some time ago). It is a locked VOIP phone to
> Joip.com. On Ubuntu with latest version as of a few days ago?
>
> I redirected the traffic to Freeswitch (basically proxy.joip.com points
> and to the Freeswitch host) and also changed the SIP port to 23768 (what the
> phone wants). Very close to this:
> http://darkskiez.co.uk/index.php?page=Use_The_Panasonic_Globarange_With_Asterisk
>
>
> I have Xlite successfully registering (SIP port 23768 and using
> proxy.joip.com not the IP address) so I?m confident the freeswitch config
> is close to being okay?
>
> Now I have intercepted the traffic during registration. The issue (I think)
> is that the SIP authentication phases use a different username and I don?t
> know how to configure it.
> I looked into this page extensively:
> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide
> And also tried to understand sip_suth_username but I cannot find the
> solution.
>
> If someone can point me in the right direction, that would be perfect.
>
> Were these the exact same credentials that you used when you tested with
x-lite? Grab a SIP trace of that registration happening and compare it to
this one.
-MC
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From carlos.talbot at gmail.com  Wed Jan 13 10:42:47 2010
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Wed, 13 Jan 2010 12:42:47 -0600
Subject: [Freeswitch-users] FSComm Windows build
In-Reply-To: <fc0ed37b1001130841k11b3ae8ekaeefb227fb4a227f@mail.gmail.com>
References: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
	<1263350125191-4295967.post@n2.nabble.com>
	<5800526b1001121918r15a336f5jbc5b724e66ac2825@mail.gmail.com>
	<1263358274180-4311097.post@n2.nabble.com>
	<fc0ed37b1001130841k11b3ae8ekaeefb227fb4a227f@mail.gmail.com>
Message-ID: <5800526b1001131042sef0ffbfvdfd8e28efe57a137@mail.gmail.com>

I just finished uploading a new version of FSComm that includes a 32 and 64
bit build with respective QT and FreeSWITCH libraries. It also now includes
the Microsoft redistributable package. The new executable is 15MB in size.
I'm not sure when the CDN sync will kick in...

regards,

Carlos

On Wed, Jan 13, 2010 at 10:41 AM, Martin Rodriguez <gmartinr82 at gmail.com>wrote:

> Hi List:
>
> Test Installation of this compilation but wanting to run it generates
> the following error. I have installed Windows XP SP3.
>
> Attached screenshot.
>
> Bye
> Martin Rodriguez
>
>
> 2010/1/13 Jeff Lenk <jeff at jefflenk.com>:
> >
> > Carlos,
> >
> > Thanks for those links!
> >
> > I am currenltly using prebuilt 4.6 32 bit libs and was hoping Nokia
> prebuilt
> > the x64 too. I guess I might have to look into building myself as well.
> >
> > -Jeff
> >
> >
> > Jeff,
> >
> > not yet. I'm still linking with a 32bit QT library of 4.5.2 I compiled
> for
> > VS2008 sometime last year. I followed a guide similar to this one:
> >
> http://dcsoft.com/community_server/blogs/dcsoft/archive/2009/03/06/how-to-setup-qt-4-5-visual-studio-integration.aspx
> >
> > At the time I didn't see a need for a 64bit version of FsGui. Since
> FSComm
> > has FreeSWITCH in the back end this kind of changes things.
> >
> > Looks like someone put up a wiki on compiling QT for Win x64:
> >
> http://en.wikibooks.org/wiki/Opticks_Developer_Guide/Getting_Started/Building_Qt_From_Source
> >
> > Now that QT 4.6 is out I might have to revisit a new library build with
> 64
> > bit in mind.
> >
> > Carlos
> >
> > --
> > View this message in context:
> http://n2.nabble.com/FSComm-Windows-build-tp4289256p4311097.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From mailinglist at fribert.dk  Wed Jan 13 11:10:02 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Wed, 13 Jan 2010 20:10:02 +0100
Subject: [Freeswitch-users] Svar: Re:  Multi-Homed setup,
	starting		 over	 - still not working
In-Reply-To: <4B4E19D3020000E1000003AB@mail.fribert.dk>
References: <4B4DC035020000E1000003A2@mail.fribert.dk>
	<CA5A491E9DEFBE4CB777DE97E21575E9079C65@prato.activenetwerx.local>
	<4B4E0C24020000E1000003A7@mail.fribert.dk>
	<4B4E19D3020000E1000003AB@mail.fribert.dk>
Message-ID: <4B4E289A020000E1000003B0@mail.fribert.dk>

hey, I got it solved. Apparently my voip provider used a G729 Codec, which freeswitch didn't like.
Of course the VOIP provider has 4, and just the one not supported in this setup is used as default :-o
 
Now I can call in, WOW, cool :-)
 
Small steps here...


>>> 13-01-2010 kl. 19:06 skrev "mailinglist" <mailinglist at fribert.dk> i meddelelsen <4B4E19D3020000E1000003AB at mail.fribert.dk>:

Hi again

Ok, got a bit further, just had a success, I called out, weeheee :-D

Now I want to call in, and it seems the SIP provider I'm using has multiple IP's, I'm not sure if that's the issue here?
I've created the dialplan part explained in the guide, and placed it in dialplan/public as explained in IN-COMING CALLS, RINGING A GROUP

<include>
  <extension name="Privat">
    <condition field="destination_number" expression="^(4692xxxx)$">  (my phonenumber)
      <action application="set" data="domain_name=10.11.12.25"/>
      <action application="transfer" data="*8201 XML default"/>
    </condition>
  </extension>
</include>
I've joined the two soft phones I have to the group.

When I call in I get this displayed:
2010-01-13 18:57:51.396370 [NOTICE] switch_channel.c:602 New Channel sofia/external/2680xxxx at 87.54.25.116 [fedccd2a-6d00-df11-aee0-000c29b7b4cb]
2010-01-13 18:57:51.396370 [NOTICE] sofia.c:3419 Hangup sofia/external/26802233 at 87.54.25.116 [CS_NEW] [INCOMPATIBLE_DESTINATION]
2010-01-13 18:57:51.399569 [NOTICE] switch_core_session.c:1085 Session 19 (sofia/external/2680xxxx at 87.54.25.116) Ended
2010-01-13 18:57:51.399569 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/2680xxxx at 87.54.25.116 [CS_DESTROY]

The 87.54.25.116 is from musimi.dk, but outgoing is handled by 87.54.25.114 (which musimi.dk translates to).

2680xxxx is my cellphones number.

Hmm, help :-)

>>> 13-01-2010 kl. 18:08 skrev "mailinglist" <mailinglist at fribert.dk> i meddelelsen <4B4E0C24020000E1000003A7 at mail.fribert.dk>:

Hi Joseph

Hmm, you were right, I hadn't set it there, as the guide explains me that I should set it in internal.xml.
After setting it in vars.xml it authenticates as well.

So I'm contiuing the quest :-)

Thankyou very much for your assistance!

/Fribse


>>> 13-01-2010 kl. 16:30 skrev "Joseph L. Casale" <jcasale at activenetwerx.com> i meddelelsen <CA5A491E9DEFBE4CB777DE97E21575E9079C65 at prato.activenetwerx.local>:

>But are you sayng that "eval ${domain}" will do more than is done when I restart the fs?
>I don't see that anywhere in the guide :-D
>
>I'll try it out asap!

What I am saying is it "evaluates" that placeholder and shows
its value. You can use it to show the current value to verify
it matches what you intend, which is for it to be the same as
your LAN ip.

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From jerry.richards at teotech.com  Wed Jan 13 11:12:54 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 13 Jan 2010 11:12:54 -0800
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <10842021.111263405710193.JavaMail.root@zimbra>
References: <21466640.91263405615737.JavaMail.root@zimbra>
	<10842021.111263405710193.JavaMail.root@zimbra>
Message-ID: <47564CF8463D463E8816123D6AF5B30F@greyhawk.tonecommander.com>

I would recommend you use out-of-band DTMF based on RFC 2833.  In band DTMF
is usually not reliable in short bursts.
 
Jerry


  _____  

From: Dan [mailto:freeswitch-users at digitaldan.com] 
Sent: Wednesday, January 13, 2010 10:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Cisco media gateway and DTMF


Hi guys, I'm having an issue with DTMF detection from our cisco 5300
gateways.  DTMF detection has been working great when someone was manually
inputing the digits (ie there are small pauses between the DTMF digits),
but when the other end is sending those digits from say a speed dial or
programmed dial string and can send the DTMF digits one right after the
other we start to drop one or more digits about 75% of the time.    

I upgraded to trunk this morning and have tried everything that was listed
on http://wiki.freeswitch.org/wiki/RTP_Issues .  I have a pcap output of the
SIP and RTP data, it looks fine to me, I can see the DTMF payloads coming as
they should be.  Any ideas?  I would be fine sending or posting the pcap
file to someone privately. 

Thanks
Dan-

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From mike at van.lammeren.net  Wed Jan 13 11:30:06 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Wed, 13 Jan 2010 14:30:06 -0500
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0F362E0@BESONE.corp.dom>
References: <191c3a031001121535t4819ee44p74cc9f7cd4fe972f@mail.gmail.com>
	<A2167AE7633A5C4DAE56EEBD1428ECC0F362E0@BESONE.corp.dom>
Message-ID: <5d2828f1001131130s6ef166c0wc98f70023420c0fd@mail.gmail.com>

Hello!

Before you remove "session:originate" from the wiki, you should take a look
at this:
http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocumented_Session_Functions.3F

<http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocumented_Session_Functions.3F>There
is, in fact, a function called "originate".

Mike van Lammeren


On Wed, Jan 13, 2010 at 5:22 AM, Cavalera Claudio Luigi <
Claudio.Cavalera at italtel.it> wrote:

>  Thanks a lot Anthony,
> some comments inline (and please forgive me for my broken email client).
>
> >>  example1: Consider this simple lua script in which i create two
> sessions:
>
>  >>  api = freeswitch.API();
>  >>  api:execute("originate", "sofia/internal/1001%192.168.1.1 &park");
>  >>  api:execute("originate", "sofia/internal/1002%192.168.1.1 &park");
>
>  >  capture the output from api:execute the uuid is in there
>
> Thx a lot,
> this was one piece i was missing although it's already on the wiki here:
> http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls
>
>
>  > because lua calls it freeswitch.bridge
>
>  > session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
>  > session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1",
> session1);
>  > freeswitch.bridge(session1, session2);
>
> good to now, there isn't any example of freeswitch.bridge in the wiki and i
> would like to add one.
> Where I could find the full api of
> freeswitch.Session( ) ?
> because I've seen this working also without "session1" in the second line:
> session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
> session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");
> freeswitch.bridge(session1, session2);
>
>  also is there any difference between  freeswitch.bridge
> and freeswitch.execute(uuid_bridge ...) ?
>
>  >> example3: yet another possibility
>  >> local session1 =
> freeswitch.Session("sofia/internal/1001%192.168.1.1");
>  >> session1:originate(session2, "sofia/internal/1002%192.168.1.1", 1000);
>  >> but it does not work either.
>
>  > The above is gibberish try:
>  > local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
>  > session1:execute("bridge", "sofia/internal/1002%192.168.1.1");
>
> Okay i will report also this bridge example on the wiki which was missing.
> But does session:originate make sense in some cases or not? Otherwise i'm
> going to remove this line on the wiki
>  http://wiki.freeswitch.org/wiki/Mod_lua#session:originate
>
> Thanks,
> Claudio
>
>
>
> Internet Email Confidentiality Footer
>
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From yehavi.bourvine at gmail.com  Wed Jan 13 11:32:17 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 13 Jan 2010 21:32:17 +0200
Subject: [Freeswitch-users] How to build FreeSwitch with non-default SSL
	library?
Message-ID: <d361ca7b1001131132o1043cce4p31af0f95f804bd7b@mail.gmail.com>

Hello,

  I suspect that the Openssl library used on fedora is a modified one
and has some interoperability problems with phones. I would like to build
freeswitch with a library from openssl.org which is installed
under /usr/local. Is there an easy way to tell  "configure" script to use
that version instead?

                                 Thanks, __Yehavi:
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From freeswitch-users at digitaldan.com  Wed Jan 13 11:36:26 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Wed, 13 Jan 2010 12:36:26 -0700 (MST)
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <47564CF8463D463E8816123D6AF5B30F@greyhawk.tonecommander.com>
Message-ID: <26415028.141263411386237.JavaMail.root@zimbra>

Sorry, I should have made that more clear, this is using 2833. As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through). 
----- Original Message ----- 
From: "Jerry Richards" <jerry.richards at teotech.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 12:12:54 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 


I would recommend you use out-of-band DTMF based on RFC 2833. In band DTMF is usually not reliable in short bursts. 

Jerry 




From: Dan [mailto:freeswitch-users at digitaldan.com] 
Sent: Wednesday, January 13, 2010 10:02 AM 
To: freeswitch-users at lists.freeswitch.org 
Subject: [Freeswitch-users] Cisco media gateway and DTMF 



Hi guys, I'm having an issue with DTMF detection from our cisco 5300 gateways. DTMF detection has been working great when someone was manually inputing the digits (ie there are small pauses between the DTMF digits), but when the other end is sending those digits from say a speed dial or programmed dial string and can send the DTMF digits one right after the other we start to drop one or more digits about 75% of the time. 


I upgraded to trunk this morning and have tried everything that was listed on http://wiki.freeswitch.org/wiki/RTP_Issues . I have a pcap output of the SIP and RTP data, it looks fine to me, I can see the DTMF payloads coming as they should be. Any ideas? I would be fine sending or posting the pcap file to someone privately. 


Thanks 
Dan- 
_______________________________________________ 
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From brian at freeswitch.org  Wed Jan 13 11:39:52 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 13:39:52 -0600
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <26415028.141263411386237.JavaMail.root@zimbra>
References: <26415028.141263411386237.JavaMail.root@zimbra>
Message-ID: <84ACB346-D901-4B2A-AFD0-CDA482D1B731@freeswitch.org>

Cisco gateways can detect DTMF at random if you have kids screaming or loud noises in the background... very unreliable.

/b

On Jan 13, 2010, at 1:36 PM, Dan wrote:

> Sorry, I should have made that more clear, this is using 2833.  As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through).  

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From freeswitch-users at digitaldan.com  Wed Jan 13 12:12:14 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Wed, 13 Jan 2010 13:12:14 -0700 (MST)
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <24348778.171263412569609.JavaMail.root@zimbra>
Message-ID: <1256032.191263413534672.JavaMail.root@zimbra>

Thanks for your reply, 


Wireshark shows all of the digits coming through cleanly, in the case below (also on pastebin) its 0951*72789, freeswitch does not seem to be processing one or more of the rtp events. Is there some specific debugging I should turn on that might give me some more info on why freeswitch is not processing some of these events? 


http://pastebin.freeswitch.org/11780 



449 12.280148 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
450 12.285108 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
451 12.290082 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
452 12.335129 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
453 12.340124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
454 12.345124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
455 12.365162 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21862, Time=359229277, Mark 
456 12.380167 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21863, Time=359229605 
461 12.385248 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
462 12.385253 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
465 12.390163 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
466 12.435211 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
467 12.440201 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
468 12.445205 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
469 12.480251 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21870, Time=359230405 
470 12.485275 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
471 12.485284 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
472 12.490251 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
475 12.535277 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
476 12.540287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
477 12.545287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
478 12.580337 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21877, Time=359231197 
484 12.585337 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
485 12.585343 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
486 12.590362 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
492 12.635393 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
493 12.640368 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
494 12.645369 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
495 12.665406 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21884, Time=359231677, Mark 
496 12.680431 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21885, Time=359232005 
497 12.680438 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
502 12.685408 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
506 12.690412 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
507 12.735450 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
508 12.740453 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
509 12.745469 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
511 12.780526 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21892, Time=359232805 
512 12.780534 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
513 12.785502 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
514 12.790492 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
515 12.835544 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
516 12.840537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
517 12.845537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
523 12.880582 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21899, Time=359233605 
524 12.885593 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
525 12.885599 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
526 12.890580 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
528 12.935618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
529 12.940618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
530 12.945618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
531 12.965656 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21906, Time=359234077, Mark 
532 12.980659 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21907, Time=359234413 
538 12.985690 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
539 12.985695 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
540 12.990657 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
541 13.035703 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
542 13.040701 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
543 13.045700 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
544 13.080762 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21914, Time=359235205 
545 13.080768 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
546 13.085736 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
547 13.090798 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
549 13.135809 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
550 13.140780 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
551 13.145793 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
554 13.180827 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21921, Time=359236005 
558 13.185847 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
559 13.185854 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
560 13.190821 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
566 13.235872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
567 13.240864 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
568 13.245872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) ----- Original Message ----- 
From: "Brian West" <brian at freeswitch.org> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 12:39:52 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

Cisco gateways can detect DTMF at random if you have kids screaming or loud noises in the background... very unreliable. 


/b 



On Jan 13, 2010, at 1:36 PM, Dan wrote: 


Sorry, I should have made that more clear, this is using 2833. As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through). 

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From david.villasmil.work at gmail.com  Wed Jan 13 12:20:31 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 13 Jan 2010 21:20:31 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <EA8724C8-84A8-44DB-9F76-EFA517E76402@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<EA8724C8-84A8-44DB-9F76-EFA517E76402@jerris.com>
Message-ID: <9853f4ff1001131220u72ab5177x287a6dcb84f9b185@mail.gmail.com>

Thanks for answering,

Well, imagine I have a content provider which will deliver custom ringbacks
via SIP INVITES, they point would be to receive A-leg then bridge to 2
B-legs and deliver both incoming rops to A-side. Another option is to fork
the call with bridge, the bad thing is that as soon as FS receives
progress/audio from 1 leg, FS discards the other one, not good for me :)


Thanks, and I hope you can enlighten me!

David


On Wed, Jan 13, 2010 at 6:21 PM, Michael Jerris <mike at jerris.com> wrote:

> Are you asking if you can mux all of the progress audio from multiple
> b-legs?  if so, no, and why would you want to?
>
> Mike
>
> On Jan 13, 2010, at 6:15 AM, David Villasmil wrote:
>
> > hello
> >
> > is it possible to bridge multiple b-legs and provide all audio
> > (progress) until there is an answer on one channel?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From david.villasmil.work at gmail.com  Wed Jan 13 12:23:20 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 13 Jan 2010 21:23:20 +0100
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0F362D9@BESONE.corp.dom>
References: <A2167AE7633A5C4DAE56EEBD1428ECC0F362D9@BESONE.corp.dom>
Message-ID: <9853f4ff1001131223g68ff7baev3622d19deac45677@mail.gmail.com>

Hello,

That's easy ;) (I'm pretty new)

var = session:getVariable("uuid")

but you need to have 2 sessions.

Hope this helps

David

On Tue, Jan 12, 2010 at 10:43 AM, Cavalera Claudio Luigi <
Claudio.Cavalera at italtel.it> wrote:

> Hello,
> this should be simple in theory therefore I'm probably missing the right
> way to do it.
> I want to play with sessions in lua, bridge them, park them, etc...
>
> example1: Consider this simple lua script in which i create two
> sessions:
>
> api = freeswitch.API();
> api:execute("originate", "sofia/internal/1001%192.168.1.1 &park");
> api:execute("originate", "sofia/internal/1002%192.168.1.1 &park");
>
> now if i want to bridge them i suppose i should use something like
>
> api:execute("uuid_bridge", "uuid_1 uuid_2");
>
> but how do i get uuid_1 and uuid_2, i.e. the uuids of the two sessions?
>
>
> example2: I could create sessions with
>
> local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
> local session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");
>
> but then there is NOT a bridge API to bridge the sessions like:
> bridge(session1, session2);
>
> I admit I have not yet understood why such bridge possibility exist in
> javascript but does not exist in lua.
> http://wiki.freeswitch.org/wiki/Javascript_Misc_bridge
> I guess there is a reason for this but I can't figure it out.
>
>
> example3: yet another possibility
> local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
> session1:originate(session2, "sofia/internal/1002%192.168.1.1", 1000);
>
> but it does not work either.
>
> Besides with this third example something strange happen:
>
> freeswitch at internal> show channels
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr
> ,dest,application,application_data,dialplan,context,read_codec,read_rate
> ,write_codec,write_rate,secure
> 1c5db2df-14ce-4516-94f2-bb7c087e0802,outbound,2010-01-12
> 10:50:59,1263289859,sofia/internal/1004,CS_REPORTING,FreeSWITCH,00000000
> 00,,1004,,,,default,PCMA,8000,PCMA,8000,
>
> 1 total.
>
> freeswitch at internal> uuid_kill 1c5db2df-14ce-4516-94f2-bb7c087e0802
> -ERR No Such Channel!
>
> freeswitch at internal> show channels
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr
> ,dest,application,application_data,dialplan,context,read_codec,read_rate
> ,write_codec,write_rate,secure
> 1c5db2df-14ce-4516-94f2-bb7c087e0802,outbound,2010-01-12
> 10:50:59,1263289859,sofia/internal/1004,CS_REPORTING,FreeSWITCH,00000000
> 00,,1004,,,,default,PCMA,8000,PCMA,8000,
>
> 1 total.
>
> freeswitch at internal>
>
> If you are interested the full log is here:
> http://pastebin.freeswitch.org/11757
> but I admit i'm not on latest trunk yet!
>
> Thanks.
> Ciao,
> Claudio
>
>
> PS: Is there a reason why there is a uuid_park command but not
> uuid_valet_park ?
>
>
> Internet Email Confidentiality Footer
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From mike at jerris.com  Wed Jan 13 12:41:51 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 15:41:51 -0500
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <1256032.191263413534672.JavaMail.root@zimbra>
References: <1256032.191263413534672.JavaMail.root@zimbra>
Message-ID: <692070C5-1915-4CDA-B3A3-C7BB3D0E2CA9@jerris.com>

What are you doing with these digits on the freeswitch side, what is it thinking is passed, and do you see anything in debug logs.  You also could uncomment the #define DEBUG_2833  from switch_rtp.c for more info.

Mike

On Jan 13, 2010, at 3:12 PM, Dan wrote:

> Thanks for your reply,
> 
> Wireshark shows all of the digits coming through cleanly, in the case below (also on pastebin)  its 0951*72789, freeswitch does not seem to be processing one or more of the rtp events.  Is there some specific debugging I should turn on that might give me some more info on why freeswitch is not processing some of these events?
> 
> http://pastebin.freeswitch.org/11780
> 
>     449 12.280148   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Zero 0
>     450 12.285108   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Zero 0
>     451 12.290082   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Zero 0
>     452 12.335129   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end)
>     453 12.340124   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end)
>     454 12.345124   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end)
>     455 12.365162   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21862, Time=359229277, Mark 
>     456 12.380167   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21863, Time=359229605 
>     461 12.385248   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9
>     462 12.385253   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9
>     465 12.390163   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9
>     466 12.435211   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end)
>     467 12.440201   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end)
>     468 12.445205   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end)
>     469 12.480251   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21870, Time=359230405 
>     470 12.485275   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Five 5
>     471 12.485284   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Five 5
>     472 12.490251   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Five 5
>     475 12.535277   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Five 5 (end)
>     476 12.540287   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Five 5 (end)
>     477 12.545287   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Five 5 (end)
>     478 12.580337   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21877, Time=359231197 
>     484 12.585337   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF One 1
>     485 12.585343   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF One 1
>     486 12.590362   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF One 1
>     492 12.635393   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF One 1 (end)
>     493 12.640368   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF One 1 (end)
>     494 12.645369   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF One 1 (end)
>     495 12.665406   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21884, Time=359231677, Mark 
>     496 12.680431   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21885, Time=359232005 
>     497 12.680438   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Star *
>     502 12.685408   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Star *
>     506 12.690412   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Star *
>     507 12.735450   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Star * (end)
>     508 12.740453   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Star * (end)
>     509 12.745469   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Star * (end)
>     511 12.780526   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21892, Time=359232805 
>     512 12.780534   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7
>     513 12.785502   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7
>     514 12.790492   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7
>     515 12.835544   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end)
>     516 12.840537   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end)
>     517 12.845537   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end)
>     523 12.880582   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21899, Time=359233605 
>     524 12.885593   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Two 2
>     525 12.885599   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Two 2
>     526 12.890580   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Two 2
>     528 12.935618   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Two 2 (end)
>     529 12.940618   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Two 2 (end)
>     530 12.945618   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Two 2 (end)
>     531 12.965656   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21906, Time=359234077, Mark 
>     532 12.980659   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21907, Time=359234413 
>     538 12.985690   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7
>     539 12.985695   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7
>     540 12.990657   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7
>     541 13.035703   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end)
>     542 13.040701   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end)
>     543 13.045700   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end)
>     544 13.080762   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21914, Time=359235205 
>     545 13.080768   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Eight 8
>     546 13.085736   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Eight 8
>     547 13.090798   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Eight 8
>     549 13.135809   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end)
>     550 13.140780   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end)
>     551 13.145793   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end)
>     554 13.180827   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21921, Time=359236005 
>     558 13.185847   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9
>     559 13.185854   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9
>     560 13.190821   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9
>     566 13.235872   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end)
>     567 13.240864   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end)
>     568 13.245872   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end)
> ----- Original Message -----
> From: "Brian West" <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wednesday, January 13, 2010 12:39:52 PM
> Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF
> 
> Cisco gateways can detect DTMF at random if you have kids screaming or loud noises in the background... very unreliable.
> 
> /b
> 
> On Jan 13, 2010, at 1:36 PM, Dan wrote:
> 
> Sorry, I should have made that more clear, this is using 2833.  As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through).  
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From mike at jerris.com  Wed Jan 13 12:43:59 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 15:43:59 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001131220u72ab5177x287a6dcb84f9b185@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<EA8724C8-84A8-44DB-9F76-EFA517E76402@jerris.com>
	<9853f4ff1001131220u72ab5177x287a6dcb84f9b185@mail.gmail.com>
Message-ID: <3BED4A88-87DB-4D9A-979D-6DBFBEA21778@jerris.com>

Muxing 2 ringtones together would result in complete nonsense, especially in the case of custom ringback.  How could this ever be usable?

Mike

On Jan 13, 2010, at 3:20 PM, David Villasmil wrote:

> Thanks for answering,
> 
> Well, imagine I have a content provider which will deliver custom ringbacks via SIP INVITES, they point would be to receive A-leg then bridge to 2 B-legs and deliver both incoming rops to A-side. Another option is to fork the call with bridge, the bad thing is that as soon as FS receives progress/audio from 1 leg, FS discards the other one, not good for me :)
> 
> 
> Thanks, and I hope you can enlighten me!
> 
> David
> 
> 
> On Wed, Jan 13, 2010 at 6:21 PM, Michael Jerris <mike at jerris.com> wrote:
> Are you asking if you can mux all of the progress audio from multiple b-legs?  if so, no, and why would you want to?
> 
> Mike
> 
> On Jan 13, 2010, at 6:15 AM, David Villasmil wrote:
> 
> > hello
> >
> > is it possible to bridge multiple b-legs and provide all audio
> > (progress) until there is an answer on one channel?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
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From freeswitch-users at digitaldan.com  Wed Jan 13 12:54:25 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Wed, 13 Jan 2010 13:54:25 -0700 (MST)
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <692070C5-1915-4CDA-B3A3-C7BB3D0E2CA9@jerris.com>
Message-ID: <18293564.271263416065447.JavaMail.root@zimbra>

Thanks, I will try and turn up the debugging on it. I have a lua script that is calling session:getDigits(20,"#",15000) . When I watch it under fs_cli I'm getting this output 

----- Original Message ----- 
From: "Michael Jerris" <mike at jerris.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 1:41:51 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

What are you doing with these digits on the freeswitch side, what is it thinking is passed, and do you see anything in debug logs. You also could uncomment the #define DEBUG_2833 from switch_rtp.c for more info. 


Mike 




On Jan 13, 2010, at 3:12 PM, Dan wrote: 




Thanks for your reply, 


Wireshark shows all of the digits coming through cleanly, in the case below (also on pastebin) its 0951*72789, freeswitch does not seem to be processing one or more of the rtp events. Is there some specific debugging I should turn on that might give me some more info on why freeswitch is not processing some of these events? 


http://pastebin.freeswitch.org/11780 



449 12.280148 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
450 12.285108 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
451 12.290082 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
452 12.335129 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
453 12.340124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
454 12.345124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
455 12.365162 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21862, Time=359229277, Mark 
456 12.380167 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21863, Time=359229605 
461 12.385248 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
462 12.385253 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
465 12.390163 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
466 12.435211 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
467 12.440201 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
468 12.445205 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
469 12.480251 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21870, Time=359230405 
470 12.485275 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
471 12.485284 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
472 12.490251 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
475 12.535277 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
476 12.540287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
477 12.545287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
478 12.580337 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21877, Time=359231197 
484 12.585337 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
485 12.585343 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
486 12.590362 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
492 12.635393 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
493 12.640368 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
494 12.645369 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
495 12.665406 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21884, Time=359231677, Mark 
496 12.680431 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21885, Time=359232005 
497 12.680438 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
502 12.685408 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
506 12.690412 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
507 12.735450 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
508 12.740453 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
509 12.745469 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
511 12.780526 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21892, Time=359232805 
512 12.780534 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
513 12.785502 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
514 12.790492 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
515 12.835544 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
516 12.840537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
517 12.845537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
523 12.880582 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21899, Time=359233605 
524 12.885593 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
525 12.885599 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
526 12.890580 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
528 12.935618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
529 12.940618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
530 12.945618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
531 12.965656 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21906, Time=359234077, Mark 
532 12.980659 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21907, Time=359234413 
538 12.985690 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
539 12.985695 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
540 12.990657 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
541 13.035703 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
542 13.040701 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
543 13.045700 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
544 13.080762 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21914, Time=359235205 
545 13.080768 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
546 13.085736 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
547 13.090798 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
549 13.135809 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
550 13.140780 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
551 13.145793 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
554 13.180827 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21921, Time=359236005 
558 13.185847 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
559 13.185854 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
560 13.190821 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
566 13.235872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
567 13.240864 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
568 13.245872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) ----- Original Message ----- 
From: "Brian West" < brian at freeswitch.org > 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 12:39:52 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

Cisco gateways can detect DTMF at random if you have kids screaming or loud noises in the background... very unreliable. 


/b 



On Jan 13, 2010, at 1:36 PM, Dan wrote: 


Sorry, I should have made that more clear, this is using 2833. As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through). 

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From freeswitch-users at digitaldan.com  Wed Jan 13 12:56:16 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Wed, 13 Jan 2010 13:56:16 -0700 (MST)
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <692070C5-1915-4CDA-B3A3-C7BB3D0E2CA9@jerris.com>
Message-ID: <8181470.301263416176200.JavaMail.root@zimbra>

Thanks, I will try and turn up the debugging on it. I have a lua script that is calling session:getDigits(20,"#",15000) . When I watch it under fs_cli I'm getting this output 



2010-01-13 10:44:15.750329 [NOTICE] switch_cpp.cpp:1129 main: Getting digits 
2010-01-13 10:44:17.586676 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 0:400 
2010-01-13 10:44:17.790705 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 5:400 
2010-01-13 10:44:17.910737 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 1:400 
2010-01-13 10:44:18.114779 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 7:400 
2010-01-13 10:44:18.234812 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 2:400 
2010-01-13 10:44:18.462832 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 8:400 
2010-01-13 10:44:18.582852 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 9:400 
2010-01-13 10:44:23.311740 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF #:400 
2010-01-13 10:44:23.311740 [DEBUG] switch_cpp.cpp:735 getDigits dtmf_buf: 0517289 




It will drop any one or more numbers from any part of that sequence. 


Thanks 
Dan 
(sorry for the duped message) ----- Original Message ----- 
From: "Michael Jerris" <mike at jerris.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 1:41:51 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

What are you doing with these digits on the freeswitch side, what is it thinking is passed, and do you see anything in debug logs. You also could uncomment the #define DEBUG_2833 from switch_rtp.c for more info. 


Mike 




On Jan 13, 2010, at 3:12 PM, Dan wrote: 




Thanks for your reply, 


Wireshark shows all of the digits coming through cleanly, in the case below (also on pastebin) its 0951*72789, freeswitch does not seem to be processing one or more of the rtp events. Is there some specific debugging I should turn on that might give me some more info on why freeswitch is not processing some of these events? 


http://pastebin.freeswitch.org/11780 



449 12.280148 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
450 12.285108 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
451 12.290082 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
452 12.335129 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
453 12.340124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
454 12.345124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
455 12.365162 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21862, Time=359229277, Mark 
456 12.380167 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21863, Time=359229605 
461 12.385248 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
462 12.385253 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
465 12.390163 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
466 12.435211 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
467 12.440201 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
468 12.445205 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
469 12.480251 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21870, Time=359230405 
470 12.485275 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
471 12.485284 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
472 12.490251 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
475 12.535277 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
476 12.540287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
477 12.545287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
478 12.580337 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21877, Time=359231197 
484 12.585337 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
485 12.585343 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
486 12.590362 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
492 12.635393 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
493 12.640368 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
494 12.645369 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
495 12.665406 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21884, Time=359231677, Mark 
496 12.680431 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21885, Time=359232005 
497 12.680438 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
502 12.685408 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
506 12.690412 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
507 12.735450 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
508 12.740453 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
509 12.745469 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
511 12.780526 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21892, Time=359232805 
512 12.780534 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
513 12.785502 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
514 12.790492 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
515 12.835544 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
516 12.840537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
517 12.845537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
523 12.880582 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21899, Time=359233605 
524 12.885593 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
525 12.885599 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
526 12.890580 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
528 12.935618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
529 12.940618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
530 12.945618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
531 12.965656 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21906, Time=359234077, Mark 
532 12.980659 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21907, Time=359234413 
538 12.985690 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
539 12.985695 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
540 12.990657 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
541 13.035703 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
542 13.040701 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
543 13.045700 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
544 13.080762 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21914, Time=359235205 
545 13.080768 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
546 13.085736 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
547 13.090798 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
549 13.135809 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
550 13.140780 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
551 13.145793 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
554 13.180827 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21921, Time=359236005 
558 13.185847 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
559 13.185854 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
560 13.190821 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
566 13.235872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
567 13.240864 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
568 13.245872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) ----- Original Message ----- 
From: "Brian West" < brian at freeswitch.org > 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 12:39:52 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

Cisco gateways can detect DTMF at random if you have kids screaming or loud noises in the background... very unreliable. 


/b 



On Jan 13, 2010, at 1:36 PM, Dan wrote: 


Sorry, I should have made that more clear, this is using 2833. As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through). 

_______________________________________________ 
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FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 
_______________________________________________ 
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 


_______________________________________________ 
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FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
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From msc at freeswitch.org  Wed Jan 13 14:36:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Jan 2010 14:36:23 -0800
Subject: [Freeswitch-users] 10 seconds delay on fallback
In-Reply-To: <201001130746.47968.sos@sokhapkin.dyndns.org>
References: <201001130746.47968.sos@sokhapkin.dyndns.org>
Message-ID: <87f2f3b91001131436k68fcc2f3kfa4ce66f4fd5c72@mail.gmail.com>

Which rev of FreeSWITCH are you running? Also, collect a full debug trace of
a working vs. non-working call so that you can compare what's happening. Put
those on pastebin so others can have a look.
-MC

On Wed, Jan 13, 2010 at 4:46 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> To implement fallback to backup PSTN routes and do LCR my dialplan executes
> the following commands:
>
> set execute_on_answer=set hangup_after_bridge=true
> set some custom channel variables
> bridge sofia/gateway1/number
> set some custom channel variables
> bridge sofia/gateway2/number
> ....
>
> This generally works fine if gateway1 returns SIP error, dialplan sets
> channel
> variables to another values and calls gateway2 immediately. But if GW1
> responds with 183 early media and SIP error after that, bad thing happens -
> dialplan continues immediately, sets channel variables, executes bridge
> application, but INVITE to GW2 is sent after 10 seconds delay. Caller gets
> 10
> extra seconds of post dial delay. I suspect the delay happens because of
> attempt to read audio frame, switch_ivr_originate() has the following
> lines:
>
> if (switch_channel_media_ready(caller_channel)) {
>        tstatus = switch_core_session_read_frame()
>      ...
>
> Any advice how to avoid extra delay in this situation?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mouncifbb at gmail.com  Wed Jan 13 15:00:32 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Wed, 13 Jan 2010 18:00:32 -0500
Subject: [Freeswitch-users] rtp-autoflush
Message-ID: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>

We are running freeswitch with bypass_media=true, if we change the following
settings :
rtp-autoflush-during-bridge and  rtp-autoflush
does it affect anything?

Thanks
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From anthony.minessale at gmail.com  Wed Jan 13 15:08:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Jan 2010 17:08:51 -0600
Subject: [Freeswitch-users] rtp-autoflush
In-Reply-To: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>
References: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>
Message-ID: <191c3a031001131508i713ef7b8r48ba21c875e4e723@mail.gmail.com>

no, as the name implies bypass_media does not touch media whatsoever and
hense rtp related settings do not come into play.


On Wed, Jan 13, 2010 at 5:00 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> We are running freeswitch with bypass_media=true, if we change the
> following settings :
> rtp-autoflush-during-bridge and  rtp-autoflush
> does it affect anything?
>
> Thanks
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mouncifbb at gmail.com  Wed Jan 13 15:20:13 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Wed, 13 Jan 2010 18:20:13 -0500
Subject: [Freeswitch-users] rtp-autoflush
In-Reply-To: <191c3a031001131508i713ef7b8r48ba21c875e4e723@mail.gmail.com>
References: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>
	<191c3a031001131508i713ef7b8r48ba21c875e4e723@mail.gmail.com>
Message-ID: <ade741d61001131520x1390ba23ud33dc4f3daba58f6@mail.gmail.com>

thank you, also is it okay to set bypass_media_after_bridge=true and
bypass_media=true at the same time?

On Wed, Jan 13, 2010 at 6:08 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> no, as the name implies bypass_media does not touch media whatsoever and
> hense rtp related settings do not come into play.
>
>
> On Wed, Jan 13, 2010 at 5:00 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>
>> We are running freeswitch with bypass_media=true, if we change the
>> following settings :
>> rtp-autoflush-during-bridge and  rtp-autoflush
>> does it affect anything?
>>
>> Thanks
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Wed Jan 13 15:30:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Jan 2010 17:30:33 -0600
Subject: [Freeswitch-users] rtp-autoflush
In-Reply-To: <ade741d61001131520x1390ba23ud33dc4f3daba58f6@mail.gmail.com>
References: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>
	<191c3a031001131508i713ef7b8r48ba21c875e4e723@mail.gmail.com>
	<ade741d61001131520x1390ba23ud33dc4f3daba58f6@mail.gmail.com>
Message-ID: <191c3a031001131530u6b105b35jcb09aa989a6ecce7@mail.gmail.com>

its better to use one of the other but it probably is ok

On Wed, Jan 13, 2010 at 5:20 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> thank you, also is it okay to set bypass_media_after_bridge=true and
> bypass_media=true at the same time?
>
>
> On Wed, Jan 13, 2010 at 6:08 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> no, as the name implies bypass_media does not touch media whatsoever and
>> hense rtp related settings do not come into play.
>>
>>
>> On Wed, Jan 13, 2010 at 5:00 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>>
>>> We are running freeswitch with bypass_media=true, if we change the
>>> following settings :
>>>  rtp-autoflush-during-bridge and  rtp-autoflush
>>> does it affect anything?
>>>
>>> Thanks
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From sos at sokhapkin.dyndns.org  Wed Jan 13 15:36:42 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 18:36:42 -0500
Subject: [Freeswitch-users] 10 seconds delay on fallback
In-Reply-To: <87f2f3b91001131436k68fcc2f3kfa4ce66f4fd5c72@mail.gmail.com>
References: <201001130746.47968.sos@sokhapkin.dyndns.org>
	<87f2f3b91001131436k68fcc2f3kfa4ce66f4fd5c72@mail.gmail.com>
Message-ID: <201001131836.42349.sos@sokhapkin.dyndns.org>

freeswitch at internal> version
FreeSWITCH Version 1.0.5pre10 (16012M)

The problem happens with bypass_media=true.

The lines in log related to the call are (sorry, need to hide IP addresses and 
numbers):

recv 1080 bytes from udp/[X.X.X.X]:5060 at 03:27:02.529350:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
...
2010-01-12 22:27:02.529311 [INFO] sofia.c:509 Update Callee ID to "XXXXX" 
<sip:XXXXXX at XXXXXX>
2010-01-12 22:27:02.529311 [DEBUG] sofia.c:3806 Channel sofia/cwu/XXXXXX at XXXXX 
entering state [proceeding][183]

2010-01-12 22:27:02.529311 [NOTICE] sofia.c:3885 Pre-Answer 2010-01-12 
22:27:02.529311 [DEBUG] switch_channel.c:2020 Send signal XXXXXXXXX [BREAK]
2010-01-12 22:27:02.529311 [DEBUG] sofia.c:3898 XXXXXXXXXXXXX receive message 
[PROGRESS]
2010-01-12 22:27:02.529311 [INFO] sofia.c:3898 Sending early media
2010-01-12 22:27:02.529311 [NOTICE] mod_sofia.c:1765 Pre-Answer XXXXXXXXXXX!
2010-01-12 22:27:02.529311 [DEBUG] switch_core_session.c:700 Send signal 
XXXXXXXXXX [BREAK]
2010-01-12 22:27:02.529311 [DEBUG] switch_core_session.c:639 Send signal 
XXXXXXXXXXXX [BREAK]
   ------------------------------------------------------------------------
recv 716 bytes from udp/[XXXXXXXXX]:5060 at 03:27:02.608352:
   ------------------------------------------------------------------------
   SIP/2.0 480 Temporarily Unavailable
....
2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXX SET 
[A2B_id]=[86339421]
EXECUTE XXXXXXXXXX set(A2B_tp_id_trunk=126)
2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXXXXXXXXX SET 
[A2B_tp_id_trunk]=[126]
EXECUTE XXXXXXXXX sched_hangup(+10800)
2010-01-12 22:27:02.611340 [DEBUG] switch_scheduler.c:214 Added task 109951 
switch_ivr_schedule_hangup (8e75053d-e864-4f34-b768-b41b6839dadd) to run at 
1263364022
EXECUTE XXXXXXXXXXX set(bypass_media=true)
2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXXXXXX SET 
[bypass_media]=[true]
EXECUTE XXXXXXXXX 
bridge([sip_contact_user=XXXXXX,sip_auth_username=,sip_auth_password=]sofia/cwu/XXXXXXX at XXXXXXX)
2010-01-12 22:27:02.612344 [DEBUG] switch_ivr.c:1199 XXXXXXXX receive message 
[MEDIA]
2010-01-12 22:27:02.612344 [DEBUG] switch_core_session.c:639 Send signal 
XXXXXXXXXXXX [BREAK]

10 seconds later:

2010-01-12 22:27:12.614323 [NOTICE] switch_channel.c:613 New Channel 
sofia/cwu/XXXXXXX at XXXXXXX [68cb4360-8da5-49fc-8fbd-f4a4e
4351a94]





On Wednesday 13 January 2010, Michael Collins wrote:
> Which rev of FreeSWITCH are you running? Also, collect a full debug trace
> of a working vs. non-working call so that you can compare what's happening.
> Put those on pastebin so others can have a look.
> -MC
>
> On Wed, Jan 13, 2010 at 4:46 AM, Sergey Okhapkin
>
> <sos at sokhapkin.dyndns.org>wrote:
> > To implement fallback to backup PSTN routes and do LCR my dialplan
> > executes the following commands:
> >
> > set execute_on_answer=set hangup_after_bridge=true
> > set some custom channel variables
> > bridge sofia/gateway1/number
> > set some custom channel variables
> > bridge sofia/gateway2/number
> > ....
> >
> > This generally works fine if gateway1 returns SIP error, dialplan sets
> > channel
> > variables to another values and calls gateway2 immediately. But if GW1
> > responds with 183 early media and SIP error after that, bad thing happens
> > - dialplan continues immediately, sets channel variables, executes bridge
> > application, but INVITE to GW2 is sent after 10 seconds delay. Caller
> > gets 10
> > extra seconds of post dial delay. I suspect the delay happens because of
> > attempt to read audio frame, switch_ivr_originate() has the following
> > lines:
> >
> > if (switch_channel_media_ready(caller_channel)) {
> >        tstatus = switch_core_session_read_frame()
> >      ...
> >
> > Any advice how to avoid extra delay in this situation?
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From david.villasmil.work at gmail.com  Wed Jan 13 18:10:23 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 14 Jan 2010 03:10:23 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <3BED4A88-87DB-4D9A-979D-6DBFBEA21778@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<EA8724C8-84A8-44DB-9F76-EFA517E76402@jerris.com>
	<9853f4ff1001131220u72ab5177x287a6dcb84f9b185@mail.gmail.com>
	<3BED4A88-87DB-4D9A-979D-6DBFBEA21778@jerris.com>
Message-ID: <9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>

MIke,

This is done on a daily basis by i.e. mobile companies, you dial a customer
number and you hear some music whilst hearing the ringing at the same time.

If it can not be done by muxing both rtps, can it be done the other way,
then?: (Another option is to fork the call with bridge, the bad thing is
that as soon as FS receives progress/audio from 1 leg, FS discards the other
one, not good for me )

Thanks


david

On Wed, Jan 13, 2010 at 9:43 PM, Michael Jerris <mike at jerris.com> wrote:

> Muxing 2 ringtones together would result in complete nonsense, especially
> in the case of custom ringback.  How could this ever be usable?
>
> Mike
>
> On Jan 13, 2010, at 3:20 PM, David Villasmil wrote:
>
> Thanks for answering,
>
> Well, imagine I have a content provider which will deliver custom ringbacks
> via SIP INVITES, they point would be to receive A-leg then bridge to 2
> B-legs and deliver both incoming rops to A-side. Another option is to fork
> the call with bridge, the bad thing is that as soon as FS receives
> progress/audio from 1 leg, FS discards the other one, not good for me :)
>
>
> Thanks, and I hope you can enlighten me!
>
> David
>
>
> On Wed, Jan 13, 2010 at 6:21 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> Are you asking if you can mux all of the progress audio from multiple
>> b-legs?  if so, no, and why would you want to?
>>
>> Mike
>>
>> On Jan 13, 2010, at 6:15 AM, David Villasmil wrote:
>>
>> > hello
>> >
>> > is it possible to bridge multiple b-legs and provide all audio
>> > (progress) until there is an answer on one channel?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Jan 13 18:42:13 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 20:42:13 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<EA8724C8-84A8-44DB-9F76-EFA517E76402@jerris.com>
	<9853f4ff1001131220u72ab5177x287a6dcb84f9b185@mail.gmail.com>
	<3BED4A88-87DB-4D9A-979D-6DBFBEA21778@jerris.com>
	<9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
Message-ID: <C3D4ADB0-7A80-45A9-91E1-FFABB987851B@freeswitch.org>

It can not be done currently and I don't expect this to ever be done.  The specs says the first target out of all invites to provide media wins and the others are canceled.  That is why it behaves the way it does.

You can ignore_early_media=true and set ringback=blah.wav if you wish to provide caller ringback.

/b

On Jan 13, 2010, at 8:10 PM, David Villasmil wrote:

> MIke,
> 
> This is done on a daily basis by i.e. mobile companies, you dial a customer number and you hear some music whilst hearing the ringing at the same time.
> 
> If it can not be done by muxing both rtps, can it be done the other way, then?: (Another option is to fork the call with bridge, the bad thing is that as soon as FS receives progress/audio from 1 leg, FS discards the other one, not good for me )
> 
> Thanks
> 
> 
> david



From mouncifbb at gmail.com  Wed Jan 13 18:46:15 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Wed, 13 Jan 2010 21:46:15 -0500
Subject: [Freeswitch-users] rtp-autoflush
In-Reply-To: <191c3a031001131530u6b105b35jcb09aa989a6ecce7@mail.gmail.com>
References: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>
	<191c3a031001131508i713ef7b8r48ba21c875e4e723@mail.gmail.com>
	<ade741d61001131520x1390ba23ud33dc4f3daba58f6@mail.gmail.com>
	<191c3a031001131530u6b105b35jcb09aa989a6ecce7@mail.gmail.com>
Message-ID: <ade741d61001131846v5379fe23ua550e7293515e84d@mail.gmail.com>

One more question, will it participate in codec negotiation sent by the
inbound voip provider?
Right now I have it setup this way:

Incoming Inbound DID (voip provider --> freeswitch(media bypass mode) ->
endpoint sip device.

Thanks again!

On Wed, Jan 13, 2010 at 6:30 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> its better to use one of the other but it probably is ok
>
>
> On Wed, Jan 13, 2010 at 5:20 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>
>> thank you, also is it okay to set bypass_media_after_bridge=true and
>> bypass_media=true at the same time?
>>
>>
>> On Wed, Jan 13, 2010 at 6:08 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> no, as the name implies bypass_media does not touch media whatsoever and
>>> hense rtp related settings do not come into play.
>>>
>>>
>>> On Wed, Jan 13, 2010 at 5:00 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>>>
>>>> We are running freeswitch with bypass_media=true, if we change the
>>>> following settings :
>>>>  rtp-autoflush-during-bridge and  rtp-autoflush
>>>> does it affect anything?
>>>>
>>>> Thanks
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sos at sokhapkin.dyndns.org  Wed Jan 13 19:05:43 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 22:05:43 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <C3D4ADB0-7A80-45A9-91E1-FFABB987851B@freeswitch.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
	<C3D4ADB0-7A80-45A9-91E1-FFABB987851B@freeswitch.org>
Message-ID: <201001132205.43306.sos@sokhapkin.dyndns.org>

Well, the question is - what is "media"? To me media is what is returned 
by "200 OK" response to INVITE. 18X provisional responses are NOT media, they 
are early media indications (well, even with RTP stream inside), which shall 
be sent back to the caller, but should NOT be accounted in the call 
processing. Early media is too early to be accounted. Only responses with SIP 
code 200 or more matter.

On Wednesday 13 January 2010, Brian West wrote:
> It can not be done currently and I don't expect this to ever be done.  The
> specs says the first target out of all invites to provide media wins and
> the others are canceled.  That is why it behaves the way it does.
>
> You can ignore_early_media=true and set ringback=blah.wav if you wish to
> provide caller ringback.
>
> /b
>
> On Jan 13, 2010, at 8:10 PM, David Villasmil wrote:
> > MIke,
> >
> > This is done on a daily basis by i.e. mobile companies, you dial a
> > customer number and you hear some music whilst hearing the ringing at the
> > same time.
> >
> > If it can not be done by muxing both rtps, can it be done the other way,
> > then?: (Another option is to fork the call with bridge, the bad thing is
> > that as soon as FS receives progress/audio from 1 leg, FS discards the
> > other one, not good for me )
> >
> > Thanks
> >
> >
> > david
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Wed Jan 13 19:11:38 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 21:11:38 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001132205.43306.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
	<C3D4ADB0-7A80-45A9-91E1-FFABB987851B@freeswitch.org>
	<201001132205.43306.sos@sokhapkin.dyndns.org>
Message-ID: <7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>

By all accounts its still Media and the first one to provide it in a forked dial is to be connected to the channel of the calling party even if its early... the call answer time is not started till the 200 is received.  

I'm not talking about billing or answered time either.. i'm talking pure early media and how it is to be handled in FreeSWITCH.

That my friend is in the specs to behave like that.

/b



On Jan 13, 2010, at 9:05 PM, Sergey Okhapkin wrote:

> Well, the question is - what is "media"? To me media is what is returned 
> by "200 OK" response to INVITE. 18X provisional responses are NOT media, they 
> are early media indications (well, even with RTP stream inside), which shall 
> be sent back to the caller, but should NOT be accounted in the call 
> processing. Early media is too early to be accounted. Only responses with SIP 
> code 200 or more matter.



From mike at jerris.com  Wed Jan 13 19:12:48 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 22:12:48 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<EA8724C8-84A8-44DB-9F76-EFA517E76402@jerris.com>
	<9853f4ff1001131220u72ab5177x287a6dcb84f9b185@mail.gmail.com>
	<3BED4A88-87DB-4D9A-979D-6DBFBEA21778@jerris.com>
	<9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
Message-ID: <FA68C6E7-732F-4104-BFF1-23333AD2A68E@jerris.com>

I have never seen a carrier do this and I still doubt that it would be  
at all usable.  What happens now is we essentially fork the dial,  
first with media wins.  Alternatively you can wait for first answer,  
and provide your own ringback tones.  Can you describe a bit more your  
use case as I just don't get it.  With what you describe, I imagine a  
call where one leg gets a sit tone due to problem with the number and  
it's muxed with a ringtone of the other b leg, or ring and busy at the  
same time.

Mike

On Jan 13, 2010, at 9:10 PM, David Villasmil <david.villasmil.work at gmail.com 
 > wrote:

> MIke,
>
> This is done on a daily basis by i.e. mobile companies, you dial a  
> customer number and you hear some music whilst hearing the ringing  
> at the same time.
>
> If it can not be done by muxing both rtps, can it be done the other  
> way, then?: (Another option is to fork the call with bridge, the bad  
> thing is that as soon as FS receives progress/audio from 1 leg, FS  
> discards the other one, not good for me )
>
> Thanks
>
>
> david
>
> On Wed, Jan 13, 2010 at 9:43 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> Muxing 2 ringtones together would result in complete nonsense,  
> especially in the case of custom ringback.  How could this ever be  
> usable?
>
> Mike
>
> On Jan 13, 2010, at 3:20 PM, David Villasmil wrote:
>
>> Thanks for answering,
>>
>> Well, imagine I have a content provider which will deliver custom  
>> ringbacks via SIP INVITES, they point would be to receive A-leg  
>> then bridge to 2 B-legs and deliver both incoming rops to A-side.  
>> Another option is to fork the call with bridge, the bad thing is  
>> that as soon as FS receives progress/audio from 1 leg, FS discards  
>> the other one, not good for me :)
>>
>>
>> Thanks, and I hope you can enlighten me!
>>
>> David
>>
>>
>> On Wed, Jan 13, 2010 at 6:21 PM, Michael Jerris <mike at jerris.com>  
>> wrote:
>> Are you asking if you can mux all of the progress audio from  
>> multiple b-legs?  if so, no, and why would you want to?
>>
>> Mike
>>
>> On Jan 13, 2010, at 6:15 AM, David Villasmil wrote:
>>
>> > hello
>> >
>> > is it possible to bridge multiple b-legs and provide all audio
>> > (progress) until there is an answer on one channel?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From null at invalid.name  Wed Jan 13 17:48:03 2010
From: null at invalid.name (Dan Lane)
Date: Thu, 14 Jan 2010 01:48:03 +0000
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <3BED4A88-87DB-4D9A-979D-6DBFBEA21778@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<EA8724C8-84A8-44DB-9F76-EFA517E76402@jerris.com>
	<9853f4ff1001131220u72ab5177x287a6dcb84f9b185@mail.gmail.com>
	<3BED4A88-87DB-4D9A-979D-6DBFBEA21778@jerris.com>
Message-ID: <b042b23b1001131748t229a2206y479e56666b037378@mail.gmail.com>

On Wed, Jan 13, 2010 at 8:43 PM, Michael Jerris <mike at jerris.com> wrote:
> Muxing 2 ringtones together would result in complete nonsense, especially in
> the case of custom ringback. ?How could this ever be usable?

It sounds like he wants to mux two audio streams so he can have the
normal ringback tone overlaid on a novelty ringback tone so as not to
confuse the caller (I understand this is quite a normal way of doing
such things)

The easiest way to do this (and the way I do it) is to record a normal
ringback tone then use sox to combine the custom ringback with the
normal one and use the resulting file as your ringback. You get fine
control over the timing and volume this way too.


From sos at sokhapkin.dyndns.org  Wed Jan 13 19:29:58 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 22:29:58 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132205.43306.sos@sokhapkin.dyndns.org>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
Message-ID: <201001132229.58830.sos@sokhapkin.dyndns.org>

If you wish FreeSWITCH to be well adopted by the community, then FS should 
follow the real world "rules" but not specs...

Don't you agree that 99% of SIP servers are set up to interconnect 
with "buggy" PSTN and should follow PSTN rules, but not SIP specs?

Some background - I did run asterisk for years, but switched to FS recently 
because of critical asterisk problems with SIP handling when asterisk is not 
in the media path. I spend a lot of time porting the billing system to FS. 
And I did it. What I got? Critical problems with FS early media handling.  
Hopefully I can switch back to asterisk if FS problems with early media will 
begin to draw customers away.

On Wednesday 13 January 2010, Brian West wrote:
> By all accounts its still Media and the first one to provide it in a forked
> dial is to be connected to the channel of the calling party even if its
> early... the call answer time is not started till the 200 is received.
>
> I'm not talking about billing or answered time either.. i'm talking pure
> early media and how it is to be handled in FreeSWITCH.
>
> That my friend is in the specs to behave like that.
>
> /b
>
> On Jan 13, 2010, at 9:05 PM, Sergey Okhapkin wrote:
> > Well, the question is - what is "media"? To me media is what is returned
> > by "200 OK" response to INVITE. 18X provisional responses are NOT media,
> > they are early media indications (well, even with RTP stream inside),
> > which shall be sent back to the caller, but should NOT be accounted in
> > the call processing. Early media is too early to be accounted. Only
> > responses with SIP code 200 or more matter.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Wed Jan 13 19:30:40 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 22:30:40 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
	<C3D4ADB0-7A80-45A9-91E1-FFABB987851B@freeswitch.org>
	<201001132205.43306.sos@sokhapkin.dyndns.org>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
Message-ID: <174D1183-2731-454B-AFEB-9F92F6101BB1@jerris.com>

And to reiterate, you can control this to wait for first answer, at  
your option.  These 2 modes are for totally different use cases.

Mike

On Jan 13, 2010, at 10:11 PM, Brian West <brian at freeswitch.org> wrote:

> By all accounts its still Media and the first one to provide it in a  
> forked dial is to be connected to the channel of the calling party  
> even if its early... the call answer time is not started till the  
> 200 is received.
>
> I'm not talking about billing or answered time either.. i'm talking  
> pure early media and how it is to be handled in FreeSWITCH.
>
> That my friend is in the specs to behave like that.
>
> /b
>
>
>
> On Jan 13, 2010, at 9:05 PM, Sergey Okhapkin wrote:
>
>> Well, the question is - what is "media"? To me media is what is  
>> returned
>> by "200 OK" response to INVITE. 18X provisional responses are NOT  
>> media, they
>> are early media indications (well, even with RTP stream inside),  
>> which shall
>> be sent back to the caller, but should NOT be accounted in the call
>> processing. Early media is too early to be accounted. Only  
>> responses with SIP
>> code 200 or more matter.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From dave at 3c.co.uk  Wed Jan 13 19:32:05 2010
From: dave at 3c.co.uk (David Knell)
Date: Thu, 14 Jan 2010 03:32:05 +0000
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
	<C3D4ADB0-7A80-45A9-91E1-FFABB987851B@freeswitch.org>
	<201001132205.43306.sos@sokhapkin.dyndns.org>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
Message-ID: <1263439925.11216.36.camel@local.freepabx.com>

What would happen if several outdials were made from a conference,
rather than using a forked dial?

--Dave

> By all accounts its still Media and the first one to provide it in a forked dial is to be connected to the channel of the calling party even if its early... the call answer time is not started till the 200 is received.  
> 
> I'm not talking about billing or answered time either.. i'm talking pure early media and how it is to be handled in FreeSWITCH.
> 
> That my friend is in the specs to behave like that.
> 
> /b
> 
> 
> 
> On Jan 13, 2010, at 9:05 PM, Sergey Okhapkin wrote:
> 
> > Well, the question is - what is "media"? To me media is what is returned 
> > by "200 OK" response to INVITE. 18X provisional responses are NOT media, they 
> > are early media indications (well, even with RTP stream inside), which shall 
> > be sent back to the caller, but should NOT be accounted in the call 
> > processing. Early media is too early to be accounted. Only responses with SIP 
> > code 200 or more matter.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From anthony.minessale at gmail.com  Wed Jan 13 19:33:15 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Jan 2010 21:33:15 -0600
Subject: [Freeswitch-users] rtp-autoflush
In-Reply-To: <ade741d61001131846v5379fe23ua550e7293515e84d@mail.gmail.com>
References: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>
	<191c3a031001131508i713ef7b8r48ba21c875e4e723@mail.gmail.com>
	<ade741d61001131520x1390ba23ud33dc4f3daba58f6@mail.gmail.com>
	<191c3a031001131530u6b105b35jcb09aa989a6ecce7@mail.gmail.com>
	<ade741d61001131846v5379fe23ua550e7293515e84d@mail.gmail.com>
Message-ID: <191c3a031001131933g2c290efds5b8342d1ce55ca0c@mail.gmail.com>

Not if you set inbound-bypass-media = true in the sip profile or set
late-negotiaation = true and set bypass_media channel var to true from the
dp

On Jan 13, 2010 8:51 PM, "Mouncif Benniane" <mouncifbb at gmail.com> wrote:

One more question, will it participate in codec negotiation sent by the
inbound voip provider?
Right now I have it setup this way:

Incoming Inbound DID (voip provider --> freeswitch(media bypass mode) ->
endpoint sip device.

Thanks again!

On Wed, Jan 13, 2010 at 6:30 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote: > > its be...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Wed Jan 13 19:39:56 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 21:39:56 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <1263439925.11216.36.camel@local.freepabx.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
	<C3D4ADB0-7A80-45A9-91E1-FFABB987851B@freeswitch.org>
	<201001132205.43306.sos@sokhapkin.dyndns.org>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
	<1263439925.11216.36.camel@local.freepabx.com>
Message-ID: <26079A18-5968-4E1E-A276-9DE889E83FA4@freeswitch.org>

That would work but its sub optimal.  What he wants to accomplish is not do able and not going to ever happen at this rate.

/b

On Jan 13, 2010, at 9:32 PM, David Knell wrote:

> What would happen if several outdials were made from a conference,
> rather than using a forked dial?
> 
> --Dave



From brian at freeswitch.org  Wed Jan 13 19:44:27 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 21:44:27 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001132229.58830.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132205.43306.sos@sokhapkin.dyndns.org>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
	<201001132229.58830.sos@sokhapkin.dyndns.org>
Message-ID: <6D061C37-43A6-48D1-A8E3-549FC3E8679E@freeswitch.org>

Can you elaborate on these "Critical" issues you seem to be having?  Why aren't you opening a jira for them if they are that critical to your needs?

/b


On Jan 13, 2010, at 9:29 PM, Sergey Okhapkin wrote:

> If you wish FreeSWITCH to be well adopted by the community, then FS should 
> follow the real world "rules" but not specs...
> 
> Don't you agree that 99% of SIP servers are set up to interconnect 
> with "buggy" PSTN and should follow PSTN rules, but not SIP specs?
> 
> Some background - I did run asterisk for years, but switched to FS recently 
> because of critical asterisk problems with SIP handling when asterisk is not 
> in the media path. I spend a lot of time porting the billing system to FS. 
> And I did it. What I got? Critical problems with FS early media handling.  
> Hopefully I can switch back to asterisk if FS problems with early media will 
> begin to draw customers away.



From sos at sokhapkin.dyndns.org  Wed Jan 13 19:45:38 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 22:45:38 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <174D1183-2731-454B-AFEB-9F92F6101BB1@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
	<174D1183-2731-454B-AFEB-9F92F6101BB1@jerris.com>
Message-ID: <201001132245.38117.sos@sokhapkin.dyndns.org>

How to wait for first answer and pass early media back to the caller?

On Wednesday 13 January 2010, Michael Jerris wrote:
> And to reiterate, you can control this to wait for first answer, at
> your option.  These 2 modes are for totally different use cases.
>
> Mike
>
> On Jan 13, 2010, at 10:11 PM, Brian West <brian at freeswitch.org> wrote:
> > By all accounts its still Media and the first one to provide it in a
> > forked dial is to be connected to the channel of the calling party
> > even if its early... the call answer time is not started till the
> > 200 is received.
> >
> > I'm not talking about billing or answered time either.. i'm talking
> > pure early media and how it is to be handled in FreeSWITCH.
> >
> > That my friend is in the specs to behave like that.
> >
> > /b
> >
> > On Jan 13, 2010, at 9:05 PM, Sergey Okhapkin wrote:
> >> Well, the question is - what is "media"? To me media is what is
> >> returned
> >> by "200 OK" response to INVITE. 18X provisional responses are NOT
> >> media, they
> >> are early media indications (well, even with RTP stream inside),
> >> which shall
> >> be sent back to the caller, but should NOT be accounted in the call
> >> processing. Early media is too early to be accounted. Only
> >> responses with SIP
> >> code 200 or more matter.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Wed Jan 13 19:47:53 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 22:47:53 -0500
Subject: [Freeswitch-users] 10 seconds delay on fallback
In-Reply-To: <201001131836.42349.sos@sokhapkin.dyndns.org>
References: <201001130746.47968.sos@sokhapkin.dyndns.org>
	<87f2f3b91001131436k68fcc2f3kfa4ce66f4fd5c72@mail.gmail.com>
	<201001131836.42349.sos@sokhapkin.dyndns.org>
Message-ID: <27ADEBDB-E9C6-42AF-9A19-1CDF918A5F6E@jerris.com>

Try out a more recent trunk, specifically, svn r16193 had an important fix for signaling handling in bypass and proxy media modes.

Mike

On Jan 13, 2010, at 6:36 PM, Sergey Okhapkin wrote:

> freeswitch at internal> version
> FreeSWITCH Version 1.0.5pre10 (16012M)
> 
> The problem happens with bypass_media=true.
> 
> The lines in log related to the call are (sorry, need to hide IP addresses and 
> numbers):
> 
> recv 1080 bytes from udp/[X.X.X.X]:5060 at 03:27:02.529350:
>   ------------------------------------------------------------------------
>   SIP/2.0 183 Session Progress
> ...
> 2010-01-12 22:27:02.529311 [INFO] sofia.c:509 Update Callee ID to "XXXXX" 
> <sip:XXXXXX at XXXXXX>
> 2010-01-12 22:27:02.529311 [DEBUG] sofia.c:3806 Channel sofia/cwu/XXXXXX at XXXXX 
> entering state [proceeding][183]
> 
> 2010-01-12 22:27:02.529311 [NOTICE] sofia.c:3885 Pre-Answer 2010-01-12 
> 22:27:02.529311 [DEBUG] switch_channel.c:2020 Send signal XXXXXXXXX [BREAK]
> 2010-01-12 22:27:02.529311 [DEBUG] sofia.c:3898 XXXXXXXXXXXXX receive message 
> [PROGRESS]
> 2010-01-12 22:27:02.529311 [INFO] sofia.c:3898 Sending early media
> 2010-01-12 22:27:02.529311 [NOTICE] mod_sofia.c:1765 Pre-Answer XXXXXXXXXXX!
> 2010-01-12 22:27:02.529311 [DEBUG] switch_core_session.c:700 Send signal 
> XXXXXXXXXX [BREAK]
> 2010-01-12 22:27:02.529311 [DEBUG] switch_core_session.c:639 Send signal 
> XXXXXXXXXXXX [BREAK]
>   ------------------------------------------------------------------------
> recv 716 bytes from udp/[XXXXXXXXX]:5060 at 03:27:02.608352:
>   ------------------------------------------------------------------------
>   SIP/2.0 480 Temporarily Unavailable
> ....
> 2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXX SET 
> [A2B_id]=[86339421]
> EXECUTE XXXXXXXXXX set(A2B_tp_id_trunk=126)
> 2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXXXXXXXXX SET 
> [A2B_tp_id_trunk]=[126]
> EXECUTE XXXXXXXXX sched_hangup(+10800)
> 2010-01-12 22:27:02.611340 [DEBUG] switch_scheduler.c:214 Added task 109951 
> switch_ivr_schedule_hangup (8e75053d-e864-4f34-b768-b41b6839dadd) to run at 
> 1263364022
> EXECUTE XXXXXXXXXXX set(bypass_media=true)
> 2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXXXXXX SET 
> [bypass_media]=[true]
> EXECUTE XXXXXXXXX 
> bridge([sip_contact_user=XXXXXX,sip_auth_username=,sip_auth_password=]sofia/cwu/XXXXXXX at XXXXXXX)
> 2010-01-12 22:27:02.612344 [DEBUG] switch_ivr.c:1199 XXXXXXXX receive message 
> [MEDIA]
> 2010-01-12 22:27:02.612344 [DEBUG] switch_core_session.c:639 Send signal 
> XXXXXXXXXXXX [BREAK]
> 
> 10 seconds later:
> 
> 2010-01-12 22:27:12.614323 [NOTICE] switch_channel.c:613 New Channel 
> sofia/cwu/XXXXXXX at XXXXXXX [68cb4360-8da5-49fc-8fbd-f4a4e
> 4351a94]
> 
> 
> 
> 
> 
> On Wednesday 13 January 2010, Michael Collins wrote:
>> Which rev of FreeSWITCH are you running? Also, collect a full debug trace
>> of a working vs. non-working call so that you can compare what's happening.
>> Put those on pastebin so others can have a look.
>> -MC
>> 
>> On Wed, Jan 13, 2010 at 4:46 AM, Sergey Okhapkin
>> 
>> <sos at sokhapkin.dyndns.org>wrote:
>>> To implement fallback to backup PSTN routes and do LCR my dialplan
>>> executes the following commands:
>>> 
>>> set execute_on_answer=set hangup_after_bridge=true
>>> set some custom channel variables
>>> bridge sofia/gateway1/number
>>> set some custom channel variables
>>> bridge sofia/gateway2/number
>>> ....
>>> 
>>> This generally works fine if gateway1 returns SIP error, dialplan sets
>>> channel
>>> variables to another values and calls gateway2 immediately. But if GW1
>>> responds with 183 early media and SIP error after that, bad thing happens
>>> - dialplan continues immediately, sets channel variables, executes bridge
>>> application, but INVITE to GW2 is sent after 10 seconds delay. Caller
>>> gets 10
>>> extra seconds of post dial delay. I suspect the delay happens because of
>>> attempt to read audio frame, switch_ivr_originate() has the following
>>> lines:
>>> 
>>> if (switch_channel_media_ready(caller_channel)) {
>>>       tstatus = switch_core_session_read_frame()
>>>     ...
>>> 
>>> Any advice how to avoid extra delay in this situation?
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Wed Jan 13 19:49:12 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 22:49:12 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001132229.58830.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132205.43306.sos@sokhapkin.dyndns.org>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
	<201001132229.58830.sos@sokhapkin.dyndns.org>
Message-ID: <464CD1B0-95D0-4D6D-B330-87B059FB76A6@jerris.com>

What exactly are your problems?   Is this just the "10 second" thread you posted today?

Mike

On Jan 13, 2010, at 10:29 PM, Sergey Okhapkin wrote:

> If you wish FreeSWITCH to be well adopted by the community, then FS should 
> follow the real world "rules" but not specs...
> 
> Don't you agree that 99% of SIP servers are set up to interconnect 
> with "buggy" PSTN and should follow PSTN rules, but not SIP specs?
> 
> Some background - I did run asterisk for years, but switched to FS recently 
> because of critical asterisk problems with SIP handling when asterisk is not 
> in the media path. I spend a lot of time porting the billing system to FS. 
> And I did it. What I got? Critical problems with FS early media handling.  
> Hopefully I can switch back to asterisk if FS problems with early media will 
> begin to draw customers away.



From brian at freeswitch.org  Wed Jan 13 19:53:23 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 21:53:23 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001132245.38117.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
	<174D1183-2731-454B-AFEB-9F92F6101BB1@jerris.com>
	<201001132245.38117.sos@sokhapkin.dyndns.org>
Message-ID: <02DBC6DF-7B8A-435A-B89D-367BE0FE80C7@freeswitch.org>

Can you elaborate what you mean?  I'm guessing you want to fork dial X calls and pass a mux version of all that early media back to the A-Leg.  In which case that is 100% impossible and impractical.  This scenario never happens on the PSTN. 

Now if you want to ignore the early media and provide your own ringback and pass media once the call is answered thats doable. 

http://wiki.freeswitch.org/wiki/Channel_Variables#ringback
&&
http://wiki.freeswitch.org/wiki/Channel_Variables#ignore_early_media

If this isn't the case on either one please clarify.

/b


On Jan 13, 2010, at 9:45 PM, Sergey Okhapkin wrote:

> How to wait for first answer and pass early media back to the caller?
> 
> On Wednesday 13 January 2010, Michael Jerris wrote:
>> And to reiterate, you can control this to wait for first answer, at
>> your option.  These 2 modes are for totally different use cases.
>> 
>> Mike
> 



From sos at sokhapkin.dyndns.org  Wed Jan 13 19:55:23 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 22:55:23 -0500
Subject: [Freeswitch-users] 10 seconds delay on fallback
In-Reply-To: <27ADEBDB-E9C6-42AF-9A19-1CDF918A5F6E@jerris.com>
References: <201001130746.47968.sos@sokhapkin.dyndns.org>
	<201001131836.42349.sos@sokhapkin.dyndns.org>
	<27ADEBDB-E9C6-42AF-9A19-1CDF918A5F6E@jerris.com>
Message-ID: <201001132255.23846.sos@sokhapkin.dyndns.org>

Thank you for the heads up, will do.

On Wednesday 13 January 2010, Michael Jerris wrote:
> Try out a more recent trunk, specifically, svn r16193 had an important fix
> for signaling handling in bypass and proxy media modes.
>
> Mike
>
> On Jan 13, 2010, at 6:36 PM, Sergey Okhapkin wrote:
> > freeswitch at internal> version
> > FreeSWITCH Version 1.0.5pre10 (16012M)
> >
> > The problem happens with bypass_media=true.
> >
> > The lines in log related to the call are (sorry, need to hide IP
> > addresses and numbers):
> >
> > recv 1080 bytes from udp/[X.X.X.X]:5060 at 03:27:02.529350:
> >  
> > ------------------------------------------------------------------------
> > SIP/2.0 183 Session Progress
> > ...
> > 2010-01-12 22:27:02.529311 [INFO] sofia.c:509 Update Callee ID to "XXXXX"
> > <sip:XXXXXX at XXXXXX>
> > 2010-01-12 22:27:02.529311 [DEBUG] sofia.c:3806 Channel
> > sofia/cwu/XXXXXX at XXXXX entering state [proceeding][183]
> >
> > 2010-01-12 22:27:02.529311 [NOTICE] sofia.c:3885 Pre-Answer 2010-01-12
> > 22:27:02.529311 [DEBUG] switch_channel.c:2020 Send signal XXXXXXXXX
> > [BREAK] 2010-01-12 22:27:02.529311 [DEBUG] sofia.c:3898 XXXXXXXXXXXXX
> > receive message [PROGRESS]
> > 2010-01-12 22:27:02.529311 [INFO] sofia.c:3898 Sending early media
> > 2010-01-12 22:27:02.529311 [NOTICE] mod_sofia.c:1765 Pre-Answer
> > XXXXXXXXXXX! 2010-01-12 22:27:02.529311 [DEBUG] switch_core_session.c:700
> > Send signal XXXXXXXXXX [BREAK]
> > 2010-01-12 22:27:02.529311 [DEBUG] switch_core_session.c:639 Send signal
> > XXXXXXXXXXXX [BREAK]
> >  
> > ------------------------------------------------------------------------
> > recv 716 bytes from udp/[XXXXXXXXX]:5060 at 03:27:02.608352:
> >  
> > ------------------------------------------------------------------------
> > SIP/2.0 480 Temporarily Unavailable
> > ....
> > 2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXX SET
> > [A2B_id]=[86339421]
> > EXECUTE XXXXXXXXXX set(A2B_tp_id_trunk=126)
> > 2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXXXXXXXXX SET
> > [A2B_tp_id_trunk]=[126]
> > EXECUTE XXXXXXXXX sched_hangup(+10800)
> > 2010-01-12 22:27:02.611340 [DEBUG] switch_scheduler.c:214 Added task
> > 109951 switch_ivr_schedule_hangup (8e75053d-e864-4f34-b768-b41b6839dadd)
> > to run at 1263364022
> > EXECUTE XXXXXXXXXXX set(bypass_media=true)
> > 2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXXXXXX SET
> > [bypass_media]=[true]
> > EXECUTE XXXXXXXXX
> > bridge([sip_contact_user=XXXXXX,sip_auth_username=,sip_auth_password=]sof
> >ia/cwu/XXXXXXX at XXXXXXX) 2010-01-12 22:27:02.612344 [DEBUG]
> > switch_ivr.c:1199 XXXXXXXX receive message [MEDIA]
> > 2010-01-12 22:27:02.612344 [DEBUG] switch_core_session.c:639 Send signal
> > XXXXXXXXXXXX [BREAK]
> >
> > 10 seconds later:
> >
> > 2010-01-12 22:27:12.614323 [NOTICE] switch_channel.c:613 New Channel
> > sofia/cwu/XXXXXXX at XXXXXXX [68cb4360-8da5-49fc-8fbd-f4a4e
> > 4351a94]
> >
> > On Wednesday 13 January 2010, Michael Collins wrote:
> >> Which rev of FreeSWITCH are you running? Also, collect a full debug
> >> trace of a working vs. non-working call so that you can compare what's
> >> happening. Put those on pastebin so others can have a look.
> >> -MC
> >>
> >> On Wed, Jan 13, 2010 at 4:46 AM, Sergey Okhapkin
> >>
> >> <sos at sokhapkin.dyndns.org>wrote:
> >>> To implement fallback to backup PSTN routes and do LCR my dialplan
> >>> executes the following commands:
> >>>
> >>> set execute_on_answer=set hangup_after_bridge=true
> >>> set some custom channel variables
> >>> bridge sofia/gateway1/number
> >>> set some custom channel variables
> >>> bridge sofia/gateway2/number
> >>> ....
> >>>
> >>> This generally works fine if gateway1 returns SIP error, dialplan sets
> >>> channel
> >>> variables to another values and calls gateway2 immediately. But if GW1
> >>> responds with 183 early media and SIP error after that, bad thing
> >>> happens - dialplan continues immediately, sets channel variables,
> >>> executes bridge application, but INVITE to GW2 is sent after 10 seconds
> >>> delay. Caller gets 10
> >>> extra seconds of post dial delay. I suspect the delay happens because
> >>> of attempt to read audio frame, switch_ivr_originate() has the
> >>> following lines:
> >>>
> >>> if (switch_channel_media_ready(caller_channel)) {
> >>>       tstatus = switch_core_session_read_frame()
> >>>     ...
> >>>
> >>> Any advice how to avoid extra delay in this situation?
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> >>>s http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mouncifbb at gmail.com  Wed Jan 13 19:54:57 2010
From: mouncifbb at gmail.com (Mouncifbb)
Date: Wed, 13 Jan 2010 22:54:57 -0500
Subject: [Freeswitch-users] rtp-autoflush
In-Reply-To: <191c3a031001131933g2c290efds5b8342d1ce55ca0c@mail.gmail.com>
References: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>
	<191c3a031001131508i713ef7b8r48ba21c875e4e723@mail.gmail.com>
	<ade741d61001131520x1390ba23ud33dc4f3daba58f6@mail.gmail.com>
	<191c3a031001131530u6b105b35jcb09aa989a6ecce7@mail.gmail.com>
	<ade741d61001131846v5379fe23ua550e7293515e84d@mail.gmail.com>
	<191c3a031001131933g2c290efds5b8342d1ce55ca0c@mail.gmail.com>
Message-ID: <86CE8043-D1F0-4D75-8EE9-777AA6BBA27B@gmail.com>

Is the external.XML profile suited to be used for my scenario when  
bridging to the sip endpoint ?

Sent from my iPhone

On Jan 13, 2010, at 10:33 PM, Anthony Minessale <anthony.minessale at gmail.com 
 > wrote:

> Not if you set inbound-bypass-media = true in the sip profile or set  
> late-negotiaation = true and set bypass_media channel var to true  
> from the dp
>
>> On Jan 13, 2010 8:51 PM, "Mouncif Benniane" <mouncifbb at gmail.com>  
>> wrote:
>>
>> One more question, will it participate in codec negotiation sent by  
>> the inbound voip provider?
>> Right now I have it setup this way:
>>
>> Incoming Inbound DID (voip provider --> freeswitch(media bypass  
>> mode) ->  endpoint sip device.
>>
>> Thanks again!
>> On Wed, Jan 13, 2010 at 6:30 PM, Anthony Minessale <anthony.minessale at gmail.com 
>> > wrote: > > its be...
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From mouncifbb at gmail.com  Wed Jan 13 20:00:51 2010
From: mouncifbb at gmail.com (Mouncifbb)
Date: Wed, 13 Jan 2010 23:00:51 -0500
Subject: [Freeswitch-users] rtp-autoflush
In-Reply-To: <191c3a031001131933g2c290efds5b8342d1ce55ca0c@mail.gmail.com>
References: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>
	<191c3a031001131508i713ef7b8r48ba21c875e4e723@mail.gmail.com>
	<ade741d61001131520x1390ba23ud33dc4f3daba58f6@mail.gmail.com>
	<191c3a031001131530u6b105b35jcb09aa989a6ecce7@mail.gmail.com>
	<ade741d61001131846v5379fe23ua550e7293515e84d@mail.gmail.com>
	<191c3a031001131933g2c290efds5b8342d1ce55ca0c@mail.gmail.com>
Message-ID: <76CD79BB-64BD-4C2F-B97E-F80DC74EAB4E@gmail.com>

I forgot to add the sip end point device is not local it's another sip  
proxy on public IP who is suppose to trust FS invite without auth

Thanks

Sent from my iPhone

On Jan 13, 2010, at 10:33 PM, Anthony Minessale <anthony.minessale at gmail.com 
 > wrote:

> Not if you set inbound-bypass-media = true in the sip profile or set  
> late-negotiaation = true and set bypass_media channel var to true  
> from the dp
>
>> On Jan 13, 2010 8:51 PM, "Mouncif Benniane" <mouncifbb at gmail.com>  
>> wrote:
>>
>> One more question, will it participate in codec negotiation sent by  
>> the inbound voip provider?
>> Right now I have it setup this way:
>>
>> Incoming Inbound DID (voip provider --> freeswitch(media bypass  
>> mode) ->  endpoint sip device.
>>
>> Thanks again!
>> On Wed, Jan 13, 2010 at 6:30 PM, Anthony Minessale <anthony.minessale at gmail.com 
>> > wrote: > > its be...
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From sos at sokhapkin.dyndns.org  Wed Jan 13 20:08:03 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 23:08:03 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <6D061C37-43A6-48D1-A8E3-549FC3E8679E@freeswitch.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132229.58830.sos@sokhapkin.dyndns.org>
	<6D061C37-43A6-48D1-A8E3-549FC3E8679E@freeswitch.org>
Message-ID: <201001132308.03170.sos@sokhapkin.dyndns.org>

Critical issues are when SIP error come after 18X provisional response.

- if bypass_media is false then dialplan stops and leg a is explicitly hang up 
(switch_ivr_bridge.c, line 513).
- if bypass_media is true, then dialplan continue, but there is 10 seconds 
delay before next bridge application sends INVITE to gateway ( 
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024354.html ) 
I didn't track down yet why this happens looking at FS sources.

Why I didn't open a bug on jira? Because FS behaves according to the design 
and specs :-) But not according to real world requirements... 


On Wednesday 13 January 2010, Brian West wrote:
> Can you elaborate on these "Critical" issues you seem to be having?  Why
> aren't you opening a jira for them if they are that critical to your needs?
>
> /b
>
> On Jan 13, 2010, at 9:29 PM, Sergey Okhapkin wrote:
> > If you wish FreeSWITCH to be well adopted by the community, then FS
> > should follow the real world "rules" but not specs...
> >
> > Don't you agree that 99% of SIP servers are set up to interconnect
> > with "buggy" PSTN and should follow PSTN rules, but not SIP specs?
> >
> > Some background - I did run asterisk for years, but switched to FS
> > recently because of critical asterisk problems with SIP handling when
> > asterisk is not in the media path. I spend a lot of time porting the
> > billing system to FS. And I did it. What I got? Critical problems with FS
> > early media handling. Hopefully I can switch back to asterisk if FS
> > problems with early media will begin to draw customers away.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From sos at sokhapkin.dyndns.org  Wed Jan 13 20:11:06 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 23:11:06 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <464CD1B0-95D0-4D6D-B330-87B059FB76A6@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132229.58830.sos@sokhapkin.dyndns.org>
	<464CD1B0-95D0-4D6D-B330-87B059FB76A6@jerris.com>
Message-ID: <201001132311.06257.sos@sokhapkin.dyndns.org>

Just "10 seconds" could easily draw away customer's $$$. And seems like the 
process already began... :-(

On Wednesday 13 January 2010, Michael Jerris wrote:
> What exactly are your problems?   Is this just the "10 second" thread you
> posted today?
>
> Mike
>
> On Jan 13, 2010, at 10:29 PM, Sergey Okhapkin wrote:
> > If you wish FreeSWITCH to be well adopted by the community, then FS
> > should follow the real world "rules" but not specs...
> >
> > Don't you agree that 99% of SIP servers are set up to interconnect
> > with "buggy" PSTN and should follow PSTN rules, but not SIP specs?
> >
> > Some background - I did run asterisk for years, but switched to FS
> > recently because of critical asterisk problems with SIP handling when
> > asterisk is not in the media path. I spend a lot of time porting the
> > billing system to FS. And I did it. What I got? Critical problems with FS
> > early media handling. Hopefully I can switch back to asterisk if FS
> > problems with early media will begin to draw customers away.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Wed Jan 13 20:13:30 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 22:13:30 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001132308.03170.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132229.58830.sos@sokhapkin.dyndns.org>
	<6D061C37-43A6-48D1-A8E3-549FC3E8679E@freeswitch.org>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
Message-ID: <5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>

This is exactly why you open a jira.  But we have zero control over the far end sending a 18X then an error.  That is NOT something we can fix. As for the others I suspect they have been fixed... what Rev are you running?

/b

On Jan 13, 2010, at 10:08 PM, Sergey Okhapkin wrote:

> Critical issues are when SIP error come after 18X provisional response.
> 
> - if bypass_media is false then dialplan stops and leg a is explicitly hang up 
> (switch_ivr_bridge.c, line 513).
> - if bypass_media is true, then dialplan continue, but there is 10 seconds 
> delay before next bridge application sends INVITE to gateway ( 
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024354.html ) 
> I didn't track down yet why this happens looking at FS sources.
> 
> Why I didn't open a bug on jira? Because FS behaves according to the design 
> and specs :-) But not according to real world requirements... 

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From sos at sokhapkin.dyndns.org  Wed Jan 13 20:26:41 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 23:26:41 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
Message-ID: <201001132326.41354.sos@sokhapkin.dyndns.org>

I run FreeSWITCH Version 1.0.5pre10 (16012M), sorry but I don't feel 
comfortable running SVN trunk on production servers... Perhaps because of my 
experience with asterisk. Neither version after 1.4.18 did work OK to me.

On Wednesday 13 January 2010, Brian West wrote:
> This is exactly why you open a jira.  But we have zero control over the far
> end sending a 18X then an error.  That is NOT something we can fix. As for
> the others I suspect they have been fixed... what Rev are you running?
>
> /b
>
> On Jan 13, 2010, at 10:08 PM, Sergey Okhapkin wrote:
> > Critical issues are when SIP error come after 18X provisional response.
> >
> > - if bypass_media is false then dialplan stops and leg a is explicitly
> > hang up (switch_ivr_bridge.c, line 513).
> > - if bypass_media is true, then dialplan continue, but there is 10
> > seconds delay before next bridge application sends INVITE to gateway (
> > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
> >4.html ) I didn't track down yet why this happens looking at FS sources.
> >
> > Why I didn't open a bug on jira? Because FS behaves according to the
> > design and specs :-) But not according to real world requirements...




From dave at 3c.co.uk  Wed Jan 13 20:28:18 2010
From: dave at 3c.co.uk (David Knell)
Date: Thu, 14 Jan 2010 04:28:18 +0000
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <26079A18-5968-4E1E-A276-9DE889E83FA4@freeswitch.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
	<C3D4ADB0-7A80-45A9-91E1-FFABB987851B@freeswitch.org>
	<201001132205.43306.sos@sokhapkin.dyndns.org>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
	<1263439925.11216.36.camel@local.freepabx.com>
	<26079A18-5968-4E1E-A276-9DE889E83FA4@freeswitch.org>
Message-ID: <1263443298.11216.45.camel@local.freepabx.com>

Well, surely if it would work, then it's feasible and there's no reason
why it shouldn't happen.  Whether or not a solution will achieve the
desired end result in an optimal fashion is - for most people -
subordinate to getting their problem solved.

Don't forget the two rules of optimisation:
1.  Don't.
2.  (for experts only) Don't yet.

--Dave

> That would work but its sub optimal.  What he wants to accomplish is not do able and not going to ever happen at this rate.
> 
> /b
> 
> On Jan 13, 2010, at 9:32 PM, David Knell wrote:
> 
> > What would happen if several outdials were made from a conference,
> > rather than using a forked dial?
> > 
> > --Dave
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From mike at jerris.com  Wed Jan 13 20:47:53 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 23:47:53 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001132308.03170.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132229.58830.sos@sokhapkin.dyndns.org>
	<6D061C37-43A6-48D1-A8E3-549FC3E8679E@freeswitch.org>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
Message-ID: <632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>


On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:

> Critical issues are when SIP error come after 18X provisional response.
> 
> - if bypass_media is false then dialplan stops and leg a is explicitly hang up 
> (switch_ivr_bridge.c, line 513).

 behavior can be modified with continue_on_fail and hangup_after_bridge channel vars, perhaps ignore_early_media as well

> - if bypass_media is true, then dialplan continue, but there is 10 seconds 
> delay before next bridge application sends INVITE to gateway ( 
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024354.html ) 
> I didn't track down yet why this happens looking at FS sources.

see response on that thread

> 
> Why I didn't open a bug on jira? Because FS behaves according to the design 
> and specs :-) But not according to real world requirements... 


Really, people are trying to help you and your going to be snarky in response?

> 
> 
> On Wednesday 13 January 2010, Brian West wrote:
>> Can you elaborate on these "Critical" issues you seem to be having?  Why
>> aren't you opening a jira for them if they are that critical to your needs?


Mike



From mike at jerris.com  Wed Jan 13 20:48:34 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 23:48:34 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001132326.41354.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
	<201001132326.41354.sos@sokhapkin.dyndns.org>
Message-ID: <52E8BE9E-B6D6-411C-9080-697AF24017B4@jerris.com>

You already are running on trunk.

Mike

On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote:

> I run FreeSWITCH Version 1.0.5pre10 (16012M), sorry but I don't feel 
> comfortable running SVN trunk on production servers... Perhaps because of my 
> experience with asterisk. Neither version after 1.4.18 did work OK to me.
> 
> On Wednesday 13 January 2010, Brian West wrote:
>> This is exactly why you open a jira.  But we have zero control over the far
>> end sending a 18X then an error.  That is NOT something we can fix. As for
>> the others I suspect they have been fixed... what Rev are you running?
>> 
>> /b



From anthony.minessale at gmail.com  Wed Jan 13 21:15:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Jan 2010 23:15:38 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001132114s18ed2baw2ff802dbd615fe01@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
	<201001132326.41354.sos@sokhapkin.dyndns.org>
	<52E8BE9E-B6D6-411C-9080-697AF24017B4@jerris.com>
	<191c3a031001132114s18ed2baw2ff802dbd615fe01@mail.gmail.com>
Message-ID: <191c3a031001132115u2d88bab7p7cbb27ab1eaad466@mail.gmail.com>

Perhaps best not to help him anymore without an apology for the snap
judgement and comparison to asterisk clearly designed to push our buttons.
We'll be here when you realize we were trying to help you but I can't
promise we will still have paitence......

On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike at jerris.com> wrote:

You already are running on trunk.

Mike

On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote: > I run FreeSWITCH
Version 1.0.5pre10 (16012M...

_______________________________________________ FreeSWITCH-users mailing
list FreeSWITCH-users at lists...
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From lakindia89 at gmail.com  Wed Jan 13 23:57:54 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 14 Jan 2010 13:27:54 +0530
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>
References: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
	<191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>
Message-ID: <7d79b3931001132357i36bbb482jdf2bbdd3aea2a583@mail.gmail.com>

I taught the signal handler will be inherited by the child process. It also
does like that.
After making a call, If I press ctrl + c, the above program printed
PARENT PID: Signal SIGINT is generated
CHILD PID: Signal SIGINT is generated.

So I think the sigal handlers will be inherited to the child.
Anyway I'll also try registering signal handlers in child also, and then
I'll come back with that result.

Thanks....
On Wed, Jan 13, 2010 at 9:48 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you would have to register signals in your child process too
>
>   On Wed, Jan 13, 2010 at 3:13 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>>  Hi all,
>>
>> I've done a sample program (In perl ESL) , which play a file to the caller
>> and then it will call recvEvent() and print the event name. I've handled
>> signals also.
>>
>> When I send SIGINT to my program (Perl), the signal handler is called and
>> I can see the print output. But in the same time,  I received
>> SERVER_DISCONNECTED from freeswitch as event.
>>
>> I don't know why I received SERVER_DISCONNECTED from freeswitch. Is it
>> because, the recvEvent() from perl internally  calls the recvevent function
>> in the Esl.c and when it waits to receive the information from socket, the
>> signal occurred???
>>
>> Please clarify me!!
>>
>> Here is my program
>> require ESL;
>> use IO::Socket::INET;
>> use Data::Dumper;
>>
>> my $ip = "192.168.1.222";
>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
>> Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>> die "Could not create socket: $!\n" unless $sock;
>> &register_Signals();
>>
>> for(;;) {
>>     my $new_sock = $sock->accept();
>>     next if (not defined ($new_sock));
>>     my $pid = fork();
>>     if ($pid) {
>>         close($new_sock);
>>         next;
>>     }
>>     print "CHILD PID: $$\n";
>>     my $host = $new_sock->sockhost();
>>     my $fd = fileno($new_sock);
>>
>>     my $con = new ESL::ESLconnection($fd);
>>     my $info = $con->getInfo();
>>
>>     my $uuid = $info->getHeader("unique-id");
>>
>>     printf "Connected call %s, from %s\n", $uuid,
>> $info->getHeader("caller-caller-id-number");
>>     my $r=$con->execute("answer");
>>     print Dumper $r;
>>     $con->events("plain","all");
>>     my
>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>
>>     while($con->connected()) {
>>         my $e = $con->recvEvent();
>>
>>         if ($e) {
>>             my $name = $e->getHeader("event-name");
>>             print "EVENT [$name]\n";
>>             if ($name eq "DTMF") {
>>                 my $digit = $e->getHeader("dtmf-digit");
>>                 print "$digit\n";
>>             }
>>         }
>>     }
>>     close($new_sock);
>> }
>> sub register_Signals() {
>>     foreach ( keys %SIG ) {
>>         $SIG{$_} = 'sig_Handler';
>>     }
>> }
>>
>> sub sig_Handler() {
>>     my $handle=$_[0];
>>     if($handle eq "INT") {
>>         print "$$: SIGNAL SIG$handle is generated\n";
>>     }
>> }
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Thu Jan 14 00:02:32 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 14 Jan 2010 13:32:32 +0530
Subject: [Freeswitch-users] Sangoma PRI installation for FreeSWITCH
In-Reply-To: <B4A27408A33F4E74A4DBEA492B906C20@greyhawk.tonecommander.com>
References: <369c72d81001122021o6885913cl618965791aec4621@mail.gmail.com>
	<B4A27408A33F4E74A4DBEA492B906C20@greyhawk.tonecommander.com>
Message-ID: <7d79b3931001140002w638d2c0bn2f1d9b6f9b51bd1d@mail.gmail.com>

Thanks Jerry, for making the wancfg_fs configuration available.

On Wed, Jan 13, 2010 at 9:45 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  Here are my instructions to install the wanpipe driver and I have not had
> your problem:
>
>
> 1)       Download the following Wanpipe driver tarball
> wanpipe-<version>.tgz
>
> 2)       Store the wanpipe-<version>.tgz file in the /opt folder.
>
> 3)       tar xvfz wanpipe-<version>.tgz
>
> 4)       cd wanpipe-<version>
>
> 5)       make openzap
>
> 6)       make install
>
> 7)       make install_pri
>
> 8)       wanrouter hwprobe     // confirms successful wanpipe installation
>
> 9)       /usr/sbin/wancfg_fs   // starts wanpipe configuration utility
>
> 10)   1=NO            // Change Freeswitch Configuration Directory
>
> 11)   1=T1            // Select Media Type
>
> 12)   1=YES (keep)    // Configure Port 1 T1, B8ZS, ESF
>
> 13)   1=NORMAL        // Select Clock
>
> 14)   1=NATIONAL      // Select Switchtype
>
> 15)   1=CPE           // Select Signaling Type
>
> 16)   1               // Input Group #
>
> 17)   1=YES           // Enable H/W DTMF Detect
>
> 18)   1=YES           // Enable FAX Detect
>
> 19)   1=YES=CONTINUE  // Configuration Complete
>
> 20)   1=YES           // Save Configuration
>
> 21)   1=YES           // Wanrouter start on reboot
>
> 22)   1=YES           // smg_ctrl start/stop on wanrouter start
>
> 23)   Done.
>
> Jerry
>
>  ------------------------------
> *From:* Magesh R [mailto:magesh.freeswitch at gmail.com]
> *Sent:* Tuesday, January 12, 2010 8:21 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Sangoma PRI installation for FreeSWITCH
>
> Dear All,
>          I have installed Sangoma PRI card in machine with following steps,
>                  * wget
> ftp://ftp.sangoma.com/linux/custom/3.5/wanpipe-3.5.8.7.tgz
>                  * tar -xvfz wanpipe-3.5.8.7.tgz
>                  * cd wanpipe-3.5.8.7
>                  * make openzap
>                  * make install
>                  * make install_pri
>
>            I have executed "wanrouter hwprobe" command it prints the
> following details,
>
>            1 . AFT-A102-SH : SLOT=4 : BUS=3 : IRQ=11 : CPU=A : PORT=1 :
> HWEC=0 : V=36
>            2 . AFT-A102-SH : SLOT=4 : BUS=3 : IRQ=11 : CPU=A : PORT=2 :
> HWEC=0 : V=36
>
>         Card Cnt: A101-2=1
>
>        Next I have executed wancfg_fs script to configure the sangoma for
> freeswitch.
> It creates the following configuration files
>                      * wanpipe1.conf
>                      * wanpipe2.conf
>                      * smg_prid.conf
>                      * openzap.conf
>                      * openzap.conf.xml
>        I have attached those files.
>        I have started the wanrouter and printed the wanrouter status,
>               Wanrouter Status:
>
> Device name | Protocol | Station | Status        |
> wanpipe1    | AFT TE1  | N/A     | Connected     |
> wanpipe2    | AFT TE1  | N/A     | Disconnected  |
>
>
>       Next I have started the smg_ctrl, but it failed to start. It prints
> the following things,
> smg_ctrl start
>
> Starting processes...
> Loading SCTP...OK
> Starting sangoma_prid...OK
> sangoma_prid failed to start
> check /var/log/sangoma_mgd.log for errors
>
> Stopping running processes...
> safe_sangoma not running...
> sangoma_prid is stopped
> Removing PID files...done
>
> I have checked /var/log/sangoma_mgd.log file. But nothing was there.
>
> Could any please tell me where I made mistake?
>
> Thanks,
> Mag
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kawarod at laposte.net  Thu Jan 14 00:17:50 2010
From: kawarod at laposte.net (rod)
Date: Thu, 14 Jan 2010 12:17:50 +0400
Subject: [Freeswitch-users] Eavesdrop in LUA
Message-ID: <4B4ED32E.30706@laposte.net>

Hi all,

I'm trying to do this in LUA:
     A call B

and I'd like to setup a new call to C with eavesdrop of A conversation 
with B.

I have no idea how to do this if someone can help.
I switched to LUA cause I see no way to achieve this with dialplan 
(snippets are welcome).

regards,
rod


From devel at thom.fr.eu.org  Thu Jan 14 00:23:09 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Thu, 14 Jan 2010 09:23:09 +0100
Subject: [Freeswitch-users]
 =?utf-8?q?playback_and_play=5Fand=5Fget=5Fdigi?=
 =?utf-8?q?ts_strange_misunderstanding?=
Message-ID: <6d698be09281bb173cf0de33c1c2e44b@thom.fr.eu.org>

 

So I could (kind of) solve this by myself. There is in vars.xml the
variable ${sound_prefix}. I did set it properly to my french sound path and
then it worked. 

However, for the sake of discussion, I did try this with
1.0.5pre8 and the result was different : 

with the extension 

FS was
trying to play the file
${FREESWITCH_PATH}/sounds/en/us/callie/misc/ringing_disabled.wav 

with the
extension 

FS was trying to load
${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/ringing_disabled.wav 

Now with
the latest snapshot, with either one of the 2 mentionned extensions, FS
tries to play the file in
${sound_prefix}/filepath/${codec_bit_rate}/filename whereas 1.0.5pre8 did
not add the codec_bit_rate in the path but took care of the
default_language variable. 

Fran?ois 

On Wed, 13 Jan 2010 16:32:16 +0100,
Fran?ois Legal wrote:  

Hello, 

trying to make so dialplan extensions
that use the playback and play_and_get_digits applications, but I'm having
trouble with the file name specification. 

The files I want to play are in
the french language (fr/fr/julie as configured in lang/fr/fr.xml) 

My
extension is as follows : 

The channel is using a bit rate of 8000 Hz, so
by the set default_language=fr I would expect freeswitch to playback the
file at
${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/8000/ringing_disabled.wav
whereas it tries to playback the file at
${FREESWITCH_PATH}/sounds/en/us/callie/misc/8000/ringing_disabled.wav 

I
have the same with play_and_get_digits application. 

What am I doing wrong
? 

Fran?ois  

 
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From kond at nstel.ru  Thu Jan 14 00:26:40 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Thu, 14 Jan 2010 11:26:40 +0300
Subject: [Freeswitch-users] sip trunk question: why call throughexternal
	profile is challenged?
In-Reply-To: <98955F81-E9C1-4926-A648-AD49FB9D38A4@jerris.com>
Message-ID: <20100114082640.2C5D311F32@mail.nstel.ru>

Mike, thanks for the reply.

 

Mmm. looks like I need more detailed instructions where to dig.

Is there a way to turn off "challenging" completely?

I thought that <param name="auth-calls" value="false"/> should do it, but
alas.

By the way should this parameter be visible in either "sofia status profile
external" or "sofia status gateway sipx4.lab.nstel.ru" ? I don't see it.

 

I attached traces of failed and successful calls.

 

Thanks and regards,

Nikolay. 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Wednesday, January 13, 2010 8:30 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] sip trunk question: why call throughexternal
profile is challenged?

 

Look at how sipx sets up the users when they build the extensions and such
for conferences, there was something special here, but I can't recall what.

 

Mike

 

On Jan 13, 2010, at 9:10 AM, Nikolay Kondratyev wrote:





Hi all!

 

I'm brand new to FreeSwitch, but have some experience with SipX.

Our company is evaluating FS.

For test purposes I set up FS on a virtual machine (vmware esxi). I use
CentOS.

The FS version I use is 1.0.5-20100110-0400.

 

I have a question regarding sip trunk between FS and SipX.

I created the following GW in external profile: 

[freeswitch at freeswitch external]$ cat sipx-lab.xml | grep -v '<!--'

<include>

  <gateway name="sipx4.lab.nstel.ru">

  <param name="username" value="zxcv"/>

  <param name="password" value="2007"/>

  <param name="register" value="false"/>

  </gateway>

</include>

 

External.xml file is not modified after installation.

 

I see this gateway via fs_cli:

freeswitch at internal> sofia status gateway sipx4.lab.nstel.ru

============================================================================
=====================

Name            sipx4.lab.nstel.ru

Profile         external

Scheme          Digest

Realm           sipx4.lab.nstel.ru

Username        zxcv

Password        yes

>From            <sip:zxcv at sipx4.lab.nstel.ru;transport=udp>

Contact
<sip:gw+sipx4.lab.nstel.ru at 172.23.22.49:5080;transport=udp;gw=sipx4.lab.nste
l.ru>

Exten           zxcv

To              sip:zxcv at sipx4.lab.nstel.ru

Proxy           sip:sipx4.lab.nstel.ru

Context         public

Expires         3600

Freq            3600

Ping            0

PingFreq        0

PingState       0/0/0

State           NOREG

Status          UP

CallsIN         0

CallsOUT        0

============================================================================
=====================

 

I created new FS extension 2853. I registers (xlite) and I can call it.

 

Now I want to call FS user from sipx.

 

>From the sipx side one can configure link to FS differently. There are two
options:

1. Call FS directly trough sipxproxy (the core part of sipx works as sip
proxy, not as b2bua)

2. Call trough embedded b2bua, named sipxbridge.

 

When a call is going trough sipxbridge, it is successfully landed at FS
extension 2853.

But when a call is going from the sipxproxy, it is challenged with Status
407 "proxy authentication required', and then call fails.

(I'm not sure if sipx should handle this challenge, but this is separate
question for the sipx forum).

At the default log level I see nothing in the freeswitch.log.

 

So the question is why one call through external profile is being challenged
while the other is not?

I suspect that the reason is in the difference in the two Invite messages:

 

Here is challenged Invite:

INVITE sip:2853 at fs.lab.nstel.ru:5080;transport=udp SIP/2.0

Record-Route:
<sip:172.23.12.104:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EOTRiZjI3MjgtYWMxN
zBjZTktMTNjNC0yZjZkZWMtNjNkNmUwN2UtMmY2ZGVj.900_ntap%2Aid%7EMjIyNjAtMQ%60%60
%213ba6d85f6946c4c6001bee1d3b54474f>

From:
"testphone3857"<sip:3857 at lab.nstel.ru>;tag=94bf2728-ac170ce9-13c4-2f6dec-63d
6e07e-2f6dec

To: <sip:2853 at lab.nstel.ru>

Call-Id: 94bed2a0-ac170ce9-13c4-2f6dec-68ffdb81-2f6dec at lab.nstel.ru

Cseq: 2 INVITE

Via: SIP/2.0/UDP
172.23.12.104;branch=z9hG4bK-sipXecs-000eaf48a36fe4029c7cde004a6f44424847

Via: SIP/2.0/TCP
172.23.12.104;branch=z9hG4bK-sipXecs-000bf3008cfa3f3e70470789c75232ba9499~ac
7fd729330fd563f83cacd941311e75;id=22260-1

Via: SIP/2.0/UDP 172.23.12.233:5060;branch=z9hG4bK-2f6dec-b9456412-7ad07784

Max-Forwards: 18

Supported: replaces

User-Agent: LG-Nortel LIP 6812 v1.2.38sp SN/00405A18B634

Contact: <sip:3857 at 172.23.12.233:5060;x-sipX-nonat>

Proxy-Authorization: Digest
username="3857",realm="lab.nstel.ru",nonce="e2152722611af1bbe59f3a8eb31b8eb8
4b4db2ca",uri="sip:2853 at lab.nstel.ru",response="a09c34fbd897783c9b9af50bd044
ccaa",algorithm=MD5

Content-Type: application/sdp

Content-Length: 301

Date: Wed, 13 Jan 2010 11:47:22 GMT

Expires: 60

X-Sipx-Handled: X172.23.12.104-81.211.30.104

 

v=0

o=LGEIPP 16246 16247 IN IP4 172.23.12.233

s=SIP Call

c=IN IP4 172.23.12.233

t=0 0

m=audio 23008 RTP/AVP 0 8 18 4 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:18 annexb=no

a=fmtp:101 0-11

a=sendrecv

 

 

And here is successful Invite:

INVITE sip:2853 at fs.lab.nstel.ru;user=phone SIP/2.0

Call-ID: 94bec3b8-ac170ce9-13c4-2f6d47-2fee09d0-2f6d47 at lab.nstel.ru.0

CSeq: 1 INVITE

From: "testphone3857" <sip:3857 at 172.23.12.104>;tag=816640414244159961

To: <sip:2853 at fs.lab.nstel.ru;user=phone>

Via: SIP/2.0/UDP
172.23.12.104:5080;branch=z9hG4bK885dde2c8680f5315845cd3350b8b605373534

Max-Forwards: 70

User-Agent: sipXecs/4.0.2 sipXecs/sipxbridge (Linux)

P-Asserted-Identity: <sip:1004 at 172.23.12.104>

Contact: <sip:1004 at 172.23.12.104:5080;transport=udp>

Route: <sip:172.23.22.49:5080;transport=udp;lr>

Session-Expires: 1800;refresher=uac

Allow: INVITE,BYE,ACK,CANCEL,OPTIONS

Content-Type: application/sdp

Content-Length: 315

 

v=0

o=sipxbridge 6640787141824452741 1 IN IP4 172.23.12.104

s=SIP Call

c=IN IP4 172.23.12.104

t=0 0

m=audio 30248 RTP/AVP 0 8 18 4 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:18 annexb=no

a=fmtp:101 0-11

a=sendrecv

 

Can somebody please if it is a FS configuration problem or a software
problem or is it a problem on the sipx side?

 

Thanks in advance,

Nikolay.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From john_re at fastmail.us  Thu Jan 14 01:06:52 2010
From: john_re at fastmail.us (giovanni_re)
Date: Thu, 14 Jan 2010 01:06:52 -0800
Subject: [Freeswitch-users] Attend CampKDE Jan 15-18 via Voice over Internet
	(VOIP), BerkeleyTIP
Message-ID: <1263460012.26773.1354547409@webmail.messagingengine.com>

This weekend is CampKDE in San Diego, California, USA.  For those who
would like to participate, but can't physically attend, the BerkeleyTIP
group will (unofficially, as a public service) provide assistance to
everyone who would like to communicate via voice about &/or with the
conference, & other KDE community members, globally.  [VOIP
communication is typically free (no money $ cost) to people with
internet access.]

Join #berkeleytip on irc.freenode.net, & we will do our best (depending
on availability of BTIP people) to help people get on VOIP in order to
talk about KDE, or whatever.

You are also invited to join the BerkeleyTIP mailing list, & ask any
help or other questions there.
http://groups.google.com/group/BerkTIPGlobal

Here's the announcement there:
CampKDE Jan 15-18 SanDiego, VOIP & Ride Share; BerkeleyTIP  
http://groups.google.com/group/BerkTIPGlobal/t/53cbf00b987b66e0

I'll be announcing this on some KDE related mailing lists.

When (if) I get time, I'll locate the CampKDE organizers, & email them
telling them BTIP is helping KDE community members with this
unofficially, & asking them if they'd like to try to do something
officially in arranging global VOIP for the KDE community interested in
CampKDE.

Here is the VOIP HowTo page for BerkeleyTIP:
http://sites.google.com/site/berkeleytip/remote-attendance

There are 2 ways to join the VOIP conference: 1) Computer VOIP globally
using a sw client & headset & mic, & 2) Dial in with land or cell phone
to a phone # in Washington state, USA (free, no $ money cost, if you
have free long distance phone service).

BerkeleyTIP:
http://sites.google.com/site/berkeleytip/

=====
So, join on the #berkeleytip freenode irc channel, or the mailing list,
& we'll do our best to help you get on the global conference to discuss
KDE via voice communication. :)


From devel at thom.fr.eu.org  Thu Jan 14 01:08:28 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Thu, 14 Jan 2010 10:08:28 +0100
Subject: [Freeswitch-users] Issue with openzap and DTMF
Message-ID: <2bf9ca9a45058c72a1f8d13070cb582e@thom.fr.eu.org>



Hello, 

yesterday, while trying some new extensions with
play_and_get_dtmf, I noticed a strange behaviour of freeswitch. 

The
configuration is as follows : 

An FXS port is bridged to an FXO (say FXO1)
port for calling (through my carrier) another FXO (say FXO2) port. The call
incoming on FXO2 reaches some extension which does answer then
play_and_get_dtmf (or read). So I dial the digits I want, and for each
dialed digit, I get it twice in the resultant variable. 

Now if I reach
the extension by calling in from a cell phone, I get exactly the digits I
dialed. 

This seems to me as a problem in openzap. 

I just wanted to
report. 

Fran?ois
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From yehavi.bourvine at gmail.com  Thu Jan 14 01:14:58 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 14 Jan 2010 11:14:58 +0200
Subject: [Freeswitch-users] How to build FreeSwitch with non-default SSL
	library?
In-Reply-To: <d361ca7b1001131132o1043cce4p31af0f95f804bd7b@mail.gmail.com>
References: <d361ca7b1001131132o1043cce4p31af0f95f804bd7b@mail.gmail.com>
Message-ID: <d361ca7b1001140114s53c25d28h2cbf496440833941@mail.gmail.com>

Answwring myself:  "make distclean", set LDFLAGS=-L/path-to-ssl-lib and then
configure and make.

                       __Yehavi:

2010/1/13 Yehavi Bourvine <yehavi.bourvine at gmail.com>

>  Hello,
>
>   I suspect that the Openssl library used on fedora is a modified one
> and has some interoperability problems with phones. I would like to build
> freeswitch with a library from openssl.org which is installed
> under /usr/local. Is there an easy way to tell  "configure" script to use
> that version instead?
>
>                                  Thanks, __Yehavi:
>
>
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From jingwei.yang at gmail.com  Thu Jan 14 01:24:04 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 14 Jan 2010 17:24:04 +0800
Subject: [Freeswitch-users]  Not able to capture the custom event
Message-ID: <13529f9d1001140124t42e9dfe5u3e70a6976aa032a8@mail.gmail.com>

Hi Guys,

I'm testing to see whether a custom event can be triggered and captured.
Here's the extension:

    <extension name="test_ivr">
      <condition field="destination_number" expression="^777$">
        <action application="event" data="Event-Name=*CUSTOM*
,Event-Subclass=myevent::ACDnotify"/>
        <action application="transfer" data="9999 XML default" />
      </condition>
    </extension>

If I telnet and use "events plain all", I can get the event. However, if I
use "events plain CUSTOM" or "event plain CUSTOM", it doesn't get captured.
Is there anything wrong with the command?

Regards,
-Jingwei
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From david.villasmil.work at gmail.com  Thu Jan 14 02:26:45 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 14 Jan 2010 11:26:45 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001132115u2d88bab7p7cbb27ab1eaad466@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
	<201001132326.41354.sos@sokhapkin.dyndns.org>
	<52E8BE9E-B6D6-411C-9080-697AF24017B4@jerris.com>
	<191c3a031001132114s18ed2baw2ff802dbd615fe01@mail.gmail.com>
	<191c3a031001132115u2d88bab7p7cbb27ab1eaad466@mail.gmail.com>
Message-ID: <9853f4ff1001140226y2b035f37rfcafb03190435ec1@mail.gmail.com>

Hello again,

Using ingore early media only ignores ALL media, that's not what I need. At
least in europe i've seen it many times, MNOs provide a service with which
you can have a song played to the caller while the call is connecting to
your cell phone. This is basically what I'm trying to achieve.

The content provider is in possesions of the media and it require us to fork
the call and send a SIP INVITE to the on one leg and the call to the
destination number on another leg. they will only provide a progress, no
answer.

I CAN do it locally by playing the file as a custom ringback but that's not
the standard in terms of commercial use of the content.

Is there any way to modify the behaviour of fs when i receives media? Let's
say i.e. it doesn't drop the other leg, but provides the first early media
it receives and just wait for some channel to answer? I will not have both
media but it would work.

thanks

David

On Thu, Jan 14, 2010 at 6:15 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Perhaps best not to help him anymore without an apology for the snap
> judgement and comparison to asterisk clearly designed to push our buttons.
> We'll be here when you realize we were trying to help you but I can't
> promise we will still have paitence......
>
> On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike at jerris.com> wrote:
>
> You already are running on trunk.
>
> Mike
>
> On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote: > I run FreeSWITCH
> Version 1.0.5pre10 (16012M...
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Claudio.Cavalera at italtel.it  Thu Jan 14 03:23:05 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Thu, 14 Jan 2010 12:23:05 +0100
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <5d2828f1001131130s6ef166c0wc98f70023420c0fd@mail.gmail.com>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F362E8@BESONE.corp.dom>

Hi Mike,
in fact i've completed that page with the list of available  session
functions.
 
I've not removed "session:originate" yet, but it would be better if
someone could provide an example in order to write an example in the
wiki.
I've added this valuable example also with the help of rupa
http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.Session
 
I would like to write something also about api_on_answer to use an api
instead of a dialplan application.
 
BRs,
Claudio


________________________________

	From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike
van Lammeren
	Sent: Wednesday, January 13, 2010 8:30 PM
	To: freeswitch-users at lists.freeswitch.org
	Subject: Re: [Freeswitch-users] playing with sessions in lua
	
	
	Hello! 

	Before you remove "session:originate" from the wiki, you should
take a look at this:
	
http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocument
ed_Session_Functions.3F

	
<http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocumen
ted_Session_Functions.3F> There is, in fact, a function called
"originate".

	Mike van Lammeren


	On Wed, Jan 13, 2010 at 5:22 AM, Cavalera Claudio Luigi
<Claudio.Cavalera at italtel.it> wrote:
	

		Thanks a lot Anthony,
		some comments inline (and please forgive me for my
broken email client).
		 
		
		>>  example1: Consider this simple lua script in which i
create two sessions:
		
		 >>  api = freeswitch.API();
		 >>  api:execute("originate",
"sofia/internal/1001%192.168.1.1 &park");
		 >>  api:execute("originate",
"sofia/internal/1002%192.168.1.1 &park");
		
		 >  capture the output from api:execute the uuid is in
there
		
		
		Thx a lot,
		this was one piece i was missing although it's already
on the wiki here:
	
http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls 
		


		 > because lua calls it freeswitch.bridge
		
		 > session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
		 > session2 =
freeswitch.Session("sofia/internal/1002%192.168.1.1", session1);
		 > freeswitch.bridge(session1, session2); 
		 
		good to now, there isn't any example of
freeswitch.bridge in the wiki and i would like to add one.
		Where I could find the full api of
		freeswitch.Session( ) ?
		because I've seen this working also without "session1"
in the second line:
		
		session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
		
		session2 =
freeswitch.Session("sofia/internal/1002%192.168.1.1");
		freeswitch.bridge(session1, session2); 
		
		 also is there any difference between  freeswitch.bridge
and freeswitch.execute(uuid_bridge ...) ?
		 
		 >> example3: yet another possibility
		 >> local session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
		 >> session1:originate(session2,
"sofia/internal/1002%192.168.1.1", 1000);
		 >> but it does not work either.
		
		 > The above is gibberish try:
		 > local session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
		 > session1:execute("bridge",
"sofia/internal/1002%192.168.1.1"); 
		 
		Okay i will report also this bridge example on the wiki
which was missing.
		But does session:originate make sense in some cases or
not? Otherwise i'm going to remove this line on the wiki
	
http://wiki.freeswitch.org/wiki/Mod_lua#session:originate
<http://wiki.freeswitch.org/wiki/Mod_lua#session:originate> 
		 
		Thanks,
		Claudio

		 

		Internet Email Confidentiality Footer

	
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		La presente comunicazione, con le informazioni in essa
contenute e ogni documento o file allegato, e' rivolta unicamente alla/e
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riceverla. Se non siete i destinatari/autorizzati siete avvisati che
qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul
contenuto di tali informazioni e' vietata e potrebbe essere contro la
legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione
dei dati personali). Se avete ricevuto questa comunicazione per errore,
vi preghiamo di darne immediata notizia al mittente e di distruggere il
messaggio originale e ogni file allegato senza farne copia alcuna o
riprodurne in alcun modo il contenuto. 
		
		This e-mail and its attachments are intended for the
addressee(s) only and are confidential and/or may contain legally
privileged information. If you have received this message by mistake or
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From sos at sokhapkin.dyndns.org  Thu Jan 14 04:19:12 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Jan 2010 07:19:12 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
Message-ID: <201001140719.12363.sos@sokhapkin.dyndns.org>

Case 1 (bypass_media is off) is already on jira, 
http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test 
installation with latest trunk to reproduce case 2, when bypass_media=true 
and 10 seconds delay happens when 18X and then error are received on leg b.

On Wednesday 13 January 2010, Michael Jerris wrote:
> On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> > Critical issues are when SIP error come after 18X provisional response.
> >
> > - if bypass_media is false then dialplan stops and leg a is explicitly
> > hang up (switch_ivr_bridge.c, line 513).
>
>  behavior can be modified with continue_on_fail and hangup_after_bridge
> channel vars, perhaps ignore_early_media as well
>
> > - if bypass_media is true, then dialplan continue, but there is 10
> > seconds delay before next bridge application sends INVITE to gateway (
> > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
> >4.html ) I didn't track down yet why this happens looking at FS sources.
>
> see response on that thread
>
> > Why I didn't open a bug on jira? Because FS behaves according to the
> > design and specs :-) But not according to real world requirements...
>
> Really, people are trying to help you and your going to be snarky in
> response?
>
> > On Wednesday 13 January 2010, Brian West wrote:
> >> Can you elaborate on these "Critical" issues you seem to be having?  Why
> >> aren't you opening a jira for them if they are that critical to your
> >> needs?
>
> Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Thu Jan 14 06:50:46 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Jan 2010 08:50:46 -0600
Subject: [Freeswitch-users] ESL Mono/Managed
Message-ID: <1E6792AC-17F2-4DE9-9798-431350A5E77D@freeswitch.org>

I need someone that knows C# to write up some examples using ESL in C#

cd libs/esl
make managedmod
cd managed

Look at the perl example single_command.pl
require ESL;

my $command = shift;
my $args = join(" ", @ARGV);

my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
my $e = $con->api($command, $args);
print $e->getBody();


Need to write the same thing in C# and commit it as an example.  Also other examples exist in the perl directory those should be translated also if possible.

If you're interested please find me on IRC (bkw_)

Thanks,
Brina



From costa.zikalala at gmail.com  Thu Jan 14 06:54:57 2010
From: costa.zikalala at gmail.com (Costa Zikalala)
Date: Thu, 14 Jan 2010 16:54:57 +0200
Subject: [Freeswitch-users] Mod_Fax
Message-ID: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>

Hi All

Has anyone worked on a email2fax script for mod_fax?
If not how much would it cost for some genius here to quickly whip-up one?

Ideally both email2fax and fax2email should come standard with mod_fax or is
this not possible?

Thanks
Costa
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From brian at freeswitch.org  Thu Jan 14 07:09:40 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Jan 2010 09:09:40 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001140719.12363.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
Message-ID: <11677234-5A33-42BE-A440-6BBC859CE91E@freeswitch.org>

The first question is why are you using Proxy Media?  Secondly I'm almost positive that issue was resolved.  Thirdly 1.0.5pre10 was just an SVN snapshot!  We have since fixed a multitude of tiny issues here and there since that time one being a bug related to proxy media mode.

Thanks,
/b

On Jan 14, 2010, at 6:19 AM, Sergey Okhapkin wrote:

> Case 1 (bypass_media is off) is already on jira, 
> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test 
> installation with latest trunk to reproduce case 2, when bypass_media=true 
> and 10 seconds delay happens when 18X and then error are received on leg b.

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From kawarod at laposte.net  Thu Jan 14 07:09:59 2010
From: kawarod at laposte.net (rod)
Date: Thu, 14 Jan 2010 19:09:59 +0400
Subject: [Freeswitch-users] Eavesdrop in LUA
In-Reply-To: <4B4ED32E.30706@laposte.net>
References: <4B4ED32E.30706@laposte.net>
Message-ID: <4B4F33C7.6020403@laposte.net>

Hi all,

I have an incomplete solution for those interested.

I did it like this in dialplan:
        <action application="set" data="api_on_answer=originate 
sofia/internal/2000 at 10.10.10.10 &eavesdrop(${uuid})"/>-->
        <action application="bridge" 
data="{ecoute=${caller_id_number}}sofia/gateway/${caller_id_number}/${destination_number}"/>

so when a call is setup, FS initiate a new call to 2000 and eavesdrop 
the call.
But I have a small problem, the callee receives no sound until the 
eavesdropper send a SIP reply, so there is a 2-3 seconds delay before 
caller and callee can talk each other.

rod


rod a ?crit :
> Hi all,
>
> I'm trying to do this in LUA:
>      A call B
>
> and I'd like to setup a new call to C with eavesdrop of A conversation 
> with B.
>
> I have no idea how to do this if someone can help.
> I switched to LUA cause I see no way to achieve this with dialplan 
> (snippets are welcome).
>
> regards,
> rod
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   


From mike at van.lammeren.net  Thu Jan 14 07:15:52 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 14 Jan 2010 10:15:52 -0500
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0F362E8@BESONE.corp.dom>
References: <5d2828f1001131130s6ef166c0wc98f70023420c0fd@mail.gmail.com>
	<A2167AE7633A5C4DAE56EEBD1428ECC0F362E8@BESONE.corp.dom>
Message-ID: <5d2828f1001140715m2aba7ed6n3cef077c64943e76@mail.gmail.com>

Hi Claudio!

Thanks for the additions to the wiki! Every little bit helps.

I don't think I explained myself well, earlier. The point I was trying to
make about the wiki is that rather than remove the section about
"originate", it would be better to make an entry like "originate -- Does
anyone know what this does?"

Mike van Lammeren


On Thu, Jan 14, 2010 at 6:23 AM, Cavalera Claudio Luigi <
Claudio.Cavalera at italtel.it> wrote:

>  Hi Mike,
> in fact i've completed that page with the list of available  session
> functions.
>
> I've not removed "session:originate" yet, but it would be better if someone
> could provide an example in order to write an example in the wiki.
> I've added this valuable example also with the help of rupa
> http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.Session
>
> I would like to write something also about api_on_answer to use an api
> instead of a dialplan application.
>
> BRs,
> Claudio
>
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Mike van
> Lammeren
> *Sent:* Wednesday, January 13, 2010 8:30 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] playing with sessions in lua
>
> Hello!
>
> Before you remove "session:originate" from the wiki, you should take a look
> at this:
>
> http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocumented_Session_Functions.3F
>
> <http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocumented_Session_Functions.3F>There
> is, in fact, a function called "originate".
>
> Mike van Lammeren
>
>
> On Wed, Jan 13, 2010 at 5:22 AM, Cavalera Claudio Luigi <
> Claudio.Cavalera at italtel.it> wrote:
>
>>  Thanks a lot Anthony,
>> some comments inline (and please forgive me for my broken email client).
>>
>>  >>  example1: Consider this simple lua script in which i create two
>> sessions:
>>
>>  >>  api = freeswitch.API();
>>  >>  api:execute("originate", "sofia/internal/1001%192.168.1.1 &park");
>>  >>  api:execute("originate", "sofia/internal/1002%192.168.1.1 &park");
>>
>>  >  capture the output from api:execute the uuid is in there
>>
>> Thx a lot,
>> this was one piece i was missing although it's already on the wiki here:
>> http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls
>>
>>
>>  > because lua calls it freeswitch.bridge
>>
>>  > session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
>>  > session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1",
>> session1);
>>  > freeswitch.bridge(session1, session2);
>>
>> good to now, there isn't any example of freeswitch.bridge in the wiki and
>> i would like to add one.
>> Where I could find the full api of
>> freeswitch.Session( ) ?
>> because I've seen this working also without "session1" in the second line:
>>  session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
>> session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");
>> freeswitch.bridge(session1, session2);
>>
>>  also is there any difference between  freeswitch.bridge
>> and freeswitch.execute(uuid_bridge ...) ?
>>
>>   >> example3: yet another possibility
>>  >> local session1 =
>> freeswitch.Session("sofia/internal/1001%192.168.1.1");
>>  >> session1:originate(session2, "sofia/internal/1002%192.168.1.1",
>> 1000);
>>  >> but it does not work either.
>>
>>  > The above is gibberish try:
>>  > local session1 =
>> freeswitch.Session("sofia/internal/1001%192.168.1.1");
>>  > session1:execute("bridge", "sofia/internal/1002%192.168.1.1");
>>
>> Okay i will report also this bridge example on the wiki which was missing.
>> But does session:originate make sense in some cases or not? Otherwise i'm
>> going to remove this line on the wiki
>>  http://wiki.freeswitch.org/wiki/Mod_lua#session:originate
>>
>> Thanks,
>> Claudio
>>
>>
>>
>> Internet Email Confidentiality Footer
>>
>>
>> ********************************************************************************************************************************************
>>
>> La presente comunicazione, con le informazioni in essa contenute e ogni
>> documento o file allegato, e' rivolta unicamente alla/e persona/e cui e'
>> indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete
>> i destinatari/autorizzati siete avvisati che qualsiasi azione, copia,
>> comunicazione, divulgazione o simili basate sul contenuto di tali
>> informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P.,
>> D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se
>> avete ricevuto questa comunicazione per errore, vi preghiamo di darne
>> immediata notizia al mittente e di distruggere il messaggio originale e ogni
>> file allegato senza farne copia alcuna o riprodurne in alcun modo il
>> contenuto.
>>
>> This e-mail and its attachments are intended for the addressee(s) only and
>> are confidential and/or may contain legally privileged information. If you
>> have received this message by mistake or are not one of the addressees
>> above, you may take no action based on it, and you may not copy or show it
>> to anyone; please reply to this e-mail and point out the error which has
>> occurred.
>>
>> ********************************************************************************************************************************************
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
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From sos at sokhapkin.dyndns.org  Thu Jan 14 07:20:06 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Jan 2010 10:20:06 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
Message-ID: <201001141020.06137.sos@sokhapkin.dyndns.org>

With trunk version and bypass_media=true the behavior is different - leg a is 
terminated after 20 seconds of wait, dialplan can't continue. Dialplan and FS 
log are at http://pastebin.freeswitch.org/11794 , note timestamps of lines 
378 and 379.

FS is running on 192.168.1.2:5060, SIP client on 192.168.1.2:5066, 
103 at 192.168.1.254 responds with early media and 480.

Let me know if you need additional information.

On Wednesday 13 January 2010, Michael Jerris wrote:
> On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> > Critical issues are when SIP error come after 18X provisional response.
> >
> > - if bypass_media is false then dialplan stops and leg a is explicitly
> > hang up (switch_ivr_bridge.c, line 513).
>
>  behavior can be modified with continue_on_fail and hangup_after_bridge
> channel vars, perhaps ignore_early_media as well
>
> > - if bypass_media is true, then dialplan continue, but there is 10
> > seconds delay before next bridge application sends INVITE to gateway (
> > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
> >4.html ) I didn't track down yet why this happens looking at FS sources.
>
> see response on that thread
>
> > Why I didn't open a bug on jira? Because FS behaves according to the
> > design and specs :-) But not according to real world requirements...
>
> Really, people are trying to help you and your going to be snarky in
> response?
>
> > On Wednesday 13 January 2010, Brian West wrote:
> >> Can you elaborate on these "Critical" issues you seem to be having?  Why
> >> aren't you opening a jira for them if they are that critical to your
> >> needs?
>
> Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at van.lammeren.net  Thu Jan 14 07:44:04 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 14 Jan 2010 10:44:04 -0500
Subject: [Freeswitch-users] Question about a1-hash
Message-ID: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>

Hello!

I have written a Lua script to connect to a database and provide directory
information for phones registering with FreeSWITCH.

My problem is that I store an MD5 hash of the passwords in the database, so
I wish there was a way to get FreeSWITCH to authenticate using the MD5 hash
of the password provided by the phone, and not the password itself.

According to the
wiki<http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide>,
it is possible to pass in a parameter called *a1-hash* instead of the
username and password. The a1-hash parameter is an MD5 hash of a string
comprising the username, domain and password, separated by
colons. Unfortunately, I can't generate that string, since I don't have the
raw password, just the MD5 hash.

I would have my Lua script do the authentication, but cannot because
FreeSWITCH doesn't pass the user's password to the script.

The best solution I can think of is to enter the MD5 hash of the password in
the phone.

Does anyone have a better idea?


Mike van Lammeren
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From sos at sokhapkin.dyndns.org  Thu Jan 14 07:46:58 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Jan 2010 10:46:58 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001141020.06137.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
	<201001141020.06137.sos@sokhapkin.dyndns.org>
Message-ID: <201001141046.58561.sos@sokhapkin.dyndns.org>

I repeated the test without bypass_media. Leg a is terminated as soon as 480 
received from leg b and dialplan doesn't continue. Let me know if you want me 
to post FS log to pastebin.

On Thursday 14 January 2010, Sergey Okhapkin wrote:
> With trunk version and bypass_media=true the behavior is different - leg a
> is terminated after 20 seconds of wait, dialplan can't continue. Dialplan
> and FS log are at http://pastebin.freeswitch.org/11794 , note timestamps of
> lines 378 and 379.
>
> FS is running on 192.168.1.2:5060, SIP client on 192.168.1.2:5066,
> 103 at 192.168.1.254 responds with early media and 480.
>
> Let me know if you need additional information.
>
> On Wednesday 13 January 2010, Michael Jerris wrote:
> > On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> > > Critical issues are when SIP error come after 18X provisional response.
> > >
> > > - if bypass_media is false then dialplan stops and leg a is explicitly
> > > hang up (switch_ivr_bridge.c, line 513).
> >
> >  behavior can be modified with continue_on_fail and hangup_after_bridge
> > channel vars, perhaps ignore_early_media as well
> >
> > > - if bypass_media is true, then dialplan continue, but there is 10
> > > seconds delay before next bridge application sends INVITE to gateway (
> > > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024
> > >35 4.html ) I didn't track down yet why this happens looking at FS
> > > sources.
> >
> > see response on that thread
> >
> > > Why I didn't open a bug on jira? Because FS behaves according to the
> > > design and specs :-) But not according to real world requirements...
> >
> > Really, people are trying to help you and your going to be snarky in
> > response?
> >
> > > On Wednesday 13 January 2010, Brian West wrote:
> > >> Can you elaborate on these "Critical" issues you seem to be having? 
> > >> Why aren't you opening a jira for them if they are that critical to
> > >> your needs?
> >
> > Mike
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Thu Jan 14 08:00:00 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Jan 2010 10:00:00 -0600
Subject: [Freeswitch-users] Question about a1-hash
In-Reply-To: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>
References: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>
Message-ID: <7677225F-6384-415B-AF98-09DED8A1F033@freeswitch.org>

We don't have the password so we can't pass it to you please read: http://en.wikipedia.org/wiki/Digest_access_authentication

Its how the authentication is done and we are never given the text of the password you are however given the details so you can calculate the response and verify it without having to know the password.

/b

On Jan 14, 2010, at 9:44 AM, Mike van Lammeren wrote:

> Hello!
> 
> I have written a Lua script to connect to a database and provide directory information for phones registering with FreeSWITCH.
> 
> My problem is that I store an MD5 hash of the passwords in the database, so I wish there was a way to get FreeSWITCH to authenticate using the MD5 hash of the password provided by the phone, and not the password itself.
> 
> According to the wiki, it is possible to pass in a parameter called a1-hash instead of the username and password. The a1-hash parameter is an MD5 hash of a string comprising the username, domain and password, separated by colons. Unfortunately, I can't generate that string, since I don't have the raw password, just the MD5 hash.
> 
> I would have my Lua script do the authentication, but cannot because FreeSWITCH doesn't pass the user's password to the script.
> 
> The best solution I can think of is to enter the MD5 hash of the password in the phone.
> 
> Does anyone have a better idea?
> 
> 
> Mike van Lammeren

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From mike at jerris.com  Thu Jan 14 08:08:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Jan 2010 11:08:37 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001141046.58561.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
	<201001141020.06137.sos@sokhapkin.dyndns.org>
	<201001141046.58561.sos@sokhapkin.dyndns.org>
Message-ID: <B2801B0F-47E5-43C8-919A-671D5733C53C@jerris.com>

Lets move this all to jira so that we can properly track the issue.  We will need full debug logs with siptrace, on current svn trunk along with dialplan and such to reproduce the issue.

Mike

On Jan 14, 2010, at 10:46 AM, Sergey Okhapkin wrote:

> I repeated the test without bypass_media. Leg a is terminated as soon as 480 
> received from leg b and dialplan doesn't continue. Let me know if you want me 
> to post FS log to pastebin.
> 
> On Thursday 14 January 2010, Sergey Okhapkin wrote:
>> With trunk version and bypass_media=true the behavior is different - leg a
>> is terminated after 20 seconds of wait, dialplan can't continue. Dialplan
>> and FS log are at http://pastebin.freeswitch.org/11794 , note timestamps of
>> lines 378 and 379.
>> 
>> FS is running on 192.168.1.2:5060, SIP client on 192.168.1.2:5066,
>> 103 at 192.168.1.254 responds with early media and 480.
>> 
>> Let me know if you need additional information.



From mike at jerris.com  Thu Jan 14 08:12:59 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Jan 2010 11:12:59 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001140226y2b035f37rfcafb03190435ec1@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
	<201001132326.41354.sos@sokhapkin.dyndns.org>
	<52E8BE9E-B6D6-411C-9080-697AF24017B4@jerris.com>
	<191c3a031001132114s18ed2baw2ff802dbd615fe01@mail.gmail.com>
	<191c3a031001132115u2d88bab7p7cbb27ab1eaad466@mail.gmail.com>
	<9853f4ff1001140226y2b035f37rfcafb03190435ec1@mail.gmail.com>
Message-ID: <43692088-738A-475C-8D60-E40D296ECEB1@jerris.com>

Okay, so that is a very interesting use case.   Seems a bizarre way to do this for the carrier, but interesting none the less.  I'd say the chances of actually muxing the early media is small, what might be possible would be something to say which b legs media to pass along to the a leg.  I have not looked at all at how complicated this is, it will be down deep somewhere in switch_ivr_originate code around where we if for ignore_early_media.  This code is pretty complex, I can't say that we will ever actually add this functionality, but at least now I won't blow it off as complete nonsense.

Mike



On Jan 14, 2010, at 5:26 AM, David Villasmil wrote:

> Hello again,
> 
> Using ingore early media only ignores ALL media, that's not what I need. At least in europe i've seen it many times, MNOs provide a service with which you can have a song played to the caller while the call is connecting to your cell phone. This is basically what I'm trying to achieve.
> 
> The content provider is in possesions of the media and it require us to fork the call and send a SIP INVITE to the on one leg and the call to the destination number on another leg. they will only provide a progress, no answer.
> 
> I CAN do it locally by playing the file as a custom ringback but that's not the standard in terms of commercial use of the content.
> 
> Is there any way to modify the behaviour of fs when i receives media? Let's say i.e. it doesn't drop the other leg, but provides the first early media it receives and just wait for some channel to answer? I will not have both media but it would work.
> 
> thanks
> 
> David
> 
> On Thu, Jan 14, 2010 at 6:15 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> Perhaps best not to help him anymore without an apology for the snap judgement and comparison to asterisk clearly designed to push our buttons.
> We'll be here when you realize we were trying to help you but I can't promise we will still have paitence......
> 
> 
>> On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike at jerris.com> wrote:
>> 
>> You already are running on trunk.
>> 
>> Mike
>> On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote: > I run FreeSWITCH Version 1.0.5pre10 (16012M...
>> 
>> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists...
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From freeswitch at aastral.net  Thu Jan 14 08:15:06 2010
From: freeswitch at aastral.net (Bill W)
Date: Thu, 14 Jan 2010 11:15:06 -0500
Subject: [Freeswitch-users] Question about a1-hash
In-Reply-To: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>
References: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>
Message-ID: <4B4F430A.8080006@aastral.net>

Why don't you just store the a1-hash in the database instead of the 
password?

-Bill W.


Mike van Lammeren wrote:
> Hello!
> 
> I have written a Lua script to connect to a database and provide 
> directory information for phones registering with FreeSWITCH.
> 
> My problem is that I store an MD5 hash of the passwords in the database, 
> so I wish there was a way to get FreeSWITCH to authenticate using the 
> MD5 hash of the password provided by the phone, and not the password itself.
> 
> According to the wiki 
> <http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide>, it is 
> possible to pass in a parameter called /a1-hash/ instead of the username 
> and password. The a1-hash parameter is an MD5 hash of a string 
> comprising the username, domain and password, separated by 
> colons. Unfortunately, I can't generate that string, since I don't have 
> the raw password, just the MD5 hash.
> 
> I would have my Lua script do the authentication, but cannot because 
> FreeSWITCH doesn't pass the user's password to the script.
> 
> The best solution I can think of is to enter the MD5 hash of the 
> password in the phone.
> 
> Does anyone have a better idea?
> 
> 
> Mike van Lammeren
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Thu Jan 14 08:22:21 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 10:22:21 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001140719.12363.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>

Sergey,

The bug you reference was closed because proxy_media mode by design send's
the B leg's codec to the A
LEG so if there is a failure condition there is no easy graceful way to back
out and try another call.
I will try to make it possible if yo post a bounty, I estimate a minimum of
$1000USD in consulting time.

The other one I might look at once you have apologized.

I can probably add some code to make the bridge exit without terminating the
A leg even when hangup_after_bridge=true in the case where the the B leg is
not answered but right now I am sort of annoyed with your attitude in this
thread.


On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Case 1 (bypass_media is off) is already on jira,
> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
> installation with latest trunk to reproduce case 2, when bypass_media=true
> and 10 seconds delay happens when 18X and then error are received on leg b.
>
> On Wednesday 13 January 2010, Michael Jerris wrote:
> > On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> > > Critical issues are when SIP error come after 18X provisional response.
> > >
> > > - if bypass_media is false then dialplan stops and leg a is explicitly
> > > hang up (switch_ivr_bridge.c, line 513).
> >
> >  behavior can be modified with continue_on_fail and hangup_after_bridge
> > channel vars, perhaps ignore_early_media as well
> >
> > > - if bypass_media is true, then dialplan continue, but there is 10
> > > seconds delay before next bridge application sends INVITE to gateway (
> > >
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
> > >4.html ) I didn't track down yet why this happens looking at FS sources.
> >
> > see response on that thread
> >
> > > Why I didn't open a bug on jira? Because FS behaves according to the
> > > design and specs :-) But not according to real world requirements...
> >
> > Really, people are trying to help you and your going to be snarky in
> > response?
> >
> > > On Wednesday 13 January 2010, Brian West wrote:
> > >> Can you elaborate on these "Critical" issues you seem to be having?
>  Why
> > >> aren't you opening a jira for them if they are that critical to your
> > >> needs?
> >
> > Mike
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at van.lammeren.net  Thu Jan 14 08:24:42 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 14 Jan 2010 11:24:42 -0500
Subject: [Freeswitch-users] Question about a1-hash
In-Reply-To: <7677225F-6384-415B-AF98-09DED8A1F033@freeswitch.org>
References: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>
	<7677225F-6384-415B-AF98-09DED8A1F033@freeswitch.org>
Message-ID: <5d2828f1001140824u416ec5ebh2b2636af4a02ae33@mail.gmail.com>

That's awesome! I should have noticed those 32-character strings in the
parameters passed to the script. Thanks!

It's a little off-topic, but I'm glad to see someone using digest
authentication. It's too bad that it was un-supported by browsers for so
long, that no one touched it for web apps. The choice is either use basic
authentication, which is plaintext, or switch to https. With https, not
everyone realizes that the web server, and any apps, can see the password in
plain text.

Mike van Lammeren


On Thu, Jan 14, 2010 at 11:00 AM, Brian West <brian at freeswitch.org> wrote:

> We don't have the password so we can't pass it to you please read:
> http://en.wikipedia.org/wiki/Digest_access_authentication
>
> Its how the authentication is done and we are never given the text of the
> password you are however given the details so you can calculate the response
> and verify it without having to know the password.
>
> /b
>
> On Jan 14, 2010, at 9:44 AM, Mike van Lammeren wrote:
>
> Hello!
>
> I have written a Lua script to connect to a database and provide directory
> information for phones registering with FreeSWITCH.
>
> My problem is that I store an MD5 hash of the passwords in the database, so
> I wish there was a way to get FreeSWITCH to authenticate using the MD5 hash
> of the password provided by the phone, and not the password itself.
>
> According to the wiki<http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide>,
> it is possible to pass in a parameter called *a1-hash* instead of the
> username and password. The a1-hash parameter is an MD5 hash of a string
> comprising the username, domain and password, separated by
> colons. Unfortunately, I can't generate that string, since I don't have the
> raw password, just the MD5 hash.
>
> I would have my Lua script do the authentication, but cannot because
> FreeSWITCH doesn't pass the user's password to the script.
>
> The best solution I can think of is to enter the MD5 hash of the password
> in the phone.
>
> Does anyone have a better idea?
>
>
> Mike van Lammeren
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Jan 14 08:26:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 10:26:52 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
	<191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
Message-ID: <191c3a031001140826q2d014484v8807bf3b98b78f63@mail.gmail.com>

David,

Your request is somewhat challenging because we have to make sure FreeSWITCH
is agnostic in protocols and that aside, we also have a very complicated and
feature-rich originate API that does not currently support sending audio
from one leg to the A leg while it's trying to call 10 other B legs.  Please
try to understand that this request is a very special side case and we would
have to do many hours of work to make it possible.  I am not sure if you are
simply asking if its possible or if you want us to implement it for you but
I am afraid it would fall under commercial support to undertake that unique
of a feature that only helps a very small percentage of our user base
considering all the work we already have to do every day.



On Thu, Jan 14, 2010 at 10:22 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Sergey,
>
> The bug you reference was closed because proxy_media mode by design send's
> the B leg's codec to the A
> LEG so if there is a failure condition there is no easy graceful way to
> back out and try another call.
> I will try to make it possible if yo post a bounty, I estimate a minimum of
> $1000USD in consulting time.
>
> The other one I might look at once you have apologized.
>
> I can probably add some code to make the bridge exit without terminating
> the A leg even when hangup_after_bridge=true in the case where the the B leg
> is not answered but right now I am sort of annoyed with your attitude in
> this thread.
>
>
>
> On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin <sos at sokhapkin.dyndns.org
> > wrote:
>
>> Case 1 (bypass_media is off) is already on jira,
>> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
>> installation with latest trunk to reproduce case 2, when bypass_media=true
>> and 10 seconds delay happens when 18X and then error are received on leg
>> b.
>>
>> On Wednesday 13 January 2010, Michael Jerris wrote:
>> > On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
>> > > Critical issues are when SIP error come after 18X provisional
>> response.
>> > >
>> > > - if bypass_media is false then dialplan stops and leg a is explicitly
>> > > hang up (switch_ivr_bridge.c, line 513).
>> >
>> >  behavior can be modified with continue_on_fail and hangup_after_bridge
>> > channel vars, perhaps ignore_early_media as well
>> >
>> > > - if bypass_media is true, then dialplan continue, but there is 10
>> > > seconds delay before next bridge application sends INVITE to gateway (
>> > >
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
>> > >4.html ) I didn't track down yet why this happens looking at FS
>> sources.
>> >
>> > see response on that thread
>> >
>> > > Why I didn't open a bug on jira? Because FS behaves according to the
>> > > design and specs :-) But not according to real world requirements...
>> >
>> > Really, people are trying to help you and your going to be snarky in
>> > response?
>> >
>> > > On Wednesday 13 January 2010, Brian West wrote:
>> > >> Can you elaborate on these "Critical" issues you seem to be having?
>>  Why
>> > >> aren't you opening a jira for them if they are that critical to your
>> > >> needs?
>> >
>> > Mike
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at van.lammeren.net  Thu Jan 14 08:27:26 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 14 Jan 2010 11:27:26 -0500
Subject: [Freeswitch-users] Question about a1-hash
In-Reply-To: <4B4F430A.8080006@aastral.net>
References: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>
	<4B4F430A.8080006@aastral.net>
Message-ID: <5d2828f1001140827w386fe2acw478ae378b0ecfe53@mail.gmail.com>

Hi Bill!

That's a perfectly reasonable suggestion, but for this solution, I need a
username/password combo that spans domains. Also, the password is used in
other places too.

Thanks for the reply!

Mike van Lammeren


On Thu, Jan 14, 2010 at 11:15 AM, Bill W <freeswitch at aastral.net> wrote:

> Why don't you just store the a1-hash in the database instead of the
> password?
>
> -Bill W.
>
>
> Mike van Lammeren wrote:
> > Hello!
> >
> > I have written a Lua script to connect to a database and provide
> > directory information for phones registering with FreeSWITCH.
> >
> > My problem is that I store an MD5 hash of the passwords in the database,
> > so I wish there was a way to get FreeSWITCH to authenticate using the
> > MD5 hash of the password provided by the phone, and not the password
> itself.
> >
> > According to the wiki
> > <http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide>, it is
> > possible to pass in a parameter called /a1-hash/ instead of the username
> > and password. The a1-hash parameter is an MD5 hash of a string
> > comprising the username, domain and password, separated by
> > colons. Unfortunately, I can't generate that string, since I don't have
> > the raw password, just the MD5 hash.
> >
> > I would have my Lua script do the authentication, but cannot because
> > FreeSWITCH doesn't pass the user's password to the script.
> >
> > The best solution I can think of is to enter the MD5 hash of the
> > password in the phone.
> >
> > Does anyone have a better idea?
> >
> >
> > Mike van Lammeren
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Thu Jan 14 08:37:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 10:37:45 -0600
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <5d2828f1001140715m2aba7ed6n3cef077c64943e76@mail.gmail.com>
References: <5d2828f1001131130s6ef166c0wc98f70023420c0fd@mail.gmail.com>
	<A2167AE7633A5C4DAE56EEBD1428ECC0F362E8@BESONE.corp.dom>
	<5d2828f1001140715m2aba7ed6n3cef077c64943e76@mail.gmail.com>
Message-ID: <191c3a031001140837m7cffcdd5w71886d6c8ba1dafe@mail.gmail.com>

Don't forget that lua,perl,python,managed,java all share the same exact C++
source file with swig so the same exact api applies to all of the above.

On Thu, Jan 14, 2010 at 9:15 AM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Hi Claudio!
>
> Thanks for the additions to the wiki! Every little bit helps.
>
> I don't think I explained myself well, earlier. The point I was trying to
> make about the wiki is that rather than remove the section about
> "originate", it would be better to make an entry like "originate -- Does
> anyone know what this does?"
>
> Mike van Lammeren
>
>
> On Thu, Jan 14, 2010 at 6:23 AM, Cavalera Claudio Luigi <
> Claudio.Cavalera at italtel.it> wrote:
>
>>  Hi Mike,
>> in fact i've completed that page with the list of available  session
>> functions.
>>
>> I've not removed "session:originate" yet, but it would be better if
>> someone could provide an example in order to write an example in the wiki.
>> I've added this valuable example also with the help of rupa
>> http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.Session
>>
>> I would like to write something also about api_on_answer to use an api
>> instead of a dialplan application.
>>
>> BRs,
>> Claudio
>>
>>  ------------------------------
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Mike van
>> Lammeren
>> *Sent:* Wednesday, January 13, 2010 8:30 PM
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] playing with sessions in lua
>>
>> Hello!
>>
>> Before you remove "session:originate" from the wiki, you should take a
>> look at this:
>>
>> http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocumented_Session_Functions.3F
>>
>> <http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocumented_Session_Functions.3F>There
>> is, in fact, a function called "originate".
>>
>> Mike van Lammeren
>>
>>
>> On Wed, Jan 13, 2010 at 5:22 AM, Cavalera Claudio Luigi <
>> Claudio.Cavalera at italtel.it> wrote:
>>
>>>  Thanks a lot Anthony,
>>> some comments inline (and please forgive me for my broken email client).
>>>
>>>  >>  example1: Consider this simple lua script in which i create two
>>> sessions:
>>>
>>>  >>  api = freeswitch.API();
>>>  >>  api:execute("originate", "sofia/internal/1001%192.168.1.1 &park");
>>>  >>  api:execute("originate", "sofia/internal/1002%192.168.1.1 &park");
>>>
>>>  >  capture the output from api:execute the uuid is in there
>>>
>>> Thx a lot,
>>> this was one piece i was missing although it's already on the wiki here:
>>> http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls
>>>
>>>
>>>  > because lua calls it freeswitch.bridge
>>>
>>>  > session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
>>>  > session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1",
>>> session1);
>>>  > freeswitch.bridge(session1, session2);
>>>
>>> good to now, there isn't any example of freeswitch.bridge in the wiki and
>>> i would like to add one.
>>> Where I could find the full api of
>>> freeswitch.Session( ) ?
>>> because I've seen this working also without "session1" in the second
>>> line:
>>>  session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
>>> session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");
>>> freeswitch.bridge(session1, session2);
>>>
>>>  also is there any difference between  freeswitch.bridge
>>> and freeswitch.execute(uuid_bridge ...) ?
>>>
>>>   >> example3: yet another possibility
>>>  >> local session1 =
>>> freeswitch.Session("sofia/internal/1001%192.168.1.1");
>>>  >> session1:originate(session2, "sofia/internal/1002%192.168.1.1",
>>> 1000);
>>>  >> but it does not work either.
>>>
>>>  > The above is gibberish try:
>>>  > local session1 =
>>> freeswitch.Session("sofia/internal/1001%192.168.1.1");
>>>  > session1:execute("bridge", "sofia/internal/1002%192.168.1.1");
>>>
>>> Okay i will report also this bridge example on the wiki which was
>>> missing.
>>> But does session:originate make sense in some cases or not? Otherwise i'm
>>> going to remove this line on the wiki
>>>  http://wiki.freeswitch.org/wiki/Mod_lua#session:originate
>>>
>>> Thanks,
>>> Claudio
>>>
>>>
>>>
>>> Internet Email Confidentiality Footer
>>>
>>>
>>> ********************************************************************************************************************************************
>>>
>>> La presente comunicazione, con le informazioni in essa contenute e ogni
>>> documento o file allegato, e' rivolta unicamente alla/e persona/e cui e'
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>>> informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P.,
>>> D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se
>>> avete ricevuto questa comunicazione per errore, vi preghiamo di darne
>>> immediata notizia al mittente e di distruggere il messaggio originale e ogni
>>> file allegato senza farne copia alcuna o riprodurne in alcun modo il
>>> contenuto.
>>>
>>> This e-mail and its attachments are intended for the addressee(s) only
>>> and are confidential and/or may contain legally privileged information. If
>>> you have received this message by mistake or are not one of the addressees
>>> above, you may take no action based on it, and you may not copy or show it
>>> to anyone; please reply to this e-mail and point out the error which has
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>>>
>>> ********************************************************************************************************************************************
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From david.villasmil.work at gmail.com  Thu Jan 14 08:41:54 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 14 Jan 2010 17:41:54 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001140826q2d014484v8807bf3b98b78f63@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
	<191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
	<191c3a031001140826q2d014484v8807bf3b98b78f63@mail.gmail.com>
Message-ID: <9853f4ff1001140841n603ddc45n972b3aba83600d01@mail.gmail.com>

Anthony,

     Thanks. I understand it's complicated. Another option is to be able to
configure whether or not to discard all other B-legs on receiving media but
on receiving an final code like 200. This way we will still get the first
media that arrived but provide an answer on the actual channel that provided
the 200.

My request comes from 2 sides, one commercial and the other personal. For
the commercial I just need to now whether it is possible by some tweaking so
the functionality can be offered.

The other, I've been involved in FS as a user from the very beginning
(although I was absent for some time), and I'd like to help in the
development of the project by letting the community know what's out there
commercially speaking.


Hope you can help.

David

On Thu, Jan 14, 2010 at 5:26 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> David,
>
> Your request is somewhat challenging because we have to make sure
> FreeSWITCH is agnostic in protocols and that aside, we also have a very
> complicated and feature-rich originate API that does not currently support
> sending audio from one leg to the A leg while it's trying to call 10 other B
> legs.  Please try to understand that this request is a very special side
> case and we would have to do many hours of work to make it possible.  I am
> not sure if you are simply asking if its possible or if you want us to
> implement it for you but I am afraid it would fall under commercial support
> to undertake that unique of a feature that only helps a very small
> percentage of our user base considering all the work we already have to do
> every day.
>
>
>
>
> On Thu, Jan 14, 2010 at 10:22 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Sergey,
>>
>> The bug you reference was closed because proxy_media mode by design send's
>> the B leg's codec to the A
>> LEG so if there is a failure condition there is no easy graceful way to
>> back out and try another call.
>> I will try to make it possible if yo post a bounty, I estimate a minimum
>> of $1000USD in consulting time.
>>
>> The other one I might look at once you have apologized.
>>
>> I can probably add some code to make the bridge exit without terminating
>> the A leg even when hangup_after_bridge=true in the case where the the B leg
>> is not answered but right now I am sort of annoyed with your attitude in
>> this thread.
>>
>>
>>
>> On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin <
>> sos at sokhapkin.dyndns.org> wrote:
>>
>>> Case 1 (bypass_media is off) is already on jira,
>>> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
>>> installation with latest trunk to reproduce case 2, when
>>> bypass_media=true
>>> and 10 seconds delay happens when 18X and then error are received on leg
>>> b.
>>>
>>> On Wednesday 13 January 2010, Michael Jerris wrote:
>>> > On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
>>> > > Critical issues are when SIP error come after 18X provisional
>>> response.
>>> > >
>>> > > - if bypass_media is false then dialplan stops and leg a is
>>> explicitly
>>> > > hang up (switch_ivr_bridge.c, line 513).
>>> >
>>> >  behavior can be modified with continue_on_fail and hangup_after_bridge
>>> > channel vars, perhaps ignore_early_media as well
>>> >
>>> > > - if bypass_media is true, then dialplan continue, but there is 10
>>> > > seconds delay before next bridge application sends INVITE to gateway
>>> (
>>> > >
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
>>> > >4.html ) I didn't track down yet why this happens looking at FS
>>> sources.
>>> >
>>> > see response on that thread
>>> >
>>> > > Why I didn't open a bug on jira? Because FS behaves according to the
>>> > > design and specs :-) But not according to real world requirements...
>>> >
>>> > Really, people are trying to help you and your going to be snarky in
>>> > response?
>>> >
>>> > > On Wednesday 13 January 2010, Brian West wrote:
>>> > >> Can you elaborate on these "Critical" issues you seem to be having?
>>>  Why
>>> > >> aren't you opening a jira for them if they are that critical to your
>>> > >> needs?
>>> >
>>> > Mike
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sos at sokhapkin.dyndns.org  Thu Jan 14 08:41:59 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Jan 2010 11:41:59 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
	<191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
Message-ID: <201001141141.59831.sos@sokhapkin.dyndns.org>

Anthony, I apology, I didn't expect you'll be annoyed. As for the issue, the 
issue is not in "proxy_media" setting, same happens when both "proxy_media" 
and "bypass_media" are not set. The code does explicit leg A hangup in 
switch_ivr_bridge.c, line 513. The condition there is "if leg A is up and leg 
A is not in answered state, then hangup leg A", this prevents dialplan to 
continue execution on early media bridge termination.

Issue when "bypass_media=true" is completely different, I'll open the issue on 
jira.

On Thursday 14 January 2010, Anthony Minessale wrote:
> Sergey,
>
> The bug you reference was closed because proxy_media mode by design send's
> the B leg's codec to the A
> LEG so if there is a failure condition there is no easy graceful way to
> back out and try another call.
> I will try to make it possible if yo post a bounty, I estimate a minimum of
> $1000USD in consulting time.
>
> The other one I might look at once you have apologized.
>
> I can probably add some code to make the bridge exit without terminating
> the A leg even when hangup_after_bridge=true in the case where the the B
> leg is not answered but right now I am sort of annoyed with your attitude
> in this thread.
>
>
> On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin
>
> <sos at sokhapkin.dyndns.org>wrote:
> > Case 1 (bypass_media is off) is already on jira,
> > http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
> > installation with latest trunk to reproduce case 2, when
> > bypass_media=true and 10 seconds delay happens when 18X and then error
> > are received on leg b.
> >
> > On Wednesday 13 January 2010, Michael Jerris wrote:
> > > On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> > > > Critical issues are when SIP error come after 18X provisional
> > > > response.
> > > >
> > > > - if bypass_media is false then dialplan stops and leg a is
> > > > explicitly hang up (switch_ivr_bridge.c, line 513).
> > >
> > >  behavior can be modified with continue_on_fail and hangup_after_bridge
> > > channel vars, perhaps ignore_early_media as well
> > >
> > > > - if bypass_media is true, then dialplan continue, but there is 10
> > > > seconds delay before next bridge application sends INVITE to gateway
> > > > (
> >
> > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
> >
> > > >4.html ) I didn't track down yet why this happens looking at FS
> > > > sources.
> > >
> > > see response on that thread
> > >
> > > > Why I didn't open a bug on jira? Because FS behaves according to the
> > > > design and specs :-) But not according to real world requirements...
> > >
> > > Really, people are trying to help you and your going to be snarky in
> > > response?
> > >
> > > > On Wednesday 13 January 2010, Brian West wrote:
> > > >> Can you elaborate on these "Critical" issues you seem to be having?
> >
> >  Why
> >
> > > >> aren't you opening a jira for them if they are that critical to your
> > > >> needs?
> > >
> > > Mike
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From anthony.minessale at gmail.com  Thu Jan 14 08:47:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 10:47:24 -0600
Subject: [Freeswitch-users] Not able to capture the custom event
In-Reply-To: <13529f9d1001140124t42e9dfe5u3e70a6976aa032a8@mail.gmail.com>
References: <13529f9d1001140124t42e9dfe5u3e70a6976aa032a8@mail.gmail.com>
Message-ID: <191c3a031001140847q1238bbcdi4d9dd84648ee2a27@mail.gmail.com>

you cant subscribe to all custom events, you have to specify a list of
subclasses
you have to say events plain custom myevent::ACDnotify

On Thu, Jan 14, 2010 at 3:24 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:

> Hi Guys,
>
> I'm testing to see whether a custom event can be triggered and captured.
> Here's the extension:
>
>     <extension name="test_ivr">
>       <condition field="destination_number" expression="^777$">
>         <action application="event" data="Event-Name=*CUSTOM*
> ,Event-Subclass=myevent::ACDnotify"/>
>         <action application="transfer" data="9999 XML default" />
>       </condition>
>     </extension>
>
> If I telnet and use "events plain all", I can get the event. However, if I
> use "events plain CUSTOM" or "event plain CUSTOM", it doesn't get captured.
> Is there anything wrong with the command?
>
> Regards,
> -Jingwei
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mrene_lists at avgs.ca  Thu Jan 14 08:47:30 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 14 Jan 2010 11:47:30 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001141141.59831.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
	<191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
	<201001141141.59831.sos@sokhapkin.dyndns.org>
Message-ID: <23BD1322-7882-4D43-80AB-27872096B4C9@avgs.ca>

One way you could fix it is set "hangup_after_bridge=false" and then  
set "execute_on_answer=set\shangup_after_bridge=true". That will make  
it continue executing the dialplan atter a failure that came in after  
183. Note that there is a side effect to this method, since audio was  
being relayed, you are forced to use the same codec on the 2nd  
carrier, best to tell them they are wrong failing a call after 183.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 14-Jan-10, at 11:41 AM, Sergey Okhapkin wrote:

> Anthony, I apology, I didn't expect you'll be annoyed. As for the  
> issue, the
> issue is not in "proxy_media" setting, same happens when both  
> "proxy_media"
> and "bypass_media" are not set. The code does explicit leg A hangup in
> switch_ivr_bridge.c, line 513. The condition there is "if leg A is  
> up and leg
> A is not in answered state, then hangup leg A", this prevents  
> dialplan to
> continue execution on early media bridge termination.
>
> Issue when "bypass_media=true" is completely different, I'll open  
> the issue on
> jira.
>
> On Thursday 14 January 2010, Anthony Minessale wrote:
>> Sergey,
>>
>> The bug you reference was closed because proxy_media mode by design  
>> send's
>> the B leg's codec to the A
>> LEG so if there is a failure condition there is no easy graceful  
>> way to
>> back out and try another call.
>> I will try to make it possible if yo post a bounty, I estimate a  
>> minimum of
>> $1000USD in consulting time.
>>
>> The other one I might look at once you have apologized.
>>
>> I can probably add some code to make the bridge exit without  
>> terminating
>> the A leg even when hangup_after_bridge=true in the case where the  
>> the B
>> leg is not answered but right now I am sort of annoyed with your  
>> attitude
>> in this thread.
>>
>>
>> On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin
>>
>> <sos at sokhapkin.dyndns.org>wrote:
>>> Case 1 (bypass_media is off) is already on jira,
>>> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
>>> installation with latest trunk to reproduce case 2, when
>>> bypass_media=true and 10 seconds delay happens when 18X and then  
>>> error
>>> are received on leg b.
>>>
>>> On Wednesday 13 January 2010, Michael Jerris wrote:
>>>> On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
>>>>> Critical issues are when SIP error come after 18X provisional
>>>>> response.
>>>>>
>>>>> - if bypass_media is false then dialplan stops and leg a is
>>>>> explicitly hang up (switch_ivr_bridge.c, line 513).
>>>>
>>>> behavior can be modified with continue_on_fail and  
>>>> hangup_after_bridge
>>>> channel vars, perhaps ignore_early_media as well
>>>>
>>>>> - if bypass_media is true, then dialplan continue, but there is 10
>>>>> seconds delay before next bridge application sends INVITE to  
>>>>> gateway
>>>>> (
>>>
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
>>>
>>>>> 4.html ) I didn't track down yet why this happens looking at FS
>>>>> sources.
>>>>
>>>> see response on that thread
>>>>
>>>>> Why I didn't open a bug on jira? Because FS behaves according to  
>>>>> the
>>>>> design and specs :-) But not according to real world  
>>>>> requirements...
>>>>
>>>> Really, people are trying to help you and your going to be snarky  
>>>> in
>>>> response?
>>>>
>>>>> On Wednesday 13 January 2010, Brian West wrote:
>>>>>> Can you elaborate on these "Critical" issues you seem to be  
>>>>>> having?
>>>
>>> Why
>>>
>>>>>> aren't you opening a jira for them if they are that critical to  
>>>>>> your
>>>>>> needs?
>>>>
>>>> Mike
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
>>>> s http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mrene_lists at avgs.ca  Thu Jan 14 08:49:06 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 14 Jan 2010 11:49:06 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001141141.59831.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
	<191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
	<201001141141.59831.sos@sokhapkin.dyndns.org>
Message-ID: <911CE0A0-5A54-4991-9642-98C236DC7678@avgs.ca>

Or try r16309, where Anthony fixed it so it doesnt hangup if the b-leg  
isnt answered yet :D

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 14-Jan-10, at 11:41 AM, Sergey Okhapkin wrote:

> Anthony, I apology, I didn't expect you'll be annoyed. As for the  
> issue, the
> issue is not in "proxy_media" setting, same happens when both  
> "proxy_media"
> and "bypass_media" are not set. The code does explicit leg A hangup in
> switch_ivr_bridge.c, line 513. The condition there is "if leg A is  
> up and leg
> A is not in answered state, then hangup leg A", this prevents  
> dialplan to
> continue execution on early media bridge termination.
>
> Issue when "bypass_media=true" is completely different, I'll open  
> the issue on
> jira.
>
> On Thursday 14 January 2010, Anthony Minessale wrote:
>> Sergey,
>>
>> The bug you reference was closed because proxy_media mode by design  
>> send's
>> the B leg's codec to the A
>> LEG so if there is a failure condition there is no easy graceful  
>> way to
>> back out and try another call.
>> I will try to make it possible if yo post a bounty, I estimate a  
>> minimum of
>> $1000USD in consulting time.
>>
>> The other one I might look at once you have apologized.
>>
>> I can probably add some code to make the bridge exit without  
>> terminating
>> the A leg even when hangup_after_bridge=true in the case where the  
>> the B
>> leg is not answered but right now I am sort of annoyed with your  
>> attitude
>> in this thread.
>>
>>
>> On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin
>>
>> <sos at sokhapkin.dyndns.org>wrote:
>>> Case 1 (bypass_media is off) is already on jira,
>>> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
>>> installation with latest trunk to reproduce case 2, when
>>> bypass_media=true and 10 seconds delay happens when 18X and then  
>>> error
>>> are received on leg b.
>>>
>>> On Wednesday 13 January 2010, Michael Jerris wrote:
>>>> On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
>>>>> Critical issues are when SIP error come after 18X provisional
>>>>> response.
>>>>>
>>>>> - if bypass_media is false then dialplan stops and leg a is
>>>>> explicitly hang up (switch_ivr_bridge.c, line 513).
>>>>
>>>> behavior can be modified with continue_on_fail and  
>>>> hangup_after_bridge
>>>> channel vars, perhaps ignore_early_media as well
>>>>
>>>>> - if bypass_media is true, then dialplan continue, but there is 10
>>>>> seconds delay before next bridge application sends INVITE to  
>>>>> gateway
>>>>> (
>>>
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
>>>
>>>>> 4.html ) I didn't track down yet why this happens looking at FS
>>>>> sources.
>>>>
>>>> see response on that thread
>>>>
>>>>> Why I didn't open a bug on jira? Because FS behaves according to  
>>>>> the
>>>>> design and specs :-) But not according to real world  
>>>>> requirements...
>>>>
>>>> Really, people are trying to help you and your going to be snarky  
>>>> in
>>>> response?
>>>>
>>>>> On Wednesday 13 January 2010, Brian West wrote:
>>>>>> Can you elaborate on these "Critical" issues you seem to be  
>>>>>> having?
>>>
>>> Why
>>>
>>>>>> aren't you opening a jira for them if they are that critical to  
>>>>>> your
>>>>>> needs?
>>>>
>>>> Mike
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
>>>> s http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From info at daccii.it  Thu Jan 14 08:50:21 2010
From: info at daccii.it (Albano Daniele Salvatore - Lavoro)
Date: Thu, 14 Jan 2010 17:50:21 +0100
Subject: [Freeswitch-users] ESL Mono/Managed
In-Reply-To: <1E6792AC-17F2-4DE9-9798-431350A5E77D@freeswitch.org>
References: <1E6792AC-17F2-4DE9-9798-431350A5E77D@freeswitch.org>
Message-ID: <4B4F4B4D.1060509@daccii.it>

Hi,

i should start to use it in short for a software i wrote in C#, so if 
you wants i can write them :)

However, i can't start before the next week, my develoment machine broke 
so i'm waiting some pieces to replace the broke stuff :\ I should be 
ready at the start of the next week.

Best Regards,
Daniele

Brian West ha scritto:
> I need someone that knows C# to write up some examples using ESL in C#
> 
> cd libs/esl
> make managedmod
> cd managed
> 
> Look at the perl example single_command.pl
> require ESL;
> 
> my $command = shift;
> my $args = join(" ", @ARGV);
> 
> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
> my $e = $con->api($command, $args);
> print $e->getBody();
> 
> 
> Need to write the same thing in C# and commit it as an example.  Also other examples exist in the perl directory those should be translated also if possible.
> 
> If you're interested please find me on IRC (bkw_)
> 
> Thanks,
> Brina
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rupa at rupa.com  Thu Jan 14 08:55:04 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 14 Jan 2010 10:55:04 -0600
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
Message-ID: <b39273531001140855i606a06a4ne55be84d825b1a74@mail.gmail.com>

I don't see an email2fax yet, but an important first step is in:

contrib/sathieu/email2pdf/email2pdf

On Thu, Jan 14, 2010 at 8:54 AM, Costa Zikalala <costa.zikalala at gmail.com>wrote:

> Hi All
>
> Has anyone worked on a email2fax script for mod_fax?
> If not how much would it cost for some genius here to quickly whip-up one?
>
> Ideally both email2fax and fax2email should come standard with mod_fax or
> is this not possible?
>
> Thanks
> Costa
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From brian at freeswitch.org  Thu Jan 14 08:57:40 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Jan 2010 10:57:40 -0600
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <191c3a031001140837m7cffcdd5w71886d6c8ba1dafe@mail.gmail.com>
References: <5d2828f1001131130s6ef166c0wc98f70023420c0fd@mail.gmail.com>
	<A2167AE7633A5C4DAE56EEBD1428ECC0F362E8@BESONE.corp.dom>
	<5d2828f1001140715m2aba7ed6n3cef077c64943e76@mail.gmail.com>
	<191c3a031001140837m7cffcdd5w71886d6c8ba1dafe@mail.gmail.com>
Message-ID: <2B0DD918-7E81-44B8-BF7F-0E9A0834A038@freeswitch.org>

And now with tcl bindings.  :P

/b

On Jan 14, 2010, at 10:37 AM, Anthony Minessale wrote:

> Don't forget that lua,perl,python,managed,java all share the same exact C++ source file with swig so the same exact api applies to all of the above.
> 
> On Thu, Jan 14, 2010 at 9:15 AM, Mike van Lammeren <mike at van.lammeren.net> wrote:
> Hi Claudio!
> 
> Thanks for the additions to the wiki! Every little bit helps.
> 
> I don't think I explained myself well, earlier. The point I was trying to make about the wiki is that rather than remove the section about "originate", it would be better to make an entry like "originate -- Does anyone know what this does?"
> 
> Mike van Lammeren

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From anthony.minessale at gmail.com  Thu Jan 14 08:58:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 10:58:24 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001141141.59831.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
	<191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
	<201001141141.59831.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031001140858i8ef1b09kf5d547914b5b8b68@mail.gmail.com>

r16310

Thank you for the apology.

On Thu, Jan 14, 2010 at 10:41 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Anthony, I apology, I didn't expect you'll be annoyed. As for the issue,
> the
> issue is not in "proxy_media" setting, same happens when both "proxy_media"
> and "bypass_media" are not set. The code does explicit leg A hangup in
> switch_ivr_bridge.c, line 513. The condition there is "if leg A is up and
> leg
> A is not in answered state, then hangup leg A", this prevents dialplan to
> continue execution on early media bridge termination.
>
> Issue when "bypass_media=true" is completely different, I'll open the issue
> on
> jira.
>
> On Thursday 14 January 2010, Anthony Minessale wrote:
> > Sergey,
> >
> > The bug you reference was closed because proxy_media mode by design
> send's
> > the B leg's codec to the A
> > LEG so if there is a failure condition there is no easy graceful way to
> > back out and try another call.
> > I will try to make it possible if yo post a bounty, I estimate a minimum
> of
> > $1000USD in consulting time.
> >
> > The other one I might look at once you have apologized.
> >
> > I can probably add some code to make the bridge exit without terminating
> > the A leg even when hangup_after_bridge=true in the case where the the B
> > leg is not answered but right now I am sort of annoyed with your attitude
> > in this thread.
> >
> >
> > On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin
> >
> > <sos at sokhapkin.dyndns.org>wrote:
> > > Case 1 (bypass_media is off) is already on jira,
> > > http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
> > > installation with latest trunk to reproduce case 2, when
> > > bypass_media=true and 10 seconds delay happens when 18X and then error
> > > are received on leg b.
> > >
> > > On Wednesday 13 January 2010, Michael Jerris wrote:
> > > > On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> > > > > Critical issues are when SIP error come after 18X provisional
> > > > > response.
> > > > >
> > > > > - if bypass_media is false then dialplan stops and leg a is
> > > > > explicitly hang up (switch_ivr_bridge.c, line 513).
> > > >
> > > >  behavior can be modified with continue_on_fail and
> hangup_after_bridge
> > > > channel vars, perhaps ignore_early_media as well
> > > >
> > > > > - if bypass_media is true, then dialplan continue, but there is 10
> > > > > seconds delay before next bridge application sends INVITE to
> gateway
> > > > > (
> > >
> > >
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
> > >
> > > > >4.html ) I didn't track down yet why this happens looking at FS
> > > > > sources.
> > > >
> > > > see response on that thread
> > > >
> > > > > Why I didn't open a bug on jira? Because FS behaves according to
> the
> > > > > design and specs :-) But not according to real world
> requirements...
> > > >
> > > > Really, people are trying to help you and your going to be snarky in
> > > > response?
> > > >
> > > > > On Wednesday 13 January 2010, Brian West wrote:
> > > > >> Can you elaborate on these "Critical" issues you seem to be
> having?
> > >
> > >  Why
> > >
> > > > >> aren't you opening a jira for them if they are that critical to
> your
> > > > >> needs?
> > > >
> > > > Mike
> > > >
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-user
> > > >s http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Thu Jan 14 09:00:09 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Jan 2010 11:00:09 -0600
Subject: [Freeswitch-users] mod_java is moving to unsupported on Jan 20th.
Message-ID: <E9807833-85C3-493C-BA11-F7C3586F3E38@freeswitch.org>

If nobody steps up to take over maintenance of mod_java by the 20th of January we'll be moving it to unsupported.  If you wish to take over maintenance please contact me off list ASAP.

Thanks,
Brian West
FreeSWITCH

From sos at sokhapkin.dyndns.org  Thu Jan 14 09:00:10 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Jan 2010 12:00:10 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <23BD1322-7882-4D43-80AB-27872096B4C9@avgs.ca>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141141.59831.sos@sokhapkin.dyndns.org>
	<23BD1322-7882-4D43-80AB-27872096B4C9@avgs.ca>
Message-ID: <201001141200.10315.sos@sokhapkin.dyndns.org>

"hangup_after_bridge=false" is the default setting.

On Thursday 14 January 2010, Mathieu Rene wrote:
> One way you could fix it is set "hangup_after_bridge=false" and then
> set "execute_on_answer=set\shangup_after_bridge=true". That will make
> it continue executing the dialplan atter a failure that came in after
> 183. Note that there is a side effect to this method, since audio was
> being relayed, you are forced to use the same codec on the 2nd
> carrier, best to tell them they are wrong failing a call after 183.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
> On 14-Jan-10, at 11:41 AM, Sergey Okhapkin wrote:
> > Anthony, I apology, I didn't expect you'll be annoyed. As for the
> > issue, the
> > issue is not in "proxy_media" setting, same happens when both
> > "proxy_media"
> > and "bypass_media" are not set. The code does explicit leg A hangup in
> > switch_ivr_bridge.c, line 513. The condition there is "if leg A is
> > up and leg
> > A is not in answered state, then hangup leg A", this prevents
> > dialplan to
> > continue execution on early media bridge termination.
> >
> > Issue when "bypass_media=true" is completely different, I'll open
> > the issue on
> > jira.
> >
> > On Thursday 14 January 2010, Anthony Minessale wrote:
> >> Sergey,
> >>
> >> The bug you reference was closed because proxy_media mode by design
> >> send's
> >> the B leg's codec to the A
> >> LEG so if there is a failure condition there is no easy graceful
> >> way to
> >> back out and try another call.
> >> I will try to make it possible if yo post a bounty, I estimate a
> >> minimum of
> >> $1000USD in consulting time.
> >>
> >> The other one I might look at once you have apologized.
> >>
> >> I can probably add some code to make the bridge exit without
> >> terminating
> >> the A leg even when hangup_after_bridge=true in the case where the
> >> the B
> >> leg is not answered but right now I am sort of annoyed with your
> >> attitude
> >> in this thread.
> >>
> >>
> >> On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin
> >>
> >> <sos at sokhapkin.dyndns.org>wrote:
> >>> Case 1 (bypass_media is off) is already on jira,
> >>> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
> >>> installation with latest trunk to reproduce case 2, when
> >>> bypass_media=true and 10 seconds delay happens when 18X and then
> >>> error
> >>> are received on leg b.
> >>>
> >>> On Wednesday 13 January 2010, Michael Jerris wrote:
> >>>> On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> >>>>> Critical issues are when SIP error come after 18X provisional
> >>>>> response.
> >>>>>
> >>>>> - if bypass_media is false then dialplan stops and leg a is
> >>>>> explicitly hang up (switch_ivr_bridge.c, line 513).
> >>>>
> >>>> behavior can be modified with continue_on_fail and
> >>>> hangup_after_bridge
> >>>> channel vars, perhaps ignore_early_media as well
> >>>>
> >>>>> - if bypass_media is true, then dialplan continue, but there is 10
> >>>>> seconds delay before next bridge application sends INVITE to
> >>>>> gateway
> >>>>> (
> >>>
> >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024
> >>>35
> >>>
> >>>>> 4.html ) I didn't track down yet why this happens looking at FS
> >>>>> sources.
> >>>>
> >>>> see response on that thread
> >>>>
> >>>>> Why I didn't open a bug on jira? Because FS behaves according to
> >>>>> the
> >>>>> design and specs :-) But not according to real world
> >>>>> requirements...
> >>>>
> >>>> Really, people are trying to help you and your going to be snarky
> >>>> in
> >>>> response?
> >>>>
> >>>>> On Wednesday 13 January 2010, Brian West wrote:
> >>>>>> Can you elaborate on these "Critical" issues you seem to be
> >>>>>> having?
> >>>
> >>> Why
> >>>
> >>>>>> aren't you opening a jira for them if they are that critical to
> >>>>>> your
> >>>>>> needs?
> >>>>
> >>>> Mike
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> >>>>r s http://www.freeswitch.org
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> >>>s http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From freeswitch at aastral.net  Thu Jan 14 09:20:23 2010
From: freeswitch at aastral.net (Bill W)
Date: Thu, 14 Jan 2010 12:20:23 -0500
Subject: [Freeswitch-users] Question about a1-hash
In-Reply-To: <5d2828f1001140827w386fe2acw478ae378b0ecfe53@mail.gmail.com>
References: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>	<4B4F430A.8080006@aastral.net>
	<5d2828f1001140827w386fe2acw478ae378b0ecfe53@mail.gmail.com>
Message-ID: <4B4F5257.6080508@aastral.net>

Hey Mike/Brian,

Okay, I'm missing something here.  Sure, you can calculate the response, 
but how are you going to validate that against what is in the database?

Thanks,
Bill





From mike at jerris.com  Thu Jan 14 09:20:31 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Jan 2010 12:20:31 -0500
Subject: [Freeswitch-users] playback and play_and_get_digits strange
	misunderstanding
In-Reply-To: <6d698be09281bb173cf0de33c1c2e44b@thom.fr.eu.org>
References: <6d698be09281bb173cf0de33c1c2e44b@thom.fr.eu.org>
Message-ID: <816ED94F-2501-46F0-ACE6-AD82011F8466@jerris.com>

Could you try this with latest trunk as well for comparison, I just changed the way the sounds base directory is set and I want to make sure I did not mess anything up.

Mike

On Jan 14, 2010, at 3:23 AM, Fran?ois Legal wrote:

> So I could (kind of) solve this by myself. There is in vars.xml the variable ${sound_prefix}. I did set it properly to my french sound path and then it worked.
> 
>  
> However, for the sake of discussion, I did try this with 1.0.5pre8 and the result was different :
> 
> with the extension
> 
>   <extension name="Disable Ringing on FXS">
>    <condition field="destination_number" expression="^#1234$">
>     <action application="set_global" data="dont_ring_fxs=true"/>
>     <action application="answer"/>
>     <action application="sleep" data="1000"/>
>     <action application="playback" data="misc/ringing_disabled.wav"/>
>     <action application="hangup" data="normal_clearing"/>
>    </condition>
>   </extension>
> 
> FS was trying to play the file ${FREESWITCH_PATH}/sounds/en/us/callie/misc/ringing_disabled.wav
> 
> with the extension
> 
>   <extension name="Disable Ringing on FXS">
>    <condition field="destination_number" expression="^#1234$">
>     <action application="set" data="default_language=fr"/>
>     <action application="set_global" data="dont_ring_fxs=true"/>
>     <action application="answer"/>
>     <action application="sleep" data="1000"/>
>     <action application="playback" data="misc/ringing_disabled.wav"/>
>     <action application="hangup" data="normal_clearing"/>
>    </condition>
>   </extension>
> 
> FS was trying to load ${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/ringing_disabled.wav
> 
> Now with the latest snapshot, with either one of the 2 mentionned extensions, FS tries to play the file in ${sound_prefix}/filepath/${codec_bit_rate}/filename whereas 1.0.5pre8 did not add the codec_bit_rate in the path but took care of the default_language variable.
> 
>  
> Fran?ois
> 
>  
> On Wed, 13 Jan 2010 16:32:16 +0100, Fran?ois Legal wrote:
> 
> Hello,
> 
>  
> trying to make so dialplan extensions that use the playback and play_and_get_digits applications, but I'm having trouble with the file name specification.
> 
>  
> The files I want to play are in the french language (fr/fr/julie as configured in lang/fr/fr.xml)
> 
> My extension is as follows :
> 
> 
>   <extension name="Disable Ringing on FXS">
>    <condition field="destination_number" expression="^#1234$">
>     <action application="set" data="default_language=fr"/>
>     <action application="set_global" data="dont_ring_fxs=true"/>
>     <action application="answer"/>
>     <action application="sleep" data="1000"/>
>     <action application="playback" data="misc/ringing_disabled.wav"/>
>     <action application="hangup" data="normal_clearing"/>
>    </condition>
>   </extension>
> 
> The channel is using a bit rate of 8000 Hz, so by the set default_language=fr I would expect freeswitch to playback the file at ${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/8000/ringing_disabled.wav whereas it tries to playback the file at ${FREESWITCH_PATH}/sounds/en/us/callie/misc/8000/ringing_disabled.wav
> 
> I have the same with play_and_get_digits application.
> 
>  
> What am I doing wrong ?
> 
>  
> Fran?ois
> 
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Thu Jan 14 09:21:35 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Jan 2010 12:21:35 -0500
Subject: [Freeswitch-users] sip trunk question: why call throughexternal
	profile is challenged?
In-Reply-To: <20100114082640.2C5D311F32@mail.nstel.ru>
References: <20100114082640.2C5D311F32@mail.nstel.ru>
Message-ID: <AD524CC0-726B-46D3-A22C-0B1F44C9B899@jerris.com>

if you look in the sample configs for the words blind and auth you will find all these settings, also you can setup acls for ip auth to not challenge.

Mike

On Jan 14, 2010, at 3:26 AM, Nikolay Kondratyev wrote:

> Mike, thanks for the reply.
>  
> Mmm? looks like I need more detailed instructions where to dig?
> Is there a way to turn off ?challenging? completely?
> I thought that <param name="auth-calls" value="false"/> should do it, but alas?
> By the way should this parameter be visible in either ?sofia status profile external? or ?sofia status gateway sipx4.lab.nstel.ru? ? I don?t see it?
>  
> I attached traces of failed and successful calls.
>  
> Thanks and regards,
> Nikolay.
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Wednesday, January 13, 2010 8:30 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] sip trunk question: why call throughexternal profile is challenged?
>  
> Look at how sipx sets up the users when they build the extensions and such for conferences, there was something special here, but I can't recall what.
>  
> Mike
>  
> On Jan 13, 2010, at 9:10 AM, Nikolay Kondratyev wrote:
> 
> 
> Hi all!
>  
> I?m brand new to FreeSwitch, but have some experience with SipX.
> Our company is evaluating FS.
> For test purposes I set up FS on a virtual machine (vmware esxi). I use CentOS.
> The FS version I use is 1.0.5-20100110-0400.
>  
> I have a question regarding sip trunk between FS and SipX.
> I created the following GW in external profile:
> [freeswitch at freeswitch external]$ cat sipx-lab.xml | grep -v '<!--'
> <include>
>   <gateway name="sipx4.lab.nstel.ru">
>   <param name="username" value="zxcv"/>
>   <param name="password" value="2007"/>
>   <param name="register" value="false"/>
>   </gateway>
> </include>
>  
> External.xml file is not modified after installation.
>  
> I see this gateway via fs_cli:
> freeswitch at internal> sofia status gateway sipx4.lab.nstel.ru
> =================================================================================================
> Name            sipx4.lab.nstel.ru
> Profile         external
> Scheme          Digest
> Realm           sipx4.lab.nstel.ru
> Username        zxcv
> Password        yes
> From            <sip:zxcv at sipx4.lab.nstel.ru;transport=udp>
> Contact         <sip:gw+sipx4.lab.nstel.ru at 172.23.22.49:5080;transport=udp;gw=sipx4.lab.nstel.ru>
> Exten           zxcv
> To              sip:zxcv at sipx4.lab.nstel.ru
> Proxy           sip:sipx4.lab.nstel.ru
> Context         public
> Expires         3600
> Freq            3600
> Ping            0
> PingFreq        0
> PingState       0/0/0
> State           NOREG
> Status          UP
> CallsIN         0
> CallsOUT        0
> =================================================================================================
>  
> I created new FS extension 2853. I registers (xlite) and I can call it.
>  
> Now I want to call FS user from sipx.
>  
> From the sipx side one can configure link to FS differently. There are two options:
> 1. Call FS directly trough sipxproxy (the core part of sipx works as sip proxy, not as b2bua)
> 2. Call trough embedded b2bua, named sipxbridge.
>  
> When a call is going trough sipxbridge, it is successfully landed at FS extension 2853.
> But when a call is going from the sipxproxy, it is challenged with Status 407 ?proxy authentication required?, and then call fails.
> (I?m not sure if sipx should handle this challenge, but this is separate question for the sipx forum).
> At the default log level I see nothing in the freeswitch.log.
>  
> So the question is why one call through external profile is being challenged while the other is not?
> I suspect that the reason is in the difference in the two Invite messages:
>  
> Here is challenged Invite:
> INVITE sip:2853 at fs.lab.nstel.ru:5080;transport=udp SIP/2.0
> Record-Route: <sip:172.23.12.104:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EOTRiZjI3MjgtYWMxNzBjZTktMTNjNC0yZjZkZWMtNjNkNmUwN2UtMmY2ZGVj.900_ntap%2Aid%7EMjIyNjAtMQ%60%60%213ba6d85f6946c4c6001bee1d3b54474f>
> From: "testphone3857"<sip:3857 at lab.nstel.ru>;tag=94bf2728-ac170ce9-13c4-2f6dec-63d6e07e-2f6dec
> To: <sip:2853 at lab.nstel.ru>
> Call-Id: 94bed2a0-ac170ce9-13c4-2f6dec-68ffdb81-2f6dec at lab.nstel.ru
> Cseq: 2 INVITE
> Via: SIP/2.0/UDP 172.23.12.104;branch=z9hG4bK-sipXecs-000eaf48a36fe4029c7cde004a6f44424847
> Via: SIP/2.0/TCP 172.23.12.104;branch=z9hG4bK-sipXecs-000bf3008cfa3f3e70470789c75232ba9499~ac7fd729330fd563f83cacd941311e75;id=22260-1
> Via: SIP/2.0/UDP 172.23.12.233:5060;branch=z9hG4bK-2f6dec-b9456412-7ad07784
> Max-Forwards: 18
> Supported: replaces
> User-Agent: LG-Nortel LIP 6812 v1.2.38sp SN/00405A18B634
> Contact: <sip:3857 at 172.23.12.233:5060;x-sipX-nonat>
> Proxy-Authorization: Digest username="3857",realm="lab.nstel.ru",nonce="e2152722611af1bbe59f3a8eb31b8eb84b4db2ca",uri="sip:2853 at lab.nstel.ru",response="a09c34fbd897783c9b9af50bd044ccaa",algorithm=MD5
> Content-Type: application/sdp
> Content-Length: 301
> Date: Wed, 13 Jan 2010 11:47:22 GMT
> Expires: 60
> X-Sipx-Handled: X172.23.12.104-81.211.30.104
>  
> v=0
> o=LGEIPP 16246 16247 IN IP4 172.23.12.233
> s=SIP Call
> c=IN IP4 172.23.12.233
> t=0 0
> m=audio 23008 RTP/AVP 0 8 18 4 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:18 annexb=no
> a=fmtp:101 0-11
> a=sendrecv
>  
>  
> And here is successful Invite:
> INVITE sip:2853 at fs.lab.nstel.ru;user=phone SIP/2.0
> Call-ID: 94bec3b8-ac170ce9-13c4-2f6d47-2fee09d0-2f6d47 at lab.nstel.ru.0
> CSeq: 1 INVITE
> From: "testphone3857" <sip:3857 at 172.23.12.104>;tag=816640414244159961
> To: <sip:2853 at fs.lab.nstel.ru;user=phone>
> Via: SIP/2.0/UDP 172.23.12.104:5080;branch=z9hG4bK885dde2c8680f5315845cd3350b8b605373534
> Max-Forwards: 70
> User-Agent: sipXecs/4.0.2 sipXecs/sipxbridge (Linux)
> P-Asserted-Identity: <sip:1004 at 172.23.12.104>
> Contact: <sip:1004 at 172.23.12.104:5080;transport=udp>
> Route: <sip:172.23.22.49:5080;transport=udp;lr>
> Session-Expires: 1800;refresher=uac
> Allow: INVITE,BYE,ACK,CANCEL,OPTIONS
> Content-Type: application/sdp
> Content-Length: 315
>  
> v=0
> o=sipxbridge 6640787141824452741 1 IN IP4 172.23.12.104
> s=SIP Call
> c=IN IP4 172.23.12.104
> t=0 0
> m=audio 30248 RTP/AVP 0 8 18 4 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:18 annexb=no
> a=fmtp:101 0-11
> a=sendrecv
>  
> Can somebody please if it is a FS configuration problem or a software problem or is it a problem on the sipx side?
>  
> Thanks in advance,
> Nikolay.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
>  
> <call_from_sipx_via_unmanaged_gw.pcap><call_from_sipx_via_siptrunk_sip-only.pcap>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Thu Jan 14 09:21:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 11:21:43 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001141200.10315.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141141.59831.sos@sokhapkin.dyndns.org>
	<23BD1322-7882-4D43-80AB-27872096B4C9@avgs.ca>
	<201001141200.10315.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031001140921n5de0a9eakeb3144aba91c78d5@mail.gmail.com>

ok both methods work now.
your jira was fixed in r16311 and the non bypass equivalent was fixed in the
previous commit

On Thu, Jan 14, 2010 at 11:00 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> "hangup_after_bridge=false" is the default setting.
>
> On Thursday 14 January 2010, Mathieu Rene wrote:
> > One way you could fix it is set "hangup_after_bridge=false" and then
> > set "execute_on_answer=set\shangup_after_bridge=true". That will make
> > it continue executing the dialplan atter a failure that came in after
> > 183. Note that there is a side effect to this method, since audio was
> > being relayed, you are forced to use the same codec on the 2nd
> > carrier, best to tell them they are wrong failing a call after 183.
> >
> > Mathieu Rene
> > Avant-Garde Solutions Inc
> > Office: + 1 (514) 664-1044 x100
> > Cell: +1 (514) 664-1044 x200
> > mrene at avgs.ca
> >
> > On 14-Jan-10, at 11:41 AM, Sergey Okhapkin wrote:
> > > Anthony, I apology, I didn't expect you'll be annoyed. As for the
> > > issue, the
> > > issue is not in "proxy_media" setting, same happens when both
> > > "proxy_media"
> > > and "bypass_media" are not set. The code does explicit leg A hangup in
> > > switch_ivr_bridge.c, line 513. The condition there is "if leg A is
> > > up and leg
> > > A is not in answered state, then hangup leg A", this prevents
> > > dialplan to
> > > continue execution on early media bridge termination.
> > >
> > > Issue when "bypass_media=true" is completely different, I'll open
> > > the issue on
> > > jira.
> > >
> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> > >> Sergey,
> > >>
> > >> The bug you reference was closed because proxy_media mode by design
> > >> send's
> > >> the B leg's codec to the A
> > >> LEG so if there is a failure condition there is no easy graceful
> > >> way to
> > >> back out and try another call.
> > >> I will try to make it possible if yo post a bounty, I estimate a
> > >> minimum of
> > >> $1000USD in consulting time.
> > >>
> > >> The other one I might look at once you have apologized.
> > >>
> > >> I can probably add some code to make the bridge exit without
> > >> terminating
> > >> the A leg even when hangup_after_bridge=true in the case where the
> > >> the B
> > >> leg is not answered but right now I am sort of annoyed with your
> > >> attitude
> > >> in this thread.
> > >>
> > >>
> > >> On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin
> > >>
> > >> <sos at sokhapkin.dyndns.org>wrote:
> > >>> Case 1 (bypass_media is off) is already on jira,
> > >>> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
> > >>> installation with latest trunk to reproduce case 2, when
> > >>> bypass_media=true and 10 seconds delay happens when 18X and then
> > >>> error
> > >>> are received on leg b.
> > >>>
> > >>> On Wednesday 13 January 2010, Michael Jerris wrote:
> > >>>> On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> > >>>>> Critical issues are when SIP error come after 18X provisional
> > >>>>> response.
> > >>>>>
> > >>>>> - if bypass_media is false then dialplan stops and leg a is
> > >>>>> explicitly hang up (switch_ivr_bridge.c, line 513).
> > >>>>
> > >>>> behavior can be modified with continue_on_fail and
> > >>>> hangup_after_bridge
> > >>>> channel vars, perhaps ignore_early_media as well
> > >>>>
> > >>>>> - if bypass_media is true, then dialplan continue, but there is 10
> > >>>>> seconds delay before next bridge application sends INVITE to
> > >>>>> gateway
> > >>>>> (
> > >>>
> > >>>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024
> > >>>35
> > >>>
> > >>>>> 4.html ) I didn't track down yet why this happens looking at FS
> > >>>>> sources.
> > >>>>
> > >>>> see response on that thread
> > >>>>
> > >>>>> Why I didn't open a bug on jira? Because FS behaves according to
> > >>>>> the
> > >>>>> design and specs :-) But not according to real world
> > >>>>> requirements...
> > >>>>
> > >>>> Really, people are trying to help you and your going to be snarky
> > >>>> in
> > >>>> response?
> > >>>>
> > >>>>> On Wednesday 13 January 2010, Brian West wrote:
> > >>>>>> Can you elaborate on these "Critical" issues you seem to be
> > >>>>>> having?
> > >>>
> > >>> Why
> > >>>
> > >>>>>> aren't you opening a jira for them if they are that critical to
> > >>>>>> your
> > >>>>>> needs?
> > >>>>
> > >>>> Mike
> > >>>>
> > >>>>
> > >>>> _______________________________________________
> > >>>> FreeSWITCH-users mailing list
> > >>>> FreeSWITCH-users at lists.freeswitch.org
> > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-use
> > >>>>r s http://www.freeswitch.org
> > >>>
> > >>> _______________________________________________
> > >>> FreeSWITCH-users mailing list
> > >>> FreeSWITCH-users at lists.freeswitch.org
> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >>>s http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Thu Jan 14 09:23:21 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 11:23:21 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001140841n603ddc45n972b3aba83600d01@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
	<191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
	<191c3a031001140826q2d014484v8807bf3b98b78f63@mail.gmail.com>
	<9853f4ff1001140841n603ddc45n972b3aba83600d01@mail.gmail.com>
Message-ID: <191c3a031001140923i72afe059ja18c6911c60c81e2@mail.gmail.com>

David I already understood your request.  Even what you consider a
compromise is a lot of work.
You really should post a bounty for something this exotic.


On Thu, Jan 14, 2010 at 10:41 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Anthony,
>
>      Thanks. I understand it's complicated. Another option is to be able to
> configure whether or not to discard all other B-legs on receiving media but
> on receiving an final code like 200. This way we will still get the first
> media that arrived but provide an answer on the actual channel that provided
> the 200.
>
> My request comes from 2 sides, one commercial and the other personal. For
> the commercial I just need to now whether it is possible by some tweaking so
> the functionality can be offered.
>
> The other, I've been involved in FS as a user from the very beginning
> (although I was absent for some time), and I'd like to help in the
> development of the project by letting the community know what's out there
> commercially speaking.
>
>
> Hope you can help.
>
> David
>
> On Thu, Jan 14, 2010 at 5:26 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> David,
>>
>> Your request is somewhat challenging because we have to make sure
>> FreeSWITCH is agnostic in protocols and that aside, we also have a very
>> complicated and feature-rich originate API that does not currently support
>> sending audio from one leg to the A leg while it's trying to call 10 other B
>> legs.  Please try to understand that this request is a very special side
>> case and we would have to do many hours of work to make it possible.  I am
>> not sure if you are simply asking if its possible or if you want us to
>> implement it for you but I am afraid it would fall under commercial support
>> to undertake that unique of a feature that only helps a very small
>> percentage of our user base considering all the work we already have to do
>> every day.
>>
>>
>>
>>
>> On Thu, Jan 14, 2010 at 10:22 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> Sergey,
>>>
>>> The bug you reference was closed because proxy_media mode by design
>>> send's the B leg's codec to the A
>>> LEG so if there is a failure condition there is no easy graceful way to
>>> back out and try another call.
>>> I will try to make it possible if yo post a bounty, I estimate a minimum
>>> of $1000USD in consulting time.
>>>
>>> The other one I might look at once you have apologized.
>>>
>>> I can probably add some code to make the bridge exit without terminating
>>> the A leg even when hangup_after_bridge=true in the case where the the B leg
>>> is not answered but right now I am sort of annoyed with your attitude in
>>> this thread.
>>>
>>>
>>>
>>> On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin <
>>> sos at sokhapkin.dyndns.org> wrote:
>>>
>>>> Case 1 (bypass_media is off) is already on jira,
>>>> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
>>>> installation with latest trunk to reproduce case 2, when
>>>> bypass_media=true
>>>> and 10 seconds delay happens when 18X and then error are received on leg
>>>> b.
>>>>
>>>> On Wednesday 13 January 2010, Michael Jerris wrote:
>>>> > On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
>>>> > > Critical issues are when SIP error come after 18X provisional
>>>> response.
>>>> > >
>>>> > > - if bypass_media is false then dialplan stops and leg a is
>>>> explicitly
>>>> > > hang up (switch_ivr_bridge.c, line 513).
>>>> >
>>>> >  behavior can be modified with continue_on_fail and
>>>> hangup_after_bridge
>>>> > channel vars, perhaps ignore_early_media as well
>>>> >
>>>> > > - if bypass_media is true, then dialplan continue, but there is 10
>>>> > > seconds delay before next bridge application sends INVITE to gateway
>>>> (
>>>> > >
>>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
>>>> > >4.html ) I didn't track down yet why this happens looking at FS
>>>> sources.
>>>> >
>>>> > see response on that thread
>>>> >
>>>> > > Why I didn't open a bug on jira? Because FS behaves according to the
>>>> > > design and specs :-) But not according to real world requirements...
>>>> >
>>>> > Really, people are trying to help you and your going to be snarky in
>>>> > response?
>>>> >
>>>> > > On Wednesday 13 January 2010, Brian West wrote:
>>>> > >> Can you elaborate on these "Critical" issues you seem to be having?
>>>>  Why
>>>> > >> aren't you opening a jira for them if they are that critical to
>>>> your
>>>> > >> needs?
>>>> >
>>>> > Mike
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From Claudio.Cavalera at italtel.it  Thu Jan 14 09:25:08 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Thu, 14 Jan 2010 18:25:08 +0100
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <5d2828f1001140715m2aba7ed6n3cef077c64943e76@mail.gmail.com>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F362EA@BESONE.corp.dom>

We are telling the same thing Mike, I've also written to elaborate the
"originate" entry on mod_lua wiki.
We could take examples from here maybe:
http://wiki.freeswitch.org/wiki/Session_originate
 
ciao,
claudio


________________________________

	From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike
van Lammeren
	Sent: Thursday, January 14, 2010 4:16 PM
	To: freeswitch-users at lists.freeswitch.org
	Subject: Re: [Freeswitch-users] playing with sessions in lua
	
	
	Hi Claudio! 

	Thanks for the additions to the wiki! Every little bit helps.

	I don't think I explained myself well, earlier. The point I was
trying to make about the wiki is that rather than remove the section
about "originate", it would be better to make an entry like "originate
-- Does anyone know what this does?"

	Mike van Lammeren


	On Thu, Jan 14, 2010 at 6:23 AM, Cavalera Claudio Luigi
<Claudio.Cavalera at italtel.it> wrote:
	

		Hi Mike,
		in fact i've completed that page with the list of
available  session functions.
		 
		I've not removed "session:originate" yet, but it would
be better if someone could provide an example in order to write an
example in the wiki.
		I've added this valuable example also with the help of
rupa
	
http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.Session
		 
		I would like to write something also about api_on_answer
to use an api instead of a dialplan application.
		 
		BRs,
		Claudio


________________________________

			From:
freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike
van Lammeren
			Sent: Wednesday, January 13, 2010 8:30 PM 

			To: freeswitch-users at lists.freeswitch.org
			Subject: Re: [Freeswitch-users] playing with
sessions in lua
			

			Hello! 

			Before you remove "session:originate" from the
wiki, you should take a look at this:
	
http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocument
ed_Session_Functions.3F

	
<http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocumen
ted_Session_Functions.3F> There is, in fact, a function called
"originate".

			Mike van Lammeren


			On Wed, Jan 13, 2010 at 5:22 AM, Cavalera
Claudio Luigi <Claudio.Cavalera at italtel.it> wrote:
			

				Thanks a lot Anthony,
				some comments inline (and please forgive
me for my broken email client).
				 
				
				>>  example1: Consider this simple lua
script in which i create two sessions:
				
				 >>  api = freeswitch.API();
				 >>  api:execute("originate",
"sofia/internal/1001%192.168.1.1 &park");
				 >>  api:execute("originate",
"sofia/internal/1002%192.168.1.1 &park");
				
				 >  capture the output from api:execute
the uuid is in there
				
				
				Thx a lot,
				this was one piece i was missing
although it's already on the wiki here:
	
http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls 
				


				 > because lua calls it
freeswitch.bridge
				
				 > session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
				 > session2 =
freeswitch.Session("sofia/internal/1002%192.168.1.1", session1);
				 > freeswitch.bridge(session1,
session2); 
				 
				good to now, there isn't any example of
freeswitch.bridge in the wiki and i would like to add one.
				Where I could find the full api of
				freeswitch.Session( ) ?
				because I've seen this working also
without "session1" in the second line:
				
				session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
				
				session2 =
freeswitch.Session("sofia/internal/1002%192.168.1.1");
				freeswitch.bridge(session1, session2); 
				
				 also is there any difference between
freeswitch.bridge and freeswitch.execute(uuid_bridge ...) ?
				 
				 >> example3: yet another possibility
				 >> local session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
				 >> session1:originate(session2,
"sofia/internal/1002%192.168.1.1", 1000);
				 >> but it does not work either.
				
				 > The above is gibberish try:
				 > local session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
				 > session1:execute("bridge",
"sofia/internal/1002%192.168.1.1"); 
				 
				Okay i will report also this bridge
example on the wiki which was missing.
				But does session:originate make sense in
some cases or not? Otherwise i'm going to remove this line on the wiki
	
http://wiki.freeswitch.org/wiki/Mod_lua#session:originate
<http://wiki.freeswitch.org/wiki/Mod_lua#session:originate> 
				 
				Thanks,
				Claudio

				 

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From brian at freeswitch.org  Thu Jan 14 09:30:55 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Jan 2010 11:30:55 -0600
Subject: [Freeswitch-users] Question about a1-hash
In-Reply-To: <4B4F5257.6080508@aastral.net>
References: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>	<4B4F430A.8080006@aastral.net>
	<5d2828f1001140827w386fe2acw478ae378b0ecfe53@mail.gmail.com>
	<4B4F5257.6080508@aastral.net>
Message-ID: <41473BE6-E347-4FCD-AD15-9964265C53EB@freeswitch.org>

You take what is in the data base and create an a1-hash if you have the parts to create it.. if not your SOL.

/b

On Jan 14, 2010, at 11:20 AM, Bill W wrote:

> Hey Mike/Brian,
> 
> Okay, I'm missing something here.  Sure, you can calculate the response, 
> but how are you going to validate that against what is in the database?
> 
> Thanks,
> Bill



From mike at van.lammeren.net  Thu Jan 14 09:36:04 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 14 Jan 2010 12:36:04 -0500
Subject: [Freeswitch-users] Question about a1-hash
In-Reply-To: <4B4F5257.6080508@aastral.net>
References: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>
	<4B4F430A.8080006@aastral.net>
	<5d2828f1001140827w386fe2acw478ae378b0ecfe53@mail.gmail.com>
	<4B4F5257.6080508@aastral.net>
Message-ID: <5d2828f1001140936t62dbebbdg85489d203e4f3295@mail.gmail.com>

Oh, I still have to go with my solution of using the MD5 hash of the
password for the phones. Since I cannot change the database for other
reasons, I am still unable to generate the a1-hash. Like FreeSWITCH, my
script doesn't know the password either!

All the same, I'm glad that FreeSWITCH uses digest authentication. It is a
good way to implement authentication.


Mike van Lammeren

On Thu, Jan 14, 2010 at 12:20 PM, Bill W <freeswitch at aastral.net> wrote:

> Hey Mike/Brian,
>
> Okay, I'm missing something here.  Sure, you can calculate the response,
> but how are you going to validate that against what is in the database?
>
> Thanks,
> Bill
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From djbinter at yahoo.com  Thu Jan 14 09:39:19 2010
From: djbinter at yahoo.com (DJB)
Date: Thu, 14 Jan 2010 09:39:19 -0800 (PST)
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <43692088-738A-475C-8D60-E40D296ECEB1@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
	<201001132326.41354.sos@sokhapkin.dyndns.org>
	<52E8BE9E-B6D6-411C-9080-697AF24017B4@jerris.com>
	<191c3a031001132114s18ed2baw2ff802dbd615fe01@mail.gmail.com>
	<191c3a031001132115u2d88bab7p7cbb27ab1eaad466@mail.gmail.com>
	<9853f4ff1001140226y2b035f37rfcafb03190435ec1@mail.gmail.com>
	<43692088-738A-475C-8D60-E40D296ECEB1@jerris.com>
Message-ID: <407935.47740.qm@web37505.mail.mud.yahoo.com>

What about sending Sip 183 with SDP (no 200OK), so that your customers can hear recordings?

djbinter 




________________________________
From: Michael Jerris <mike at jerris.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, January 14, 2010 8:12:59 AM
Subject: Re: [Freeswitch-users] multiple leg and multiple rtp

Okay, so that is a very interesting use case.   Seems a bizarre way to do this for the carrier, but interesting none the less.  I'd say the chances of actually muxing the early media is small, what might be possible would be something to say which b legs media to pass along to the a leg.  I have not looked at all at how complicated this is, it will be down deep somewhere in switch_ivr_originate code around where we if for ignore_early_media.  This code is pretty complex, I can't say that we will ever actually add this functionality, but at least now I won't blow it off as complete nonsense.

Mike





On Jan 14, 2010, at 5:26 AM, David Villasmil wrote:

Hello again,
>
>Using ingore early media only ignores ALL media, that's not what I need. At least in europe i've seen it many times, MNOs provide a service with which you can have a song played to the caller while the call is connecting to your cell phone. This is basically what I'm trying to achieve.
>
>The content provider is in possesions of the media and it require us to fork the call and send a SIP INVITE to the on one leg and the call to the destination number on another leg. they will only provide a progress, no answer.
>
>I CAN do it locally by playing the file as a custom ringback but that's not the standard in terms of commercial use of the content.
>
>Is there any way to modify the behaviour of fs when i receives media? Let's say i.e. it doesn't drop the other leg, but provides the first early media it receives and just wait for some channel to answer? I will not have both media but it would work.
>
>thanks
>
>David
>
>
>On Thu, Jan 14, 2010 at 6:15 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
>Perhaps best not to help him anymore without an apology for the snap judgement and comparison to asterisk clearly designed to push our buttons.
>>>>
>>We'll be here when you realize we were trying to help you but I can't promise we will still have paitence......
>>
>>
>>On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike at jerris.com> wrote:
>>>
>>>You already are running on trunk.
>>>
>>>>>>Mike
>>>
>>>>>>On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote:
>>>
>>>> I run FreeSWITCH Version 1.0.5pre10 (16012M...
>>>_______________________________________________
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists...
>>_______________________________________________
>>>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>_______________________________________________
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>



      
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From vgribanovsky at rnd.stcnet.ru  Thu Jan 14 08:58:14 2010
From: vgribanovsky at rnd.stcnet.ru (Vadim Gribanovsky)
Date: Thu, 14 Jan 2010 19:58:14 +0300
Subject: [Freeswitch-users] FS can't direct call to SBC
Message-ID: <1263488294.2918.26.camel@localhost.localdomain>

Hi all!
        
        Let me ask FS experts and users to help to resolve following
        problem:
        
        /////////////////////////////////////////////////////////////////////////////////////////////////
        1.VOIP test environment:
        =>local (10.10.10.235/24) FS(v1.0.4) installation + 
        =>local IP-Phone (GXP2020) +
        =>SBC (acts as stateless proxy) provides secure access (TLS
        +SRTP) to FS
        for +
        =>external NATed IP-PHONE (PJSIPua)
        as depicted below:
        PHONE(10.10.10.31) <-udp-> FS(10.10.10.235) <-tcp-> (Local
        10.10.10.238)SBC(Public P.U.B.LS)<-tls-> INTERNET<-tls->
        NAT(Public
        P.U.B.LN) <-tls-> PHONE (10.10.10.21)
        
        2.Problem:
        => local IP-Phone get registration to FS, -OK!
        => external NATed IP-PHONE get registration to FS via SBC, -OK!
        => FS try to send NOTIFY request to external NATed IP-PHONE
        using
        [sip_contact_host:port] record instead
        [sip_received_ip:port](SBC) -
        FAIL!
        => external NATed IP-PHONE calls local IP-Phone - OK!
        => local IP-Phone couldn't call external NATed IP-PHONE - FS
        redirect to
        voicemail - FAIL!
        
        3. Logs:
        
        =>FS CLI
        API CALL [sofia(status)] output:
                             Name          Type
        Data      State
        =================================================================================================
                     vfsp_profile       profile
        sip:mod_sofia at 10.10.10.235:5090      RUNNING (0)
                          vfsp_gw       gateway
        sip:FS_VFSP at 10.10.10.238      NOREG
                         internal       profile
        sip:mod_sofia at 10.10.10.235:5060      RUNNING (0)
                    internal-ipv6       profile
        sip:mod_sofia@[::1]:5060      RUNNING (0)
                     10.10.10.235         alias
        internal      ALIASED
        =================================================================================================
        3 profiles 1 alias
        
        API CALL [sofia(status profile internal)] output:
        =================================================================================================
        Name                    internal
        Domain Name             N/A
        DBName                  sofia_reg_internal
        Pres Hosts       
        Dialplan                XML
        Context                 public
        Challenge Realm         auto_from
        RTP-IP                  10.10.10.235
        Ext-RTP-IP              10.10.10.235
        SIP-IP                  10.10.10.235
        Ext-SIP-IP              10.10.10.235
        URL                     sip:mod_sofia at 10.10.10.235:5060
        BIND-URL                sip:mod_sofia at 10.10.10.235:5060
        HOLD-MUSIC              local_stream://moh
        OUTBOUND-PROXY          N/A
        CODECS
        G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
        TEL-EVENT               101
        DTMF-MODE               rfc2833
        CNG                     13
        SESSION-TO              0
        MAX-DIALOG              0
        NOMEDIA                 false
        LATE-NEG                false
        PROXY-MEDIA             false
        AGGRESSIVENAT           false
        STUN-ENABLED            true
        STUN-AUTO-DISABLE       false
        CALLS-IN                3
        FAILED-CALLS-IN         0
        CALLS-OUT               0
        FAILED-CALLS-OUT        0
        
        Registrations:
        =================================================================================================
        Call-ID:        db639b76d2884388 at 10.10.10.31
        User:           1010 at 10.10.10.235
        Contact:        "user"
        <sip:1010 at 10.10.10.31:5060;transport=udp;user=phone>
        Agent:          Grandstream GXP2020 1.2.1.4
        Status:         Registered(UDP)(unknown) EXP(2010-01-14
        16:00:09)
        Host:           aq
        IP:             10.10.10.31
        Port:           5060
        Auth-User:      1010
        Auth-Realm:     10.10.10.235
        
        =================================================================================================
        
        
        API CALL [sofia(status profile vfsp_profile)] output:
        =================================================================================================
        Name                    vfsp_profile
        Domain Name             N/A
        DBName                  sofia_reg_vfsp_profile
        Pres Hosts       
        Dialplan                XML
        Context                 public
        Challenge Realm         auto_to
        RTP-IP                  10.10.10.235
        Ext-RTP-IP              stun:stun.freeswitch.org
        SIP-IP                  10.10.10.235
        Ext-SIP-IP              P.U.B.L
        URL                     sip:mod_sofia at 10.10.10.235:5090
        BIND-URL                sip:mod_sofia at 10.10.10.235:5090
        HOLD-MUSIC              local_stream://moh
        OUTBOUND-PROXY          N/A
        CODECS                  PCMU,PCMA,GSM
        TEL-EVENT               101
        DTMF-MODE               rfc2833
        CNG                     13
        SESSION-TO              0
        MAX-DIALOG              0
        NOMEDIA                 false
        LATE-NEG                false
        PROXY-MEDIA             false
        AGGRESSIVENAT           true
        STUN-ENABLED            true
        STUN-AUTO-DISABLE       false
        CALLS-IN                0
        FAILED-CALLS-IN         0
        CALLS-OUT               0
        FAILED-CALLS-OUT        0
        
        Registrations:
        =================================================================================================
        Call-ID:        6197eb4f-72fa-428f-8c0e-5c5551d185e2
        User:           1007 at 10.10.10.235
        Contact:        "user" <sip:1007 at P.U.B.LN:43459;transport=tcp>
        Agent:          PJSUA v1.0.3/i686-pc-linux-gnu
        Status:         Registered(TCP)(unknown) EXP(2010-01-14
        16:04:46)
        Host:           aq
        IP:             10.10.10.238
        Port:           38782
        Auth-User:      1007
        Auth-Realm:     10.10.10.235
        
        =================================================================================================
        
        => sniffer
        pkt1. SBC -> FS
        REGISTER sip:10.10.10.235;transport=tcp SIP/2.0
        Record-Route: <sip:10.10.10.238:5060;transport=tcp;lr>
        Via: SIP/2.0/tcp
        10.10.10.238:36707;rport;branch=z9hG4bKPj66cc4b809a6bb1510af258991c1f0587
        Via: SIP/2.0/tcp
        10.10.10.21:32780;rport=43459;received=P.U.B.L;branch=z9hG4bKPj71ce8307-a505-4477-9c47-348cb947b1be
        Max-Forwards: 69
        From:
        <sip:1007 at 10.10.10.235>;tag=a702b4fb-d565-4627-9cff-5fb6bfed2922
        To: <sip:1007 at 10.10.10.235>
        Call-ID: b618fe07-a1ef-4eee-926c-ff8e8f3c9096
        CSeq: 49600 REGISTER
        User-Agent: PJSUA v1.0.3/i686-pc-linux-gnu
        Contact: <sip:1007 at P.U.B.LN:43459;transport=tcp>
        Expires: 300
        
        pkt2. FS -> SBC
        SIP/2.0 200 OK
        Via: SIP/2.0/TCP
        10.10.10.238:36707;rport=36707;branch=z9hG4bKPj66cc4b809a6bb1510af258991c1f0587
        Via: SIP/2.0/TCP
        10.10.10.21:32780;rport=43459;received=P.U.B.LN;branch=z9hG4bKPj71ce8307-a505-4477-9c47-348cb947b1be
        From:
        <sip:1007 at 10.10.10.235>;tag=a702b4fb-d565-4627-9cff-5fb6bfed2922
        To: <sip:1007 at 10.10.10.235>;tag=Brpemg1NcteDc
        Call-ID: b618fe07-a1ef-4eee-926c-ff8e8f3c9096
        CSeq: 49600 REGISTER
        Contact: <sip:1007 at P.U.B.LN:43459;transport=tcp>;expires=300
        Date: Thu, 14 Jan 2010 13:15:37 GMT
        User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
        
        pkt3. FS -> external NATed IP-PHONE (trying to send NOTIFY via
        default
        gateway instead SBC!!!)
        Internet Protocol, Src: 10.10.10.235 (10.10.10.235), Dst:
        P.U.B.LN
        (P.U.B.LN)
        Transmission Control Protocol, Src Port: 52304 (52304), Dst
        Port: 43459
        (43459), Seq: 0, Len: 0
        Flags: 0x02 (SYN)
        
        4. default gateway -> FS
        Internet Protocol, Src: P.U.B.LN (P.U.B.LN), Dst: 10.10.10.235
        (10.10.10.235)
        Transmission Control Protocol, Src Port: 43459 (43459), Dst
        Port: 52304
        (52304), Seq: 1, Ack: 1, Len: 0
        Flags: 0x14 (RST, ACK)
        ///////////////////////////////////////////////////////////////////////////////////////////////////////////
        
        The same happened when i use default FS configuration (with
        external profile).
        Can somebody push me to the right way?
        
        Thanks in advance,
        
        Vadim
        
        
        
        
        



From daletrub at gmail.com  Thu Jan 14 09:49:15 2010
From: daletrub at gmail.com (Dale Trub)
Date: Thu, 14 Jan 2010 09:49:15 -0800
Subject: [Freeswitch-users] ISP with SIP trunking for 10k+ ports?
Message-ID: <c23fe9f21001140949o372a2ae1kaaca6d43ba7a6fe1@mail.gmail.com>

We're looking for an ISP with SIP trunking and DIDs to host our FS servers,
and are having a hard time locating an ISP who meet these two requirements
a) can handle 10,000+ simultaneous inbound ports
b) returns our calls in timely manner

We have a six-figure annual budget for this service ... and so far the "big
guys" sales teams can't get it together to put together a bid.

The ideal partner would share with us the backend telcom revenue in the
neighborhood of .5c/min, but we're open to other arrangements.

Does anyone have any recommendations for ISPs with SIP trunking and DIDs
that they can share?  Specific salespeople contact info especially
appreciated!

Thanks,

Dale
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From freeswitch at aastral.net  Thu Jan 14 09:50:56 2010
From: freeswitch at aastral.net (Bill W)
Date: Thu, 14 Jan 2010 12:50:56 -0500
Subject: [Freeswitch-users] Question about a1-hash
In-Reply-To: <41473BE6-E347-4FCD-AD15-9964265C53EB@freeswitch.org>
References: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>	<4B4F430A.8080006@aastral.net>	<5d2828f1001140827w386fe2acw478ae378b0ecfe53@mail.gmail.com>	<4B4F5257.6080508@aastral.net>
	<41473BE6-E347-4FCD-AD15-9964265C53EB@freeswitch.org>
Message-ID: <4B4F5980.5000006@aastral.net>

Ah okay, it is just as I thought.  Either I store the a1-hash in the 
database, or store the password in clear text so I can calculate a1-hash 
on the fly.

Or as Mike says, have the phone password be the actual hash value of the 
password in the database, and then calculate an a1-hash with that hashed 
  password.


Thanks,
Bill W.


Brian West wrote:
> You take what is in the data base and create an a1-hash if you have the parts to create it.. if not your SOL.
> 
> /b
> 
> On Jan 14, 2010, at 11:20 AM, Bill W wrote:
> 
>


From costa.zikalala at gmail.com  Thu Jan 14 09:58:10 2010
From: costa.zikalala at gmail.com (Costa Zikalala)
Date: Thu, 14 Jan 2010 19:58:10 +0200
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <b39273531001140855i606a06a4ne55be84d825b1a74@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com> 
	<b39273531001140855i606a06a4ne55be84d825b1a74@mail.gmail.com>
Message-ID: <59daa2cd1001140958x49f64247r554d76d596d1264f@mail.gmail.com>

Thank you so much for this Rupa.



2010/1/14 Rupa Schomaker <rupa at rupa.com>

> I don't see an email2fax yet, but an important first step is in:
>
> contrib/sathieu/email2pdf/email2pdf
>
> On Thu, Jan 14, 2010 at 8:54 AM, Costa Zikalala <costa.zikalala at gmail.com>wrote:
>
>> Hi All
>>
>> Has anyone worked on a email2fax script for mod_fax?
>> If not how much would it cost for some genius here to quickly whip-up one?
>>
>> Ideally both email2fax and fax2email should come standard with mod_fax or
>> is this not possible?
>>
>> Thanks
>> Costa
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Jan 14 09:59:21 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 11:59:21 -0600
Subject: [Freeswitch-users] ISP with SIP trunking for 10k+ ports?
In-Reply-To: <c23fe9f21001140949o372a2ae1kaaca6d43ba7a6fe1@mail.gmail.com>
References: <c23fe9f21001140949o372a2ae1kaaca6d43ba7a6fe1@mail.gmail.com>
Message-ID: <191c3a031001140959n59342638r280b950ef6884da4@mail.gmail.com>

Maybe you could direct some of that budget to the FS project or sponsor this
summer's ClueCon MMX  =D
Also, If you call us at 877 742 2583 we can probably connect you some telco
resources.



On Thu, Jan 14, 2010 at 11:49 AM, Dale Trub <daletrub at gmail.com> wrote:

> We're looking for an ISP with SIP trunking and DIDs to host our FS servers,
> and are having a hard time locating an ISP who meet these two requirements
> a) can handle 10,000+ simultaneous inbound ports
> b) returns our calls in timely manner
>
> We have a six-figure annual budget for this service ... and so far the "big
> guys" sales teams can't get it together to put together a bid.
>
> The ideal partner would share with us the backend telcom revenue in the
> neighborhood of .5c/min, but we're open to other arrangements.
>
> Does anyone have any recommendations for ISPs with SIP trunking and DIDs
> that they can share?  Specific salespeople contact info especially
> appreciated!
>
> Thanks,
>
> Dale
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Thu Jan 14 10:01:19 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 12:01:19 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <407935.47740.qm@web37505.mail.mud.yahoo.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
	<201001132326.41354.sos@sokhapkin.dyndns.org>
	<52E8BE9E-B6D6-411C-9080-697AF24017B4@jerris.com>
	<191c3a031001132114s18ed2baw2ff802dbd615fe01@mail.gmail.com>
	<191c3a031001132115u2d88bab7p7cbb27ab1eaad466@mail.gmail.com>
	<9853f4ff1001140226y2b035f37rfcafb03190435ec1@mail.gmail.com>
	<43692088-738A-475C-8D60-E40D296ECEB1@jerris.com>
	<407935.47740.qm@web37505.mail.mud.yahoo.com>
Message-ID: <191c3a031001141001q3b164652s87c1561e2a2daf2c@mail.gmail.com>

he wants to call 3 people at once and let the A leg hear early media from
call #1 while call #2 and #3 still are progressing which is not simple to do
without doing thousands of dollars in development.


On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:

> What about sending Sip 183 with SDP (no 200OK), so that your customers can
> hear recordings?
>
> djbinter
>
>
> ------------------------------
> *From:* Michael Jerris <mike at jerris.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thu, January 14, 2010 8:12:59 AM
> *Subject:* Re: [Freeswitch-users] multiple leg and multiple rtp
>
> Okay, so that is a very interesting use case.   Seems a bizarre way to do
> this for the carrier, but interesting none the less.  I'd say the chances of
> actually muxing the early media is small, what might be possible would be
> something to say which b legs media to pass along to the a leg.  I have not
> looked at all at how complicated this is, it will be down deep somewhere in
> switch_ivr_originate code around where we if for ignore_early_media.  This
> code is pretty complex, I can't say that we will ever actually add this
> functionality, but at least now I won't blow it off as complete nonsense.
>
> Mike
>
>
>
> On Jan 14, 2010, at 5:26 AM, David Villasmil wrote:
>
> Hello again,
>
> Using ingore early media only ignores ALL media, that's not what I need. At
> least in europe i've seen it many times, MNOs provide a service with which
> you can have a song played to the caller while the call is connecting to
> your cell phone. This is basically what I'm trying to achieve.
>
> The content provider is in possesions of the media and it require us to
> fork the call and send a SIP INVITE to the on one leg and the call to the
> destination number on another leg. they will only provide a progress, no
> answer.
>
> I CAN do it locally by playing the file as a custom ringback but that's not
> the standard in terms of commercial use of the content.
>
> Is there any way to modify the behaviour of fs when i receives media? Let's
> say i.e. it doesn't drop the other leg, but provides the first early media
> it receives and just wait for some channel to answer? I will not have both
> media but it would work.
>
> thanks
>
> David
>
> On Thu, Jan 14, 2010 at 6:15 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Perhaps best not to help him anymore without an apology for the snap
>> judgement and comparison to asterisk clearly designed to push our buttons.
>> We'll be here when you realize we were trying to help you but I can't
>> promise we will still have paitence......
>>
>> On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike at jerris.com> wrote:
>>
>> You already are running on trunk.
>>
>> Mike
>>
>> On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote: > I run FreeSWITCH
>> Version 1.0.5pre10 (16012M...
>>
>> _______________________________________________ FreeSWITCH-users mailing
>> list FreeSWITCH-users at lists...
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mrene_lists at avgs.ca  Thu Jan 14 10:08:49 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 14 Jan 2010 13:08:49 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001141001q3b164652s87c1561e2a2daf2c@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
	<201001132326.41354.sos@sokhapkin.dyndns.org>
	<52E8BE9E-B6D6-411C-9080-697AF24017B4@jerris.com>
	<191c3a031001132114s18ed2baw2ff802dbd615fe01@mail.gmail.com>
	<191c3a031001132115u2d88bab7p7cbb27ab1eaad466@mail.gmail.com>
	<9853f4ff1001140226y2b035f37rfcafb03190435ec1@mail.gmail.com>
	<43692088-738A-475C-8D60-E40D296ECEB1@jerris.com>
	<407935.47740.qm@web37505.mail.mud.yahoo.com>
	<191c3a031001141001q3b164652s87c1561e2a2daf2c@mail.gmail.com>
Message-ID: <C3EB94B1-BA8B-4D89-A781-0E47E4C1C8AB@avgs.ca>

For the music I guess you could use "uuid_displace" to mix it in  
before you call bridge but even then you can't fork-dial multiple  
destinations and have early media at the same time, it doesnt make  
much sense.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 14-Jan-10, at 1:01 PM, Anthony Minessale wrote:

> he wants to call 3 people at once and let the A leg hear early media  
> from call #1 while call #2 and #3 still are progressing which is not  
> simple to do without doing thousands of dollars in development.
>
>
> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
> What about sending Sip 183 with SDP (no 200OK), so that your  
> customers can hear recordings?
>
> djbinter
>
>
> From: Michael Jerris <mike at jerris.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thu, January 14, 2010 8:12:59 AM
> Subject: Re: [Freeswitch-users] multiple leg and multiple rtp
>
> Okay, so that is a very interesting use case.   Seems a bizarre way  
> to do this for the carrier, but interesting none the less.  I'd say  
> the chances of actually muxing the early media is small, what might  
> be possible would be something to say which b legs media to pass  
> along to the a leg.  I have not looked at all at how complicated  
> this is, it will be down deep somewhere in switch_ivr_originate code  
> around where we if for ignore_early_media.  This code is pretty  
> complex, I can't say that we will ever actually add this  
> functionality, but at least now I won't blow it off as complete  
> nonsense.
>
> Mike
>
>
>
> On Jan 14, 2010, at 5:26 AM, David Villasmil wrote:
>
>> Hello again,
>>
>> Using ingore early media only ignores ALL media, that's not what I  
>> need. At least in europe i've seen it many times, MNOs provide a  
>> service with which you can have a song played to the caller while  
>> the call is connecting to your cell phone. This is basically what  
>> I'm trying to achieve.
>>
>> The content provider is in possesions of the media and it require  
>> us to fork the call and send a SIP INVITE to the on one leg and the  
>> call to the destination number on another leg. they will only  
>> provide a progress, no answer.
>>
>> I CAN do it locally by playing the file as a custom ringback but  
>> that's not the standard in terms of commercial use of the content.
>>
>> Is there any way to modify the behaviour of fs when i receives  
>> media? Let's say i.e. it doesn't drop the other leg, but provides  
>> the first early media it receives and just wait for some channel to  
>> answer? I will not have both media but it would work.
>>
>> thanks
>>
>> David
>>
>> On Thu, Jan 14, 2010 at 6:15 AM, Anthony Minessale <anthony.minessale at gmail.com 
>> > wrote:
>> Perhaps best not to help him anymore without an apology for the  
>> snap judgement and comparison to asterisk clearly designed to push  
>> our buttons.
>> We'll be here when you realize we were trying to help you but I  
>> can't promise we will still have paitence......
>>
>>
>>> On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike at jerris.com> wrote:
>>>
>>> You already are running on trunk.
>>>
>>> Mike
>>> On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote: > I run  
>>> FreeSWITCH Version 1.0.5pre10 (16012M...
>>>
>>> _______________________________________________ FreeSWITCH-users  
>>> mailing list FreeSWITCH-users at lists...
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kokoska.rokoska at post.cz  Thu Jan 14 10:10:07 2010
From: kokoska.rokoska at post.cz (kokoska.rokoska)
Date: Thu, 14 Jan 2010 19:10:07 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <23BD1322-7882-4D43-80AB-27872096B4C9@avgs.ca>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>	<201001140719.12363.sos@sokhapkin.dyndns.org>	<191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>	<201001141141.59831.sos@sokhapkin.dyndns.org>
	<23BD1322-7882-4D43-80AB-27872096B4C9@avgs.ca>
Message-ID: <4B4F5DFF.6000504@post.cz>

Dne 14.1.2010 17:47, Mathieu Rene napsal(a):
> One way you could fix it is set "hangup_after_bridge=false" and then  
> set "execute_on_answer=set\shangup_after_bridge=true". That will make  
> it continue executing the dialplan atter a failure that came in after  
> 183. Note that there is a side effect to this method, since audio was  
> being relayed, you are forced to use the same codec on the 2nd  
> carrier, best to tell them they are wrong failing a call after 183.
> 
> Mathieu Rene

Hello Mathieu, are you sure, that "failing call" after 183 is wrong?
I don't think so, but may be I'm wrong, so could you, please, point me
to the docs?

BTW: I'm prety sure, that this behaviour is VERY common in EU. IMO
nearly all big carriers in case of call failure (according to ETSI
recommendations) send first call-progress with media ("The number your
are calling is unavailable" etc.) and than reject call with proper "Q"
reason. And I think it should be mapped to SIP like 183+SDP and than
4XX/5XX/6XX.

Just my 2c :-)

Best regards,

kokoska.rokoska


From sos at sokhapkin.dyndns.org  Thu Jan 14 10:18:34 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Jan 2010 13:18:34 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001141001q3b164652s87c1561e2a2daf2c@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<407935.47740.qm@web37505.mail.mud.yahoo.com>
	<191c3a031001141001q3b164652s87c1561e2a2daf2c@mail.gmail.com>
Message-ID: <201001141318.34958.sos@sokhapkin.dyndns.org>

The issue here is when "originate" routine should return and 
set "originate_status" variable. Current behavior is to return on early 
media, but what if to introduce a variable "originate_wait_for_answer" with 
default value "false" and use the variable in originate code to decide when 
to return - on 18X or "200 OK"?

On Thursday 14 January 2010, Anthony Minessale wrote:
> he wants to call 3 people at once and let the A leg hear early media from
> call #1 while call #2 and #3 still are progressing which is not simple to
> do without doing thousands of dollars in development.
>
> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
> > What about sending Sip 183 with SDP (no 200OK), so that your customers
> > can hear recordings?
> >
> > djbinter
> >
> >
> > ------------------------------
> > *From:* Michael Jerris <mike at jerris.com>
> > *To:* freeswitch-users at lists.freeswitch.org
> > *Sent:* Thu, January 14, 2010 8:12:59 AM
> > *Subject:* Re: [Freeswitch-users] multiple leg and multiple rtp
> >
> > Okay, so that is a very interesting use case.   Seems a bizarre way to do
> > this for the carrier, but interesting none the less.  I'd say the chances
> > of actually muxing the early media is small, what might be possible would
> > be something to say which b legs media to pass along to the a leg.  I
> > have not looked at all at how complicated this is, it will be down deep
> > somewhere in switch_ivr_originate code around where we if for
> > ignore_early_media.  This code is pretty complex, I can't say that we
> > will ever actually add this functionality, but at least now I won't blow
> > it off as complete nonsense.
> >
> > Mike
> >
> >
> >
> > On Jan 14, 2010, at 5:26 AM, David Villasmil wrote:
> >
> > Hello again,
> >
> > Using ingore early media only ignores ALL media, that's not what I need.
> > At least in europe i've seen it many times, MNOs provide a service with
> > which you can have a song played to the caller while the call is
> > connecting to your cell phone. This is basically what I'm trying to
> > achieve.
> >
> > The content provider is in possesions of the media and it require us to
> > fork the call and send a SIP INVITE to the on one leg and the call to the
> > destination number on another leg. they will only provide a progress, no
> > answer.
> >
> > I CAN do it locally by playing the file as a custom ringback but that's
> > not the standard in terms of commercial use of the content.
> >
> > Is there any way to modify the behaviour of fs when i receives media?
> > Let's say i.e. it doesn't drop the other leg, but provides the first
> > early media it receives and just wait for some channel to answer? I will
> > not have both media but it would work.
> >
> > thanks
> >
> > David
> >
> > On Thu, Jan 14, 2010 at 6:15 AM, Anthony Minessale <
> >
> > anthony.minessale at gmail.com> wrote:
> >> Perhaps best not to help him anymore without an apology for the snap
> >> judgement and comparison to asterisk clearly designed to push our
> >> buttons. We'll be here when you realize we were trying to help you but I
> >> can't promise we will still have paitence......
> >>
> >> On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike at jerris.com> wrote:
> >>
> >> You already are running on trunk.
> >>
> >> Mike
> >>
> >> On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote: > I run FreeSWITCH
> >> Version 1.0.5pre10 (16012M...
> >>
> >> _______________________________________________ FreeSWITCH-users mailing
> >> list FreeSWITCH-users at lists...
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From sos at sokhapkin.dyndns.org  Thu Jan 14 10:27:40 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Jan 2010 13:27:40 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001140858i8ef1b09kf5d547914b5b8b68@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141141.59831.sos@sokhapkin.dyndns.org>
	<191c3a031001140858i8ef1b09kf5d547914b5b8b68@mail.gmail.com>
Message-ID: <201001141327.40713.sos@sokhapkin.dyndns.org>

Confirmed. r16311 has the issue resolved in both bypass_media=true/false 
modes.

On Thursday 14 January 2010, Anthony Minessale wrote:
> r16310
>
> Thank you for the apology.
>
> On Thu, Jan 14, 2010 at 10:41 AM, Sergey Okhapkin
>
> <sos at sokhapkin.dyndns.org>wrote:
> > Anthony, I apology, I didn't expect you'll be annoyed. As for the issue,
> > the
> > issue is not in "proxy_media" setting, same happens when both
> > "proxy_media" and "bypass_media" are not set. The code does explicit leg
> > A hangup in switch_ivr_bridge.c, line 513. The condition there is "if leg
> > A is up and leg
> > A is not in answered state, then hangup leg A", this prevents dialplan to
> > continue execution on early media bridge termination.
> >
> > Issue when "bypass_media=true" is completely different, I'll open the
> > issue on
> > jira.
> >
> > On Thursday 14 January 2010, Anthony Minessale wrote:
> > > Sergey,
> > >
> > > The bug you reference was closed because proxy_media mode by design
> >
> > send's
> >
> > > the B leg's codec to the A
> > > LEG so if there is a failure condition there is no easy graceful way to
> > > back out and try another call.
> > > I will try to make it possible if yo post a bounty, I estimate a
> > > minimum
> >
> > of
> >
> > > $1000USD in consulting time.
> > >
> > > The other one I might look at once you have apologized.
> > >
> > > I can probably add some code to make the bridge exit without
> > > terminating the A leg even when hangup_after_bridge=true in the case
> > > where the the B leg is not answered but right now I am sort of annoyed
> > > with your attitude in this thread.
> > >
> > >
> > > On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin
> > >
> > > <sos at sokhapkin.dyndns.org>wrote:
> > > > Case 1 (bypass_media is off) is already on jira,
> > > > http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
> > > > installation with latest trunk to reproduce case 2, when
> > > > bypass_media=true and 10 seconds delay happens when 18X and then
> > > > error are received on leg b.
> > > >
> > > > On Wednesday 13 January 2010, Michael Jerris wrote:
> > > > > On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> > > > > > Critical issues are when SIP error come after 18X provisional
> > > > > > response.
> > > > > >
> > > > > > - if bypass_media is false then dialplan stops and leg a is
> > > > > > explicitly hang up (switch_ivr_bridge.c, line 513).
> > > > >
> > > > >  behavior can be modified with continue_on_fail and
> >
> > hangup_after_bridge
> >
> > > > > channel vars, perhaps ignore_early_media as well
> > > > >
> > > > > > - if bypass_media is true, then dialplan continue, but there is
> > > > > > 10 seconds delay before next bridge application sends INVITE to
> >
> > gateway
> >
> > > > > > (
> >
> > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
> >
> > > > > >4.html ) I didn't track down yet why this happens looking at FS
> > > > > > sources.
> > > > >
> > > > > see response on that thread
> > > > >
> > > > > > Why I didn't open a bug on jira? Because FS behaves according to
> >
> > the
> >
> > > > > > design and specs :-) But not according to real world
> >
> > requirements...
> >
> > > > > Really, people are trying to help you and your going to be snarky
> > > > > in response?
> > > > >
> > > > > > On Wednesday 13 January 2010, Brian West wrote:
> > > > > >> Can you elaborate on these "Critical" issues you seem to be
> >
> > having?
> >
> > > >  Why
> > > >
> > > > > >> aren't you opening a jira for them if they are that critical to
> >
> > your
> >
> > > > > >> needs?
> > > > >
> > > > > Mike
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-user
> >
> > > > >s http://www.freeswitch.org
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From david.villasmil.work at gmail.com  Thu Jan 14 10:29:19 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 14 Jan 2010 19:29:19 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <C3EB94B1-BA8B-4D89-A781-0E47E4C1C8AB@avgs.ca>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132326.41354.sos@sokhapkin.dyndns.org>
	<52E8BE9E-B6D6-411C-9080-697AF24017B4@jerris.com>
	<191c3a031001132114s18ed2baw2ff802dbd615fe01@mail.gmail.com>
	<191c3a031001132115u2d88bab7p7cbb27ab1eaad466@mail.gmail.com>
	<9853f4ff1001140226y2b035f37rfcafb03190435ec1@mail.gmail.com>
	<43692088-738A-475C-8D60-E40D296ECEB1@jerris.com>
	<407935.47740.qm@web37505.mail.mud.yahoo.com>
	<191c3a031001141001q3b164652s87c1561e2a2daf2c@mail.gmail.com>
	<C3EB94B1-BA8B-4D89-A781-0E47E4C1C8AB@avgs.ca>
Message-ID: <9853f4ff1001141029i4be469bek779fc2921099be8c@mail.gmail.com>

Anthony,

It's actually

2 call fork, 1 has the audio the other is a regular call, a-side
should hear audio from whichever comes first BUT not discarding both
b-legs and bridging to whichever answers.




On Thu, Jan 14, 2010 at 7:08 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>
> For the music I guess you could use "uuid_displace" to mix it in before you call bridge but even then you can't fork-dial multiple destinations and have early media at the same time, it doesnt make much sense.
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
> On 14-Jan-10, at 1:01 PM, Anthony Minessale wrote:
>
> he wants to call 3 people at once and let the A leg hear early media from call #1 while call #2 and #3 still are progressing which is not simple to do without doing thousands of dollars in development.
>
>
> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>>
>> What about sending Sip 183 with SDP (no 200OK), so that your customers?can hear recordings?
>> djbinter
>>
>>
>> ________________________________
>> From: Michael Jerris <mike at jerris.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Thu, January 14, 2010 8:12:59 AM
>> Subject: Re: [Freeswitch-users] multiple leg and multiple rtp
>>
>> Okay, so that is a very interesting use case. ? Seems a bizarre way to do this for the carrier, but interesting none the less. ?I'd say the chances of actually muxing the early media is small, what might be possible would be something to say which b legs media to pass along to the a leg. ?I have not looked at all at how complicated this is, it will be down deep somewhere in switch_ivr_originate code around where we if for ignore_early_media. ?This code is pretty complex, I can't say that we will ever actually add this functionality, but at least now I won't blow it off as complete nonsense.
>> Mike
>>
>>
>> On Jan 14, 2010, at 5:26 AM, David Villasmil wrote:
>>
>> Hello again,
>>
>> Using ingore early media only ignores ALL media, that's not what I need. At least in europe i've seen it many times, MNOs provide a service with which you can have a song played to the caller while the call is connecting to your cell phone. This is basically what I'm trying to achieve.
>>
>> The content provider is in possesions of the media and it require us to fork the call and send a SIP INVITE to the on one leg and the call to the destination number on another leg. they will only provide a progress, no answer.
>>
>> I CAN do it locally by playing the file as a custom ringback but that's not the standard in terms of commercial use of the content.
>>
>> Is there any way to modify the behaviour of fs when i receives media? Let's say i.e. it doesn't drop the other leg, but provides the first early media it receives and just wait for some channel to answer? I will not have both media but it would work.
>>
>> thanks
>>
>> David
>>
>> On Thu, Jan 14, 2010 at 6:15 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>>
>>> Perhaps best not to help him anymore without an apology for the snap judgement and comparison to asterisk clearly designed to push our buttons.
>>> We'll be here when you realize we were trying to help you but I can't promise we will still have paitence......
>>>
>>> On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike at jerris.com> wrote:
>>>
>>> You already are running on trunk.
>>>
>>> Mike
>>>
>>> On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote: > I run FreeSWITCH Version 1.0.5pre10 (16012M...
>>>
>>> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists...
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mike at jerris.com  Thu Jan 14 10:34:07 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Jan 2010 13:34:07 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001141318.34958.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<407935.47740.qm@web37505.mail.mud.yahoo.com>
	<191c3a031001141001q3b164652s87c1561e2a2daf2c@mail.gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
Message-ID: <A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>

this is exactly what ignore_early_media does now.

Mike

On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:

> The issue here is when "originate" routine should return and 
> set "originate_status" variable. Current behavior is to return on early 
> media, but what if to introduce a variable "originate_wait_for_answer" with 
> default value "false" and use the variable in originate code to decide when 
> to return - on 18X or "200 OK"?
> 
> On Thursday 14 January 2010, Anthony Minessale wrote:
>> he wants to call 3 people at once and let the A leg hear early media from
>> call #1 while call #2 and #3 still are progressing which is not simple to
>> do without doing thousands of dollars in development.
>> 
>> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>>> What about sending Sip 183 with SDP (no 200OK), so that your customers
>>> can hear recordings?



From sos at sokhapkin.dyndns.org  Thu Jan 14 10:42:02 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Jan 2010 13:42:02 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
Message-ID: <201001141342.02178.sos@sokhapkin.dyndns.org>

No, not exactly. ignore_early_media doesn't pass early media to the caller if 
bypass_media is false.

On Thursday 14 January 2010, Michael Jerris wrote:
> this is exactly what ignore_early_media does now.
>
> Mike
>
> On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> > The issue here is when "originate" routine should return and
> > set "originate_status" variable. Current behavior is to return on early
> > media, but what if to introduce a variable "originate_wait_for_answer"
> > with default value "false" and use the variable in originate code to
> > decide when to return - on 18X or "200 OK"?
> >
> > On Thursday 14 January 2010, Anthony Minessale wrote:
> >> he wants to call 3 people at once and let the A leg hear early media
> >> from call #1 while call #2 and #3 still are progressing which is not
> >> simple to do without doing thousands of dollars in development.
> >>
> >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
> >>> What about sending Sip 183 with SDP (no 200OK), so that your customers
> >>> can hear recordings?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From devel at thom.fr.eu.org  Thu Jan 14 10:57:22 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Thu, 14 Jan 2010 19:57:22 +0100
Subject: [Freeswitch-users] =?utf-8?q?Fwd=3A_Re=3A__playback_and_play=5Fan?=
 =?utf-8?q?d=5Fget=5Fdigits_strange_misunderstanding?=
Message-ID: <71bf5a5d66375b2d7c333d26b29917f1@thom.fr.eu.org>

 

I did with FreeSWITCH Version 1.0.5-20100112-0400 (hacked) shall I
download another one (I don't have svn). 

Fran?ois 

On Thu, 14 Jan 2010
12:20:31 -0500, Michael Jerris wrote: Could you try this with latest trunk
as well for comparison, I just changed the way the sounds base directory is
set and I want to make sure I did not mess anything up. Mike 
  On Jan 14,
2010, at 3:23 AM, Fran?ois Legal wrote: 

So I could (kind of) solve this
by myself. There is in vars.xml the variable ${sound_prefix}. I did set it
properly to my french sound path and then it worked.  

However, for the
sake of discussion, I did try this with 1.0.5pre8 and the result was
different : 

with the extension 

FS was trying to play the file
${FREESWITCH_PATH}/sounds/en/us/callie/misc/ringing_disabled.wav 

with the
extension 

FS was trying to load
${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/ringing_disabled.wav 

Now with
the latest snapshot, with either one of the 2 mentionned extensions, FS
tries to play the file in
${sound_prefix}/filepath/${codec_bit_rate}/filename whereas 1.0.5pre8 did
not add the codec_bit_rate in the path but took care of the
default_language variable.  

Fran?ois  

On Wed, 13 Jan 2010 16:32:16
+0100, Fran?ois Legal wrote:  

Hello,  

trying to make so dialplan
extensions that use the playback and play_and_get_digits applications, but
I'm having trouble with the file name specification.  

The files I want to
play are in the french language (fr/fr/julie as configured in
lang/fr/fr.xml) 

My extension is as follows : 

The channel is using a bit
rate of 8000 Hz, so by the set default_language=fr I would expect
freeswitch to playback the file at
${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/8000/ringing_disabled.wav
whereas it tries to playback the file at
${FREESWITCH_PATH}/sounds/en/us/callie/misc/8000/ringing_disabled.wav 

I
have the same with play_and_get_digits application.  

What am I doing
wrong ?  

Fran?ois   

_______________________________________________
FreeSWITCH-users mailing
list
FreeSWITCH-users at lists.freeswitch.org
[1]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




Links:
------
[1] mailto:FreeSWITCH-users at lists.freeswitch.org
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From mustafa.pk at gmail.com  Thu Jan 14 11:09:52 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Fri, 15 Jan 2010 00:09:52 +0500
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
Message-ID: <8213d6071001141109l1d475b1j3af35b5586708e42@mail.gmail.com>

fax2email is quiet easy you just need to keep a mapping of did's with
respective email addresses of persons in a database or something, and
an api call on hangup can trigger an mail script to convert and send
fax in a more user friendly format like pfd or doc etc.

for email2fax you would need to setup an smtp service fax.example.com,
your smtp service can pipe the email  to a custom script to
extract file attachments using a mime decoder and send them to
freeswitch. for all document format conversion things you can use
openoffice macros functionality which is callable from command line.

-m

On Thu, Jan 14, 2010 at 7:54 PM, Costa Zikalala
<costa.zikalala at gmail.com> wrote:
> Hi All
>
> Has anyone worked on a email2fax script for mod_fax?
> If not how much would it cost for some genius here to quickly whip-up one?
>
> Ideally both email2fax and fax2email should come standard with mod_fax or is
> this not possible?
>
> Thanks
> Costa
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com


From mouncifbb at gmail.com  Thu Jan 14 11:23:53 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Thu, 14 Jan 2010 14:23:53 -0500
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <8213d6071001141109l1d475b1j3af35b5586708e42@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
	<8213d6071001141109l1d475b1j3af35b5586708e42@mail.gmail.com>
Message-ID: <ade741d61001141123x393512f6x81768de82a84c214@mail.gmail.com>

Curious! How do you trigger an api call on hangup?

On Thu, Jan 14, 2010 at 2:09 PM, Ghulam Mustafa <mustafa.pk at gmail.com>wrote:

> fax2email is quiet easy you just need to keep a mapping of did's with
> respective email addresses of persons in a database or something, and
> an api call on hangup can trigger an mail script to convert and send
> fax in a more user friendly format like pfd or doc etc.
>
> for email2fax you would need to setup an smtp service fax.example.com,
> your smtp service can pipe the email  to a custom script to
> extract file attachments using a mime decoder and send them to
> freeswitch. for all document format conversion things you can use
> openoffice macros functionality which is callable from command line.
>
> -m
>
> On Thu, Jan 14, 2010 at 7:54 PM, Costa Zikalala
> <costa.zikalala at gmail.com> wrote:
> > Hi All
> >
> > Has anyone worked on a email2fax script for mod_fax?
> > If not how much would it cost for some genius here to quickly whip-up
> one?
> >
> > Ideally both email2fax and fax2email should come standard with mod_fax or
> is
> > this not possible?
> >
> > Thanks
> > Costa
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mrene_lists at avgs.ca  Thu Jan 14 11:26:30 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 14 Jan 2010 14:26:30 -0500
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <ade741d61001141123x393512f6x81768de82a84c214@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
	<8213d6071001141109l1d475b1j3af35b5586708e42@mail.gmail.com>
	<ade741d61001141123x393512f6x81768de82a84c214@mail.gmail.com>
Message-ID: <50B0D531-15AC-4718-ABD1-BBA74D6E30C1@avgs.ca>

You set the api_hangup_hook variable to what you want to run.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 14-Jan-10, at 2:23 PM, Mouncif Benniane wrote:

> Curious! How do you trigger an api call on hangup?
>
> On Thu, Jan 14, 2010 at 2:09 PM, Ghulam Mustafa  
> <mustafa.pk at gmail.com> wrote:
> fax2email is quiet easy you just need to keep a mapping of did's with
> respective email addresses of persons in a database or something, and
> an api call on hangup can trigger an mail script to convert and send
> fax in a more user friendly format like pfd or doc etc.
>
> for email2fax you would need to setup an smtp service fax.example.com,
> your smtp service can pipe the email  to a custom script to
> extract file attachments using a mime decoder and send them to
> freeswitch. for all document format conversion things you can use
> openoffice macros functionality which is callable from command line.
>
> -m
>
> On Thu, Jan 14, 2010 at 7:54 PM, Costa Zikalala
> <costa.zikalala at gmail.com> wrote:
> > Hi All
> >
> > Has anyone worked on a email2fax script for mod_fax?
> > If not how much would it cost for some genius here to quickly whip- 
> up one?
> >
> > Ideally both email2fax and fax2email should come standard with  
> mod_fax or is
> > this not possible?
> >
> > Thanks
> > Costa
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mustafa.pk at gmail.com  Thu Jan 14 11:31:27 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Fri, 15 Jan 2010 00:31:27 +0500
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <ade741d61001141123x393512f6x81768de82a84c214@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
	<8213d6071001141109l1d475b1j3af35b5586708e42@mail.gmail.com>
	<ade741d61001141123x393512f6x81768de82a84c214@mail.gmail.com>
Message-ID: <8213d6071001141131m3a6ab686xf716dc26983ade5a@mail.gmail.com>

you can do it within dialplan, or you can trigger an event or
something, there are several ways to get it done!

On Fri, Jan 15, 2010 at 12:23 AM, Mouncif Benniane <mouncifbb at gmail.com> wrote:
> Curious! How do you trigger an api call on hangup?
>
> On Thu, Jan 14, 2010 at 2:09 PM, Ghulam Mustafa <mustafa.pk at gmail.com>
> wrote:
>>
>> fax2email is quiet easy you just need to keep a mapping of did's with
>> respective email addresses of persons in a database or something, and
>> an api call on hangup can trigger an mail script to convert and send
>> fax in a more user friendly format like pfd or doc etc.
>>
>> for email2fax you would need to setup an smtp service fax.example.com,
>> your smtp service can pipe the email ?to a custom script to
>> extract file attachments using a mime decoder and send them to
>> freeswitch. for all document format conversion things you can use
>> openoffice macros functionality which is callable from command line.
>>
>> -m
>>
>> On Thu, Jan 14, 2010 at 7:54 PM, Costa Zikalala
>> <costa.zikalala at gmail.com> wrote:
>> > Hi All
>> >
>> > Has anyone worked on a email2fax script for mod_fax?
>> > If not how much would it cost for some genius here to quickly whip-up
>> > one?
>> >
>> > Ideally both email2fax and fax2email should come standard with mod_fax
>> > or is
>> > this not possible?
>> >
>> > Thanks
>> > Costa
>> >
>> >
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Ghulam Mustafa
>> cell: +92 333.611.7681
>> sip: cyrenity at ekiga.net
>> mail: mustafa.pk at gmail.com
>> web: cyrenity.wordpress.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com


From msc at freeswitch.org  Thu Jan 14 11:51:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Jan 2010 11:51:37 -0800
Subject: [Freeswitch-users] Announcement: FreeSWITCH Event Socket Library
	Supports TCL
Message-ID: <87f2f3b91001141151w361de47fgc6e1db6f9929ea06@mail.gmail.com>

Just to let everyone know, we've added TCL support for ESL. More info here:
http://www.freeswitch.org/node/228

Let us know if you do anything interesting with it and we'll add it to the
wiki.
-Michael
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From mouncifbb at gmail.com  Thu Jan 14 11:54:03 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Thu, 14 Jan 2010 14:54:03 -0500
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <8213d6071001141131m3a6ab686xf716dc26983ade5a@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
	<8213d6071001141109l1d475b1j3af35b5586708e42@mail.gmail.com>
	<ade741d61001141123x393512f6x81768de82a84c214@mail.gmail.com>
	<8213d6071001141131m3a6ab686xf716dc26983ade5a@mail.gmail.com>
Message-ID: <ade741d61001141154m75b9fcbak76e874e80d023ac0@mail.gmail.com>

thanks, how would it look in dialplan if I have to call a javascript?



On Thu, Jan 14, 2010 at 2:31 PM, Ghulam Mustafa <mustafa.pk at gmail.com>wrote:

> you can do it within dialplan, or you can trigger an event or
> something, there are several ways to get it done!
>
> On Fri, Jan 15, 2010 at 12:23 AM, Mouncif Benniane <mouncifbb at gmail.com>
> wrote:
> > Curious! How do you trigger an api call on hangup?
> >
> > On Thu, Jan 14, 2010 at 2:09 PM, Ghulam Mustafa <mustafa.pk at gmail.com>
> > wrote:
> >>
> >> fax2email is quiet easy you just need to keep a mapping of did's with
> >> respective email addresses of persons in a database or something, and
> >> an api call on hangup can trigger an mail script to convert and send
> >> fax in a more user friendly format like pfd or doc etc.
> >>
> >> for email2fax you would need to setup an smtp service fax.example.com,
> >> your smtp service can pipe the email  to a custom script to
> >> extract file attachments using a mime decoder and send them to
> >> freeswitch. for all document format conversion things you can use
> >> openoffice macros functionality which is callable from command line.
> >>
> >> -m
> >>
> >> On Thu, Jan 14, 2010 at 7:54 PM, Costa Zikalala
> >> <costa.zikalala at gmail.com> wrote:
> >> > Hi All
> >> >
> >> > Has anyone worked on a email2fax script for mod_fax?
> >> > If not how much would it cost for some genius here to quickly whip-up
> >> > one?
> >> >
> >> > Ideally both email2fax and fax2email should come standard with mod_fax
> >> > or is
> >> > this not possible?
> >> >
> >> > Thanks
> >> > Costa
> >> >
> >> >
> >> >
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Ghulam Mustafa
> >> cell: +92 333.611.7681
> >> sip: cyrenity at ekiga.net
> >> mail: mustafa.pk at gmail.com
> >> web: cyrenity.wordpress.com
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch-users at digitaldan.com  Thu Jan 14 12:30:34 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Thu, 14 Jan 2010 13:30:34 -0700 (MST)
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <32185789.01263487832057.JavaMail.root@zimbra>
Message-ID: <13392952.131263501034741.JavaMail.root@zimbra>

So I did figure it out, it ended up being the CISCO_SKIP_MARK_BIT_2833 issue mentioned on the RTP troubleshooting wiki page. I had incorrectly added this to my dialplan rather then to the external sip profile where it belonged. 


Thanks 
Dan- 

----- Original Message ----- 
From: "Dan" <freeswitch-users at digitaldan.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 1:56:16 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 


Thanks, I will try and turn up the debugging on it. I have a lua script that is calling session:getDigits(20,"#",15000) . When I watch it under fs_cli I'm getting this output 



2010-01-13 10:44:15.750329 [NOTICE] switch_cpp.cpp:1129 main: Getting digits 
2010-01-13 10:44:17.586676 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 0:400 
2010-01-13 10:44:17.790705 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 5:400 
2010-01-13 10:44:17.910737 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 1:400 
2010-01-13 10:44:18.114779 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 7:400 
2010-01-13 10:44:18.234812 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 2:400 
2010-01-13 10:44:18.462832 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 8:400 
2010-01-13 10:44:18.582852 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 9:400 
2010-01-13 10:44:23.311740 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF #:400 
2010-01-13 10:44:23.311740 [DEBUG] switch_cpp.cpp:735 getDigits dtmf_buf: 0517289 




It will drop any one or more numbers from any part of that sequence. 


Thanks 
Dan 
(sorry for the duped message) ----- Original Message ----- 
From: "Michael Jerris" <mike at jerris.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 1:41:51 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

What are you doing with these digits on the freeswitch side, what is it thinking is passed, and do you see anything in debug logs. You also could uncomment the #define DEBUG_2833 from switch_rtp.c for more info. 


Mike 




On Jan 13, 2010, at 3:12 PM, Dan wrote: 




Thanks for your reply, 


Wireshark shows all of the digits coming through cleanly, in the case below (also on pastebin) its 0951*72789, freeswitch does not seem to be processing one or more of the rtp events. Is there some specific debugging I should turn on that might give me some more info on why freeswitch is not processing some of these events? 


http://pastebin.freeswitch.org/11780 



449 12.280148 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
450 12.285108 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
451 12.290082 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
452 12.335129 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
453 12.340124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
454 12.345124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
455 12.365162 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21862, Time=359229277, Mark 
456 12.380167 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21863, Time=359229605 
461 12.385248 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
462 12.385253 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
465 12.390163 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
466 12.435211 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
467 12.440201 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
468 12.445205 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
469 12.480251 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21870, Time=359230405 
470 12.485275 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
471 12.485284 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
472 12.490251 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
475 12.535277 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
476 12.540287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
477 12.545287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
478 12.580337 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21877, Time=359231197 
484 12.585337 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
485 12.585343 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
486 12.590362 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
492 12.635393 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
493 12.640368 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
494 12.645369 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
495 12.665406 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21884, Time=359231677, Mark 
496 12.680431 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21885, Time=359232005 
497 12.680438 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
502 12.685408 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
506 12.690412 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
507 12.735450 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
508 12.740453 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
509 12.745469 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
511 12.780526 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21892, Time=359232805 
512 12.780534 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
513 12.785502 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
514 12.790492 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
515 12.835544 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
516 12.840537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
517 12.845537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
523 12.880582 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21899, Time=359233605 
524 12.885593 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
525 12.885599 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
526 12.890580 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
528 12.935618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
529 12.940618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
530 12.945618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
531 12.965656 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21906, Time=359234077, Mark 
532 12.980659 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21907, Time=359234413 
538 12.985690 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
539 12.985695 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
540 12.990657 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
541 13.035703 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
542 13.040701 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
543 13.045700 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
544 13.080762 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21914, Time=359235205 
545 13.080768 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
546 13.085736 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
547 13.090798 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
549 13.135809 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
550 13.140780 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
551 13.145793 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
554 13.180827 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21921, Time=359236005 
558 13.185847 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
559 13.185854 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
560 13.190821 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
566 13.235872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
567 13.240864 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
568 13.245872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) ----- Original Message ----- 
From: "Brian West" < brian at freeswitch.org > 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 12:39:52 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

Cisco gateways can detect DTMF at random if you have kids screaming or loud noises in the background... very unreliable. 


/b 



On Jan 13, 2010, at 1:36 PM, Dan wrote: 


Sorry, I should have made that more clear, this is using 2833. As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through). 

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From anthony.minessale at gmail.com  Thu Jan 14 13:04:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 15:04:00 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001141342.02178.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>

I added bridge_early_media=true to do the best I can do.
This is the most I will do, especially for free, nobody can take a hint that
you should be paying for all these custom requests so take it or leave it
but this thread is done.........



On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> No, not exactly. ignore_early_media doesn't pass early media to the caller
> if
> bypass_media is false.
>
> On Thursday 14 January 2010, Michael Jerris wrote:
> > this is exactly what ignore_early_media does now.
> >
> > Mike
> >
> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> > > The issue here is when "originate" routine should return and
> > > set "originate_status" variable. Current behavior is to return on early
> > > media, but what if to introduce a variable "originate_wait_for_answer"
> > > with default value "false" and use the variable in originate code to
> > > decide when to return - on 18X or "200 OK"?
> > >
> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> > >> he wants to call 3 people at once and let the A leg hear early media
> > >> from call #1 while call #2 and #3 still are progressing which is not
> > >> simple to do without doing thousands of dollars in development.
> > >>
> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
> customers
> > >>> can hear recordings?
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Thu Jan 14 14:01:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 16:01:52 -0600
Subject: [Freeswitch-users] sip trunk question: why call throughexternal
	profile is challenged?
In-Reply-To: <AD524CC0-726B-46D3-A22C-0B1F44C9B899@jerris.com>
References: <20100114082640.2C5D311F32@mail.nstel.ru>
	<AD524CC0-726B-46D3-A22C-0B1F44C9B899@jerris.com>
Message-ID: <191c3a031001141401h4cef36a7p84bac827bc94352f@mail.gmail.com>

try setting param accept-blind-auth to true in your sofia profile config
internal.xml
iirc it was made just for sipX who feels the need to send auth headers even
when nobody asked for them.
so even when auth-calls is false we will still try to parse the auth if one
is sent.


On Thu, Jan 14, 2010 at 11:21 AM, Michael Jerris <mike at jerris.com> wrote:

> if you look in the sample configs for the words blind and auth you will
> find all these settings, also you can setup acls for ip auth to not
> challenge.
>
> Mike
>
> On Jan 14, 2010, at 3:26 AM, Nikolay Kondratyev wrote:
>
>  Mike, thanks for the reply.
>
>
>
> Mmm? looks like I need more detailed instructions where to dig?
>
> Is there a way to turn off ?challenging? completely?
>
> I thought that <param name="auth-calls" value="false"/> should do it, but
> alas?
>
> By the way should this parameter be visible in either ?sofia status profile
> external? or ?sofia status gateway sipx4.lab.nstel.ru? ? I don?t see it?
>
>
>
> I attached traces of failed and successful calls.
>
>
>
> Thanks and regards,
>
> Nikolay.
>   ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Jerris
> *Sent:* Wednesday, January 13, 2010 8:30 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] sip trunk question: why call
> throughexternal profile is challenged?
>
>
>
> Look at how sipx sets up the users when they build the extensions and such
> for conferences, there was something special here, but I can't recall what.
>
>
>
> Mike
>
>
>
> On Jan 13, 2010, at 9:10 AM, Nikolay Kondratyev wrote:
>
>
>
>  Hi all!
>
>
>
> I?m brand new to FreeSwitch, but have some experience with SipX.
>
> Our company is evaluating FS.
>
> For test purposes I set up FS on a virtual machine (vmware esxi). I use
> CentOS.
>
> The FS version I use is 1.0.5-20100110-0400.
>
>
>
> I have a question regarding sip trunk between FS and SipX.
>
> I created the following GW in external profile:
>
> [freeswitch at freeswitch external]$ cat sipx-lab.xml | grep -v '<!--'
>
> <include>
>
>   <gateway name="sipx4.lab.nstel.ru">
>
>   <param name="username" value="zxcv"/>
>
>   <param name="password" value="2007"/>
>
>   <param name="register" value="false"/>
>
>   </gateway>
>
> </include>
>
>
>
> External.xml file is not modified after installation.
>
>
>
> I see this gateway via fs_cli:
>
> freeswitch at internal> sofia status gateway sipx4.lab.nstel.ru
>
>
> =================================================================================================
>
> Name            sipx4.lab.nstel.ru
>
> Profile         external
>
> Scheme          Digest
>
> Realm           sipx4.lab.nstel.ru
>
> Username        zxcv
>
> Password        yes
>
> From            <sip:zxcv at sipx4.lab.nstel.ru;transport=udp>
>
> Contact         <
> sip:gw+sipx4.lab.nstel.ru at 172.23.22.49:5080;transport=udp;gw=sipx4.lab.nstel.ru
> >
>
> Exten           zxcv
>
> To              sip:zxcv at sipx4.lab.nstel.ru
>
> Proxy           sip:sipx4.lab.nstel.ru
>
> Context         public
>
> Expires         3600
>
> Freq            3600
>
> Ping            0
>
> PingFreq        0
>
> PingState       0/0/0
>
> State           NOREG
>
> Status          UP
>
> CallsIN         0
>
> CallsOUT        0
>
>
> =================================================================================================
>
>
>
> I created new FS extension 2853. I registers (xlite) and I can call it.
>
>
>
> Now I want to call FS user from sipx.
>
>
>
> From the sipx side one can configure link to FS differently. There are two
> options:
>
> 1. Call FS directly trough sipxproxy (the core part of sipx works as sip
> proxy, not as b2bua)
>
> 2. Call trough embedded b2bua, named sipxbridge.
>
>
>
> When a call is going trough sipxbridge, it is successfully landed at FS
> extension 2853.
>
> But when a call is going from the sipxproxy, it is challenged with Status
> 407 ?proxy authentication required?, and then call fails.
>
> (I?m not sure if sipx should handle this challenge, but this is separate
> question for the sipx forum).
>
> At the default log level I see nothing in the freeswitch.log.
>
>
>
> So the question is why one call through external profile is being
> challenged while the other is not?
>
> I suspect that the reason is in the difference in the two Invite messages:
>
>
>
> Here is challenged Invite:
>
> INVITE sip:2853 at fs.lab.nstel.ru:5080;transport=udp SIP/2.0
>
> Record-Route: <
> sip:172.23.12.104:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EOTRiZjI3MjgtYWMxNzBjZTktMTNjNC0yZjZkZWMtNjNkNmUwN2UtMmY2ZGVj.900_ntap%2Aid%7EMjIyNjAtMQ%60%60%213ba6d85f6946c4c6001bee1d3b54474f
> >
>
> From: "testphone3857"<sip:3857 at lab.nstel.ru
> >;tag=94bf2728-ac170ce9-13c4-2f6dec-63d6e07e-2f6dec
>
> To: <sip:2853 at lab.nstel.ru>
>
> Call-Id: 94bed2a0-ac170ce9-13c4-2f6dec-68ffdb81-2f6dec at lab.nstel.ru
>
> Cseq: 2 INVITE
>
> Via: SIP/2.0/UDP
> 172.23.12.104;branch=z9hG4bK-sipXecs-000eaf48a36fe4029c7cde004a6f44424847
>
> Via: SIP/2.0/TCP
> 172.23.12.104;branch=z9hG4bK-sipXecs-000bf3008cfa3f3e70470789c75232ba9499~ac7fd729330fd563f83cacd941311e75;id=22260-1
>
> Via: SIP/2.0/UDP 172.23.12.233:5060
> ;branch=z9hG4bK-2f6dec-b9456412-7ad07784
>
> Max-Forwards: 18
>
> Supported: replaces
>
> User-Agent: LG-Nortel LIP 6812 v1.2.38sp SN/00405A18B634
>
> Contact: <sip:3857 at 172.23.12.233:5060;x-sipX-nonat>
>
> Proxy-Authorization: Digest username="3857",realm="lab.nstel.ru
> ",nonce="e2152722611af1bbe59f3a8eb31b8eb84b4db2ca",uri="
> sip:2853 at lab.nstel.ru
> ",response="a09c34fbd897783c9b9af50bd044ccaa",algorithm=MD5
>
> Content-Type: application/sdp
>
> Content-Length: 301
>
> Date: Wed, 13 Jan 2010 11:47:22 GMT
>
> Expires: 60
>
> X-Sipx-Handled: X172.23.12.104-81.211.30.104
>
>
>
> v=0
>
> o=LGEIPP 16246 16247 IN IP4 172.23.12.233
>
> s=SIP Call
>
> c=IN IP4 172.23.12.233
>
> t=0 0
>
> m=audio 23008 RTP/AVP 0 8 18 4 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:18 G729/8000
>
> a=rtpmap:4 G723/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:18 annexb=no
>
> a=fmtp:101 0-11
>
> a=sendrecv
>
>
>
>
>
> And here is successful Invite:
>
> INVITE sip:2853 at fs.lab.nstel.ru;user=phone SIP/2.0
>
> Call-ID: 94bec3b8-ac170ce9-13c4-2f6d47-2fee09d0-2f6d47 at lab.nstel.ru.0
>
> CSeq: 1 INVITE
>
> From: "testphone3857" <sip:3857 at 172.23.12.104>;tag=816640414244159961
>
> To: <sip:2853 at fs.lab.nstel.ru;user=phone>
>
> Via: SIP/2.0/UDP 172.23.12.104:5080
> ;branch=z9hG4bK885dde2c8680f5315845cd3350b8b605373534
>
> Max-Forwards: 70
>
> User-Agent: sipXecs/4.0.2 sipXecs/sipxbridge (Linux)
>
> P-Asserted-Identity: <sip:1004 at 172.23.12.104>
>
> Contact: <sip:1004 at 172.23.12.104:5080;transport=udp>
>
> Route: <sip:172.23.22.49:5080;transport=udp;lr>
>
> Session-Expires: 1800;refresher=uac
>
> Allow: INVITE,BYE,ACK,CANCEL,OPTIONS
>
> Content-Type: application/sdp
>
> Content-Length: 315
>
>
>
> v=0
>
> o=sipxbridge 6640787141824452741 1 IN IP4 172.23.12.104
>
> s=SIP Call
>
> c=IN IP4 172.23.12.104
>
> t=0 0
>
> m=audio 30248 RTP/AVP 0 8 18 4 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:18 G729/8000
>
> a=rtpmap:4 G723/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:18 annexb=no
>
> a=fmtp:101 0-11
>
> a=sendrecv
>
>
>
> Can somebody please if it is a FS configuration problem or a software
> problem or is it a problem on the sipx side?
>
>
>
> Thanks in advance,
>
> Nikolay.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   <call_from_sipx_via_unmanaged_gw.pcap>
> <call_from_sipx_via_siptrunk_sip-only.pcap>
> _______________________________________________
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From mike at jerris.com  Thu Jan 14 14:22:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Jan 2010 17:22:18 -0500
Subject: [Freeswitch-users] sip trunk question: why call throughexternal
	profile is challenged?
In-Reply-To: <191c3a031001141401h4cef36a7p84bac827bc94352f@mail.gmail.com>
References: <20100114082640.2C5D311F32@mail.nstel.ru>
	<AD524CC0-726B-46D3-A22C-0B1F44C9B899@jerris.com>
	<191c3a031001141401h4cef36a7p84bac827bc94352f@mail.gmail.com>
Message-ID: <1BD8DFE1-E3DD-46C3-A383-C9627939BB65@jerris.com>

This now rings a bell, also, if you create a gateway on that profile who's gateway name matches the realm on the incoming auth header, I think that also was working.

Mike

On Jan 14, 2010, at 5:01 PM, Anthony Minessale wrote:

> try setting param accept-blind-auth to true in your sofia profile config internal.xml
> iirc it was made just for sipX who feels the need to send auth headers even when nobody asked for them.
> so even when auth-calls is false we will still try to parse the auth if one is sent.
> 
> 
> On Thu, Jan 14, 2010 at 11:21 AM, Michael Jerris <mike at jerris.com> wrote:
> if you look in the sample configs for the words blind and auth you will find all these settings, also you can setup acls for ip auth to not challenge.
> 
> Mike
> 
> On Jan 14, 2010, at 3:26 AM, Nikolay Kondratyev wrote:
> 
>> Mike, thanks for the reply.
>> 
>>  
>> Mmm? looks like I need more detailed instructions where to dig?
>> 
>> Is there a way to turn off ?challenging? completely?
>> 
>> I thought that <param name="auth-calls" value="false"/> should do it, but alas?
>> 
>> By the way should this parameter be visible in either ?sofia status profile external? or ?sofia status gateway sipx4.lab.nstel.ru? ? I don?t see it?
>> 
>>  
>> I attached traces of failed and successful calls.
>> 

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From harry at vangberg.name  Thu Jan 14 15:06:07 2010
From: harry at vangberg.name (Harry Vangberg)
Date: Fri, 15 Jan 2010 00:06:07 +0100
Subject: [Freeswitch-users] No ringback.
Message-ID: <74d41a3d1001141506oc23ccc0ka41ea0a07ef2c3ce@mail.gmail.com>

Hello.

I am controlling my calls with Freeswitcher via the event socket. When
a call comes in I set the following channel variables:

hangup_after_bridge=true
ignore_early_media=true
call_timeout=20
originate_continue_on_timeout=true
instant_ringback=true
ringback=%(2000,4000,440,480)
transfer_ringback=%(2000,4000,440,480)

Then I pre_answer and bridge to one or more endpoints. The trouble is,
there's no ringback. This is with trunk at 16305. For reference, here's
excerpt from the log:

2010-01-14 22:47:14.540747 [NOTICE] switch_channel.c:642 New Channel
sofia/external/31354228 at 129.142.224.250
[c27529c2-015e-11df-8d0d-c7205d2c96c7]
2010-01-14 22:47:14.540747 [DEBUG] sofia.c:3897 Channel
sofia/external/31354228 at 129.142.224.250 entering state [received][100]
2010-01-14 22:47:14.540747 [DEBUG] sofia.c:3908 Remote SDP:
v=0
o=root 22377 22377 IN IP4 129.142.224.250
s=session
c=IN IP4 129.142.224.250
t=0 0
m=audio 14962 RTP/AVP 8 0 3 97 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

2010-01-14 22:47:14.540747 [DEBUG] sofia_glue.c:3344 Audio Codec
Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
2010-01-14 22:47:14.540747 [DEBUG] sofia_glue.c:2171 Set Codec
sofia/external/31354228 at 129.142.224.250 PCMA/8000 20 ms 160 samples
2010-01-14 22:47:14.540747 [DEBUG] sofia_glue.c:3300 Set 2833 dtmf
payload to 101
2010-01-14 22:47:14.540747 [DEBUG] sofia.c:4056
(sofia/external/31354228 at 129.142.224.250) State Change CS_NEW ->
CS_INIT
2010-01-14 22:47:14.540747 [DEBUG] switch_core_session.c:1012 Send
signal sofia/external/31354228 at 129.142.224.250 [BREAK]
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:314
(sofia/external/31354228 at 129.142.224.250) Running State Change CS_INIT
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:338
(sofia/external/31354228 at 129.142.224.250) State INIT
2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:83
sofia/external/31354228 at 129.142.224.250 SOFIA INIT
2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:111
(sofia/external/31354228 at 129.142.224.250) State Change CS_INIT ->
CS_ROUTING
2010-01-14 22:47:14.540747 [DEBUG] switch_core_session.c:1012 Send
signal sofia/external/31354228 at 129.142.224.250 [BREAK]
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:338
(sofia/external/31354228 at 129.142.224.250) State INIT going to sleep
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:314
(sofia/external/31354228 at 129.142.224.250) Running State Change
CS_ROUTING
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:341
(sofia/external/31354228 at 129.142.224.250) State ROUTING
2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:132
sofia/external/31354228 at 129.142.224.250 SOFIA ROUTING
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:78
sofia/external/31354228 at 129.142.224.250 Standard ROUTING
2010-01-14 22:47:14.540747 [INFO] mod_dialplan_xml.c:408 Processing
31354228->77344536 in context public
Dialplan: sofia/external/31354228 at 129.142.224.250 parsing
[public->ff-phone-sound-recorder] continue=false
Dialplan: sofia/external/31354228 at 129.142.224.250 Regex (FAIL)
[ff-phone-sound-recorder] destination_number(77344536) =~ /^77348624$/
break=on-false
Dialplan: sofia/external/31354228 at 129.142.224.250 parsing
[public->ff-ivr] continue=false
Dialplan: sofia/external/31354228 at 129.142.224.250 Regex (PASS)
[ff-ivr] destination_number(77344536) =~ /^(\d+)$/ break=on-false
Dialplan: sofia/external/31354228 at 129.142.224.250 Action
socket(127.0.0.1:8084 full)
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:122
(sofia/external/31354228 at 129.142.224.250) State Change CS_ROUTING ->
CS_EXECUTE
2010-01-14 22:47:14.540747 [DEBUG] switch_core_session.c:1012 Send
signal sofia/external/31354228 at 129.142.224.250 [BREAK]
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:341
(sofia/external/31354228 at 129.142.224.250) State ROUTING going to sleep
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:314
(sofia/external/31354228 at 129.142.224.250) Running State Change
CS_EXECUTE
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:348
(sofia/external/31354228 at 129.142.224.250) State EXECUTE
2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:181
sofia/external/31354228 at 129.142.224.250 SOFIA EXECUTE
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:159
sofia/external/31354228 at 129.142.224.250 Standard EXECUTE
EXECUTE sofia/external/31354228 at 129.142.224.250 socket(127.0.0.1:8084 full)
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
set(hangup_after_bridge=true)
EXECUTE sofia/external/31354228 at 129.142.224.250 set(hangup_after_bridge=true)
2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
sofia/external/31354228 at 129.142.224.250 SET
[hangup_after_bridge]=[true]
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
set(ignore_early_media=true)
EXECUTE sofia/external/31354228 at 129.142.224.250 set(ignore_early_media=true)
2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
sofia/external/31354228 at 129.142.224.250 SET
[ignore_early_media]=[true]
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
set(call_timeout=20)
EXECUTE sofia/external/31354228 at 129.142.224.250 set(call_timeout=20)
2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
sofia/external/31354228 at 129.142.224.250 SET [call_timeout]=[20]
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
set(originate_continue_on_timeout=true)
EXECUTE sofia/external/31354228 at 129.142.224.250
set(originate_continue_on_timeout=true)
2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
sofia/external/31354228 at 129.142.224.250 SET
[originate_continue_on_timeout]=[true]
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
set(instant_ringback=true)
EXECUTE sofia/external/31354228 at 129.142.224.250 set(instant_ringback=true)
2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
sofia/external/31354228 at 129.142.224.250 SET [instant_ringback]=[true]
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
set(ringback=%(2000,4000,440,480))
EXECUTE sofia/external/31354228 at 129.142.224.250
set(ringback=%(2000,4000,440,480))
2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
sofia/external/31354228 at 129.142.224.250 SET
[ringback]=[%(2000,4000,440,480)]
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
set(transfer_ringback=%(2000,4000,440,480))
EXECUTE sofia/external/31354228 at 129.142.224.250
set(transfer_ringback=%(2000,4000,440,480))
2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
sofia/external/31354228 at 129.142.224.250 SET
[transfer_ringback]=[%(2000,4000,440,480)]
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
bind_meta_app(6 b ai
transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
public)
EXECUTE sofia/external/31354228 at 129.142.224.250 bind_meta_app(6 b ai
transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
public)
2010-01-14 22:47:14.792781 [INFO] switch_ivr_async.c:2322 Bound B-Leg:
6 transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
public
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
bind_meta_app(7 b ai transfer::ff-transfer- XML public)
EXECUTE sofia/external/31354228 at 129.142.224.250 bind_meta_app(7 b ai
transfer::ff-transfer- XML public)
2010-01-14 22:47:14.792781 [INFO] switch_ivr_async.c:2322 Bound B-Leg:
7 transfer::ff-transfer- XML public
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
bind_meta_app(1 b ai
transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
public)
EXECUTE sofia/external/31354228 at 129.142.224.250 bind_meta_app(1 b ai
transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
public)
2010-01-14 22:47:14.792781 [INFO] switch_ivr_async.c:2322 Bound B-Leg:
1 transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
public
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute pre_answer()
EXECUTE sofia/external/31354228 at 129.142.224.250 pre_answer()
2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:736
sofia/external/31354228 at 129.142.224.250 receive message [PROGRESS]
2010-01-14 22:47:14.792781 [INFO] mod_dptools.c:736 Sending early media
2010-01-14 22:47:14.792781 [DEBUG] sofia_glue.c:2409 AUDIO RTP
[sofia/external/31354228 at 129.142.224.250] 10.224.54.102 port 22822 ->
129.142.224.250 port 14962 codec: 8 ms: 20
2010-01-14 22:47:14.792781 [DEBUG] switch_rtp.c:1168 Starting timer
[soft] 160 bytes per 20ms
2010-01-14 22:47:14.807902 [INFO] mod_sofia.c:1779 Ring SDP:
v=0
o=FreeSWITCH 1263486412 1263486413 IN IP4 79.125.35.134
s=FreeSWITCH
c=IN IP4 79.125.35.134
t=0 0
m=audio 22822 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2010-01-14 22:47:14.807902 [NOTICE] mod_sofia.c:1782 Pre-Answer
sofia/external/31354228 at 129.142.224.250!
2010-01-14 22:47:14.807902 [DEBUG] switch_core_session.c:633 Send
signal sofia/external/31354228 at 129.142.224.250 [BREAK]
2010-01-14 22:47:14.827576 [DEBUG] sofia.c:3892 Channel
sofia/external/31354228 at 129.142.224.250 skipping state [early][183]
2010-01-14 22:47:14.827576 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
bridge({sip_cid_type=rpid,origination_caller_id_name=77344536,origination_caller_id_number=77344536}sofia/gateway/gw1.fonet.dk/004546934488|loopback/ff-record-77344536)
EXECUTE sofia/external/31354228 at 129.142.224.250
bridge({sip_cid_type=rpid,origination_caller_id_name=77344536,origination_caller_id_number=77344536}sofia/gateway/gw1.fonet.dk/004546934488|loopback/ff-record-77344536)
2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:1738
variable string 0 = [sip_cid_type=rpid]
2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:1738
variable string 1 = [origination_caller_id_name=77344536]
2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:1738
variable string 2 = [origination_caller_id_number=77344536]
2010-01-14 22:47:14.827576 [NOTICE] switch_channel.c:642 New Channel
sofia/external/004546934488 [c2a1095c-015e-11df-8d0e-c7205d2c96c7]
2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:3227
(sofia/external/004546934488) State Change CS_NEW -> CS_INIT
2010-01-14 22:47:14.827576 [DEBUG] switch_core_session.c:1012 Send
signal sofia/external/004546934488 [BREAK]
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:314
(sofia/external/004546934488) Running State Change CS_INIT
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:338
(sofia/external/004546934488) State INIT
2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:83
sofia/external/004546934488 SOFIA INIT
2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:111
(sofia/external/004546934488) State Change CS_INIT -> CS_ROUTING
2010-01-14 22:47:14.827576 [DEBUG] switch_core_session.c:1012 Send
signal sofia/external/004546934488 [BREAK]
2010-01-14 22:47:14.827576 [DEBUG] sofia.c:3897 Channel
sofia/external/004546934488 entering state [calling][0]
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:338
(sofia/external/004546934488) State INIT going to sleep
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:314
(sofia/external/004546934488) Running State Change CS_ROUTING
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:341
(sofia/external/004546934488) State ROUTING
2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:132
sofia/external/004546934488 SOFIA ROUTING
2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:66
(sofia/external/004546934488) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2010-01-14 22:47:14.827576 [DEBUG] switch_core_session.c:1012 Send
signal sofia/external/004546934488 [BREAK]
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:341
(sofia/external/004546934488) State ROUTING going to sleep
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:314
(sofia/external/004546934488) Running State Change CS_CONSUME_MEDIA
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:360
(sofia/external/004546934488) State CONSUME_MEDIA
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:360
(sofia/external/004546934488) State CONSUME_MEDIA going to sleep
2010-01-14 22:47:14.876945 [DEBUG] switch_ivr_originate.c:1057 Raw
Codec Activation Success L16 at 8000hz 1 channel 20ms
2010-01-14 22:47:14.876945 [DEBUG] switch_core_codec.c:112
sofia/external/31354228 at 129.142.224.250 Push codec L16:10
2010-01-14 22:47:14.876945 [DEBUG] switch_ivr_originate.c:1119 Play
Ringback Tone [%(2000,4000,440,480)]
2010-01-14 22:47:14.876945 [DEBUG] switch_core_io.c:652
sofia/external/31354228 at 129.142.224.250 receive message
[TRANSCODING_NECESSARY]
2010-01-14 22:47:14.876945 [DEBUG] sofia.c:3897 Channel
sofia/external/004546934488 entering state [calling][0]
2010-01-14 22:47:14.912774 [DEBUG] switch_rtp.c:2022 Correct ip/port confirmed.
2010-01-14 22:47:15.668817 [INFO] sofia.c:514 Update Callee ID to
"004546934488" <004546934488>
2010-01-14 22:47:15.668817 [DEBUG] sofia.c:3897 Channel
sofia/external/004546934488 entering state [proceeding][180]
2010-01-14 22:47:15.668817 [NOTICE] sofia.c:3967 Ring-Ready
sofia/external/004546934488!
2010-01-14 22:47:15.668817 [DEBUG] sofia.c:3897 Channel
sofia/external/004546934488 entering state [proceeding][183]
2010-01-14 22:47:15.668817 [DEBUG] sofia.c:3908 Remote SDP:
v=0
o=root 22377 22377 IN IP4 129.142.224.250
s=session
c=IN IP4 129.142.224.250
t=0 0
m=audio 11676 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20

2010-01-14 22:47:15.668817 [DEBUG] sofia_glue.c:3344 Audio Codec
Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
2010-01-14 22:47:15.668817 [DEBUG] sofia_glue.c:2171 Set Codec
sofia/external/004546934488 PCMA/8000 20 ms 160 samples
2010-01-14 22:47:15.668817 [DEBUG] sofia_glue.c:2409 AUDIO RTP
[sofia/external/004546934488] 10.224.54.102 port 17204 ->
129.142.224.250 port 11676 codec: 8 ms: 20
2010-01-14 22:47:15.668817 [DEBUG] switch_rtp.c:1168 Starting timer
[soft] 160 bytes per 20ms
2010-01-14 22:47:15.668817 [NOTICE] sofia_glue.c:2937 Pre-Answer
sofia/external/004546934488!
2010-01-14 22:47:15.668817 [DEBUG] switch_channel.c:2053 Send signal
sofia/external/31354228 at 129.142.224.250 [BREAK]

-- 
Harry Vangberg  <harry at vangberg.name>  http://harry.vangberg.name


From david.villasmil.work at gmail.com  Thu Jan 14 15:06:59 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 00:06:59 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
Message-ID: <9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>

Anthony,

I did take the "hint", don't worry. We will probably ask for a bounty
but first we need to know:
1.- whether this is possible
2.- how long it would take
3.- how will it exactly work
4.- of course, what's the bounty (be gentle ;) )

We would of course give this back to the community.

in the meantime, bridge_early_media=true will discard the 2nd B-leg
and conversely "false" it will keep on trying to connect and if it
connects the other B-leg if will bridge to that one?

Thanks

David

On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I added bridge_early_media=true to do the best I can do.
> This is the most I will do, especially for free, nobody can take a hint that
> you should be paying for all these custom requests so take it or leave it
> but this thread is done.........
>
>
>
> On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
> wrote:
>>
>> No, not exactly. ignore_early_media doesn't pass early media to the caller
>> if
>> bypass_media is false.
>>
>> On Thursday 14 January 2010, Michael Jerris wrote:
>> > this is exactly what ignore_early_media does now.
>> >
>> > Mike
>> >
>> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> > > The issue here is when "originate" routine should return and
>> > > set "originate_status" variable. Current behavior is to return on
>> > > early
>> > > media, but what if to introduce a variable "originate_wait_for_answer"
>> > > with default value "false" and use the variable in originate code to
>> > > decide when to return - on 18X or "200 OK"?
>> > >
>> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> > >> he wants to call 3 people at once and let the A leg hear early media
>> > >> from call #1 while call #2 and #3 still are progressing which is not
>> > >> simple to do without doing thousands of dollars in development.
>> > >>
>> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>> > >>> customers
>> > >>> can hear recordings?
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From freeswitch-users at digitaldan.com  Thu Jan 14 15:28:47 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Thu, 14 Jan 2010 16:28:47 -0700 (MST)
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <13392952.131263501034741.JavaMail.root@zimbra>
Message-ID: <13986368.261263511726829.JavaMail.root@zimbra>

Actually that was not it, I had also commented out <param name="rtp-timer-name" value="soft"/> in the external.xml sip profile which is what fixed my issue. 


My question then is what does disabling the setting do? Is there any harm in not having it? 





With it disabled, I get all the DTMF digits, but I also get the following " Failed DTMF payload check." lines in my log. 



2010-01-14 16:20:02.430026 [NOTICE] switch_cpp.cpp:1129 main: Getting digits 
2010-01-14 16:20:04.246261 [ERR] switch_rtp.c:2196 Failed DTMF payload check. 
2010-01-14 16:20:04.251201 [ERR] switch_rtp.c:2196 Failed DTMF payload check. 
2010-01-14 16:20:04.256198 [ERR] switch_rtp.c:2196 Failed DTMF payload check. 
2010-01-14 16:20:04.296267 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 0:400 
2010-01-14 16:20:04.396331 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 9:400 
2010-01-14 16:20:04.496411 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 5:400 
2010-01-14 16:20:04.596505 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 1:400 
2010-01-14 16:20:04.696581 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF *:400 
2010-01-14 16:20:04.796660 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 7:400 
2010-01-14 16:20:04.896769 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 2:400 
2010-01-14 16:20:04.996830 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 7:400 
2010-01-14 16:20:05.096906 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 8:400 
2010-01-14 16:20:05.196993 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 9:400 
2010-01-14 16:20:09.950844 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF #:400 
2010-01-14 16:20:09.951064 [DEBUG] switch_cpp.cpp:735 getDigits dtmf_buf: 0951*72789 

----- Original Message ----- 
From: "Dan" <freeswitch-users at digitaldan.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Thursday, January 14, 2010 1:30:34 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 


So I did figure it out, it ended up being the CISCO_SKIP_MARK_BIT_2833 issue mentioned on the RTP troubleshooting wiki page. I had incorrectly added this to my dialplan rather then to the external sip profile where it belonged. 


Thanks 
Dan- 

----- Original Message ----- 
From: "Dan" <freeswitch-users at digitaldan.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 1:56:16 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 


Thanks, I will try and turn up the debugging on it. I have a lua script that is calling session:getDigits(20,"#",15000) . When I watch it under fs_cli I'm getting this output 



2010-01-13 10:44:15.750329 [NOTICE] switch_cpp.cpp:1129 main: Getting digits 
2010-01-13 10:44:17.586676 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 0:400 
2010-01-13 10:44:17.790705 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 5:400 
2010-01-13 10:44:17.910737 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 1:400 
2010-01-13 10:44:18.114779 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 7:400 
2010-01-13 10:44:18.234812 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 2:400 
2010-01-13 10:44:18.462832 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 8:400 
2010-01-13 10:44:18.582852 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 9:400 
2010-01-13 10:44:23.311740 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF #:400 
2010-01-13 10:44:23.311740 [DEBUG] switch_cpp.cpp:735 getDigits dtmf_buf: 0517289 




It will drop any one or more numbers from any part of that sequence. 


Thanks 
Dan 
(sorry for the duped message) ----- Original Message ----- 
From: "Michael Jerris" <mike at jerris.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 1:41:51 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

What are you doing with these digits on the freeswitch side, what is it thinking is passed, and do you see anything in debug logs. You also could uncomment the #define DEBUG_2833 from switch_rtp.c for more info. 


Mike 




On Jan 13, 2010, at 3:12 PM, Dan wrote: 




Thanks for your reply, 


Wireshark shows all of the digits coming through cleanly, in the case below (also on pastebin) its 0951*72789, freeswitch does not seem to be processing one or more of the rtp events. Is there some specific debugging I should turn on that might give me some more info on why freeswitch is not processing some of these events? 


http://pastebin.freeswitch.org/11780 



449 12.280148 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
450 12.285108 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
451 12.290082 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
452 12.335129 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
453 12.340124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
454 12.345124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
455 12.365162 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21862, Time=359229277, Mark 
456 12.380167 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21863, Time=359229605 
461 12.385248 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
462 12.385253 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
465 12.390163 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
466 12.435211 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
467 12.440201 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
468 12.445205 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
469 12.480251 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21870, Time=359230405 
470 12.485275 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
471 12.485284 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
472 12.490251 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
475 12.535277 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
476 12.540287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
477 12.545287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
478 12.580337 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21877, Time=359231197 
484 12.585337 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
485 12.585343 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
486 12.590362 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
492 12.635393 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
493 12.640368 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
494 12.645369 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
495 12.665406 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21884, Time=359231677, Mark 
496 12.680431 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21885, Time=359232005 
497 12.680438 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
502 12.685408 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
506 12.690412 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
507 12.735450 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
508 12.740453 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
509 12.745469 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
511 12.780526 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21892, Time=359232805 
512 12.780534 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
513 12.785502 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
514 12.790492 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
515 12.835544 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
516 12.840537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
517 12.845537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
523 12.880582 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21899, Time=359233605 
524 12.885593 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
525 12.885599 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
526 12.890580 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
528 12.935618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
529 12.940618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
530 12.945618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
531 12.965656 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21906, Time=359234077, Mark 
532 12.980659 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21907, Time=359234413 
538 12.985690 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
539 12.985695 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
540 12.990657 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
541 13.035703 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
542 13.040701 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
543 13.045700 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
544 13.080762 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21914, Time=359235205 
545 13.080768 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
546 13.085736 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
547 13.090798 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
549 13.135809 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
550 13.140780 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
551 13.145793 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
554 13.180827 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21921, Time=359236005 
558 13.185847 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
559 13.185854 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
560 13.190821 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
566 13.235872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
567 13.240864 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
568 13.245872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) ----- Original Message ----- 
From: "Brian West" < brian at freeswitch.org > 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 12:39:52 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

Cisco gateways can detect DTMF at random if you have kids screaming or loud noises in the background... very unreliable. 


/b 



On Jan 13, 2010, at 1:36 PM, Dan wrote: 


Sorry, I should have made that more clear, this is using 2833. As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through). 

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From harry at vangberg.name  Thu Jan 14 15:44:08 2010
From: harry at vangberg.name (Harry Vangberg)
Date: Fri, 15 Jan 2010 00:44:08 +0100
Subject: [Freeswitch-users] No ringback.
In-Reply-To: <74d41a3d1001141506oc23ccc0ka41ea0a07ef2c3ce@mail.gmail.com>
References: <74d41a3d1001141506oc23ccc0ka41ea0a07ef2c3ce@mail.gmail.com>
Message-ID: <74d41a3d1001141544h496c9412s69e50fd76b70ce84@mail.gmail.com>

Oh, damn. I need answer, not pre_answer. Yay.

2010/1/15 Harry Vangberg <harry at vangberg.name>:
> Hello.
>
> I am controlling my calls with Freeswitcher via the event socket. When
> a call comes in I set the following channel variables:
>
> hangup_after_bridge=true
> ignore_early_media=true
> call_timeout=20
> originate_continue_on_timeout=true
> instant_ringback=true
> ringback=%(2000,4000,440,480)
> transfer_ringback=%(2000,4000,440,480)
>
> Then I pre_answer and bridge to one or more endpoints. The trouble is,
> there's no ringback. This is with trunk at 16305. For reference, here's
> excerpt from the log:
>
> 2010-01-14 22:47:14.540747 [NOTICE] switch_channel.c:642 New Channel
> sofia/external/31354228 at 129.142.224.250
> [c27529c2-015e-11df-8d0d-c7205d2c96c7]
> 2010-01-14 22:47:14.540747 [DEBUG] sofia.c:3897 Channel
> sofia/external/31354228 at 129.142.224.250 entering state [received][100]
> 2010-01-14 22:47:14.540747 [DEBUG] sofia.c:3908 Remote SDP:
> v=0
> o=root 22377 22377 IN IP4 129.142.224.250
> s=session
> c=IN IP4 129.142.224.250
> t=0 0
> m=audio 14962 RTP/AVP 8 0 3 97 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
> 2010-01-14 22:47:14.540747 [DEBUG] sofia_glue.c:3344 Audio Codec
> Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
> 2010-01-14 22:47:14.540747 [DEBUG] sofia_glue.c:2171 Set Codec
> sofia/external/31354228 at 129.142.224.250 PCMA/8000 20 ms 160 samples
> 2010-01-14 22:47:14.540747 [DEBUG] sofia_glue.c:3300 Set 2833 dtmf
> payload to 101
> 2010-01-14 22:47:14.540747 [DEBUG] sofia.c:4056
> (sofia/external/31354228 at 129.142.224.250) State Change CS_NEW ->
> CS_INIT
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_session.c:1012 Send
> signal sofia/external/31354228 at 129.142.224.250 [BREAK]
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/31354228 at 129.142.224.250) Running State Change CS_INIT
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:338
> (sofia/external/31354228 at 129.142.224.250) State INIT
> 2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:83
> sofia/external/31354228 at 129.142.224.250 SOFIA INIT
> 2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:111
> (sofia/external/31354228 at 129.142.224.250) State Change CS_INIT ->
> CS_ROUTING
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_session.c:1012 Send
> signal sofia/external/31354228 at 129.142.224.250 [BREAK]
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:338
> (sofia/external/31354228 at 129.142.224.250) State INIT going to sleep
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/31354228 at 129.142.224.250) Running State Change
> CS_ROUTING
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/31354228 at 129.142.224.250) State ROUTING
> 2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:132
> sofia/external/31354228 at 129.142.224.250 SOFIA ROUTING
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:78
> sofia/external/31354228 at 129.142.224.250 Standard ROUTING
> 2010-01-14 22:47:14.540747 [INFO] mod_dialplan_xml.c:408 Processing
> 31354228->77344536 in context public
> Dialplan: sofia/external/31354228 at 129.142.224.250 parsing
> [public->ff-phone-sound-recorder] continue=false
> Dialplan: sofia/external/31354228 at 129.142.224.250 Regex (FAIL)
> [ff-phone-sound-recorder] destination_number(77344536) =~ /^77348624$/
> break=on-false
> Dialplan: sofia/external/31354228 at 129.142.224.250 parsing
> [public->ff-ivr] continue=false
> Dialplan: sofia/external/31354228 at 129.142.224.250 Regex (PASS)
> [ff-ivr] destination_number(77344536) =~ /^(\d+)$/ break=on-false
> Dialplan: sofia/external/31354228 at 129.142.224.250 Action
> socket(127.0.0.1:8084 full)
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:122
> (sofia/external/31354228 at 129.142.224.250) State Change CS_ROUTING ->
> CS_EXECUTE
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_session.c:1012 Send
> signal sofia/external/31354228 at 129.142.224.250 [BREAK]
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/31354228 at 129.142.224.250) State ROUTING going to sleep
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/31354228 at 129.142.224.250) Running State Change
> CS_EXECUTE
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:348
> (sofia/external/31354228 at 129.142.224.250) State EXECUTE
> 2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:181
> sofia/external/31354228 at 129.142.224.250 SOFIA EXECUTE
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:159
> sofia/external/31354228 at 129.142.224.250 Standard EXECUTE
> EXECUTE sofia/external/31354228 at 129.142.224.250 socket(127.0.0.1:8084 full)
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> set(hangup_after_bridge=true)
> EXECUTE sofia/external/31354228 at 129.142.224.250 set(hangup_after_bridge=true)
> 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
> sofia/external/31354228 at 129.142.224.250 SET
> [hangup_after_bridge]=[true]
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> set(ignore_early_media=true)
> EXECUTE sofia/external/31354228 at 129.142.224.250 set(ignore_early_media=true)
> 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
> sofia/external/31354228 at 129.142.224.250 SET
> [ignore_early_media]=[true]
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> set(call_timeout=20)
> EXECUTE sofia/external/31354228 at 129.142.224.250 set(call_timeout=20)
> 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
> sofia/external/31354228 at 129.142.224.250 SET [call_timeout]=[20]
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> set(originate_continue_on_timeout=true)
> EXECUTE sofia/external/31354228 at 129.142.224.250
> set(originate_continue_on_timeout=true)
> 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
> sofia/external/31354228 at 129.142.224.250 SET
> [originate_continue_on_timeout]=[true]
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> set(instant_ringback=true)
> EXECUTE sofia/external/31354228 at 129.142.224.250 set(instant_ringback=true)
> 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
> sofia/external/31354228 at 129.142.224.250 SET [instant_ringback]=[true]
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> set(ringback=%(2000,4000,440,480))
> EXECUTE sofia/external/31354228 at 129.142.224.250
> set(ringback=%(2000,4000,440,480))
> 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
> sofia/external/31354228 at 129.142.224.250 SET
> [ringback]=[%(2000,4000,440,480)]
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> set(transfer_ringback=%(2000,4000,440,480))
> EXECUTE sofia/external/31354228 at 129.142.224.250
> set(transfer_ringback=%(2000,4000,440,480))
> 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
> sofia/external/31354228 at 129.142.224.250 SET
> [transfer_ringback]=[%(2000,4000,440,480)]
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> bind_meta_app(6 b ai
> transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
> public)
> EXECUTE sofia/external/31354228 at 129.142.224.250 bind_meta_app(6 b ai
> transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
> public)
> 2010-01-14 22:47:14.792781 [INFO] switch_ivr_async.c:2322 Bound B-Leg:
> 6 transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
> public
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> bind_meta_app(7 b ai transfer::ff-transfer- XML public)
> EXECUTE sofia/external/31354228 at 129.142.224.250 bind_meta_app(7 b ai
> transfer::ff-transfer- XML public)
> 2010-01-14 22:47:14.792781 [INFO] switch_ivr_async.c:2322 Bound B-Leg:
> 7 transfer::ff-transfer- XML public
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> bind_meta_app(1 b ai
> transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
> public)
> EXECUTE sofia/external/31354228 at 129.142.224.250 bind_meta_app(1 b ai
> transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
> public)
> 2010-01-14 22:47:14.792781 [INFO] switch_ivr_async.c:2322 Bound B-Leg:
> 1 transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
> public
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute pre_answer()
> EXECUTE sofia/external/31354228 at 129.142.224.250 pre_answer()
> 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:736
> sofia/external/31354228 at 129.142.224.250 receive message [PROGRESS]
> 2010-01-14 22:47:14.792781 [INFO] mod_dptools.c:736 Sending early media
> 2010-01-14 22:47:14.792781 [DEBUG] sofia_glue.c:2409 AUDIO RTP
> [sofia/external/31354228 at 129.142.224.250] 10.224.54.102 port 22822 ->
> 129.142.224.250 port 14962 codec: 8 ms: 20
> 2010-01-14 22:47:14.792781 [DEBUG] switch_rtp.c:1168 Starting timer
> [soft] 160 bytes per 20ms
> 2010-01-14 22:47:14.807902 [INFO] mod_sofia.c:1779 Ring SDP:
> v=0
> o=FreeSWITCH 1263486412 1263486413 IN IP4 79.125.35.134
> s=FreeSWITCH
> c=IN IP4 79.125.35.134
> t=0 0
> m=audio 22822 RTP/AVP 8
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> 2010-01-14 22:47:14.807902 [NOTICE] mod_sofia.c:1782 Pre-Answer
> sofia/external/31354228 at 129.142.224.250!
> 2010-01-14 22:47:14.807902 [DEBUG] switch_core_session.c:633 Send
> signal sofia/external/31354228 at 129.142.224.250 [BREAK]
> 2010-01-14 22:47:14.827576 [DEBUG] sofia.c:3892 Channel
> sofia/external/31354228 at 129.142.224.250 skipping state [early][183]
> 2010-01-14 22:47:14.827576 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> bridge({sip_cid_type=rpid,origination_caller_id_name=77344536,origination_caller_id_number=77344536}sofia/gateway/gw1.fonet.dk/004546934488|loopback/ff-record-77344536)
> EXECUTE sofia/external/31354228 at 129.142.224.250
> bridge({sip_cid_type=rpid,origination_caller_id_name=77344536,origination_caller_id_number=77344536}sofia/gateway/gw1.fonet.dk/004546934488|loopback/ff-record-77344536)
> 2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:1738
> variable string 0 = [sip_cid_type=rpid]
> 2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:1738
> variable string 1 = [origination_caller_id_name=77344536]
> 2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:1738
> variable string 2 = [origination_caller_id_number=77344536]
> 2010-01-14 22:47:14.827576 [NOTICE] switch_channel.c:642 New Channel
> sofia/external/004546934488 [c2a1095c-015e-11df-8d0e-c7205d2c96c7]
> 2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:3227
> (sofia/external/004546934488) State Change CS_NEW -> CS_INIT
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_session.c:1012 Send
> signal sofia/external/004546934488 [BREAK]
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/004546934488) Running State Change CS_INIT
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:338
> (sofia/external/004546934488) State INIT
> 2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:83
> sofia/external/004546934488 SOFIA INIT
> 2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:111
> (sofia/external/004546934488) State Change CS_INIT -> CS_ROUTING
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_session.c:1012 Send
> signal sofia/external/004546934488 [BREAK]
> 2010-01-14 22:47:14.827576 [DEBUG] sofia.c:3897 Channel
> sofia/external/004546934488 entering state [calling][0]
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:338
> (sofia/external/004546934488) State INIT going to sleep
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/004546934488) Running State Change CS_ROUTING
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/004546934488) State ROUTING
> 2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:132
> sofia/external/004546934488 SOFIA ROUTING
> 2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:66
> (sofia/external/004546934488) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_session.c:1012 Send
> signal sofia/external/004546934488 [BREAK]
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/004546934488) State ROUTING going to sleep
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/004546934488) Running State Change CS_CONSUME_MEDIA
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:360
> (sofia/external/004546934488) State CONSUME_MEDIA
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:360
> (sofia/external/004546934488) State CONSUME_MEDIA going to sleep
> 2010-01-14 22:47:14.876945 [DEBUG] switch_ivr_originate.c:1057 Raw
> Codec Activation Success L16 at 8000hz 1 channel 20ms
> 2010-01-14 22:47:14.876945 [DEBUG] switch_core_codec.c:112
> sofia/external/31354228 at 129.142.224.250 Push codec L16:10
> 2010-01-14 22:47:14.876945 [DEBUG] switch_ivr_originate.c:1119 Play
> Ringback Tone [%(2000,4000,440,480)]
> 2010-01-14 22:47:14.876945 [DEBUG] switch_core_io.c:652
> sofia/external/31354228 at 129.142.224.250 receive message
> [TRANSCODING_NECESSARY]
> 2010-01-14 22:47:14.876945 [DEBUG] sofia.c:3897 Channel
> sofia/external/004546934488 entering state [calling][0]
> 2010-01-14 22:47:14.912774 [DEBUG] switch_rtp.c:2022 Correct ip/port confirmed.
> 2010-01-14 22:47:15.668817 [INFO] sofia.c:514 Update Callee ID to
> "004546934488" <004546934488>
> 2010-01-14 22:47:15.668817 [DEBUG] sofia.c:3897 Channel
> sofia/external/004546934488 entering state [proceeding][180]
> 2010-01-14 22:47:15.668817 [NOTICE] sofia.c:3967 Ring-Ready
> sofia/external/004546934488!
> 2010-01-14 22:47:15.668817 [DEBUG] sofia.c:3897 Channel
> sofia/external/004546934488 entering state [proceeding][183]
> 2010-01-14 22:47:15.668817 [DEBUG] sofia.c:3908 Remote SDP:
> v=0
> o=root 22377 22377 IN IP4 129.142.224.250
> s=session
> c=IN IP4 129.142.224.250
> t=0 0
> m=audio 11676 RTP/AVP 8
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
>
> 2010-01-14 22:47:15.668817 [DEBUG] sofia_glue.c:3344 Audio Codec
> Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
> 2010-01-14 22:47:15.668817 [DEBUG] sofia_glue.c:2171 Set Codec
> sofia/external/004546934488 PCMA/8000 20 ms 160 samples
> 2010-01-14 22:47:15.668817 [DEBUG] sofia_glue.c:2409 AUDIO RTP
> [sofia/external/004546934488] 10.224.54.102 port 17204 ->
> 129.142.224.250 port 11676 codec: 8 ms: 20
> 2010-01-14 22:47:15.668817 [DEBUG] switch_rtp.c:1168 Starting timer
> [soft] 160 bytes per 20ms
> 2010-01-14 22:47:15.668817 [NOTICE] sofia_glue.c:2937 Pre-Answer
> sofia/external/004546934488!
> 2010-01-14 22:47:15.668817 [DEBUG] switch_channel.c:2053 Send signal
> sofia/external/31354228 at 129.142.224.250 [BREAK]
>
> --
> Harry Vangberg ?<harry at vangberg.name> ?http://harry.vangberg.name
>



-- 
Harry Vangberg  <harry at vangberg.name>  http://harry.vangberg.name


From anthony.minessale at gmail.com  Thu Jan 14 15:54:22 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 17:54:22 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
Message-ID: <191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>

{bridge_early_media=true}
in the dial string in place of ignore_early_media=true

the first b leg in the list who sends 183 will become the ringback device
for A leg it will hear the early media
for that leg while the other legs still ring.  If some other leg answers the
final call will still be bridged to the leg who answered.


I would estimate it at $500 payable on the big paypal button on
http://www.freeswitch.org
but, I already added the patch to tree earlier today so I guess it's up to
you to pay it or not.




On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Anthony,
>
> I did take the "hint", don't worry. We will probably ask for a bounty
> but first we need to know:
> 1.- whether this is possible
> 2.- how long it would take
> 3.- how will it exactly work
> 4.- of course, what's the bounty (be gentle ;) )
>
> We would of course give this back to the community.
>
> in the meantime, bridge_early_media=true will discard the 2nd B-leg
> and conversely "false" it will keep on trying to connect and if it
> connects the other B-leg if will bridge to that one?
>
> Thanks
>
> David
>
> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > I added bridge_early_media=true to do the best I can do.
> > This is the most I will do, especially for free, nobody can take a hint
> that
> > you should be paying for all these custom requests so take it or leave it
> > but this thread is done.........
> >
> >
> >
> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin <
> sos at sokhapkin.dyndns.org>
> > wrote:
> >>
> >> No, not exactly. ignore_early_media doesn't pass early media to the
> caller
> >> if
> >> bypass_media is false.
> >>
> >> On Thursday 14 January 2010, Michael Jerris wrote:
> >> > this is exactly what ignore_early_media does now.
> >> >
> >> > Mike
> >> >
> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> >> > > The issue here is when "originate" routine should return and
> >> > > set "originate_status" variable. Current behavior is to return on
> >> > > early
> >> > > media, but what if to introduce a variable
> "originate_wait_for_answer"
> >> > > with default value "false" and use the variable in originate code to
> >> > > decide when to return - on 18X or "200 OK"?
> >> > >
> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> >> > >> he wants to call 3 people at once and let the A leg hear early
> media
> >> > >> from call #1 while call #2 and #3 still are progressing which is
> not
> >> > >> simple to do without doing thousands of dollars in development.
> >> > >>
> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
> >> > >>> customers
> >> > >>> can hear recordings?
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
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From dave at 3c.co.uk  Thu Jan 14 16:50:05 2010
From: dave at 3c.co.uk (David Knell)
Date: Fri, 15 Jan 2010 00:50:05 +0000
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
Message-ID: <1263516605.11216.91.camel@local.freepabx.com>

Hi David,

Excuse me if I'm being dumb (which is, sadly, a pretty common
occurrence) but there's a bunch of cases where what you're seeking fails
badly.  For example, if one of the calls ends up on a message along the
lines of "The number you have dialled has not been recognised.  Please
check, and try again", and that's the first early media (not unlikely,
all other things being equal), then the information passed back to the
caller is worse than useless.  Isn't it?

Cheers --

Dave

> Anthony,
> 
> I did take the "hint", don't worry. We will probably ask for a bounty
> but first we need to know:
> 1.- whether this is possible
> 2.- how long it would take
> 3.- how will it exactly work
> 4.- of course, what's the bounty (be gentle ;) )
> 
> We would of course give this back to the community.
> 
> in the meantime, bridge_early_media=true will discard the 2nd B-leg
> and conversely "false" it will keep on trying to connect and if it
> connects the other B-leg if will bridge to that one?
> 
> Thanks
> 
> David
> 
> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > I added bridge_early_media=true to do the best I can do.
> > This is the most I will do, especially for free, nobody can take a hint that
> > you should be paying for all these custom requests so take it or leave it
> > but this thread is done.........
> >
> >
> >
> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
> > wrote:
> >>
> >> No, not exactly. ignore_early_media doesn't pass early media to the caller
> >> if
> >> bypass_media is false.
> >>
> >> On Thursday 14 January 2010, Michael Jerris wrote:
> >> > this is exactly what ignore_early_media does now.
> >> >
> >> > Mike
> >> >
> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> >> > > The issue here is when "originate" routine should return and
> >> > > set "originate_status" variable. Current behavior is to return on
> >> > > early
> >> > > media, but what if to introduce a variable "originate_wait_for_answer"
> >> > > with default value "false" and use the variable in originate code to
> >> > > decide when to return - on 18X or "200 OK"?
> >> > >
> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> >> > >> he wants to call 3 people at once and let the A leg hear early media
> >> > >> from call #1 while call #2 and #3 still are progressing which is not
> >> > >> simple to do without doing thousands of dollars in development.
> >> > >>
> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
> >> > >>> customers
> >> > >>> can hear recordings?
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From jingwei.yang at gmail.com  Thu Jan 14 17:09:31 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Fri, 15 Jan 2010 09:09:31 +0800
Subject: [Freeswitch-users] Not able to capture the custom event
In-Reply-To: <191c3a031001140847q1238bbcdi4d9dd84648ee2a27@mail.gmail.com>
References: <13529f9d1001140124t42e9dfe5u3e70a6976aa032a8@mail.gmail.com>
	<191c3a031001140847q1238bbcdi4d9dd84648ee2a27@mail.gmail.com>
Message-ID: <13529f9d1001141709v6ed055f7i4b99a68517f58f38@mail.gmail.com>

Yes, it is the one. Thanks a lot for the help!

On Fri, Jan 15, 2010 at 12:47 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you cant subscribe to all custom events, you have to specify a list of
> subclasses
> you have to say events plain custom myevent::ACDnotify
>
> On Thu, Jan 14, 2010 at 3:24 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>
>> Hi Guys,
>>
>> I'm testing to see whether a custom event can be triggered and captured.
>> Here's the extension:
>>
>>     <extension name="test_ivr">
>>       <condition field="destination_number" expression="^777$">
>>         <action application="event" data="Event-Name=*CUSTOM*
>> ,Event-Subclass=myevent::ACDnotify"/>
>>         <action application="transfer" data="9999 XML default" />
>>       </condition>
>>     </extension>
>>
>> If I telnet and use "events plain all", I can get the event. However, if I
>> use "events plain CUSTOM" or "event plain CUSTOM", it doesn't get captured.
>> Is there anything wrong with the command?
>>
>> Regards,
>> -Jingwei
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Jan 14 17:23:35 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Jan 2010 17:23:35 -0800
Subject: [Freeswitch-users] No hangup tone after zap channel closed,
	tones 	in general
In-Reply-To: <8b18ffe229516c90285de8a54a00e845@thom.fr.eu.org>
References: <8b18ffe229516c90285de8a54a00e845@thom.fr.eu.org>
Message-ID: <87f2f3b91001141723u4b0b12e1rd0f922db493492cd@mail.gmail.com>

On Wed, Jan 13, 2010 at 10:25 AM, Fran?ois Legal <devel at thom.fr.eu.org>wrote:

> Hello,
>
>
>
> How shall I do to get a hangup tone on an FXS port after the channel is
> closed ?
>
> Is there something specific to configure in openzap ?
>
>
>
> For instance, if I call the following extension from an FXS port :
>
>     <!-- Send a 180 and let the far end generate ringback. -->
>     <extension name="ringback_180">
>       <condition field="destination_number" expression="^9980$">
>     <action application="ring_ready"/>
>     <action application="sleep" data="20000"/>
>     <action application="answer"/>
>     <action application="sleep" data="1000"/>
>     <action application="playback" data="voicemail/vm-goodbye.wav"/>
>     <action application="hangup"/>
>       </condition>
>     </extension>
>
> I get no tone after the hangup application is called.
>
>
>
> I also wonder if there is some documentation on the tones.conf file format,
> and about the variables uk-ring, us-ring, bong-ring, sit in vars.xml (when
> are they used, what is the syntax). I could not find any info on wiki.
>
>
>
http://wiki.freeswitch.org/wiki/TGML
-MC
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From mike at jerris.com  Thu Jan 14 17:45:55 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Jan 2010 20:45:55 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <1263516605.11216.91.camel@local.freepabx.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<1263516605.11216.91.camel@local.freepabx.com>
Message-ID: <38A2A04D-1950-4195-9C81-B6A6AE386161@jerris.com>

such are the perils of tdm voip interop.  on isdn we get indication of the failure with progress indicator, when doing sip interop, we should be able to choose this at a tdm gateway, unfortunately, there is no way in sip to indicate failure with inband progress so no way we can act on that.

Mike

On Jan 14, 2010, at 7:50 PM, David Knell wrote:

> Hi David,
> 
> Excuse me if I'm being dumb (which is, sadly, a pretty common
> occurrence) but there's a bunch of cases where what you're seeking fails
> badly.  For example, if one of the calls ends up on a message along the
> lines of "The number you have dialled has not been recognised.  Please
> check, and try again", and that's the first early media (not unlikely,
> all other things being equal), then the information passed back to the
> caller is worse than useless.  Isn't it?
> 
> Cheers --
> 
> Dave
> 
>> Anthony,
>> 
>> I did take the "hint", don't worry. We will probably ask for a bounty
>> but first we need to know:
>> 1.- whether this is possible
>> 2.- how long it would take
>> 3.- how will it exactly work
>> 4.- of course, what's the bounty (be gentle ;) )
>> 
>> We would of course give this back to the community.
>> 
>> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>> and conversely "false" it will keep on trying to connect and if it
>> connects the other B-leg if will bridge to that one?
>> 
>> Thanks
>> 
>> David
>> 
>> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> I added bridge_early_media=true to do the best I can do.
>>> This is the most I will do, especially for free, nobody can take a hint that
>>> you should be paying for all these custom requests so take it or leave it
>>> but this thread is done.........
>>> 
>>> 
>>> 
>>> On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
>>> wrote:
>>>> 
>>>> No, not exactly. ignore_early_media doesn't pass early media to the caller
>>>> if
>>>> bypass_media is false.
>>>> 
>>>> On Thursday 14 January 2010, Michael Jerris wrote:
>>>>> this is exactly what ignore_early_media does now.
>>>>> 
>>>>> Mike
>>>>> 
>>>>> On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>>>>>> The issue here is when "originate" routine should return and
>>>>>> set "originate_status" variable. Current behavior is to return on
>>>>>> early
>>>>>> media, but what if to introduce a variable "originate_wait_for_answer"
>>>>>> with default value "false" and use the variable in originate code to
>>>>>> decide when to return - on 18X or "200 OK"?
>>>>>> 
>>>>>> On Thursday 14 January 2010, Anthony Minessale wrote:
>>>>>>> he wants to call 3 people at once and let the A leg hear early media
>>>>>>> from call #1 while call #2 and #3 still are progressing which is not
>>>>>>> simple to do without doing thousands of dollars in development.
>>>>>>> 
>>>>>>> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>>>>>>>> What about sending Sip 183 with SDP (no 200OK), so that your
>>>>>>>> customers
>>>>>>>> can hear recordings?
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> --
>>> Anthony Minessale II
>>> 
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>> 
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>> 
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From david.villasmil.work at gmail.com  Thu Jan 14 17:54:15 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 02:54:15 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <1263516605.11216.91.camel@local.freepabx.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<1263516605.11216.91.camel@local.freepabx.com>
Message-ID: <9853f4ff1001141754r1a642814sacaa67be2cedbf4a@mail.gmail.com>

Dave,

In that case the early media is good also, the a-leg will hear the guy
is not available.

cheers

David

On Fri, Jan 15, 2010 at 1:50 AM, David Knell <dave at 3c.co.uk> wrote:
> Hi David,
>
> Excuse me if I'm being dumb (which is, sadly, a pretty common
> occurrence) but there's a bunch of cases where what you're seeking fails
> badly. ?For example, if one of the calls ends up on a message along the
> lines of "The number you have dialled has not been recognised. ?Please
> check, and try again", and that's the first early media (not unlikely,
> all other things being equal), then the information passed back to the
> caller is worse than useless. ?Isn't it?
>
> Cheers --
>
> Dave
>
>> Anthony,
>>
>> I did take the "hint", don't worry. We will probably ask for a bounty
>> but first we need to know:
>> 1.- whether this is possible
>> 2.- how long it would take
>> 3.- how will it exactly work
>> 4.- of course, what's the bounty (be gentle ;) )
>>
>> We would of course give this back to the community.
>>
>> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>> and conversely "false" it will keep on trying to connect and if it
>> connects the other B-leg if will bridge to that one?
>>
>> Thanks
>>
>> David
>>
>> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > I added bridge_early_media=true to do the best I can do.
>> > This is the most I will do, especially for free, nobody can take a hint that
>> > you should be paying for all these custom requests so take it or leave it
>> > but this thread is done.........
>> >
>> >
>> >
>> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
>> > wrote:
>> >>
>> >> No, not exactly. ignore_early_media doesn't pass early media to the caller
>> >> if
>> >> bypass_media is false.
>> >>
>> >> On Thursday 14 January 2010, Michael Jerris wrote:
>> >> > this is exactly what ignore_early_media does now.
>> >> >
>> >> > Mike
>> >> >
>> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> >> > > The issue here is when "originate" routine should return and
>> >> > > set "originate_status" variable. Current behavior is to return on
>> >> > > early
>> >> > > media, but what if to introduce a variable "originate_wait_for_answer"
>> >> > > with default value "false" and use the variable in originate code to
>> >> > > decide when to return - on 18X or "200 OK"?
>> >> > >
>> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> >> > >> he wants to call 3 people at once and let the A leg hear early media
>> >> > >> from call #1 while call #2 and #3 still are progressing which is not
>> >> > >> simple to do without doing thousands of dollars in development.
>> >> > >>
>> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>> >> > >>> customers
>> >> > >>> can hear recordings?
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From david.villasmil.work at gmail.com  Thu Jan 14 17:55:39 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 02:55:39 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
Message-ID: <9853f4ff1001141755o7045257cn8562d17a85d52450@mail.gmail.com>

Anthony,

     What about mixing the RTPs? what's the bounty for that?

Cheers

David

On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> {bridge_early_media=true}
> in the dial string in place of ignore_early_media=true
>
> the first b leg in the list who sends 183 will become the ringback device
> for A leg it will hear the early media
> for that leg while the other legs still ring.? If some other leg answers the
> final call will still be bridged to the leg who answered.
>
>
> I would estimate it at $500 payable on the big paypal button on
> http://www.freeswitch.org
> but, I already added the patch to tree earlier today so I guess it's up to
> you to pay it or not.
>
>
>
>
> On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> Anthony,
>>
>> I did take the "hint", don't worry. We will probably ask for a bounty
>> but first we need to know:
>> 1.- whether this is possible
>> 2.- how long it would take
>> 3.- how will it exactly work
>> 4.- of course, what's the bounty (be gentle ;) )
>>
>> We would of course give this back to the community.
>>
>> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>> and conversely "false" it will keep on trying to connect and if it
>> connects the other B-leg if will bridge to that one?
>>
>> Thanks
>>
>> David
>>
>> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > I added bridge_early_media=true to do the best I can do.
>> > This is the most I will do, especially for free, nobody can take a hint
>> > that
>> > you should be paying for all these custom requests so take it or leave
>> > it
>> > but this thread is done.........
>> >
>> >
>> >
>> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>> > <sos at sokhapkin.dyndns.org>
>> > wrote:
>> >>
>> >> No, not exactly. ignore_early_media doesn't pass early media to the
>> >> caller
>> >> if
>> >> bypass_media is false.
>> >>
>> >> On Thursday 14 January 2010, Michael Jerris wrote:
>> >> > this is exactly what ignore_early_media does now.
>> >> >
>> >> > Mike
>> >> >
>> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> >> > > The issue here is when "originate" routine should return and
>> >> > > set "originate_status" variable. Current behavior is to return on
>> >> > > early
>> >> > > media, but what if to introduce a variable
>> >> > > "originate_wait_for_answer"
>> >> > > with default value "false" and use the variable in originate code
>> >> > > to
>> >> > > decide when to return - on 18X or "200 OK"?
>> >> > >
>> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> >> > >> he wants to call 3 people at once and let the A leg hear early
>> >> > >> media
>> >> > >> from call #1 while call #2 and #3 still are progressing which is
>> >> > >> not
>> >> > >> simple to do without doing thousands of dollars in development.
>> >> > >>
>> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>> >> > >>> customers
>> >> > >>> can hear recordings?
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Thu Jan 14 18:00:34 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Jan 2010 18:00:34 -0800
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <ade741d61001141154m75b9fcbak76e874e80d023ac0@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
	<8213d6071001141109l1d475b1j3af35b5586708e42@mail.gmail.com>
	<ade741d61001141123x393512f6x81768de82a84c214@mail.gmail.com>
	<8213d6071001141131m3a6ab686xf716dc26983ade5a@mail.gmail.com>
	<ade741d61001141154m75b9fcbak76e874e80d023ac0@mail.gmail.com>
Message-ID: <87f2f3b91001141800q68550eeei310351bd4f7fcdab@mail.gmail.com>

On Thu, Jan 14, 2010 at 11:54 AM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> thanks, how would it look in dialplan if I have to call a javascript?
>
>
>
<action application="set" data="api_hangup_hook=jsrun /path/to/my_script.js
arg1 arg2 arg3"/>

-MC
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From msc at freeswitch.org  Thu Jan 14 18:05:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Jan 2010 18:05:32 -0800
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001141754r1a642814sacaa67be2cedbf4a@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<1263516605.11216.91.camel@local.freepabx.com>
	<9853f4ff1001141754r1a642814sacaa67be2cedbf4a@mail.gmail.com>
Message-ID: <87f2f3b91001141805t3a50b114ib7098d0efd04f4a9@mail.gmail.com>

On Thu, Jan 14, 2010 at 5:54 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Dave,
>
> In that case the early media is good also, the a-leg will hear the guy
> is not available.
>
> But what if he is available on one of the other b-legs? What should happen
in that scenario?

-MC


> cheers
>
> David
>
> On Fri, Jan 15, 2010 at 1:50 AM, David Knell <dave at 3c.co.uk> wrote:
> > Hi David,
> >
> > Excuse me if I'm being dumb (which is, sadly, a pretty common
> > occurrence) but there's a bunch of cases where what you're seeking fails
> > badly.  For example, if one of the calls ends up on a message along the
> > lines of "The number you have dialled has not been recognised.  Please
> > check, and try again", and that's the first early media (not unlikely,
> > all other things being equal), then the information passed back to the
> > caller is worse than useless.  Isn't it?
> >
> > Cheers --
> >
> > Dave
> >
> >> Anthony,
> >>
> >> I did take the "hint", don't worry. We will probably ask for a bounty
> >> but first we need to know:
> >> 1.- whether this is possible
> >> 2.- how long it would take
> >> 3.- how will it exactly work
> >> 4.- of course, what's the bounty (be gentle ;) )
> >>
> >> We would of course give this back to the community.
> >>
> >> in the meantime, bridge_early_media=true will discard the 2nd B-leg
> >> and conversely "false" it will keep on trying to connect and if it
> >> connects the other B-leg if will bridge to that one?
> >>
> >> Thanks
> >>
> >> David
> >>
> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >> > I added bridge_early_media=true to do the best I can do.
> >> > This is the most I will do, especially for free, nobody can take a
> hint that
> >> > you should be paying for all these custom requests so take it or leave
> it
> >> > but this thread is done.........
> >> >
> >> >
> >> >
> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin <
> sos at sokhapkin.dyndns.org>
> >> > wrote:
> >> >>
> >> >> No, not exactly. ignore_early_media doesn't pass early media to the
> caller
> >> >> if
> >> >> bypass_media is false.
> >> >>
> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
> >> >> > this is exactly what ignore_early_media does now.
> >> >> >
> >> >> > Mike
> >> >> >
> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> >> >> > > The issue here is when "originate" routine should return and
> >> >> > > set "originate_status" variable. Current behavior is to return on
> >> >> > > early
> >> >> > > media, but what if to introduce a variable
> "originate_wait_for_answer"
> >> >> > > with default value "false" and use the variable in originate code
> to
> >> >> > > decide when to return - on 18X or "200 OK"?
> >> >> > >
> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> >> >> > >> he wants to call 3 people at once and let the A leg hear early
> media
> >> >> > >> from call #1 while call #2 and #3 still are progressing which is
> not
> >> >> > >> simple to do without doing thousands of dollars in development.
> >> >> > >>
> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com>
> wrote:
> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
> >> >> > >>> customers
> >> >> > >>> can hear recordings?
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >>
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
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From david.villasmil.work at gmail.com  Thu Jan 14 18:13:32 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 03:13:32 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <87f2f3b91001141805t3a50b114ib7098d0efd04f4a9@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<1263516605.11216.91.camel@local.freepabx.com>
	<9853f4ff1001141754r1a642814sacaa67be2cedbf4a@mail.gmail.com>
	<87f2f3b91001141805t3a50b114ib7098d0efd04f4a9@mail.gmail.com>
Message-ID: <9853f4ff1001141813n58c50df9q9e97d352a8d97199@mail.gmail.com>

Michael,

     At least in my case, there will only be 2 legs, 1 providing the
music/audio, the other is the terminating side.

David

On Fri, Jan 15, 2010 at 3:05 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Thu, Jan 14, 2010 at 5:54 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> Dave,
>>
>> In that case the early media is good also, the a-leg will hear the guy
>> is not available.
>>
> But what if he is available on one of the other b-legs? What should happen
> in that scenario?
>
> -MC
>
>>
>> cheers
>>
>> David
>>
>> On Fri, Jan 15, 2010 at 1:50 AM, David Knell <dave at 3c.co.uk> wrote:
>> > Hi David,
>> >
>> > Excuse me if I'm being dumb (which is, sadly, a pretty common
>> > occurrence) but there's a bunch of cases where what you're seeking fails
>> > badly. ?For example, if one of the calls ends up on a message along the
>> > lines of "The number you have dialled has not been recognised. ?Please
>> > check, and try again", and that's the first early media (not unlikely,
>> > all other things being equal), then the information passed back to the
>> > caller is worse than useless. ?Isn't it?
>> >
>> > Cheers --
>> >
>> > Dave
>> >
>> >> Anthony,
>> >>
>> >> I did take the "hint", don't worry. We will probably ask for a bounty
>> >> but first we need to know:
>> >> 1.- whether this is possible
>> >> 2.- how long it would take
>> >> 3.- how will it exactly work
>> >> 4.- of course, what's the bounty (be gentle ;) )
>> >>
>> >> We would of course give this back to the community.
>> >>
>> >> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>> >> and conversely "false" it will keep on trying to connect and if it
>> >> connects the other B-leg if will bridge to that one?
>> >>
>> >> Thanks
>> >>
>> >> David
>> >>
>> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> >> <anthony.minessale at gmail.com> wrote:
>> >> > I added bridge_early_media=true to do the best I can do.
>> >> > This is the most I will do, especially for free, nobody can take a
>> >> > hint that
>> >> > you should be paying for all these custom requests so take it or
>> >> > leave it
>> >> > but this thread is done.........
>> >> >
>> >> >
>> >> >
>> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>> >> > <sos at sokhapkin.dyndns.org>
>> >> > wrote:
>> >> >>
>> >> >> No, not exactly. ignore_early_media doesn't pass early media to the
>> >> >> caller
>> >> >> if
>> >> >> bypass_media is false.
>> >> >>
>> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
>> >> >> > this is exactly what ignore_early_media does now.
>> >> >> >
>> >> >> > Mike
>> >> >> >
>> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> >> >> > > The issue here is when "originate" routine should return and
>> >> >> > > set "originate_status" variable. Current behavior is to return
>> >> >> > > on
>> >> >> > > early
>> >> >> > > media, but what if to introduce a variable
>> >> >> > > "originate_wait_for_answer"
>> >> >> > > with default value "false" and use the variable in originate
>> >> >> > > code to
>> >> >> > > decide when to return - on 18X or "200 OK"?
>> >> >> > >
>> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> >> >> > >> he wants to call 3 people at once and let the A leg hear early
>> >> >> > >> media
>> >> >> > >> from call #1 while call #2 and #3 still are progressing which
>> >> >> > >> is not
>> >> >> > >> simple to do without doing thousands of dollars in development.
>> >> >> > >>
>> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com>
>> >> >> > >> wrote:
>> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>> >> >> > >>> customers
>> >> >> > >>> can hear recordings?
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >>
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> > Anthony Minessale II
>> >> >
>> >> > FreeSWITCH http://www.freeswitch.org/
>> >> > ClueCon http://www.cluecon.com/
>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >
>> >> > AIM: anthm
>> >> > MSN:anthony_minessale at hotmail.com
>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> > IRC: irc.freenode.net #freeswitch
>> >> >
>> >> > FreeSWITCH Developer Conference
>> >> > sip:888 at conference.freeswitch.org
>> >> > iax:guest at conference.freeswitch.org/888
>> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> > pstn:+19193869900
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Thu Jan 14 18:20:18 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Jan 2010 20:20:18 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001141813n58c50df9q9e97d352a8d97199@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<1263516605.11216.91.camel@local.freepabx.com>
	<9853f4ff1001141754r1a642814sacaa67be2cedbf4a@mail.gmail.com>
	<87f2f3b91001141805t3a50b114ib7098d0efd04f4a9@mail.gmail.com>
	<9853f4ff1001141813n58c50df9q9e97d352a8d97199@mail.gmail.com>
Message-ID: <5BA86C46-D725-457C-A6EA-6C2650EE2FAE@freeswitch.org>

Are you just trying to provide music ringback?

/b

On Jan 14, 2010, at 8:13 PM, David Villasmil wrote:

> Michael,
> 
>     At least in my case, there will only be 2 legs, 1 providing the
> music/audio, the other is the terminating side.
> 
> David



From david.villasmil.work at gmail.com  Thu Jan 14 19:04:29 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 04:04:29 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <5BA86C46-D725-457C-A6EA-6C2650EE2FAE@freeswitch.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<1263516605.11216.91.camel@local.freepabx.com>
	<9853f4ff1001141754r1a642814sacaa67be2cedbf4a@mail.gmail.com>
	<87f2f3b91001141805t3a50b114ib7098d0efd04f4a9@mail.gmail.com>
	<9853f4ff1001141813n58c50df9q9e97d352a8d97199@mail.gmail.com>
	<5BA86C46-D725-457C-A6EA-6C2650EE2FAE@freeswitch.org>
Message-ID: <9853f4ff1001141904h386bfe19kb70dcdb281c3e3b8@mail.gmail.com>

Brian,

     Coming from an external content provider via SIP, yes. That's why
I would need the rtps mixed.

Thanks

David

On Fri, Jan 15, 2010 at 3:20 AM, Brian West <brian at freeswitch.org> wrote:
> Are you just trying to provide music ringback?
>
> /b
>
> On Jan 14, 2010, at 8:13 PM, David Villasmil wrote:
>
>> Michael,
>>
>> ? ? At least in my case, there will only be 2 legs, 1 providing the
>> music/audio, the other is the terminating side.
>>
>> David
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Thu Jan 14 19:17:31 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 21:17:31 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001141917s4d679a7fqe9d410b407e27573@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
	<9853f4ff1001141755o7045257cn8562d17a85d52450@mail.gmail.com>
	<191c3a031001141917s4d679a7fqe9d410b407e27573@mail.gmail.com>
Message-ID: <191c3a031001141917o4dfe19b9h4f9b50c3ab184281@mail.gmail.com>

If you want to mix them it would cost double.

On Jan 14, 2010 8:01 PM, "David Villasmil" <david.villasmil.work at gmail.com>
wrote:

Anthony,

    What about mixing the RTPs? what's the bounty for that?

Cheers

David

On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale

<anthony.minessale at gmail.com> wrote: > {bridge_early_media=true} > in the
dial string in place of ig...
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From pete at privateconnect.com  Fri Jan 15 00:21:24 2010
From: pete at privateconnect.com (Pete Mueller)
Date: Fri, 15 Jan 2010 01:21:24 -0700
Subject: [Freeswitch-users] Eavesdrop in LUA
In-Reply-To: <4B4F33C7.6020403@laposte.net>
References: <4B4ED32E.30706@laposte.net> <4B4F33C7.6020403@laposte.net>
Message-ID: <007201ca95bb$ba673770$2f35a650$@com>

I had a similar problem.  I solved it by first making bridging the call
between A and B.
Then originate C with a LUA script, the last line of which is:

session:execute("eavesdrop", uuid_of_a_leg)

The down side here is that A and B can talk while C is ringing, but in my
case that is not a problem.
-p

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod
Sent: Thursday, January 14, 2010 8:10 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Eavesdrop in LUA

Hi all,

I have an incomplete solution for those interested.

I did it like this in dialplan:
        <action application="set" data="api_on_answer=originate 
sofia/internal/2000 at 10.10.10.10 &eavesdrop(${uuid})"/>-->
        <action application="bridge" 
data="{ecoute=${caller_id_number}}sofia/gateway/${caller_id_number}/${destin
ation_number}"/>

so when a call is setup, FS initiate a new call to 2000 and eavesdrop 
the call.
But I have a small problem, the callee receives no sound until the 
eavesdropper send a SIP reply, so there is a 2-3 seconds delay before 
caller and callee can talk each other.

rod


rod a ?crit :
> Hi all,
>
> I'm trying to do this in LUA:
>      A call B
>
> and I'd like to setup a new call to C with eavesdrop of A conversation 
> with B.
>
> I have no idea how to do this if someone can help.
> I switched to LUA cause I see no way to achieve this with dialplan 
> (snippets are welcome).
>
> regards,
> rod
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From jingwei.yang at gmail.com  Fri Jan 15 02:51:44 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Fri, 15 Jan 2010 18:51:44 +0800
Subject: [Freeswitch-users]  Questions about mod_fifo
Message-ID: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>

Hi Guys,

I'm implementing an ACD system using ESL and mod_fifo. Based on what Anthony
suggested in this post:
http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html

*You can make an event socket application that listens for FIFO events and
keeps track of what FIFOs are currently busy and when there are people
waiting you can have that script generate a call to a group of SIP phones so
when the first one answers, it sends them in as an agent where they can
field the calls.
*

1. How should I handle the concurrent issues? If I bridge a user to two
agents and both of them answers, how does FS take care of this situation?
Will a slower agent get a busy tone automatically?

2. If the socket application is brought up after some users have called in,
what command should I use to check the busy queues? fifo list?

3. Am I using fifo list and fifo count correctly?

here's the testing dialplan:

    <extension name="in_fifo">
      <condition field="destination_number" expression="^779$">
        <action application="set" data="fifo_music=$${hold_music}"/>
        <action application="answer"/>
        <action application="fifo" data="*myq *in"/>
      </condition>
    </extension>

when a call comes in and gets queued, these are the results of some commands
I tried.

freeswitch at localhost.localdomain> fifo list
API CALL [fifo(list)] output:
<fifo_report>
  <fifo name="cool_fifo at 192.168.1.104" consumer_count="0" caller_count="0"
waiting_count="0" importance="0">
    <callers></callers>
    <consumers></consumers>
  </fifo>
</fifo_report>


freeswitch at localhost.localdomain> fifo list myq
API CALL [fifo(list myq)] output:
<fifo_report></fifo_report>


freeswitch at localhost.localdomain> fifo count myq
API CALL [fifo(count myq)] output:
none

It seems *myq* doesn't get created at all? Please enlighten.

Thanks and best regards,
-Jingwei
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From devel at thom.fr.eu.org  Fri Jan 15 02:53:48 2010
From: devel at thom.fr.eu.org (devel at thom.fr.eu.org)
Date: Fri, 15 Jan 2010 11:53:48 +0100
Subject: [Freeswitch-users] No hangup tone after zap channel closed,
	tones 	in general
In-Reply-To: <87f2f3b91001141723u4b0b12e1rd0f922db493492cd@mail.gmail.com>
References: <8b18ffe229516c90285de8a54a00e845@thom.fr.eu.org>
	<87f2f3b91001141723u4b0b12e1rd0f922db493492cd@mail.gmail.com>
Message-ID: <001201ca95d1$045eb6e0$0d1c24a0$@fr.eu.org>

Thank you for the link. I googled through but could not find anything
relevant.

 

So then with my FXS port, do I have to, when a call is over, bridge the
channel (which is either A or B leg depending on the cases) to an extension
with for instance

<action application="playback" data="tone_stream://%(500,500,440)"/>

 

Thanks

 

Fran?ois

 

 

De : freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael
Collins
Envoy? : vendredi 15 janvier 2010 02:24
? : freeswitch-users at lists.freeswitch.org
Objet : Re: [Freeswitch-users] No hangup tone after zap channel closed,
tones in general

 

 

On Wed, Jan 13, 2010 at 10:25 AM, Fran?ois Legal <devel at thom.fr.eu.org>
wrote:

Hello,

 

How shall I do to get a hangup tone on an FXS port after the channel is
closed ?

Is there something specific to configure in openzap ?

 

For instance, if I call the following extension from an FXS port :

    <!-- Send a 180 and let the far end generate ringback. -->
    <extension name="ringback_180">
      <condition field="destination_number" expression="^9980$">
    <action application="ring_ready"/>
    <action application="sleep" data="20000"/>
    <action application="answer"/>
    <action application="sleep" data="1000"/>
    <action application="playback" data="voicemail/vm-goodbye.wav"/>
    <action application="hangup"/>
      </condition>
    </extension>

I get no tone after the hangup application is called.

 

I also wonder if there is some documentation on the tones.conf file format,
and about the variables uk-ring, us-ring, bong-ring, sit in vars.xml (when
are they used, what is the syntax). I could not find any info on wiki.

 

http://wiki.freeswitch.org/wiki/TGML
-MC

 

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From david.villasmil.work at gmail.com  Fri Jan 15 03:14:42 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 12:14:42 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001141917o4dfe19b9h4f9b50c3ab184281@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
	<9853f4ff1001141755o7045257cn8562d17a85d52450@mail.gmail.com>
	<191c3a031001141917s4d679a7fqe9d410b407e27573@mail.gmail.com>
	<191c3a031001141917o4dfe19b9h4f9b50c3ab184281@mail.gmail.com>
Message-ID: <9853f4ff1001150314s4ab3b34dya96ce14fd7bac43a@mail.gmail.com>

Anthony,

Thanks a lot, I believe you can count on it. Be aware this will not be
done immediately, though. It will be in 1-2 months.

David

On Fri, Jan 15, 2010 at 4:17 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> If you want to mix them it would cost double.
>
> On Jan 14, 2010 8:01 PM, "David Villasmil" <david.villasmil.work at gmail.com>
> wrote:
>
> Anthony,
>
> ? ? What about mixing the RTPs? what's the bounty for that?
>
> Cheers
>
> David
>
> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>
> <anthony.minessale at gmail.com> wrote: > {bridge_early_media=true} > in the
> dial string in place of ig...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From david.villasmil.work at gmail.com  Fri Jan 15 03:21:16 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 12:21:16 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
Message-ID: <9853f4ff1001150321v1fb0a793s9836016aa8eb39e@mail.gmail.com>

Hello again Anthony,

I just tested it, and although functionality does not, first incoming
audio is coming in all garbled... do you know why?

David

On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> {bridge_early_media=true}
> in the dial string in place of ignore_early_media=true
>
> the first b leg in the list who sends 183 will become the ringback device
> for A leg it will hear the early media
> for that leg while the other legs still ring.? If some other leg answers the
> final call will still be bridged to the leg who answered.
>
>
> I would estimate it at $500 payable on the big paypal button on
> http://www.freeswitch.org
> but, I already added the patch to tree earlier today so I guess it's up to
> you to pay it or not.
>
>
>
>
> On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> Anthony,
>>
>> I did take the "hint", don't worry. We will probably ask for a bounty
>> but first we need to know:
>> 1.- whether this is possible
>> 2.- how long it would take
>> 3.- how will it exactly work
>> 4.- of course, what's the bounty (be gentle ;) )
>>
>> We would of course give this back to the community.
>>
>> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>> and conversely "false" it will keep on trying to connect and if it
>> connects the other B-leg if will bridge to that one?
>>
>> Thanks
>>
>> David
>>
>> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > I added bridge_early_media=true to do the best I can do.
>> > This is the most I will do, especially for free, nobody can take a hint
>> > that
>> > you should be paying for all these custom requests so take it or leave
>> > it
>> > but this thread is done.........
>> >
>> >
>> >
>> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>> > <sos at sokhapkin.dyndns.org>
>> > wrote:
>> >>
>> >> No, not exactly. ignore_early_media doesn't pass early media to the
>> >> caller
>> >> if
>> >> bypass_media is false.
>> >>
>> >> On Thursday 14 January 2010, Michael Jerris wrote:
>> >> > this is exactly what ignore_early_media does now.
>> >> >
>> >> > Mike
>> >> >
>> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> >> > > The issue here is when "originate" routine should return and
>> >> > > set "originate_status" variable. Current behavior is to return on
>> >> > > early
>> >> > > media, but what if to introduce a variable
>> >> > > "originate_wait_for_answer"
>> >> > > with default value "false" and use the variable in originate code
>> >> > > to
>> >> > > decide when to return - on 18X or "200 OK"?
>> >> > >
>> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> >> > >> he wants to call 3 people at once and let the A leg hear early
>> >> > >> media
>> >> > >> from call #1 while call #2 and #3 still are progressing which is
>> >> > >> not
>> >> > >> simple to do without doing thousands of dollars in development.
>> >> > >>
>> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>> >> > >>> customers
>> >> > >>> can hear recordings?
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From sos at sokhapkin.dyndns.org  Fri Jan 15 03:51:05 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 15 Jan 2010 06:51:05 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001150321v1fb0a793s9836016aa8eb39e@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
	<9853f4ff1001150321v1fb0a793s9836016aa8eb39e@mail.gmail.com>
Message-ID: <201001150651.05983.sos@sokhapkin.dyndns.org>

Which audio? Early media or after 200 OK?

On Friday 15 January 2010, David Villasmil wrote:
> Hello again Anthony,
>
> I just tested it, and although functionality does not, first incoming
> audio is coming in all garbled... do you know why?
>
> David
>
> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>
> <anthony.minessale at gmail.com> wrote:
> > {bridge_early_media=true}
> > in the dial string in place of ignore_early_media=true
> >
> > the first b leg in the list who sends 183 will become the ringback device
> > for A leg it will hear the early media
> > for that leg while the other legs still ring.? If some other leg answers
> > the final call will still be bridged to the leg who answered.
> >
> >
> > I would estimate it at $500 payable on the big paypal button on
> > http://www.freeswitch.org
> > but, I already added the patch to tree earlier today so I guess it's up
> > to you to pay it or not.
> >
> >
> >
> >
> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> >
> > <david.villasmil.work at gmail.com> wrote:
> >> Anthony,
> >>
> >> I did take the "hint", don't worry. We will probably ask for a bounty
> >> but first we need to know:
> >> 1.- whether this is possible
> >> 2.- how long it would take
> >> 3.- how will it exactly work
> >> 4.- of course, what's the bounty (be gentle ;) )
> >>
> >> We would of course give this back to the community.
> >>
> >> in the meantime, bridge_early_media=true will discard the 2nd B-leg
> >> and conversely "false" it will keep on trying to connect and if it
> >> connects the other B-leg if will bridge to that one?
> >>
> >> Thanks
> >>
> >> David
> >>
> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> >>
> >> <anthony.minessale at gmail.com> wrote:
> >> > I added bridge_early_media=true to do the best I can do.
> >> > This is the most I will do, especially for free, nobody can take a
> >> > hint that
> >> > you should be paying for all these custom requests so take it or leave
> >> > it
> >> > but this thread is done.........
> >> >
> >> >
> >> >
> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
> >> > <sos at sokhapkin.dyndns.org>
> >> >
> >> > wrote:
> >> >> No, not exactly. ignore_early_media doesn't pass early media to the
> >> >> caller
> >> >> if
> >> >> bypass_media is false.
> >> >>
> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
> >> >> > this is exactly what ignore_early_media does now.
> >> >> >
> >> >> > Mike
> >> >> >
> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> >> >> > > The issue here is when "originate" routine should return and
> >> >> > > set "originate_status" variable. Current behavior is to return on
> >> >> > > early
> >> >> > > media, but what if to introduce a variable
> >> >> > > "originate_wait_for_answer"
> >> >> > > with default value "false" and use the variable in originate code
> >> >> > > to
> >> >> > > decide when to return - on 18X or "200 OK"?
> >> >> > >
> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> >> >> > >> he wants to call 3 people at once and let the A leg hear early
> >> >> > >> media
> >> >> > >> from call #1 while call #2 and #3 still are progressing which is
> >> >> > >> not
> >> >> > >> simple to do without doing thousands of dollars in development.
> >> >> > >>
> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
> >> >> > >>> customers
> >> >> > >>> can hear recordings?
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> >> >users http://www.freeswitch.org
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
> >> >>ers http://www.freeswitch.org
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> >> >rs http://www.freeswitch.org
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From david.villasmil.work at gmail.com  Fri Jan 15 04:15:46 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 13:15:46 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001150651.05983.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
	<9853f4ff1001150321v1fb0a793s9836016aa8eb39e@mail.gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
Message-ID: <9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>

Yeah, sorry. Early media. Audio after 200 is fine. Early media was ok
before the change.

David

On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> Which audio? Early media or after 200 OK?
>
> On Friday 15 January 2010, David Villasmil wrote:
>> Hello again Anthony,
>>
>> I just tested it, and although functionality does not, first incoming
>> audio is coming in all garbled... do you know why?
>>
>> David
>>
>> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>>
>> <anthony.minessale at gmail.com> wrote:
>> > {bridge_early_media=true}
>> > in the dial string in place of ignore_early_media=true
>> >
>> > the first b leg in the list who sends 183 will become the ringback device
>> > for A leg it will hear the early media
>> > for that leg while the other legs still ring.? If some other leg answers
>> > the final call will still be bridged to the leg who answered.
>> >
>> >
>> > I would estimate it at $500 payable on the big paypal button on
>> > http://www.freeswitch.org
>> > but, I already added the patch to tree earlier today so I guess it's up
>> > to you to pay it or not.
>> >
>> >
>> >
>> >
>> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
>> >
>> > <david.villasmil.work at gmail.com> wrote:
>> >> Anthony,
>> >>
>> >> I did take the "hint", don't worry. We will probably ask for a bounty
>> >> but first we need to know:
>> >> 1.- whether this is possible
>> >> 2.- how long it would take
>> >> 3.- how will it exactly work
>> >> 4.- of course, what's the bounty (be gentle ;) )
>> >>
>> >> We would of course give this back to the community.
>> >>
>> >> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>> >> and conversely "false" it will keep on trying to connect and if it
>> >> connects the other B-leg if will bridge to that one?
>> >>
>> >> Thanks
>> >>
>> >> David
>> >>
>> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> >>
>> >> <anthony.minessale at gmail.com> wrote:
>> >> > I added bridge_early_media=true to do the best I can do.
>> >> > This is the most I will do, especially for free, nobody can take a
>> >> > hint that
>> >> > you should be paying for all these custom requests so take it or leave
>> >> > it
>> >> > but this thread is done.........
>> >> >
>> >> >
>> >> >
>> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>> >> > <sos at sokhapkin.dyndns.org>
>> >> >
>> >> > wrote:
>> >> >> No, not exactly. ignore_early_media doesn't pass early media to the
>> >> >> caller
>> >> >> if
>> >> >> bypass_media is false.
>> >> >>
>> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
>> >> >> > this is exactly what ignore_early_media does now.
>> >> >> >
>> >> >> > Mike
>> >> >> >
>> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> >> >> > > The issue here is when "originate" routine should return and
>> >> >> > > set "originate_status" variable. Current behavior is to return on
>> >> >> > > early
>> >> >> > > media, but what if to introduce a variable
>> >> >> > > "originate_wait_for_answer"
>> >> >> > > with default value "false" and use the variable in originate code
>> >> >> > > to
>> >> >> > > decide when to return - on 18X or "200 OK"?
>> >> >> > >
>> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> >> >> > >> he wants to call 3 people at once and let the A leg hear early
>> >> >> > >> media
>> >> >> > >> from call #1 while call #2 and #3 still are progressing which is
>> >> >> > >> not
>> >> >> > >> simple to do without doing thousands of dollars in development.
>> >> >> > >>
>> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>> >> >> > >>> customers
>> >> >> > >>> can hear recordings?
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >> >> >users http://www.freeswitch.org
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> >> >>ers http://www.freeswitch.org
>> >> >
>> >> > --
>> >> > Anthony Minessale II
>> >> >
>> >> > FreeSWITCH http://www.freeswitch.org/
>> >> > ClueCon http://www.cluecon.com/
>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >
>> >> > AIM: anthm
>> >> > MSN:anthony_minessale at hotmail.com
>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> > IRC: irc.freenode.net #freeswitch
>> >> >
>> >> > FreeSWITCH Developer Conference
>> >> > sip:888 at conference.freeswitch.org
>> >> > iax:guest at conference.freeswitch.org/888
>> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> > pstn:+19193869900
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> >> >rs http://www.freeswitch.org
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From sos at sokhapkin.dyndns.org  Fri Jan 15 04:26:17 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 15 Jan 2010 07:26:17 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
Message-ID: <201001150726.17430.sos@sokhapkin.dyndns.org>

Is bypass_media on or off?

On Friday 15 January 2010, David Villasmil wrote:
> Yeah, sorry. Early media. Audio after 200 is fine. Early media was ok
> before the change.
>
> David
>
> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
>
> <sos at sokhapkin.dyndns.org> wrote:
> > Which audio? Early media or after 200 OK?
> >
> > On Friday 15 January 2010, David Villasmil wrote:
> >> Hello again Anthony,
> >>
> >> I just tested it, and although functionality does not, first incoming
> >> audio is coming in all garbled... do you know why?
> >>
> >> David
> >>
> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
> >>
> >> <anthony.minessale at gmail.com> wrote:
> >> > {bridge_early_media=true}
> >> > in the dial string in place of ignore_early_media=true
> >> >
> >> > the first b leg in the list who sends 183 will become the ringback
> >> > device for A leg it will hear the early media
> >> > for that leg while the other legs still ring.? If some other leg
> >> > answers the final call will still be bridged to the leg who answered.
> >> >
> >> >
> >> > I would estimate it at $500 payable on the big paypal button on
> >> > http://www.freeswitch.org
> >> > but, I already added the patch to tree earlier today so I guess it's
> >> > up to you to pay it or not.
> >> >
> >> >
> >> >
> >> >
> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> >> >
> >> > <david.villasmil.work at gmail.com> wrote:
> >> >> Anthony,
> >> >>
> >> >> I did take the "hint", don't worry. We will probably ask for a bounty
> >> >> but first we need to know:
> >> >> 1.- whether this is possible
> >> >> 2.- how long it would take
> >> >> 3.- how will it exactly work
> >> >> 4.- of course, what's the bounty (be gentle ;) )
> >> >>
> >> >> We would of course give this back to the community.
> >> >>
> >> >> in the meantime, bridge_early_media=true will discard the 2nd B-leg
> >> >> and conversely "false" it will keep on trying to connect and if it
> >> >> connects the other B-leg if will bridge to that one?
> >> >>
> >> >> Thanks
> >> >>
> >> >> David
> >> >>
> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> >> >>
> >> >> <anthony.minessale at gmail.com> wrote:
> >> >> > I added bridge_early_media=true to do the best I can do.
> >> >> > This is the most I will do, especially for free, nobody can take a
> >> >> > hint that
> >> >> > you should be paying for all these custom requests so take it or
> >> >> > leave it
> >> >> > but this thread is done.........
> >> >> >
> >> >> >
> >> >> >
> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
> >> >> > <sos at sokhapkin.dyndns.org>
> >> >> >
> >> >> > wrote:
> >> >> >> No, not exactly. ignore_early_media doesn't pass early media to
> >> >> >> the caller
> >> >> >> if
> >> >> >> bypass_media is false.
> >> >> >>
> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
> >> >> >> > this is exactly what ignore_early_media does now.
> >> >> >> >
> >> >> >> > Mike
> >> >> >> >
> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> >> >> >> > > The issue here is when "originate" routine should return and
> >> >> >> > > set "originate_status" variable. Current behavior is to return
> >> >> >> > > on early
> >> >> >> > > media, but what if to introduce a variable
> >> >> >> > > "originate_wait_for_answer"
> >> >> >> > > with default value "false" and use the variable in originate
> >> >> >> > > code to
> >> >> >> > > decide when to return - on 18X or "200 OK"?
> >> >> >> > >
> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> >> >> >> > >> he wants to call 3 people at once and let the A leg hear
> >> >> >> > >> early media
> >> >> >> > >> from call #1 while call #2 and #3 still are progressing which
> >> >> >> > >> is not
> >> >> >> > >> simple to do without doing thousands of dollars in
> >> >> >> > >> development.
> >> >> >> > >>
> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> 
wrote:
> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
> >> >> >> > >>> customers
> >> >> >> > >>> can hear recordings?
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswit
> >> >> >> >ch- users http://www.freeswitch.org
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch
> >> >> >>-us ers http://www.freeswitch.org
> >> >> >
> >> >> > --
> >> >> > Anthony Minessale II
> >> >> >
> >> >> > FreeSWITCH http://www.freeswitch.org/
> >> >> > ClueCon http://www.cluecon.com/
> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >> >
> >> >> > AIM: anthm
> >> >> > MSN:anthony_minessale at hotmail.com
> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> >> > IRC: irc.freenode.net #freeswitch
> >> >> >
> >> >> > FreeSWITCH Developer Conference
> >> >> > sip:888 at conference.freeswitch.org
> >> >> > iax:guest at conference.freeswitch.org/888
> >> >> > googletalk:conf+888 at conference.freeswitch.org
> >> >> > pstn:+19193869900
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> >> >use rs http://www.freeswitch.org
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
> >> >>ers http://www.freeswitch.org
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> >> >rs http://www.freeswitch.org
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From david.villasmil.work at gmail.com  Fri Jan 15 04:38:36 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 13:38:36 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001150726.17430.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
	<201001150726.17430.sos@sokhapkin.dyndns.org>
Message-ID: <9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>

Default, haven't touched it i suppose it's off, i haven't set it anywhere

On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> Is bypass_media on or off?
>
> On Friday 15 January 2010, David Villasmil wrote:
>> Yeah, sorry. Early media. Audio after 200 is fine. Early media was ok
>> before the change.
>>
>> David
>>
>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
>>
>> <sos at sokhapkin.dyndns.org> wrote:
>> > Which audio? Early media or after 200 OK?
>> >
>> > On Friday 15 January 2010, David Villasmil wrote:
>> >> Hello again Anthony,
>> >>
>> >> I just tested it, and although functionality does not, first incoming
>> >> audio is coming in all garbled... do you know why?
>> >>
>> >> David
>> >>
>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>> >>
>> >> <anthony.minessale at gmail.com> wrote:
>> >> > {bridge_early_media=true}
>> >> > in the dial string in place of ignore_early_media=true
>> >> >
>> >> > the first b leg in the list who sends 183 will become the ringback
>> >> > device for A leg it will hear the early media
>> >> > for that leg while the other legs still ring.? If some other leg
>> >> > answers the final call will still be bridged to the leg who answered.
>> >> >
>> >> >
>> >> > I would estimate it at $500 payable on the big paypal button on
>> >> > http://www.freeswitch.org
>> >> > but, I already added the patch to tree earlier today so I guess it's
>> >> > up to you to pay it or not.
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
>> >> >
>> >> > <david.villasmil.work at gmail.com> wrote:
>> >> >> Anthony,
>> >> >>
>> >> >> I did take the "hint", don't worry. We will probably ask for a bounty
>> >> >> but first we need to know:
>> >> >> 1.- whether this is possible
>> >> >> 2.- how long it would take
>> >> >> 3.- how will it exactly work
>> >> >> 4.- of course, what's the bounty (be gentle ;) )
>> >> >>
>> >> >> We would of course give this back to the community.
>> >> >>
>> >> >> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>> >> >> and conversely "false" it will keep on trying to connect and if it
>> >> >> connects the other B-leg if will bridge to that one?
>> >> >>
>> >> >> Thanks
>> >> >>
>> >> >> David
>> >> >>
>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> >> >>
>> >> >> <anthony.minessale at gmail.com> wrote:
>> >> >> > I added bridge_early_media=true to do the best I can do.
>> >> >> > This is the most I will do, especially for free, nobody can take a
>> >> >> > hint that
>> >> >> > you should be paying for all these custom requests so take it or
>> >> >> > leave it
>> >> >> > but this thread is done.........
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>> >> >> > <sos at sokhapkin.dyndns.org>
>> >> >> >
>> >> >> > wrote:
>> >> >> >> No, not exactly. ignore_early_media doesn't pass early media to
>> >> >> >> the caller
>> >> >> >> if
>> >> >> >> bypass_media is false.
>> >> >> >>
>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
>> >> >> >> > this is exactly what ignore_early_media does now.
>> >> >> >> >
>> >> >> >> > Mike
>> >> >> >> >
>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> >> >> >> > > The issue here is when "originate" routine should return and
>> >> >> >> > > set "originate_status" variable. Current behavior is to return
>> >> >> >> > > on early
>> >> >> >> > > media, but what if to introduce a variable
>> >> >> >> > > "originate_wait_for_answer"
>> >> >> >> > > with default value "false" and use the variable in originate
>> >> >> >> > > code to
>> >> >> >> > > decide when to return - on 18X or "200 OK"?
>> >> >> >> > >
>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> >> >> >> > >> he wants to call 3 people at once and let the A leg hear
>> >> >> >> > >> early media
>> >> >> >> > >> from call #1 while call #2 and #3 still are progressing which
>> >> >> >> > >> is not
>> >> >> >> > >> simple to do without doing thousands of dollars in
>> >> >> >> > >> development.
>> >> >> >> > >>
>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com>
> wrote:
>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>> >> >> >> > >>> customers
>> >> >> >> > >>> can hear recordings?
>> >> >> >> >
>> >> >> >> > _______________________________________________
>> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >
>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswit
>> >> >> >> >ch- users http://www.freeswitch.org
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch
>> >> >> >>-us ers http://www.freeswitch.org
>> >> >> >
>> >> >> > --
>> >> >> > Anthony Minessale II
>> >> >> >
>> >> >> > FreeSWITCH http://www.freeswitch.org/
>> >> >> > ClueCon http://www.cluecon.com/
>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >> >
>> >> >> > AIM: anthm
>> >> >> > MSN:anthony_minessale at hotmail.com
>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> >> > IRC: irc.freenode.net #freeswitch
>> >> >> >
>> >> >> > FreeSWITCH Developer Conference
>> >> >> > sip:888 at conference.freeswitch.org
>> >> >> > iax:guest at conference.freeswitch.org/888
>> >> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> >> > pstn:+19193869900
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >> >> >use rs http://www.freeswitch.org
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> >> >>ers http://www.freeswitch.org
>> >> >
>> >> > --
>> >> > Anthony Minessale II
>> >> >
>> >> > FreeSWITCH http://www.freeswitch.org/
>> >> > ClueCon http://www.cluecon.com/
>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >
>> >> > AIM: anthm
>> >> > MSN:anthony_minessale at hotmail.com
>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> > IRC: irc.freenode.net #freeswitch
>> >> >
>> >> > FreeSWITCH Developer Conference
>> >> > sip:888 at conference.freeswitch.org
>> >> > iax:guest at conference.freeswitch.org/888
>> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> > pstn:+19193869900
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> >> >rs http://www.freeswitch.org
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From david.villasmil.work at gmail.com  Fri Jan 15 04:42:44 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 13:42:44 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001150321v1fb0a793s9836016aa8eb39e@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
	<9853f4ff1001150321v1fb0a793s9836016aa8eb39e@mail.gmail.com>
Message-ID: <9853f4ff1001150442v1644e9e8jd512a64d902ae78c@mail.gmail.com>

If I set bridge_early_media to "false", early audio comes in OK, if I
set it to true it's garbled.

David

On Fri, Jan 15, 2010 at 12:21 PM, David Villasmil
<david.villasmil.work at gmail.com> wrote:
> Hello again Anthony,
>
> I just tested it, and although functionality does not, first incoming
> audio is coming in all garbled... do you know why?
>
> David
>
> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> {bridge_early_media=true}
>> in the dial string in place of ignore_early_media=true
>>
>> the first b leg in the list who sends 183 will become the ringback device
>> for A leg it will hear the early media
>> for that leg while the other legs still ring.? If some other leg answers the
>> final call will still be bridged to the leg who answered.
>>
>>
>> I would estimate it at $500 payable on the big paypal button on
>> http://www.freeswitch.org
>> but, I already added the patch to tree earlier today so I guess it's up to
>> you to pay it or not.
>>
>>
>>
>>
>> On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
>> <david.villasmil.work at gmail.com> wrote:
>>>
>>> Anthony,
>>>
>>> I did take the "hint", don't worry. We will probably ask for a bounty
>>> but first we need to know:
>>> 1.- whether this is possible
>>> 2.- how long it would take
>>> 3.- how will it exactly work
>>> 4.- of course, what's the bounty (be gentle ;) )
>>>
>>> We would of course give this back to the community.
>>>
>>> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>>> and conversely "false" it will keep on trying to connect and if it
>>> connects the other B-leg if will bridge to that one?
>>>
>>> Thanks
>>>
>>> David
>>>
>>> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>> > I added bridge_early_media=true to do the best I can do.
>>> > This is the most I will do, especially for free, nobody can take a hint
>>> > that
>>> > you should be paying for all these custom requests so take it or leave
>>> > it
>>> > but this thread is done.........
>>> >
>>> >
>>> >
>>> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>>> > <sos at sokhapkin.dyndns.org>
>>> > wrote:
>>> >>
>>> >> No, not exactly. ignore_early_media doesn't pass early media to the
>>> >> caller
>>> >> if
>>> >> bypass_media is false.
>>> >>
>>> >> On Thursday 14 January 2010, Michael Jerris wrote:
>>> >> > this is exactly what ignore_early_media does now.
>>> >> >
>>> >> > Mike
>>> >> >
>>> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>>> >> > > The issue here is when "originate" routine should return and
>>> >> > > set "originate_status" variable. Current behavior is to return on
>>> >> > > early
>>> >> > > media, but what if to introduce a variable
>>> >> > > "originate_wait_for_answer"
>>> >> > > with default value "false" and use the variable in originate code
>>> >> > > to
>>> >> > > decide when to return - on 18X or "200 OK"?
>>> >> > >
>>> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>>> >> > >> he wants to call 3 people at once and let the A leg hear early
>>> >> > >> media
>>> >> > >> from call #1 while call #2 and #3 still are progressing which is
>>> >> > >> not
>>> >> > >> simple to do without doing thousands of dollars in development.
>>> >> > >>
>>> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>>> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>>> >> > >>> customers
>>> >> > >>> can hear recordings?
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >
>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com
>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org
>>> > iax:guest at conference.freeswitch.org/888
>>> > googletalk:conf+888 at conference.freeswitch.org
>>> > pstn:+19193869900
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From david.villasmil.work at gmail.com  Fri Jan 15 04:43:28 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 13:43:28 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
	<201001150726.17430.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>
Message-ID: <9853f4ff1001150443t5a18cc6bxa5032f391ab6f0ed@mail.gmail.com>

I set it to "off" just in case, same thing.

On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
<david.villasmil.work at gmail.com> wrote:
> Default, haven't touched it i suppose it's off, i haven't set it anywhere
>
> On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
> <sos at sokhapkin.dyndns.org> wrote:
>> Is bypass_media on or off?
>>
>> On Friday 15 January 2010, David Villasmil wrote:
>>> Yeah, sorry. Early media. Audio after 200 is fine. Early media was ok
>>> before the change.
>>>
>>> David
>>>
>>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
>>>
>>> <sos at sokhapkin.dyndns.org> wrote:
>>> > Which audio? Early media or after 200 OK?
>>> >
>>> > On Friday 15 January 2010, David Villasmil wrote:
>>> >> Hello again Anthony,
>>> >>
>>> >> I just tested it, and although functionality does not, first incoming
>>> >> audio is coming in all garbled... do you know why?
>>> >>
>>> >> David
>>> >>
>>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>>> >>
>>> >> <anthony.minessale at gmail.com> wrote:
>>> >> > {bridge_early_media=true}
>>> >> > in the dial string in place of ignore_early_media=true
>>> >> >
>>> >> > the first b leg in the list who sends 183 will become the ringback
>>> >> > device for A leg it will hear the early media
>>> >> > for that leg while the other legs still ring.? If some other leg
>>> >> > answers the final call will still be bridged to the leg who answered.
>>> >> >
>>> >> >
>>> >> > I would estimate it at $500 payable on the big paypal button on
>>> >> > http://www.freeswitch.org
>>> >> > but, I already added the patch to tree earlier today so I guess it's
>>> >> > up to you to pay it or not.
>>> >> >
>>> >> >
>>> >> >
>>> >> >
>>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
>>> >> >
>>> >> > <david.villasmil.work at gmail.com> wrote:
>>> >> >> Anthony,
>>> >> >>
>>> >> >> I did take the "hint", don't worry. We will probably ask for a bounty
>>> >> >> but first we need to know:
>>> >> >> 1.- whether this is possible
>>> >> >> 2.- how long it would take
>>> >> >> 3.- how will it exactly work
>>> >> >> 4.- of course, what's the bounty (be gentle ;) )
>>> >> >>
>>> >> >> We would of course give this back to the community.
>>> >> >>
>>> >> >> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>>> >> >> and conversely "false" it will keep on trying to connect and if it
>>> >> >> connects the other B-leg if will bridge to that one?
>>> >> >>
>>> >> >> Thanks
>>> >> >>
>>> >> >> David
>>> >> >>
>>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>>> >> >>
>>> >> >> <anthony.minessale at gmail.com> wrote:
>>> >> >> > I added bridge_early_media=true to do the best I can do.
>>> >> >> > This is the most I will do, especially for free, nobody can take a
>>> >> >> > hint that
>>> >> >> > you should be paying for all these custom requests so take it or
>>> >> >> > leave it
>>> >> >> > but this thread is done.........
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>>> >> >> > <sos at sokhapkin.dyndns.org>
>>> >> >> >
>>> >> >> > wrote:
>>> >> >> >> No, not exactly. ignore_early_media doesn't pass early media to
>>> >> >> >> the caller
>>> >> >> >> if
>>> >> >> >> bypass_media is false.
>>> >> >> >>
>>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
>>> >> >> >> > this is exactly what ignore_early_media does now.
>>> >> >> >> >
>>> >> >> >> > Mike
>>> >> >> >> >
>>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>>> >> >> >> > > The issue here is when "originate" routine should return and
>>> >> >> >> > > set "originate_status" variable. Current behavior is to return
>>> >> >> >> > > on early
>>> >> >> >> > > media, but what if to introduce a variable
>>> >> >> >> > > "originate_wait_for_answer"
>>> >> >> >> > > with default value "false" and use the variable in originate
>>> >> >> >> > > code to
>>> >> >> >> > > decide when to return - on 18X or "200 OK"?
>>> >> >> >> > >
>>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>>> >> >> >> > >> he wants to call 3 people at once and let the A leg hear
>>> >> >> >> > >> early media
>>> >> >> >> > >> from call #1 while call #2 and #3 still are progressing which
>>> >> >> >> > >> is not
>>> >> >> >> > >> simple to do without doing thousands of dollars in
>>> >> >> >> > >> development.
>>> >> >> >> > >>
>>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com>
>> wrote:
>>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>>> >> >> >> > >>> customers
>>> >> >> >> > >>> can hear recordings?
>>> >> >> >> >
>>> >> >> >> > _______________________________________________
>>> >> >> >> > FreeSWITCH-users mailing list
>>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >> >
>>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswit
>>> >> >> >> >ch- users http://www.freeswitch.org
>>> >> >> >>
>>> >> >> >> _______________________________________________
>>> >> >> >> FreeSWITCH-users mailing list
>>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >>
>>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch
>>> >> >> >>-us ers http://www.freeswitch.org
>>> >> >> >
>>> >> >> > --
>>> >> >> > Anthony Minessale II
>>> >> >> >
>>> >> >> > FreeSWITCH http://www.freeswitch.org/
>>> >> >> > ClueCon http://www.cluecon.com/
>>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >> >> >
>>> >> >> > AIM: anthm
>>> >> >> > MSN:anthony_minessale at hotmail.com
>>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> >> >> > IRC: irc.freenode.net #freeswitch
>>> >> >> >
>>> >> >> > FreeSWITCH Developer Conference
>>> >> >> > sip:888 at conference.freeswitch.org
>>> >> >> > iax:guest at conference.freeswitch.org/888
>>> >> >> > googletalk:conf+888 at conference.freeswitch.org
>>> >> >> > pstn:+19193869900
>>> >> >> >
>>> >> >> > _______________________________________________
>>> >> >> > FreeSWITCH-users mailing list
>>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> >> >> >use rs http://www.freeswitch.org
>>> >> >>
>>> >> >> _______________________________________________
>>> >> >> FreeSWITCH-users mailing list
>>> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>>> >> >>ers http://www.freeswitch.org
>>> >> >
>>> >> > --
>>> >> > Anthony Minessale II
>>> >> >
>>> >> > FreeSWITCH http://www.freeswitch.org/
>>> >> > ClueCon http://www.cluecon.com/
>>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >> >
>>> >> > AIM: anthm
>>> >> > MSN:anthony_minessale at hotmail.com
>>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> >> > IRC: irc.freenode.net #freeswitch
>>> >> >
>>> >> > FreeSWITCH Developer Conference
>>> >> > sip:888 at conference.freeswitch.org
>>> >> > iax:guest at conference.freeswitch.org/888
>>> >> > googletalk:conf+888 at conference.freeswitch.org
>>> >> > pstn:+19193869900
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>> >> >rs http://www.freeswitch.org
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From kond at nstel.ru  Fri Jan 15 06:33:39 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Fri, 15 Jan 2010 17:33:39 +0300
Subject: [Freeswitch-users] eavesdrop problem?
Message-ID: <20100115143339.ED51B11A9D@mail.nstel.ru>

Hi all,

 

I want to use eavesdrop application.

Playing with it I found that when one tries to eavesdrop caller the feature
works ok.

But when trying to eavesdrop callee eavesdrop attempt failes.

I just updated to the latest version from http://latest.freeswitch.org
<http://latest.freeswitch.org/>  

[freeswitch at freeswitch log]$ fs_cli -x version

FreeSWITCH Version 1.0.5-20100115-0400 (16318M)

 

My setup is as following:

I have 3 internal extensions: 2853, 2840, 2879 (all are xlite).

1. 2879 calls 2840. Then 2853 dials 882879 - eavesdrop worked as expected.

2. 2840 calls 2879. Then 2853 dials 882879 - eavesdrop failed. 

 

I attached logs for both cases.

 

I don't believe it's intended behavior. 

 

Can anybody please advise if it is a configuration or a software problem?

 

Thanks and regards,

Nikolay.

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From brian at freeswitch.org  Fri Jan 15 06:47:18 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 15 Jan 2010 08:47:18 -0600
Subject: [Freeswitch-users] eavesdrop problem?
In-Reply-To: <20100115143339.ED51B11A9D@mail.nstel.ru>
References: <20100115143339.ED51B11A9D@mail.nstel.ru>
Message-ID: <452BA845-A9A9-4465-ACF3-EF34AFBA159D@freeswitch.org>

Bugs do not belong on the mailing list.  http://jira.freeswitch.org, also do not attach zip files gz files or anything that will require us to download unpack and view them locally.  Doing this will delay attention to your issue.

/b

On Jan 15, 2010, at 8:33 AM, Nikolay Kondratyev wrote:

> Hi all,
>  
> I want to use eavesdrop application.
> Playing with it I found that when one tries to eavesdrop caller the feature works ok.
> But when trying to eavesdrop callee eavesdrop attempt failes.
> I just updated to the latest version from http://latest.freeswitch.org
> [freeswitch at freeswitch log]$ fs_cli -x version
> FreeSWITCH Version 1.0.5-20100115-0400 (16318M)
>  
> My setup is as following:
> I have 3 internal extensions: 2853, 2840, 2879 (all are xlite).
> 1. 2879 calls 2840. Then 2853 dials 882879 ? eavesdrop worked as expected.
> 2. 2840 calls 2879. Then 2853 dials 882879 ? eavesdrop failed.
>  
> I attached logs for both cases.
>  
> I don?t believe it?s intended behavior.
>  
> Can anybody please advise if it is a configuration or a software problem?
>  
> Thanks and regards,
> Nikolay.
> <successful_eavesdrop.log.gz><failed_eavesdrop.log.gz>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kond at nstel.ru  Fri Jan 15 06:57:43 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Fri, 15 Jan 2010 17:57:43 +0300
Subject: [Freeswitch-users] sip trunk question: why call
	throughexternalprofile is challenged?
In-Reply-To: <1BD8DFE1-E3DD-46C3-A383-C9627939BB65@jerris.com>
Message-ID: <20100115145743.6291912003@mail.nstel.ru>

Mike, Anthony, thanks for the advice.

I set accept-blind-auth in my external profile and FS does not challenge the
invite any more.

Did I understand right that without accept-blind-auth FS challenged incoming
Invite because of the presence of Proxy-Authorization header in the Invite?

Do I understand right that if I place accept-blind-auth inside
<gateway></gateway> it will work for that gateway only?

Thank and regards,

Nikolay.

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Friday, January 15, 2010 1:22 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] sip trunk question: why call
throughexternalprofile is challenged?

 

This now rings a bell, also, if you create a gateway on that profile who's
gateway name matches the realm on the incoming auth header, I think that
also was working.

 

Mike

 

On Jan 14, 2010, at 5:01 PM, Anthony Minessale wrote:





try setting param accept-blind-auth to true in your sofia profile config
internal.xml
iirc it was made just for sipX who feels the need to send auth headers even
when nobody asked for them.
so even when auth-calls is false we will still try to parse the auth if one
is sent.



On Thu, Jan 14, 2010 at 11:21 AM, Michael Jerris <mike at jerris.com> wrote:



if you look in the sample configs for the words blind and auth you will find
all these settings, also you can setup acls for ip auth to not challenge.

 

Mike

 

On Jan 14, 2010, at 3:26 AM, Nikolay Kondratyev wrote:

 

Mike, thanks for the reply.

 

Mmm. looks like I need more detailed instructions where to dig.

Is there a way to turn off "challenging" completely?

I thought that <param name="auth-calls" value="false"/> should do it, but
alas.

By the way should this parameter be visible in either "sofia status profile
external" or "sofia status gateway sipx4.lab.nstel.ru
<http://sipx4.lab.nstel.ru/> " ? I don't see it.

 

I attached traces of failed and successful calls.

 

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From null at invalid.name  Fri Jan 15 06:58:36 2010
From: null at invalid.name (Dan Lane)
Date: Fri, 15 Jan 2010 14:58:36 +0000
Subject: [Freeswitch-users] Nibble_total_billed variable missing when using
	nibblebill on b-leg
Message-ID: <b042b23b1001150658j26891262qa396e761c54ba2c8@mail.gmail.com>

 I'm doing something similar to the example below in order to bill on
the b-leg. Billing is working but the variable nibble_total_billed
isn't being set once the call is finished.

<action application="bridge"
data="${sofia_contact(internal/user@$${domain})},[enable_heartbeat_events=60,nibble_account=1,nibble_rate=0.01]sofia/gateway/blah/1234"/>

I see that a few others have experienced this issue (including Jira
MODAPP-385) so has anyone found a work-around to coerce this into
working?

If not, what will it take to get the issue resolved?


From kond at nstel.ru  Fri Jan 15 07:18:32 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Fri, 15 Jan 2010 18:18:32 +0300
Subject: [Freeswitch-users] eavesdrop problem?
In-Reply-To: <452BA845-A9A9-4465-ACF3-EF34AFBA159D@freeswitch.org>
Message-ID: <20100115151832.E848811F55@mail.nstel.ru>

Brian,

 

Should I open an issue in the jira (since it's a bug)?

 

In general: I'm new to FS. When I'm not sure if my problem is a bug or
misconfiguration, I think, I should better first discuss the problem in the
list before opening an issue.

Because if everybody will write everything into jira, jira will turn into
mail list, while it is not intended for that purpose.

 

Regarding attaching gzipped files: what is the size limit for attachments in
the list?

 

Thanks and regards,

Nikolay. 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Friday, January 15, 2010 5:47 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] eavesdrop problem?

 

Bugs do not belong on the mailing list.  http://jira.freeswitch.org, also do
not attach zip files gz files or anything that will require us to download
unpack and view them locally.  Doing this will delay attention to your
issue.

 

/b

 

On Jan 15, 2010, at 8:33 AM, Nikolay Kondratyev wrote:





Hi all,

 

I want to use eavesdrop application.

Playing with it I found that when one tries to eavesdrop caller the feature
works ok.

But when trying to eavesdrop callee eavesdrop attempt failes.

I just updated to the latest version from http://latest.freeswitch.org
<http://latest.freeswitch.org/> 

[freeswitch at freeswitch log]$ fs_cli -x version

FreeSWITCH Version 1.0.5-20100115-0400 (16318M)

 

My setup is as following:

I have 3 internal extensions: 2853, 2840, 2879 (all are xlite).

1. 2879 calls 2840. Then 2853 dials 882879 - eavesdrop worked as expected.

2. 2840 calls 2879. Then 2853 dials 882879 - eavesdrop failed.

 

I attached logs for both cases.

 

I don't believe it's intended behavior.

 

Can anybody please advise if it is a configuration or a software problem?

 

Thanks and regards,

Nikolay.

<successful_eavesdrop.log.gz><failed_eavesdrop.log.gz>______________________
_________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From lawwton at gmail.com  Fri Jan 15 07:00:17 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 10:00:17 -0500
Subject: [Freeswitch-users] Conference Questions
Message-ID: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>

Hello:

I've been using asterisk for a little bit over three years now. A
couple of months ago I found out about freeswitch, took a look at it,
thought it was interesting and moved on. A few weeks ago, I started
looking at a project I've been wanting to work on for quite a while
using conferences and started exploring systems and different
approaches. Based on the requirements I have, I decided to use
freeswitch. It seemed like it had the best support for conferencing so
I went for it. According to some documentation I found it also seems
to allow for more concurrent calls than asterisk which is an added
bonus.

I got a server ready, installed FC8 on it which is what I have in
production now, unpacked freeswitch there and so far it's running
beautifully. Very painless process really to get it installed, I was
happy to see that. Configuration seems a bit different since it's XML;
but being a developer myself I can see many advantages to having done
that in the future as the system scales and grows in complexity.

Sorry for the long introduction, getting to my question now. So ...
What I want to be able to do is the following:

Create and control conferences via the HTTP API. I've been reading a
bit for the past two days the documentation and I am becoming more
familiar now with how things are done using ESL, the support for PHP,
perl and I believe others.

a) It seemed to me like the way to setup the moderator of the
conference is by setting a parameter in the DialPlan and specifying
based on a condition who the moderator is, say for instance the
destination number. That's fine and it makes sense, however, say that
I am creating a new conference and I want to have 3 participants where
one of them is the moderator. What would I have to do to specify that
person A dialing for example number xxx-xxx-xxxx is the moderator (via
HTTP)? Would I have to create my own call to the system and add say an
entry to DialPlan with the right parameter for the moderator, then
create the conference?

b) When a conference is created, or when I go to create a new
conference via HTTP using the API, does it allow for example for all
numbers that will be added to be dialed at once? Or should the process
be dial each participant, sending say 3 http requests via the API? The
API command "conference dial" seems to only take one argument for
destination number; but I am asking just in case I missed something.

Thanks in advance for the help and I apologize for the long email.

Alfredo


From mike at jerris.com  Fri Jan 15 07:44:15 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 Jan 2010 10:44:15 -0500
Subject: [Freeswitch-users] sip trunk question: why call
	throughexternalprofile is challenged?
In-Reply-To: <20100115145743.6291912003@mail.nstel.ru>
References: <20100115145743.6291912003@mail.nstel.ru>
Message-ID: <24747A07-BB64-4331-955D-09051591804D@jerris.com>

I am pretty sure that does not work, as we have not matched the gateway yet.

Mike

On Jan 15, 2010, at 9:57 AM, Nikolay Kondratyev wrote:

> Mike, Anthony, thanks for the advice.
> I set accept-blind-auth in my external profile and FS does not challenge the invite any more.
> Did I understand right that without accept-blind-auth FS challenged incoming Invite because of the presence of Proxy-Authorization header in the Invite?
> Do I understand right that if I place accept-blind-auth inside <gateway></gateway> it will work for that gateway only?
> Thank and regards,
> Nikolay.
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Friday, January 15, 2010 1:22 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] sip trunk question: why call throughexternalprofile is challenged?
>  
> This now rings a bell, also, if you create a gateway on that profile who's gateway name matches the realm on the incoming auth header, I think that also was working.
>  
> Mike
>  
> On Jan 14, 2010, at 5:01 PM, Anthony Minessale wrote:
> 
> 
> try setting param accept-blind-auth to true in your sofia profile config internal.xml
> iirc it was made just for sipX who feels the need to send auth headers even when nobody asked for them.
> so even when auth-calls is false we will still try to parse the auth if one is sent.
> 
> 
> On Thu, Jan 14, 2010 at 11:21 AM, Michael Jerris <mike at jerris.com> wrote:
> 
> if you look in the sample configs for the words blind and auth you will find all these settings, also you can setup acls for ip auth to not challenge.
>  
> Mike
>  
> On Jan 14, 2010, at 3:26 AM, Nikolay Kondratyev wrote:
>  
>> Mike, thanks for the reply.
>>  
>> Mmm? looks like I need more detailed instructions where to dig?
>> Is there a way to turn off ?challenging? completely?
>> I thought that <param name="auth-calls" value="false"/> should do it, but alas?
>> By the way should this parameter be visible in either ?sofia status profile external? or ?sofia status gateway sipx4.lab.nstel.ru? ? I don?t see it?
>>  
>> I attached traces of failed and successful calls.
> 
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Fri Jan 15 08:05:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 10:05:03 -0600
Subject: [Freeswitch-users] eavesdrop problem?
In-Reply-To: <20100115143339.ED51B11A9D@mail.nstel.ru>
References: <20100115143339.ED51B11A9D@mail.nstel.ru>
Message-ID: <191c3a031001150805y2c6f0bd0pcea8a18d9d866b4b@mail.gmail.com>

don't bother,

only inbound legs are added to the db that is used to lookup for eavesdrop
because the action is in the dialplan.
The extensions to eavesdrop you are using are just a demo to show you how to
work it.
you need to know the uuid of the channel you are trying to eavesdrop on
before you can do what you want.




On Fri, Jan 15, 2010 at 8:33 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

>  Hi all,
>
>
>
> I want to use eavesdrop application.
>
> Playing with it I found that when one tries to eavesdrop caller the feature
> works ok.
>
> But when trying to eavesdrop callee eavesdrop attempt failes.
>
> I just updated to the latest version from http://latest.freeswitch.org
>
> [freeswitch at freeswitch log]$ fs_cli -x version
>
> FreeSWITCH Version 1.0.5-20100115-0400 (16318M)
>
>
>
> My setup is as following:
>
> I have 3 internal extensions: 2853, 2840, 2879 (all are xlite).
>
> 1. 2879 calls 2840. Then 2853 dials 882879 ? eavesdrop worked as expected.
>
> 2. 2840 calls 2879. Then 2853 dials 882879 ? eavesdrop failed.
>
>
>
> I attached logs for both cases.
>
>
>
> I don?t believe it?s intended behavior.
>
>
>
> Can anybody please advise if it is a configuration or a software problem?
>
>
>
> Thanks and regards,
>
> Nikolay.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Jan 15 08:08:53 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 10:08:53 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001150443t5a18cc6bxa5032f391ab6f0ed@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
	<201001150726.17430.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>
	<9853f4ff1001150443t5a18cc6bxa5032f391ab6f0ed@mail.gmail.com>
Message-ID: <191c3a031001150808i69b3937boe0e903057f984e96@mail.gmail.com>

one of the many reasons its a bad idea.
Probably the leg with the bad audio is a different ptime.
Now the amount of work I have to do escalates I would prefer you commit to
commercial support by emailing me at consulting at freeswitch.org to continue
with this.


On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> I set it to "off" just in case, same thing.
>
> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
> > Default, haven't touched it i suppose it's off, i haven't set it anywhere
> >
> > On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
> > <sos at sokhapkin.dyndns.org> wrote:
> >> Is bypass_media on or off?
> >>
> >> On Friday 15 January 2010, David Villasmil wrote:
> >>> Yeah, sorry. Early media. Audio after 200 is fine. Early media was ok
> >>> before the change.
> >>>
> >>> David
> >>>
> >>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
> >>>
> >>> <sos at sokhapkin.dyndns.org> wrote:
> >>> > Which audio? Early media or after 200 OK?
> >>> >
> >>> > On Friday 15 January 2010, David Villasmil wrote:
> >>> >> Hello again Anthony,
> >>> >>
> >>> >> I just tested it, and although functionality does not, first
> incoming
> >>> >> audio is coming in all garbled... do you know why?
> >>> >>
> >>> >> David
> >>> >>
> >>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
> >>> >>
> >>> >> <anthony.minessale at gmail.com> wrote:
> >>> >> > {bridge_early_media=true}
> >>> >> > in the dial string in place of ignore_early_media=true
> >>> >> >
> >>> >> > the first b leg in the list who sends 183 will become the ringback
> >>> >> > device for A leg it will hear the early media
> >>> >> > for that leg while the other legs still ring.  If some other leg
> >>> >> > answers the final call will still be bridged to the leg who
> answered.
> >>> >> >
> >>> >> >
> >>> >> > I would estimate it at $500 payable on the big paypal button on
> >>> >> > http://www.freeswitch.org
> >>> >> > but, I already added the patch to tree earlier today so I guess
> it's
> >>> >> > up to you to pay it or not.
> >>> >> >
> >>> >> >
> >>> >> >
> >>> >> >
> >>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> >>> >> >
> >>> >> > <david.villasmil.work at gmail.com> wrote:
> >>> >> >> Anthony,
> >>> >> >>
> >>> >> >> I did take the "hint", don't worry. We will probably ask for a
> bounty
> >>> >> >> but first we need to know:
> >>> >> >> 1.- whether this is possible
> >>> >> >> 2.- how long it would take
> >>> >> >> 3.- how will it exactly work
> >>> >> >> 4.- of course, what's the bounty (be gentle ;) )
> >>> >> >>
> >>> >> >> We would of course give this back to the community.
> >>> >> >>
> >>> >> >> in the meantime, bridge_early_media=true will discard the 2nd
> B-leg
> >>> >> >> and conversely "false" it will keep on trying to connect and if
> it
> >>> >> >> connects the other B-leg if will bridge to that one?
> >>> >> >>
> >>> >> >> Thanks
> >>> >> >>
> >>> >> >> David
> >>> >> >>
> >>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> >>> >> >>
> >>> >> >> <anthony.minessale at gmail.com> wrote:
> >>> >> >> > I added bridge_early_media=true to do the best I can do.
> >>> >> >> > This is the most I will do, especially for free, nobody can
> take a
> >>> >> >> > hint that
> >>> >> >> > you should be paying for all these custom requests so take it
> or
> >>> >> >> > leave it
> >>> >> >> > but this thread is done.........
> >>> >> >> >
> >>> >> >> >
> >>> >> >> >
> >>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
> >>> >> >> > <sos at sokhapkin.dyndns.org>
> >>> >> >> >
> >>> >> >> > wrote:
> >>> >> >> >> No, not exactly. ignore_early_media doesn't pass early media
> to
> >>> >> >> >> the caller
> >>> >> >> >> if
> >>> >> >> >> bypass_media is false.
> >>> >> >> >>
> >>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
> >>> >> >> >> > this is exactly what ignore_early_media does now.
> >>> >> >> >> >
> >>> >> >> >> > Mike
> >>> >> >> >> >
> >>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> >>> >> >> >> > > The issue here is when "originate" routine should return
> and
> >>> >> >> >> > > set "originate_status" variable. Current behavior is to
> return
> >>> >> >> >> > > on early
> >>> >> >> >> > > media, but what if to introduce a variable
> >>> >> >> >> > > "originate_wait_for_answer"
> >>> >> >> >> > > with default value "false" and use the variable in
> originate
> >>> >> >> >> > > code to
> >>> >> >> >> > > decide when to return - on 18X or "200 OK"?
> >>> >> >> >> > >
> >>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> >>> >> >> >> > >> he wants to call 3 people at once and let the A leg hear
> >>> >> >> >> > >> early media
> >>> >> >> >> > >> from call #1 while call #2 and #3 still are progressing
> which
> >>> >> >> >> > >> is not
> >>> >> >> >> > >> simple to do without doing thousands of dollars in
> >>> >> >> >> > >> development.
> >>> >> >> >> > >>
> >>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <
> djbinter at yahoo.com>
> >> wrote:
> >>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that
> your
> >>> >> >> >> > >>> customers
> >>> >> >> >> > >>> can hear recordings?
> >>> >> >> >> >
> >>> >> >> >> > _______________________________________________
> >>> >> >> >> > FreeSWITCH-users mailing list
> >>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >> >> >
> >>> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswit
> >>> >> >> >> >ch- users http://www.freeswitch.org
> >>> >> >> >>
> >>> >> >> >> _______________________________________________
> >>> >> >> >> FreeSWITCH-users mailing list
> >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >> >>
> >>> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch
> >>> >> >> >>-us ers http://www.freeswitch.org
> >>> >> >> >
> >>> >> >> > --
> >>> >> >> > Anthony Minessale II
> >>> >> >> >
> >>> >> >> > FreeSWITCH http://www.freeswitch.org/
> >>> >> >> > ClueCon http://www.cluecon.com/
> >>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>> >> >> >
> >>> >> >> > AIM: anthm
> >>> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> >> >> > IRC: irc.freenode.net #freeswitch
> >>> >> >> >
> >>> >> >> > FreeSWITCH Developer Conference
> >>> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> >> >> > iax:guest at conference.freeswitch.org/888
> >>> >> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> >> >> > pstn:+19193869900
> >>> >> >> >
> >>> >> >> > _______________________________________________
> >>> >> >> > FreeSWITCH-users mailing list
> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-
> >>> >> >> >use rs http://www.freeswitch.org
> >>> >> >>
> >>> >> >> _______________________________________________
> >>> >> >> FreeSWITCH-users mailing list
> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-us
> >>> >> >>ers http://www.freeswitch.org
> >>> >> >
> >>> >> > --
> >>> >> > Anthony Minessale II
> >>> >> >
> >>> >> > FreeSWITCH http://www.freeswitch.org/
> >>> >> > ClueCon http://www.cluecon.com/
> >>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>> >> >
> >>> >> > AIM: anthm
> >>> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> >> > IRC: irc.freenode.net #freeswitch
> >>> >> >
> >>> >> > FreeSWITCH Developer Conference
> >>> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> >> > iax:guest at conference.freeswitch.org/888
> >>> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> >> > pstn:+19193869900
> >>> >> >
> >>> >> > _______________________________________________
> >>> >> > FreeSWITCH-users mailing list
> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-use
> >>> >> >rs http://www.freeswitch.org
> >>> >>
> >>> >> _______________________________________________
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Jan 15 08:13:28 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 10:13:28 -0600
Subject: [Freeswitch-users] Conference Questions
In-Reply-To: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>
References: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>
Message-ID: <191c3a031001150813g69a86fb9w3a1601d06df5517e@mail.gmail.com>

look at the "mad boss" extension in the default dialplan
conf/dialplan/default.xml to see how to craft an all-hands conference.
otherwise individual calls to originate to send people to the conference is
also ok.


On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil <lawwton at gmail.com
> wrote:

> Hello:
>
> I've been using asterisk for a little bit over three years now. A
> couple of months ago I found out about freeswitch, took a look at it,
> thought it was interesting and moved on. A few weeks ago, I started
> looking at a project I've been wanting to work on for quite a while
> using conferences and started exploring systems and different
> approaches. Based on the requirements I have, I decided to use
> freeswitch. It seemed like it had the best support for conferencing so
> I went for it. According to some documentation I found it also seems
> to allow for more concurrent calls than asterisk which is an added
> bonus.
>
> I got a server ready, installed FC8 on it which is what I have in
> production now, unpacked freeswitch there and so far it's running
> beautifully. Very painless process really to get it installed, I was
> happy to see that. Configuration seems a bit different since it's XML;
> but being a developer myself I can see many advantages to having done
> that in the future as the system scales and grows in complexity.
>
> Sorry for the long introduction, getting to my question now. So ...
> What I want to be able to do is the following:
>
> Create and control conferences via the HTTP API. I've been reading a
> bit for the past two days the documentation and I am becoming more
> familiar now with how things are done using ESL, the support for PHP,
> perl and I believe others.
>
> a) It seemed to me like the way to setup the moderator of the
> conference is by setting a parameter in the DialPlan and specifying
> based on a condition who the moderator is, say for instance the
> destination number. That's fine and it makes sense, however, say that
> I am creating a new conference and I want to have 3 participants where
> one of them is the moderator. What would I have to do to specify that
> person A dialing for example number xxx-xxx-xxxx is the moderator (via
> HTTP)? Would I have to create my own call to the system and add say an
> entry to DialPlan with the right parameter for the moderator, then
> create the conference?
>
> b) When a conference is created, or when I go to create a new
> conference via HTTP using the API, does it allow for example for all
> numbers that will be added to be dialed at once? Or should the process
> be dial each participant, sending say 3 http requests via the API? The
> API command "conference dial" seems to only take one argument for
> destination number; but I am asking just in case I missed something.
>
> Thanks in advance for the help and I apologize for the long email.
>
> Alfredo
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From null at invalid.name  Fri Jan 15 08:13:30 2010
From: null at invalid.name (Dan Lane)
Date: Fri, 15 Jan 2010 16:13:30 +0000
Subject: [Freeswitch-users] Nibble_total_billed variable missing when
	using nibblebill on b-leg
In-Reply-To: <b042b23b1001150658j26891262qa396e761c54ba2c8@mail.gmail.com>
References: <b042b23b1001150658j26891262qa396e761c54ba2c8@mail.gmail.com>
Message-ID: <b042b23b1001150813r4950e607w111afb727746ce5c@mail.gmail.com>

On Fri, Jan 15, 2010 at 2:58 PM, Dan Lane <null at invalid.name> wrote:
> ?I'm doing something similar to the example below in order to bill on
> the b-leg. Billing is working but the variable nibble_total_billed
> isn't being set once the call is finished.
>
> <action application="bridge"
> data="${sofia_contact(internal/user@$${domain})},[enable_heartbeat_events=60,nibble_account=1,nibble_rate=0.01]sofia/gateway/blah/1234"/>
>
> I see that a few others have experienced this issue (including Jira
> MODAPP-385) so has anyone found a work-around to coerce this into
> working?
>
> If not, what will it take to get the issue resolved?
>

Of course, if I set log-b-leg=true in mod_xml_cdr then I can see the
variable because it's only going to be set on the b-leg! *slaps
forehead*

Carry on, nothing to see here.


From lawwton at gmail.com  Fri Jan 15 08:42:15 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 11:42:15 -0500
Subject: [Freeswitch-users] Conference Questions
In-Reply-To: <191c3a031001150813g69a86fb9w3a1601d06df5517e@mail.gmail.com>
References: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>
	<191c3a031001150813g69a86fb9w3a1601d06df5517e@mail.gmail.com>
Message-ID: <5fe6fa8f1001150842v71c2eadeie7ff38231c88789d@mail.gmail.com>

Appreciate the fast response Anthony.

Response or ideas on how to implement a) ?

a) It seemed to me like the way to setup the moderator of the
conference is by setting a parameter in the DialPlan and specifying
based on a condition who the moderator is, say for instance the
destination number. That's fine and it makes sense, however, say that
I am creating a new conference and I want to have 3 participants where
one of them is the moderator. What would I have to do to specify that
person A dialing for example number xxx-xxx-xxxx is the moderator (via
HTTP)? Would I have to create my own call to the system and add say an
entry to DialPlan with the right parameter for the moderator, then
create the conference?

Thanks in advance,

Alfredo Q-V

On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> look at the "mad boss" extension in the default dialplan
> conf/dialplan/default.xml to see how to craft an all-hands conference.
> otherwise individual calls to originate to send people to the conference is
> also ok.
>
>
> On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> Hello:
>>
>> I've been using asterisk for a little bit over three years now. A
>> couple of months ago I found out about freeswitch, took a look at it,
>> thought it was interesting and moved on. A few weeks ago, I started
>> looking at a project I've been wanting to work on for quite a while
>> using conferences and started exploring systems and different
>> approaches. Based on the requirements I have, I decided to use
>> freeswitch. It seemed like it had the best support for conferencing so
>> I went for it. According to some documentation I found it also seems
>> to allow for more concurrent calls than asterisk which is an added
>> bonus.
>>
>> I got a server ready, installed FC8 on it which is what I have in
>> production now, unpacked freeswitch there and so far it's running
>> beautifully. Very painless process really to get it installed, I was
>> happy to see that. Configuration seems a bit different since it's XML;
>> but being a developer myself I can see many advantages to having done
>> that in the future as the system scales and grows in complexity.
>>
>> Sorry for the long introduction, getting to my question now. So ...
>> What I want to be able to do is the following:
>>
>> Create and control conferences via the HTTP API. I've been reading a
>> bit for the past two days the documentation and I am becoming more
>> familiar now with how things are done using ESL, the support for PHP,
>> perl and I believe others.
>>
>> a) It seemed to me like the way to setup the moderator of the
>> conference is by setting a parameter in the DialPlan and specifying
>> based on a condition who the moderator is, say for instance the
>> destination number. That's fine and it makes sense, however, say that
>> I am creating a new conference and I want to have 3 participants where
>> one of them is the moderator. What would I have to do to specify that
>> person A dialing for example number xxx-xxx-xxxx is the moderator (via
>> HTTP)? Would I have to create my own call to the system and add say an
>> entry to DialPlan with the right parameter for the moderator, then
>> create the conference?
>>
>> b) When a conference is created, or when I go to create a new
>> conference via HTTP using the API, does it allow for example for all
>> numbers that will be added to be dialed at once? Or should the process
>> be dial each participant, sending say 3 http requests via the API? The
>> API command "conference dial" seems to only take one argument for
>> destination number; but I am asking just in case I missed something.
>>
>> Thanks in advance for the help and I apologize for the long email.
>>
>> Alfredo
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Fri Jan 15 08:57:05 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Jan 2010 08:57:05 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly!
Message-ID: <87f2f3b91001150857r5d1ad490ga8d29c3f8bf856b9@mail.gmail.com>

The agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_05

Come join us and let's talk about FreeSWITCH, VoIP, and all things
telephonic!

-Michael
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From jbr at consiglia.dk  Fri Jan 15 09:02:29 2010
From: jbr at consiglia.dk (Jon Bruel)
Date: Fri, 15 Jan 2010 18:02:29 +0100
Subject: [Freeswitch-users] RTCP information
Message-ID: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE09AC568494@SBS2008SERVER.consiglia.local>

In a real setup with 5-20 VoIP calls a day, every now and then there are some problems with sound quality, and I need some tools to investigate the cause.
The phones support RTCP, and I would like to hear if I can get the FS to relay those packets to some kind of logger, including the signalling information?
/Jon

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From anthony.minessale at gmail.com  Fri Jan 15 09:02:09 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 11:02:09 -0600
Subject: [Freeswitch-users] Conference Questions
In-Reply-To: <5fe6fa8f1001150842v71c2eadeie7ff38231c88789d@mail.gmail.com>
References: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>
	<191c3a031001150813g69a86fb9w3a1601d06df5517e@mail.gmail.com>
	<5fe6fa8f1001150842v71c2eadeie7ff38231c88789d@mail.gmail.com>
Message-ID: <191c3a031001150902l26298d7aq80d284a9e04f59fc@mail.gmail.com>

the flags are set as part of the dial string so you can easily choose that,
int the example I told you to look at notice the +flags{} bit at the end of
some of the dial strings.


On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil <
lawwton at gmail.com> wrote:

> Appreciate the fast response Anthony.
>
> Response or ideas on how to implement a) ?
>
> a) It seemed to me like the way to setup the moderator of the
> conference is by setting a parameter in the DialPlan and specifying
> based on a condition who the moderator is, say for instance the
> destination number. That's fine and it makes sense, however, say that
> I am creating a new conference and I want to have 3 participants where
> one of them is the moderator. What would I have to do to specify that
> person A dialing for example number xxx-xxx-xxxx is the moderator (via
> HTTP)? Would I have to create my own call to the system and add say an
> entry to DialPlan with the right parameter for the moderator, then
> create the conference?
>
> Thanks in advance,
>
> Alfredo Q-V
>
> On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > look at the "mad boss" extension in the default dialplan
> > conf/dialplan/default.xml to see how to craft an all-hands conference.
> > otherwise individual calls to originate to send people to the conference
> is
> > also ok.
> >
> >
> > On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil
> > <lawwton at gmail.com> wrote:
> >>
> >> Hello:
> >>
> >> I've been using asterisk for a little bit over three years now. A
> >> couple of months ago I found out about freeswitch, took a look at it,
> >> thought it was interesting and moved on. A few weeks ago, I started
> >> looking at a project I've been wanting to work on for quite a while
> >> using conferences and started exploring systems and different
> >> approaches. Based on the requirements I have, I decided to use
> >> freeswitch. It seemed like it had the best support for conferencing so
> >> I went for it. According to some documentation I found it also seems
> >> to allow for more concurrent calls than asterisk which is an added
> >> bonus.
> >>
> >> I got a server ready, installed FC8 on it which is what I have in
> >> production now, unpacked freeswitch there and so far it's running
> >> beautifully. Very painless process really to get it installed, I was
> >> happy to see that. Configuration seems a bit different since it's XML;
> >> but being a developer myself I can see many advantages to having done
> >> that in the future as the system scales and grows in complexity.
> >>
> >> Sorry for the long introduction, getting to my question now. So ...
> >> What I want to be able to do is the following:
> >>
> >> Create and control conferences via the HTTP API. I've been reading a
> >> bit for the past two days the documentation and I am becoming more
> >> familiar now with how things are done using ESL, the support for PHP,
> >> perl and I believe others.
> >>
> >> a) It seemed to me like the way to setup the moderator of the
> >> conference is by setting a parameter in the DialPlan and specifying
> >> based on a condition who the moderator is, say for instance the
> >> destination number. That's fine and it makes sense, however, say that
> >> I am creating a new conference and I want to have 3 participants where
> >> one of them is the moderator. What would I have to do to specify that
> >> person A dialing for example number xxx-xxx-xxxx is the moderator (via
> >> HTTP)? Would I have to create my own call to the system and add say an
> >> entry to DialPlan with the right parameter for the moderator, then
> >> create the conference?
> >>
> >> b) When a conference is created, or when I go to create a new
> >> conference via HTTP using the API, does it allow for example for all
> >> numbers that will be added to be dialed at once? Or should the process
> >> be dial each participant, sending say 3 http requests via the API? The
> >> API command "conference dial" seems to only take one argument for
> >> destination number; but I am asking just in case I missed something.
> >>
> >> Thanks in advance for the help and I apologize for the long email.
> >>
> >> Alfredo
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From kond at nstel.ru  Fri Jan 15 09:41:47 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Fri, 15 Jan 2010 20:41:47 +0300
Subject: [Freeswitch-users] eavesdrop problem?
In-Reply-To: <191c3a031001150805y2c6f0bd0pcea8a18d9d866b4b@mail.gmail.com>
Message-ID: <20100115174147.30E6A11F5A@mail.nstel.ru>

Anthony,

Thanks for the reply.

Can you please point me to the document where I could read about it?

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop does not say
anything about it.

But let me guess: I should add 

<action application="hash"
data="insert/${domain_name}-spymap/${destination_number}/${uuid}"/>

Into <extension name="global" continue="true"> in the dialplan.

Am I close?

Thanks and regards,

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Friday, January 15, 2010 7:05 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] eavesdrop problem?

 

don't bother,

only inbound legs are added to the db that is used to lookup for eavesdrop
because the action is in the dialplan.
The extensions to eavesdrop you are using are just a demo to show you how to
work it.
you need to know the uuid of the channel you are trying to eavesdrop on
before you can do what you want.





On Fri, Jan 15, 2010 at 8:33 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Hi all,

 

I want to use eavesdrop application.

Playing with it I found that when one tries to eavesdrop caller the feature
works ok.

But when trying to eavesdrop callee eavesdrop attempt failes.

I just updated to the latest version from http://latest.freeswitch.org
<http://latest.freeswitch.org/>  

[freeswitch at freeswitch log]$ fs_cli -x version

FreeSWITCH Version 1.0.5-20100115-0400 (16318M)

 

My setup is as following:

I have 3 internal extensions: 2853, 2840, 2879 (all are xlite).

1. 2879 calls 2840. Then 2853 dials 882879 - eavesdrop worked as expected.

2. 2840 calls 2879. Then 2853 dials 882879 - eavesdrop failed. 

 

I attached logs for both cases.

 

I don't believe it's intended behavior. 

 

Can anybody please advise if it is a configuration or a software problem?

 

Thanks and regards,

Nikolay.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From lawwton at gmail.com  Fri Jan 15 09:53:48 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 12:53:48 -0500
Subject: [Freeswitch-users] Conference Questions
In-Reply-To: <191c3a031001150902l26298d7aq80d284a9e04f59fc@mail.gmail.com>
References: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>
	<191c3a031001150813g69a86fb9w3a1601d06df5517e@mail.gmail.com>
	<5fe6fa8f1001150842v71c2eadeie7ff38231c88789d@mail.gmail.com>
	<191c3a031001150902l26298d7aq80d284a9e04f59fc@mail.gmail.com>
Message-ID: <5fe6fa8f1001150953s15320d34od960a6ad828dc501@mail.gmail.com>

Awesome! That's even nicer.

Appreciate it.

Alfredo Q-V

On Fri, Jan 15, 2010 at 12:02 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> the flags are set as part of the dial string so you can easily choose that,
> int the example I told you to look at notice the +flags{} bit at the end of
> some of the dial strings.
>
>
> On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> Appreciate the fast response Anthony.
>>
>> Response or ideas on how to implement a) ?
>>
>> a) It seemed to me like the way to setup the moderator of the
>> conference is by setting a parameter in the DialPlan and specifying
>> based on a condition who the moderator is, say for instance the
>> destination number. That's fine and it makes sense, however, say that
>> I am creating a new conference and I want to have 3 participants where
>> one of them is the moderator. What would I have to do to specify that
>> person A dialing for example number xxx-xxx-xxxx is the moderator (via
>> HTTP)? Would I have to create my own call to the system and add say an
>> entry to DialPlan with the right parameter for the moderator, then
>> create the conference?
>>
>> Thanks in advance,
>>
>> Alfredo Q-V
>>
>> On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > look at the "mad boss" extension in the default dialplan
>> > conf/dialplan/default.xml to see how to craft an all-hands conference.
>> > otherwise individual calls to originate to send people to the conference
>> > is
>> > also ok.
>> >
>> >
>> > On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil
>> > <lawwton at gmail.com> wrote:
>> >>
>> >> Hello:
>> >>
>> >> I've been using asterisk for a little bit over three years now. A
>> >> couple of months ago I found out about freeswitch, took a look at it,
>> >> thought it was interesting and moved on. A few weeks ago, I started
>> >> looking at a project I've been wanting to work on for quite a while
>> >> using conferences and started exploring systems and different
>> >> approaches. Based on the requirements I have, I decided to use
>> >> freeswitch. It seemed like it had the best support for conferencing so
>> >> I went for it. According to some documentation I found it also seems
>> >> to allow for more concurrent calls than asterisk which is an added
>> >> bonus.
>> >>
>> >> I got a server ready, installed FC8 on it which is what I have in
>> >> production now, unpacked freeswitch there and so far it's running
>> >> beautifully. Very painless process really to get it installed, I was
>> >> happy to see that. Configuration seems a bit different since it's XML;
>> >> but being a developer myself I can see many advantages to having done
>> >> that in the future as the system scales and grows in complexity.
>> >>
>> >> Sorry for the long introduction, getting to my question now. So ...
>> >> What I want to be able to do is the following:
>> >>
>> >> Create and control conferences via the HTTP API. I've been reading a
>> >> bit for the past two days the documentation and I am becoming more
>> >> familiar now with how things are done using ESL, the support for PHP,
>> >> perl and I believe others.
>> >>
>> >> a) It seemed to me like the way to setup the moderator of the
>> >> conference is by setting a parameter in the DialPlan and specifying
>> >> based on a condition who the moderator is, say for instance the
>> >> destination number. That's fine and it makes sense, however, say that
>> >> I am creating a new conference and I want to have 3 participants where
>> >> one of them is the moderator. What would I have to do to specify that
>> >> person A dialing for example number xxx-xxx-xxxx is the moderator (via
>> >> HTTP)? Would I have to create my own call to the system and add say an
>> >> entry to DialPlan with the right parameter for the moderator, then
>> >> create the conference?
>> >>
>> >> b) When a conference is created, or when I go to create a new
>> >> conference via HTTP using the API, does it allow for example for all
>> >> numbers that will be added to be dialed at once? Or should the process
>> >> be dial each participant, sending say 3 http requests via the API? The
>> >> API command "conference dial" seems to only take one argument for
>> >> destination number; but I am asking just in case I missed something.
>> >>
>> >> Thanks in advance for the help and I apologize for the long email.
>> >>
>> >> Alfredo
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mailinglist at fribert.dk  Fri Jan 15 09:56:42 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Fri, 15 Jan 2010 18:56:42 +0100
Subject: [Freeswitch-users] TimeOfDay for company phone
Message-ID: <4B50BA6A020000E1000003BB@mail.fribert.dk>

Hi Guys
 
Still working on my pfsense based freeswitch.
So far it's actually working really good, thanks to all the help on the mailinglist and the excellent tutorials, thankyou very much to all!
 
I'm trying to have one of the phonenumbers react on the TimeOfDay. I followed the guide in the wiki.
 
In public I have set it up to dial the local extension 8203 when the company phone rings.
So I put all this in the default.xml to have it process it:
 
 
   <extension name="Company-x8203">
     <condition field="destination_number" expression="^8203$">
       <action application="transfer" data="CompanyIn"/>
     </condition>
   </extension>
 
   <extension name="CompanyIn" continue="true">
     <condition field="destination_number" expression="^CompanyIn$"/>
     <condition wday="2-6"/>
     <condition minute-of-day="540-1020">
       <action application="set" data="Company_open=true"/>
       <action application="transfer" data="xfer-to-company"/>
       <anti-action application="set" data="Company_open=false"/>
       <anti-action application="transfer" data="xfer-to-company"/>
     </condition>
   </extension>
 
   <extension name="xfer-to-company">
     <condition field="destination_number" expression="^xfer-to-company$"/>
     <condition field="${Company_open}" expression="^true$">
       <action application="bridge" data="group/company@${domain_name}"/ ( 'mailto:group/company@$domain_name"/' )>
       <action application="answer"/>
       <action application="sleep" data="2000"/>
       <action application="voicemail" data="default ${domain_name} 1011"/>
       <anti-action application="voicemail" data="default ${domain_name} 1011"/>
     </condition>
   </extension>
But it doesn't react on the time. I'm wondering if wday and minute-of-day is working on the pfsense package, as far as I know it's version 0.9.5?
 
Any input on this?
 
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From jerry.richards at teotech.com  Fri Jan 15 10:37:06 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 15 Jan 2010 10:37:06 -0800
Subject: [Freeswitch-users] INVITE From Caller Spawned 2 INVITEs to Callee
Message-ID: <4187A3F7A21540DBAE8367C24919EBC9@greyhawk.tonecommander.com>


I posted a trace to http://pastebin.freeswitch.org/11810 that shows one
INVITE spawning 2 INVITEs with two different Call-IDs to the same Callee.

I can't tell from the trace why FS created two calls?

Best Regards,
Jerry



From brian at freeswitch.org  Fri Jan 15 10:43:46 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 15 Jan 2010 12:43:46 -0600
Subject: [Freeswitch-users] INVITE From Caller Spawned 2 INVITEs to
	Callee
In-Reply-To: <4187A3F7A21540DBAE8367C24919EBC9@greyhawk.tonecommander.com>
References: <4187A3F7A21540DBAE8367C24919EBC9@greyhawk.tonecommander.com>
Message-ID: <59FD1F5B-3A75-4710-AF6D-FF479B4EBDC5@freeswitch.org>

You have two registrations in the sip registration table for the same endpoint... I suspect one is stale .

/b

On Jan 15, 2010, at 12:37 PM, Jerry Richards wrote:

> 
> I posted a trace to http://pastebin.freeswitch.org/11810 that shows one
> INVITE spawning 2 INVITEs with two different Call-IDs to the same Callee.
> 
> I can't tell from the trace why FS created two calls?
> 
> Best Regards,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From robert.hadley at teotech.com  Fri Jan 15 11:11:37 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 15 Jan 2010 11:11:37 -0800
Subject: [Freeswitch-users] How do I communicate with FS when running as
	daemon?
Message-ID: <0067A4C2F6794E2A92339A7431B7EFA3@greyhawk.tonecommander.com>

Hi,

 

I have Freeswitch running as a daemon on CentOS 5.3

 

Other than the service freeswitch start|stop|status|reload|restart commands,
how do I communicate with FS when running as daemon?

 

Also, in the example file the reload command is commented out, how would I
tell FS to reloadxml?

 

Thanks,

Robert

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From lawwton at gmail.com  Fri Jan 15 11:18:27 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 14:18:27 -0500
Subject: [Freeswitch-users] Conference Questions
In-Reply-To: <5fe6fa8f1001150953s15320d34od960a6ad828dc501@mail.gmail.com>
References: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>
	<191c3a031001150813g69a86fb9w3a1601d06df5517e@mail.gmail.com>
	<5fe6fa8f1001150842v71c2eadeie7ff38231c88789d@mail.gmail.com>
	<191c3a031001150902l26298d7aq80d284a9e04f59fc@mail.gmail.com>
	<5fe6fa8f1001150953s15320d34od960a6ad828dc501@mail.gmail.com>
Message-ID: <5fe6fa8f1001151118x748970a4y49acfcf845be8739@mail.gmail.com>

Anthony:

I think I spoke too soon. I looked at the example; but I was under the
impression based on your previous comment that I would be able to
invoke the conference dial api command over HTTP and specify the flags
as part of the dial string. The wiki page has the following for the
API call I am thinking:

dial

Dial a destination via a specific endpoint (ie. call mom from the conference).

Usage: conference <confname> dial [{dial string
options}]<endpoint_module_name>/<destination> [<callerid_number>
[<callerid_name>]]

I would like to specify the privilege in this api call. Is that
doable? If not how could I accomplish it? Would I be able to pass the
flags in {dial string options}?

Thanks in advance for the help,

Alfredo

On Fri, Jan 15, 2010 at 12:53 PM, Alfredo Quiroga-Villamil
<lawwton at gmail.com> wrote:
> Awesome! That's even nicer.
>
> Appreciate it.
>
> Alfredo Q-V
>
> On Fri, Jan 15, 2010 at 12:02 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> the flags are set as part of the dial string so you can easily choose that,
>> int the example I told you to look at notice the +flags{} bit at the end of
>> some of the dial strings.
>>
>>
>> On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil
>> <lawwton at gmail.com> wrote:
>>>
>>> Appreciate the fast response Anthony.
>>>
>>> Response or ideas on how to implement a) ?
>>>
>>> a) It seemed to me like the way to setup the moderator of the
>>> conference is by setting a parameter in the DialPlan and specifying
>>> based on a condition who the moderator is, say for instance the
>>> destination number. That's fine and it makes sense, however, say that
>>> I am creating a new conference and I want to have 3 participants where
>>> one of them is the moderator. What would I have to do to specify that
>>> person A dialing for example number xxx-xxx-xxxx is the moderator (via
>>> HTTP)? Would I have to create my own call to the system and add say an
>>> entry to DialPlan with the right parameter for the moderator, then
>>> create the conference?
>>>
>>> Thanks in advance,
>>>
>>> Alfredo Q-V
>>>
>>> On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>> > look at the "mad boss" extension in the default dialplan
>>> > conf/dialplan/default.xml to see how to craft an all-hands conference.
>>> > otherwise individual calls to originate to send people to the conference
>>> > is
>>> > also ok.
>>> >
>>> >
>>> > On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil
>>> > <lawwton at gmail.com> wrote:
>>> >>
>>> >> Hello:
>>> >>
>>> >> I've been using asterisk for a little bit over three years now. A
>>> >> couple of months ago I found out about freeswitch, took a look at it,
>>> >> thought it was interesting and moved on. A few weeks ago, I started
>>> >> looking at a project I've been wanting to work on for quite a while
>>> >> using conferences and started exploring systems and different
>>> >> approaches. Based on the requirements I have, I decided to use
>>> >> freeswitch. It seemed like it had the best support for conferencing so
>>> >> I went for it. According to some documentation I found it also seems
>>> >> to allow for more concurrent calls than asterisk which is an added
>>> >> bonus.
>>> >>
>>> >> I got a server ready, installed FC8 on it which is what I have in
>>> >> production now, unpacked freeswitch there and so far it's running
>>> >> beautifully. Very painless process really to get it installed, I was
>>> >> happy to see that. Configuration seems a bit different since it's XML;
>>> >> but being a developer myself I can see many advantages to having done
>>> >> that in the future as the system scales and grows in complexity.
>>> >>
>>> >> Sorry for the long introduction, getting to my question now. So ...
>>> >> What I want to be able to do is the following:
>>> >>
>>> >> Create and control conferences via the HTTP API. I've been reading a
>>> >> bit for the past two days the documentation and I am becoming more
>>> >> familiar now with how things are done using ESL, the support for PHP,
>>> >> perl and I believe others.
>>> >>
>>> >> a) It seemed to me like the way to setup the moderator of the
>>> >> conference is by setting a parameter in the DialPlan and specifying
>>> >> based on a condition who the moderator is, say for instance the
>>> >> destination number. That's fine and it makes sense, however, say that
>>> >> I am creating a new conference and I want to have 3 participants where
>>> >> one of them is the moderator. What would I have to do to specify that
>>> >> person A dialing for example number xxx-xxx-xxxx is the moderator (via
>>> >> HTTP)? Would I have to create my own call to the system and add say an
>>> >> entry to DialPlan with the right parameter for the moderator, then
>>> >> create the conference?
>>> >>
>>> >> b) When a conference is created, or when I go to create a new
>>> >> conference via HTTP using the API, does it allow for example for all
>>> >> numbers that will be added to be dialed at once? Or should the process
>>> >> be dial each participant, sending say 3 http requests via the API? The
>>> >> API command "conference dial" seems to only take one argument for
>>> >> destination number; but I am asking just in case I missed something.
>>> >>
>>> >> Thanks in advance for the help and I apologize for the long email.
>>> >>
>>> >> Alfredo
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com
>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org
>>> > iax:guest at conference.freeswitch.org/888
>>> > googletalk:conf+888 at conference.freeswitch.org
>>> > pstn:+19193869900
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From gmaruzz at celliax.org  Fri Jan 15 11:22:03 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 15 Jan 2010 20:22:03 +0100
Subject: [Freeswitch-users] How do I communicate with FS when running as
	daemon?
In-Reply-To: <0067A4C2F6794E2A92339A7431B7EFA3@greyhawk.tonecommander.com>
References: <0067A4C2F6794E2A92339A7431B7EFA3@greyhawk.tonecommander.com>
Message-ID: <7b197bef1001151122h10aa87aavcb8d59d8659d7477@mail.gmail.com>

/usr/local/freeswitch/bin/fs_cli

On Fri, Jan 15, 2010 at 8:11 PM, Robert Hadley
<robert.hadley at teotech.com> wrote:
> Hi,
>
>
>
> I have Freeswitch running as a daemon on CentOS 5.3
>
>
>
> Other than the service freeswitch start|stop|status|reload|restart commands,
> how do I communicate with FS when running as daemon?
>
>
>
> Also, in the example file the reload command is commented out, how would I
> tell FS to reloadxml?
>
>
>
> Thanks,
>
> Robert
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From msc at freeswitch.org  Fri Jan 15 11:24:06 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Jan 2010 11:24:06 -0800
Subject: [Freeswitch-users] No hangup tone after zap channel closed,
	tones 	in general
In-Reply-To: <001201ca95d1$045eb6e0$0d1c24a0$@fr.eu.org>
References: <8b18ffe229516c90285de8a54a00e845@thom.fr.eu.org>
	<87f2f3b91001141723u4b0b12e1rd0f922db493492cd@mail.gmail.com>
	<001201ca95d1$045eb6e0$0d1c24a0$@fr.eu.org>
Message-ID: <87f2f3b91001151124r596418abta126ebdaae10465@mail.gmail.com>

On Fri, Jan 15, 2010 at 2:53 AM, <devel at thom.fr.eu.org> wrote:

>  Thank you for the link. I googled through but could not find anything
> relevant.
>
>
>
> So then with my FXS port, do I have to, when a call is over, bridge the
> channel (which is either A or B leg depending on the cases) to an extension
> with for instance
>
> <action application="playback" data="tone_stream://%(500,500,440)"/>
>
if you're just trying to manually send out that tone then yes, you can just
add the line in your dialplan. You can then hangup after playing the tone.
The other end will have to decide what to do on its own.
-MC
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From msc at freeswitch.org  Fri Jan 15 11:49:42 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Jan 2010 11:49:42 -0800
Subject: [Freeswitch-users] Nibble_total_billed variable missing when
	using nibblebill on b-leg
In-Reply-To: <b042b23b1001150813r4950e607w111afb727746ce5c@mail.gmail.com>
References: <b042b23b1001150658j26891262qa396e761c54ba2c8@mail.gmail.com>
	<b042b23b1001150813r4950e607w111afb727746ce5c@mail.gmail.com>
Message-ID: <87f2f3b91001151149w1fe3c74focea7753e1df38919@mail.gmail.com>

On Fri, Jan 15, 2010 at 8:13 AM, Dan Lane <null at invalid.name> wrote:

> On Fri, Jan 15, 2010 at 2:58 PM, Dan Lane <null at invalid.name> wrote:
> >  I'm doing something similar to the example below in order to bill on
> > the b-leg. Billing is working but the variable nibble_total_billed
> > isn't being set once the call is finished.
> >
> > <action application="bridge"
> > data="${sofia_contact(internal/user@
> $${domain})},[enable_heartbeat_events=60,nibble_account=1,nibble_rate=0.01]sofia/gateway/blah/1234"/>
> >
> > I see that a few others have experienced this issue (including Jira
> > MODAPP-385) so has anyone found a work-around to coerce this into
> > working?
> >
> > If not, what will it take to get the issue resolved?
> >
>
> Of course, if I set log-b-leg=true in mod_xml_cdr then I can see the
> variable because it's only going to be set on the b-leg! *slaps
> forehead*
>
> You answered your own question nicely. You're hired! Now you can answer all
the other questions on the list. :)
-MC
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From Prometheus001 at gmx.net  Fri Jan 15 11:51:51 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 15 Jan 2010 20:51:51 +0100
Subject: [Freeswitch-users] Multiple registrations: Only calling
 the	first element in the list in this mode
In-Reply-To: <b39273531001130450k16af27d2v209a300f441d39d8@mail.gmail.com>
References: <4B4DADD5.3010507@gmx.net>
	<b39273531001130450k16af27d2v209a300f441d39d8@mail.gmail.com>
Message-ID: <4B50C757.3050901@gmx.net>

Thanks Rupa,

this worked. I have documented this in the wiki:
    http://wiki.freeswitch.org/wiki/Ring_group

Best regards
Peter

Rupa Schomaker schrieb:
> Try:
>
> bridge user/100 at domain:_:user/101 at domain:_:user/102 at domain
>
> Then document it up if it works.
>
> On Wed, Jan 13, 2010 at 5:26 AM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     Hello,
>
>     I habe the following behaviour
>
>     when I call a user which is registered twice with 2 phones via
>        bridge user/100 at domain
>     both phones are ringing. This is correct as I allow multiple
>     registrations in a profile
>
>     However when I call multiple endpoints via
>        bridge user/100 at domain,user/101 at domain,user/102 at domain
>     only one phone with number100 is ringing.
>
>     Console log shows "Only calling the first element in the list in this
>     mode.":
>     2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734
>     variable
>     string 0 = [presence_id=100 at domain]
>     2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734
>     variable
>     string 1 = [transfer_fallback_extension=100]
>     2010-01-12 19:52:18.236361 [WARNING] switch_ivr_originate.c:2048 Only
>     calling the first element in the list in this mode.
>     2010-01-12 19:52:18.236361 [NOTICE] switch_channel.c:613 New Channel
>     sofia/internal/sip:100 at 10.11.12.203:2048
>     <http://sip:100 at 10.11.12.203:2048>
>     [9b95fcdc-ffab-11de-9ba2-13b2daa7ce61]
>
>     Is there any way to work around this? I need all phones to be
>     ringing in
>     this scenario.
>
>     Best regards
>     Peter
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> -Rupa
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From msc at freeswitch.org  Fri Jan 15 12:09:06 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Jan 2010 12:09:06 -0800
Subject: [Freeswitch-users] How do I communicate with FS when running as
	daemon?
In-Reply-To: <7b197bef1001151122h10aa87aavcb8d59d8659d7477@mail.gmail.com>
References: <0067A4C2F6794E2A92339A7431B7EFA3@greyhawk.tonecommander.com>
	<7b197bef1001151122h10aa87aavcb8d59d8659d7477@mail.gmail.com>
Message-ID: <87f2f3b91001151209y5d07e75ge0a9487788f7c152@mail.gmail.com>

On Fri, Jan 15, 2010 at 11:22 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org>wrote:

> /usr/local/freeswitch/bin/fs_cli
>

fs_cli lets you connect kinda like asterisk -r on steroids.

you can also do stuff like fs_cli -x "show channels" | grep my_extension

Have fun!
-MC
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From fvillarroel at yahoo.com  Fri Jan 15 12:26:13 2010
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Fri, 15 Jan 2010 12:26:13 -0800 (PST)
Subject: [Freeswitch-users] Domains.
Message-ID: <2083.99622.qm@web34302.mail.mud.yahoo.com>

Dear.

I installed FS FreeSWITCH Version 1.0.trunk (16144)

I have a problem when i send traffic from a external gateway, the calls are rejected:

2010-01-05 16:52:37.254194 [DEBUG] sofia.c:5359 0 acls to check for proxy
2010-01-05 16:52:37.254194 [DEBUG] sofia.c:5377 network ip is a proxy [0]
2010-01-05 16:52:37.254194 [DEBUG] sofia.c:5405 IP XXX.XXX.XX.125 Rejected by acl "domains". Falling back to Digest auth.


Anyone could me explain like i can do.

Regards.

Fernando


      


From anthony.minessale at gmail.com  Fri Jan 15 12:28:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 14:28:07 -0600
Subject: [Freeswitch-users] Conference Questions
In-Reply-To: <5fe6fa8f1001151118x748970a4y49acfcf845be8739@mail.gmail.com>
References: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>
	<191c3a031001150813g69a86fb9w3a1601d06df5517e@mail.gmail.com>
	<5fe6fa8f1001150842v71c2eadeie7ff38231c88789d@mail.gmail.com>
	<191c3a031001150902l26298d7aq80d284a9e04f59fc@mail.gmail.com>
	<5fe6fa8f1001150953s15320d34od960a6ad828dc501@mail.gmail.com>
	<5fe6fa8f1001151118x748970a4y49acfcf845be8739@mail.gmail.com>
Message-ID: <191c3a031001151228p406600d3h342529e4ac98bf68@mail.gmail.com>

look up the originate api instead of conference dial

originate /dial/string conference:myconf+flags{foo} inline


On Fri, Jan 15, 2010 at 1:18 PM, Alfredo Quiroga-Villamil <lawwton at gmail.com
> wrote:

> Anthony:
>
> I think I spoke too soon. I looked at the example; but I was under the
> impression based on your previous comment that I would be able to
> invoke the conference dial api command over HTTP and specify the flags
> as part of the dial string. The wiki page has the following for the
> API call I am thinking:
>
> dial
>
> Dial a destination via a specific endpoint (ie. call mom from the
> conference).
>
> Usage: conference <confname> dial [{dial string
> options}]<endpoint_module_name>/<destination> [<callerid_number>
> [<callerid_name>]]
>
> I would like to specify the privilege in this api call. Is that
> doable? If not how could I accomplish it? Would I be able to pass the
> flags in {dial string options}?
>
> Thanks in advance for the help,
>
> Alfredo
>
> On Fri, Jan 15, 2010 at 12:53 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
> > Awesome! That's even nicer.
> >
> > Appreciate it.
> >
> > Alfredo Q-V
> >
> > On Fri, Jan 15, 2010 at 12:02 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >> the flags are set as part of the dial string so you can easily choose
> that,
> >> int the example I told you to look at notice the +flags{} bit at the end
> of
> >> some of the dial strings.
> >>
> >>
> >> On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil
> >> <lawwton at gmail.com> wrote:
> >>>
> >>> Appreciate the fast response Anthony.
> >>>
> >>> Response or ideas on how to implement a) ?
> >>>
> >>> a) It seemed to me like the way to setup the moderator of the
> >>> conference is by setting a parameter in the DialPlan and specifying
> >>> based on a condition who the moderator is, say for instance the
> >>> destination number. That's fine and it makes sense, however, say that
> >>> I am creating a new conference and I want to have 3 participants where
> >>> one of them is the moderator. What would I have to do to specify that
> >>> person A dialing for example number xxx-xxx-xxxx is the moderator (via
> >>> HTTP)? Would I have to create my own call to the system and add say an
> >>> entry to DialPlan with the right parameter for the moderator, then
> >>> create the conference?
> >>>
> >>> Thanks in advance,
> >>>
> >>> Alfredo Q-V
> >>>
> >>> On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale
> >>> <anthony.minessale at gmail.com> wrote:
> >>> > look at the "mad boss" extension in the default dialplan
> >>> > conf/dialplan/default.xml to see how to craft an all-hands
> conference.
> >>> > otherwise individual calls to originate to send people to the
> conference
> >>> > is
> >>> > also ok.
> >>> >
> >>> >
> >>> > On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil
> >>> > <lawwton at gmail.com> wrote:
> >>> >>
> >>> >> Hello:
> >>> >>
> >>> >> I've been using asterisk for a little bit over three years now. A
> >>> >> couple of months ago I found out about freeswitch, took a look at
> it,
> >>> >> thought it was interesting and moved on. A few weeks ago, I started
> >>> >> looking at a project I've been wanting to work on for quite a while
> >>> >> using conferences and started exploring systems and different
> >>> >> approaches. Based on the requirements I have, I decided to use
> >>> >> freeswitch. It seemed like it had the best support for conferencing
> so
> >>> >> I went for it. According to some documentation I found it also seems
> >>> >> to allow for more concurrent calls than asterisk which is an added
> >>> >> bonus.
> >>> >>
> >>> >> I got a server ready, installed FC8 on it which is what I have in
> >>> >> production now, unpacked freeswitch there and so far it's running
> >>> >> beautifully. Very painless process really to get it installed, I was
> >>> >> happy to see that. Configuration seems a bit different since it's
> XML;
> >>> >> but being a developer myself I can see many advantages to having
> done
> >>> >> that in the future as the system scales and grows in complexity.
> >>> >>
> >>> >> Sorry for the long introduction, getting to my question now. So ...
> >>> >> What I want to be able to do is the following:
> >>> >>
> >>> >> Create and control conferences via the HTTP API. I've been reading a
> >>> >> bit for the past two days the documentation and I am becoming more
> >>> >> familiar now with how things are done using ESL, the support for
> PHP,
> >>> >> perl and I believe others.
> >>> >>
> >>> >> a) It seemed to me like the way to setup the moderator of the
> >>> >> conference is by setting a parameter in the DialPlan and specifying
> >>> >> based on a condition who the moderator is, say for instance the
> >>> >> destination number. That's fine and it makes sense, however, say
> that
> >>> >> I am creating a new conference and I want to have 3 participants
> where
> >>> >> one of them is the moderator. What would I have to do to specify
> that
> >>> >> person A dialing for example number xxx-xxx-xxxx is the moderator
> (via
> >>> >> HTTP)? Would I have to create my own call to the system and add say
> an
> >>> >> entry to DialPlan with the right parameter for the moderator, then
> >>> >> create the conference?
> >>> >>
> >>> >> b) When a conference is created, or when I go to create a new
> >>> >> conference via HTTP using the API, does it allow for example for all
> >>> >> numbers that will be added to be dialed at once? Or should the
> process
> >>> >> be dial each participant, sending say 3 http requests via the API?
> The
> >>> >> API command "conference dial" seems to only take one argument for
> >>> >> destination number; but I am asking just in case I missed something.
> >>> >>
> >>> >> Thanks in advance for the help and I apologize for the long email.
> >>> >>
> >>> >> Alfredo
> >>> >>
> >>> >> _______________________________________________
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>
> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >
> >>> >
> >>> >
> >>> > --
> >>> > Anthony Minessale II
> >>> >
> >>> > FreeSWITCH http://www.freeswitch.org/
> >>> > ClueCon http://www.cluecon.com/
> >>> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>> >
> >>> > AIM: anthm
> >>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> > IRC: irc.freenode.net #freeswitch
> >>> >
> >>> > FreeSWITCH Developer Conference
> >>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> > iax:guest at conference.freeswitch.org/888
> >>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> > pstn:+19193869900
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From kristian.kielhofner at gmail.com  Fri Jan 15 12:33:52 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 15 Jan 2010 15:33:52 -0500
Subject: [Freeswitch-users] Changing the username portion of the RURI with
	registered devices
Message-ID: <2d9149cd1001151233h6a468169yd213b1379940e1ae@mail.gmail.com>

Hello everyone,

  Is there (or can we get) a way to set the username portion of the
request URI (along with To:) arbitrarily?  Obviously this can be done
when specifying a bridge string but I'm wondering if it would be
possible with registered contacts.  So...

  Let's say I have another FS/Asterisk/etc system registered with the
username "gw".  Let's say I want to direct multiple DIDs to that
registered endpoint.  I'd bridge to:

sofia/internal/gw%domain.local

(or whatever my domain was)

  The INVITE will go out with a destination URI of the registered
contact.  Probably something like:

gw at 192.168.1.10 (assuming that's where I was registered)

  I'd like to be able to do something like:

{user_uri=9415551212}sofia/internal/gw%local.domain

  Before sending the INVITE to the device, FS/Sofia would replace the
username portion in the RURI/To: with 9415551212, something like:

9415551212 at 192.168.1.10

  Is this currently possible?

  Basically I don't want to have to independently register each DID in
this scenario.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From brian at freeswitch.org  Fri Jan 15 12:40:04 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 15 Jan 2010 14:40:04 -0600
Subject: [Freeswitch-users] Changing the username portion of the RURI
	with registered devices
In-Reply-To: <2d9149cd1001151233h6a468169yd213b1379940e1ae@mail.gmail.com>
References: <2d9149cd1001151233h6a468169yd213b1379940e1ae@mail.gmail.com>
Message-ID: <5BFE0182-1CC6-43BA-A591-6B6F2A7A0C31@freeswitch.org>

you can set the sip-force-user param on the user in the directory to force it.

or

    <!--                                                                                                                                                        
        This is an example of how to override the RURI on an outgoing invite to a registered contact.                                                           
    -->                                                                                                                                                         

    <extension name="ruri">                                                                                                                                     
      <condition field="destination_number" expression="^ruri$">                                                                                                
        <action application="bridge" data="sofia/${ruri_profile}/${ruri_user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/> 
      </condition>                                                                                                                                              
    </extension>                                                                                                                                                
                                                                                                                                                                
    <extension name="7004">                                                                                                                                     
      <condition field="destination_number" expression="^7004$">                                                                                                
        <action application="set" data="ruri_profile=default"/>                                                                                                 
        <action application="set" data="ruri_user=2000"/>                                                                                                       
        <action application="set" data="ruri_contact=1001@${domain_name}"/>                                                                                     
        <action application="execute_extension" data="ruri"/>                                                                                                   
      </condition>                                                                                                                                              
    </extension>                                                                                                                                                

/b


On Jan 15, 2010, at 2:33 PM, Kristian Kielhofner wrote:

> Hello everyone,
> 
>  Is there (or can we get) a way to set the username portion of the
> request URI (along with To:) arbitrarily?  Obviously this can be done
> when specifying a bridge string but I'm wondering if it would be
> possible with registered contacts.  So...
> 
>  Let's say I have another FS/Asterisk/etc system registered with the
> username "gw".  Let's say I want to direct multiple DIDs to that
> registered endpoint.  I'd bridge to:
> 
> sofia/internal/gw%domain.local
> 
> (or whatever my domain was)
> 
>  The INVITE will go out with a destination URI of the registered
> contact.  Probably something like:
> 
> gw at 192.168.1.10 (assuming that's where I was registered)
> 
>  I'd like to be able to do something like:
> 
> {user_uri=9415551212}sofia/internal/gw%local.domain
> 
>  Before sending the INVITE to the device, FS/Sofia would replace the
> username portion in the RURI/To: with 9415551212, something like:
> 
> 9415551212 at 192.168.1.10
> 
>  Is this currently possible?
> 
>  Basically I don't want to have to independently register each DID in
> this scenario.
> 
> -- 
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From Mailings at kh-dev.de  Fri Jan 15 12:57:57 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Fri, 15 Jan 2010 21:57:57 +0100
Subject: [Freeswitch-users] proxy_media seems to be broken
In-Reply-To: <194CB7A1-D382-4F98-AB9A-21AFFABFFD6B@freeswitch.org>
References: <A2F1C0E212BBBA4E8C0E7BE568361F0403B145DC1941@srv02.khdev.corp>
	<608345.82340.qm@web37502.mail.mud.yahoo.com>
	<194CB7A1-D382-4F98-AB9A-21AFFABFFD6B@freeswitch.org>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403B145DC1948@srv02.khdev.corp>

I saw the discussion, but the discussions were about bypass_media and I wasn't sure if that also applied to proxy_media.
That's why I was asking.


Ok, but now I got another problem with proxy_media/bypass_media in combination with T38 (latest tarball).

When using proxy_media FreeSWITCH hangs up immediately after my fax took the call with: [CS_EXCHANGE_MEDIA] [INCOMPATIBLE_DESTINATION]

In the log of my Cisco SPA2102 there's the message:
Peer Confirm T38
Peer Confirm T38
No T38 in SDP
No T38 in SDP


Then I tried bypass_media instead of proxy_media. There FS doesn't hang up, but the same log entries in my Cisco.
Maybe the SDP isn't copied after the reinvite for T38???

Thanks, Klaus



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Tuesday, January 12, 2010 4:53 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] proxy_media seems to be broken

And the tarball is updated already automatically too.  Please update to the latest FreeSWITCH... report any issues to jira if you have them in the future.

In the future please read thru the mailing list as this was discussed in two different threads yesterday with the details and the rev where it was fixed.

Thanks,
/b

On Jan 11, 2010, at 9:34 PM, DJB wrote:


Yes, it has been fixed in 16250.


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From lawwton at gmail.com  Fri Jan 15 13:07:33 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 16:07:33 -0500
Subject: [Freeswitch-users] Conference Questions
In-Reply-To: <191c3a031001151228p406600d3h342529e4ac98bf68@mail.gmail.com>
References: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>
	<191c3a031001150813g69a86fb9w3a1601d06df5517e@mail.gmail.com>
	<5fe6fa8f1001150842v71c2eadeie7ff38231c88789d@mail.gmail.com>
	<191c3a031001150902l26298d7aq80d284a9e04f59fc@mail.gmail.com>
	<5fe6fa8f1001150953s15320d34od960a6ad828dc501@mail.gmail.com>
	<5fe6fa8f1001151118x748970a4y49acfcf845be8739@mail.gmail.com>
	<191c3a031001151228p406600d3h342529e4ac98bf68@mail.gmail.com>
Message-ID: <5fe6fa8f1001151307l4f1f680fn71896f4be0373887@mail.gmail.com>

Great!

Thanks Anthony, really appreciate the help.

Alfredo Q-V

On Fri, Jan 15, 2010 at 3:28 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> look up the originate api instead of conference dial
>
> originate /dial/string conference:myconf+flags{foo} inline
>
>
> On Fri, Jan 15, 2010 at 1:18 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> Anthony:
>>
>> I think I spoke too soon. I looked at the example; but I was under the
>> impression based on your previous comment that I would be able to
>> invoke the conference dial api command over HTTP and specify the flags
>> as part of the dial string. The wiki page has the following for the
>> API call I am thinking:
>>
>> dial
>>
>> Dial a destination via a specific endpoint (ie. call mom from the
>> conference).
>>
>> Usage: conference <confname> dial [{dial string
>> options}]<endpoint_module_name>/<destination> [<callerid_number>
>> [<callerid_name>]]
>>
>> I would like to specify the privilege in this api call. Is that
>> doable? If not how could I accomplish it? Would I be able to pass the
>> flags in {dial string options}?
>>
>> Thanks in advance for the help,
>>
>> Alfredo
>>
>> On Fri, Jan 15, 2010 at 12:53 PM, Alfredo Quiroga-Villamil
>> <lawwton at gmail.com> wrote:
>> > Awesome! That's even nicer.
>> >
>> > Appreciate it.
>> >
>> > Alfredo Q-V
>> >
>> > On Fri, Jan 15, 2010 at 12:02 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >> the flags are set as part of the dial string so you can easily choose
>> >> that,
>> >> int the example I told you to look at notice the +flags{} bit at the
>> >> end of
>> >> some of the dial strings.
>> >>
>> >>
>> >> On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil
>> >> <lawwton at gmail.com> wrote:
>> >>>
>> >>> Appreciate the fast response Anthony.
>> >>>
>> >>> Response or ideas on how to implement a) ?
>> >>>
>> >>> a) It seemed to me like the way to setup the moderator of the
>> >>> conference is by setting a parameter in the DialPlan and specifying
>> >>> based on a condition who the moderator is, say for instance the
>> >>> destination number. That's fine and it makes sense, however, say that
>> >>> I am creating a new conference and I want to have 3 participants where
>> >>> one of them is the moderator. What would I have to do to specify that
>> >>> person A dialing for example number xxx-xxx-xxxx is the moderator (via
>> >>> HTTP)? Would I have to create my own call to the system and add say an
>> >>> entry to DialPlan with the right parameter for the moderator, then
>> >>> create the conference?
>> >>>
>> >>> Thanks in advance,
>> >>>
>> >>> Alfredo Q-V
>> >>>
>> >>> On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale
>> >>> <anthony.minessale at gmail.com> wrote:
>> >>> > look at the "mad boss" extension in the default dialplan
>> >>> > conf/dialplan/default.xml to see how to craft an all-hands
>> >>> > conference.
>> >>> > otherwise individual calls to originate to send people to the
>> >>> > conference
>> >>> > is
>> >>> > also ok.
>> >>> >
>> >>> >
>> >>> > On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil
>> >>> > <lawwton at gmail.com> wrote:
>> >>> >>
>> >>> >> Hello:
>> >>> >>
>> >>> >> I've been using asterisk for a little bit over three years now. A
>> >>> >> couple of months ago I found out about freeswitch, took a look at
>> >>> >> it,
>> >>> >> thought it was interesting and moved on. A few weeks ago, I started
>> >>> >> looking at a project I've been wanting to work on for quite a while
>> >>> >> using conferences and started exploring systems and different
>> >>> >> approaches. Based on the requirements I have, I decided to use
>> >>> >> freeswitch. It seemed like it had the best support for conferencing
>> >>> >> so
>> >>> >> I went for it. According to some documentation I found it also
>> >>> >> seems
>> >>> >> to allow for more concurrent calls than asterisk which is an added
>> >>> >> bonus.
>> >>> >>
>> >>> >> I got a server ready, installed FC8 on it which is what I have in
>> >>> >> production now, unpacked freeswitch there and so far it's running
>> >>> >> beautifully. Very painless process really to get it installed, I
>> >>> >> was
>> >>> >> happy to see that. Configuration seems a bit different since it's
>> >>> >> XML;
>> >>> >> but being a developer myself I can see many advantages to having
>> >>> >> done
>> >>> >> that in the future as the system scales and grows in complexity.
>> >>> >>
>> >>> >> Sorry for the long introduction, getting to my question now. So ...
>> >>> >> What I want to be able to do is the following:
>> >>> >>
>> >>> >> Create and control conferences via the HTTP API. I've been reading
>> >>> >> a
>> >>> >> bit for the past two days the documentation and I am becoming more
>> >>> >> familiar now with how things are done using ESL, the support for
>> >>> >> PHP,
>> >>> >> perl and I believe others.
>> >>> >>
>> >>> >> a) It seemed to me like the way to setup the moderator of the
>> >>> >> conference is by setting a parameter in the DialPlan and specifying
>> >>> >> based on a condition who the moderator is, say for instance the
>> >>> >> destination number. That's fine and it makes sense, however, say
>> >>> >> that
>> >>> >> I am creating a new conference and I want to have 3 participants
>> >>> >> where
>> >>> >> one of them is the moderator. What would I have to do to specify
>> >>> >> that
>> >>> >> person A dialing for example number xxx-xxx-xxxx is the moderator
>> >>> >> (via
>> >>> >> HTTP)? Would I have to create my own call to the system and add say
>> >>> >> an
>> >>> >> entry to DialPlan with the right parameter for the moderator, then
>> >>> >> create the conference?
>> >>> >>
>> >>> >> b) When a conference is created, or when I go to create a new
>> >>> >> conference via HTTP using the API, does it allow for example for
>> >>> >> all
>> >>> >> numbers that will be added to be dialed at once? Or should the
>> >>> >> process
>> >>> >> be dial each participant, sending say 3 http requests via the API?
>> >>> >> The
>> >>> >> API command "conference dial" seems to only take one argument for
>> >>> >> destination number; but I am asking just in case I missed
>> >>> >> something.
>> >>> >>
>> >>> >> Thanks in advance for the help and I apologize for the long email.
>> >>> >>
>> >>> >> Alfredo
>> >>> >>
>> >>> >> _______________________________________________
>> >>> >> FreeSWITCH-users mailing list
>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>
>> >>> >>
>> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>> >
>> >>> > --
>> >>> > Anthony Minessale II
>> >>> >
>> >>> > FreeSWITCH http://www.freeswitch.org/
>> >>> > ClueCon http://www.cluecon.com/
>> >>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>> >
>> >>> > AIM: anthm
>> >>> > MSN:anthony_minessale at hotmail.com
>> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>> > IRC: irc.freenode.net #freeswitch
>> >>> >
>> >>> > FreeSWITCH Developer Conference
>> >>> > sip:888 at conference.freeswitch.org
>> >>> > iax:guest at conference.freeswitch.org/888
>> >>> > googletalk:conf+888 at conference.freeswitch.org
>> >>> > pstn:+19193869900
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> iax:guest at conference.freeswitch.org/888
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From robert.hadley at teotech.com  Fri Jan 15 13:07:43 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 15 Jan 2010 13:07:43 -0800
Subject: [Freeswitch-users] xset warning message starting FS as daemon
Message-ID: <10273ADF13B842FE8527E8FF93BD6C28@greyhawk.tonecommander.com>

Hi,

 

Running trunk on CentOS 5.3, I don't get any warning messages starting FS
manually with the -nc option.

 

I get this message when I start FS as a daemon.

 

[root at roberth-c53 fstrkbld]# service freeswitch start

Starting freeswitch: xset:  unable to open display ""

xset:  unable to open display ""

                                                           [  OK  ]

 

 

Is this due to something I am doing or is it FS?

 

Thanks,

Robert

 

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From jcasale at activenetwerx.com  Fri Jan 15 13:16:17 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 15 Jan 2010 21:16:17 +0000
Subject: [Freeswitch-users] OT: Voip presentation
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E907D1A2@prato.activenetwerx.local>

I have to do a voip presentation to a group and need something elaborating on the
whole process, where does a call go once it leaves it the building on the wire, how
it re-enters the pstn, wtf is a pstn clec etc...

Rather than make something from scratch (although prolly not hard as the level of
detail needed is low) I thought I would ask if anyone knew of something in the public
domain I could snag and start with?

Hopefully the next project is dropping 1/2 or a dozen ports into a Nortel Meridian
dinosaur with fs behind it...

Thanks!
jlc


From anthony.minessale at gmail.com  Fri Jan 15 13:17:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 15:17:52 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001150808i69b3937boe0e903057f984e96@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
	<201001150726.17430.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>
	<9853f4ff1001150443t5a18cc6bxa5032f391ab6f0ed@mail.gmail.com>
	<191c3a031001150808i69b3937boe0e903057f984e96@mail.gmail.com>
Message-ID: <191c3a031001151317j408794a3pa2a0a5d21d63aa62@mail.gmail.com>

Try latest trunk,

you should have exactly what you want with the same parameter, again my
paypal addr is cleary displayed as a big button on the website.


On Fri, Jan 15, 2010 at 10:08 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> one of the many reasons its a bad idea.
> Probably the leg with the bad audio is a different ptime.
> Now the amount of work I have to do escalates I would prefer you commit to
> commercial support by emailing me at consulting at freeswitch.org to continue
> with this.
>
>
>
> On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> I set it to "off" just in case, same thing.
>>
>> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
>> <david.villasmil.work at gmail.com> wrote:
>> > Default, haven't touched it i suppose it's off, i haven't set it
>> anywhere
>> >
>> > On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
>> > <sos at sokhapkin.dyndns.org> wrote:
>> >> Is bypass_media on or off?
>> >>
>> >> On Friday 15 January 2010, David Villasmil wrote:
>> >>> Yeah, sorry. Early media. Audio after 200 is fine. Early media was ok
>> >>> before the change.
>> >>>
>> >>> David
>> >>>
>> >>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
>> >>>
>> >>> <sos at sokhapkin.dyndns.org> wrote:
>> >>> > Which audio? Early media or after 200 OK?
>> >>> >
>> >>> > On Friday 15 January 2010, David Villasmil wrote:
>> >>> >> Hello again Anthony,
>> >>> >>
>> >>> >> I just tested it, and although functionality does not, first
>> incoming
>> >>> >> audio is coming in all garbled... do you know why?
>> >>> >>
>> >>> >> David
>> >>> >>
>> >>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>> >>> >>
>> >>> >> <anthony.minessale at gmail.com> wrote:
>> >>> >> > {bridge_early_media=true}
>> >>> >> > in the dial string in place of ignore_early_media=true
>> >>> >> >
>> >>> >> > the first b leg in the list who sends 183 will become the
>> ringback
>> >>> >> > device for A leg it will hear the early media
>> >>> >> > for that leg while the other legs still ring.  If some other leg
>> >>> >> > answers the final call will still be bridged to the leg who
>> answered.
>> >>> >> >
>> >>> >> >
>> >>> >> > I would estimate it at $500 payable on the big paypal button on
>> >>> >> > http://www.freeswitch.org
>> >>> >> > but, I already added the patch to tree earlier today so I guess
>> it's
>> >>> >> > up to you to pay it or not.
>> >>> >> >
>> >>> >> >
>> >>> >> >
>> >>> >> >
>> >>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
>> >>> >> >
>> >>> >> > <david.villasmil.work at gmail.com> wrote:
>> >>> >> >> Anthony,
>> >>> >> >>
>> >>> >> >> I did take the "hint", don't worry. We will probably ask for a
>> bounty
>> >>> >> >> but first we need to know:
>> >>> >> >> 1.- whether this is possible
>> >>> >> >> 2.- how long it would take
>> >>> >> >> 3.- how will it exactly work
>> >>> >> >> 4.- of course, what's the bounty (be gentle ;) )
>> >>> >> >>
>> >>> >> >> We would of course give this back to the community.
>> >>> >> >>
>> >>> >> >> in the meantime, bridge_early_media=true will discard the 2nd
>> B-leg
>> >>> >> >> and conversely "false" it will keep on trying to connect and if
>> it
>> >>> >> >> connects the other B-leg if will bridge to that one?
>> >>> >> >>
>> >>> >> >> Thanks
>> >>> >> >>
>> >>> >> >> David
>> >>> >> >>
>> >>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> >>> >> >>
>> >>> >> >> <anthony.minessale at gmail.com> wrote:
>> >>> >> >> > I added bridge_early_media=true to do the best I can do.
>> >>> >> >> > This is the most I will do, especially for free, nobody can
>> take a
>> >>> >> >> > hint that
>> >>> >> >> > you should be paying for all these custom requests so take it
>> or
>> >>> >> >> > leave it
>> >>> >> >> > but this thread is done.........
>> >>> >> >> >
>> >>> >> >> >
>> >>> >> >> >
>> >>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>> >>> >> >> > <sos at sokhapkin.dyndns.org>
>> >>> >> >> >
>> >>> >> >> > wrote:
>> >>> >> >> >> No, not exactly. ignore_early_media doesn't pass early media
>> to
>> >>> >> >> >> the caller
>> >>> >> >> >> if
>> >>> >> >> >> bypass_media is false.
>> >>> >> >> >>
>> >>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
>> >>> >> >> >> > this is exactly what ignore_early_media does now.
>> >>> >> >> >> >
>> >>> >> >> >> > Mike
>> >>> >> >> >> >
>> >>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> >>> >> >> >> > > The issue here is when "originate" routine should return
>> and
>> >>> >> >> >> > > set "originate_status" variable. Current behavior is to
>> return
>> >>> >> >> >> > > on early
>> >>> >> >> >> > > media, but what if to introduce a variable
>> >>> >> >> >> > > "originate_wait_for_answer"
>> >>> >> >> >> > > with default value "false" and use the variable in
>> originate
>> >>> >> >> >> > > code to
>> >>> >> >> >> > > decide when to return - on 18X or "200 OK"?
>> >>> >> >> >> > >
>> >>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> >>> >> >> >> > >> he wants to call 3 people at once and let the A leg hear
>> >>> >> >> >> > >> early media
>> >>> >> >> >> > >> from call #1 while call #2 and #3 still are progressing
>> which
>> >>> >> >> >> > >> is not
>> >>> >> >> >> > >> simple to do without doing thousands of dollars in
>> >>> >> >> >> > >> development.
>> >>> >> >> >> > >>
>> >>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <
>> djbinter at yahoo.com>
>> >> wrote:
>> >>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that
>> your
>> >>> >> >> >> > >>> customers
>> >>> >> >> >> > >>> can hear recordings?
>> >>> >> >> >> >
>> >>> >> >> >> > _______________________________________________
>> >>> >> >> >> > FreeSWITCH-users mailing list
>> >>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >> >> >> >
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> >> >> >
>> >>> >> >> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswit
>> >>> >> >> >> >ch- users http://www.freeswitch.org
>> >>> >> >> >>
>> >>> >> >> >> _______________________________________________
>> >>> >> >> >> FreeSWITCH-users mailing list
>> >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> >> >>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> >> >>
>> >>> >> >> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch
>> >>> >> >> >>-us ers http://www.freeswitch.org
>> >>> >> >> >
>> >>> >> >> > --
>> >>> >> >> > Anthony Minessale II
>> >>> >> >> >
>> >>> >> >> > FreeSWITCH http://www.freeswitch.org/
>> >>> >> >> > ClueCon http://www.cluecon.com/
>> >>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>> >> >> >
>> >>> >> >> > AIM: anthm
>> >>> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >>> >> >> > IRC: irc.freenode.net #freeswitch
>> >>> >> >> >
>> >>> >> >> > FreeSWITCH Developer Conference
>> >>> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >>> >> >> > iax:guest at conference.freeswitch.org/888
>> >>> >> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >>> >> >> > pstn:+19193869900
>> >>> >> >> >
>> >>> >> >> > _______________________________________________
>> >>> >> >> > FreeSWITCH-users mailing list
>> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-
>> >>> >> >> >use rs http://www.freeswitch.org
>> >>> >> >>
>> >>> >> >> _______________________________________________
>> >>> >> >> FreeSWITCH-users mailing list
>> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-us
>> >>> >> >>ers http://www.freeswitch.org
>> >>> >> >
>> >>> >> > --
>> >>> >> > Anthony Minessale II
>> >>> >> >
>> >>> >> > FreeSWITCH http://www.freeswitch.org/
>> >>> >> > ClueCon http://www.cluecon.com/
>> >>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>> >> >
>> >>> >> > AIM: anthm
>> >>> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >>> >> > IRC: irc.freenode.net #freeswitch
>> >>> >> >
>> >>> >> > FreeSWITCH Developer Conference
>> >>> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >>> >> > iax:guest at conference.freeswitch.org/888
>> >>> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >>> >> > pstn:+19193869900
>> >>> >> >
>> >>> >> > _______________________________________________
>> >>> >> > FreeSWITCH-users mailing list
>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-use
>> >>> >> >rs http://www.freeswitch.org
>> >>> >>
>> >>> >> _______________________________________________
>> >>> >> FreeSWITCH-users mailing list
>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> http://www.freeswitch.org
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at jerris.com  Fri Jan 15 13:24:04 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 Jan 2010 16:24:04 -0500
Subject: [Freeswitch-users] xset warning message starting FS as daemon
In-Reply-To: <10273ADF13B842FE8527E8FF93BD6C28@greyhawk.tonecommander.com>
References: <10273ADF13B842FE8527E8FF93BD6C28@greyhawk.tonecommander.com>
Message-ID: <E1A9E241-30F9-4470-BEAC-869E265341D6@jerris.com>

Its something in the init script.  are you using one from the freeswitch tree or your own?

Mike

On Jan 15, 2010, at 4:07 PM, Robert Hadley wrote:

> Hi,
>  
> Running trunk on CentOS 5.3, I don?t get any warning messages starting FS manually with the ?nc option.
>  
> I get this message when I start FS as a daemon.
>  
> [root at roberth-c53 fstrkbld]# service freeswitch start
> Starting freeswitch: xset:  unable to open display ""
> xset:  unable to open display ""
>                                                            [  OK  ]
>  
>  
> Is this due to something I am doing or is it FS?
>  
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From lawwton at gmail.com  Fri Jan 15 14:37:24 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 17:37:24 -0500
Subject: [Freeswitch-users] Issue Compiling ESL (PHP)
Message-ID: <5fe6fa8f1001151437x491dade6q8b316f312556ff07@mail.gmail.com>

All:

System:

Linux usnc-rtp-01-fs 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT
2007 i686 i686 i386 GNU/Linux

I am trying to compile ESL, following the following steps:

1- cd to my libs/esl directory as the wiki page indicates.
2- run make

I then get right away the following:

cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o
cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o
src/esl_event.o
cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o
src/esl_threadmutex.o
cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o
src/esl_config.o
g++  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
-Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o
ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o
src/esl_config.o src/esl_oop.o
ranlib libesl.a
cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L.
-L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit
cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient
-L. -lncurses -lpthread -lesl
cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver
-L. -lncurses -lpthread -lesl
cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes ivrd.c -o ivrd -L. -lncurses
-lpthread -lesl

3- I try typing then:

make phpmod and get the following:

make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
CFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes"
CXXFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
-Wno-unused-variable" CXX_CFLAGS="" -C php
make[1]: Entering directory
`/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
g++  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
-Wno-unused-variable -I/usr/include/php -I/usr/include/php/main
-I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext
-I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function
-c esl_wrap.cpp -o esl_wrap.o
g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/kerberos/lib
-lcrypt -lcrypt -lncurses -laspell -lpspell -lcurl -lresolv -lm -ldl
-lnsl -lm -ldl -lcurl -ldl -lm -lcrypt -lm -lm -lcrypt -lpthread -o
ESL.so -L.
make[1]: Leaving directory
`/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'

I've installed I think all the -dev dependencies listed in the wiki. Any ideas?

Thanks in advance,

Alfredo Q-V


From anthony.minessale at gmail.com  Fri Jan 15 14:45:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 16:45:52 -0600
Subject: [Freeswitch-users] eavesdrop problem?
In-Reply-To: <20100115174147.30E6A11F5A@mail.nstel.ru>
References: <191c3a031001150805y2c6f0bd0pcea8a18d9d866b4b@mail.gmail.com>
	<20100115174147.30E6A11F5A@mail.nstel.ru>
Message-ID: <191c3a031001151445n51ba1514rb387179bb837c558@mail.gmail.com>

yes, exactly.
That is a demo of how you could possibly store a uuid by inserting them into
the db keyed from your user extension in the caller id.

if you do show channel and you see the uuid for each leg that is the
argument eavesdrop takes.
you can also do "all" in place of a uuid so you can cycle all the calls with
DTMF



On Fri, Jan 15, 2010 at 11:41 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

>  Anthony,
>
> Thanks for the reply.
>
> Can you please point me to the document where I could read about it?
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop does not
> say anything about it?
>
> But let me guess: I should add
>
> <action application="hash"
> data="insert/${domain_name}-spymap/${destination_number}/${uuid}"/>
>
> Into <extension name="global" continue="true"> in the dialplan.
>
> Am I close?
>
> Thanks and regards,
>
> Nikolay.
>
>
>   ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Friday, January 15, 2010 7:05 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] eavesdrop problem?
>
>
>
> don't bother,
>
>
> only inbound legs are added to the db that is used to lookup for eavesdrop
> because the action is in the dialplan.
> The extensions to eavesdrop you are using are just a demo to show you how
> to work it.
> you need to know the uuid of the channel you are trying to eavesdrop on
> before you can do what you want.
>
>
>
>  On Fri, Jan 15, 2010 at 8:33 AM, Nikolay Kondratyev <kond at nstel.ru>
> wrote:
>
> Hi all,
>
>
>
> I want to use eavesdrop application.
>
> Playing with it I found that when one tries to eavesdrop caller the feature
> works ok.
>
> But when trying to eavesdrop callee eavesdrop attempt failes.
>
> I just updated to the latest version from http://latest.freeswitch.org
>
> [freeswitch at freeswitch log]$ fs_cli -x version
>
> FreeSWITCH Version 1.0.5-20100115-0400 (16318M)
>
>
>
> My setup is as following:
>
> I have 3 internal extensions: 2853, 2840, 2879 (all are xlite).
>
> 1. 2879 calls 2840. Then 2853 dials 882879 ? eavesdrop worked as expected.
>
> 2. 2840 calls 2879. Then 2853 dials 882879 ? eavesdrop failed.
>
>
>
> I attached logs for both cases.
>
>
>
> I don?t believe it?s intended behavior.
>
>
>
> Can anybody please advise if it is a configuration or a software problem?
>
>
>
> Thanks and regards,
>
> Nikolay.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From david.villasmil.work at gmail.com  Fri Jan 15 14:45:59 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 23:45:59 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001151317j408794a3pa2a0a5d21d63aa62@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
	<201001150726.17430.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>
	<9853f4ff1001150443t5a18cc6bxa5032f391ab6f0ed@mail.gmail.com>
	<191c3a031001150808i69b3937boe0e903057f984e96@mail.gmail.com>
	<191c3a031001151317j408794a3pa2a0a5d21d63aa62@mail.gmail.com>
Message-ID: <9853f4ff1001151445s6a4dff0qf8c9e17c702b7f86@mail.gmail.com>

Anthony,

     Trying, Thanks. Is there anyway we can communicate directly?


David

On Fri, Jan 15, 2010 at 10:17 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Try latest trunk,
>
> you should have exactly what you want with the same parameter, again my
> paypal addr is cleary displayed as a big button on the website.
>
>
> On Fri, Jan 15, 2010 at 10:08 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> one of the many reasons its a bad idea.
>> Probably the leg with the bad audio is a different ptime.
>> Now the amount of work I have to do escalates I would prefer you commit to
>> commercial support by emailing me at consulting at freeswitch.org to continue
>> with this.
>>
>>
>> On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil
>> <david.villasmil.work at gmail.com> wrote:
>>>
>>> I set it to "off" just in case, same thing.
>>>
>>> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
>>> <david.villasmil.work at gmail.com> wrote:
>>> > Default, haven't touched it i suppose it's off, i haven't set it
>>> > anywhere
>>> >
>>> > On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
>>> > <sos at sokhapkin.dyndns.org> wrote:
>>> >> Is bypass_media on or off?
>>> >>
>>> >> On Friday 15 January 2010, David Villasmil wrote:
>>> >>> Yeah, sorry. Early media. Audio after 200 is fine. Early media was ok
>>> >>> before the change.
>>> >>>
>>> >>> David
>>> >>>
>>> >>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
>>> >>>
>>> >>> <sos at sokhapkin.dyndns.org> wrote:
>>> >>> > Which audio? Early media or after 200 OK?
>>> >>> >
>>> >>> > On Friday 15 January 2010, David Villasmil wrote:
>>> >>> >> Hello again Anthony,
>>> >>> >>
>>> >>> >> I just tested it, and although functionality does not, first
>>> >>> >> incoming
>>> >>> >> audio is coming in all garbled... do you know why?
>>> >>> >>
>>> >>> >> David
>>> >>> >>
>>> >>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>>> >>> >>
>>> >>> >> <anthony.minessale at gmail.com> wrote:
>>> >>> >> > {bridge_early_media=true}
>>> >>> >> > in the dial string in place of ignore_early_media=true
>>> >>> >> >
>>> >>> >> > the first b leg in the list who sends 183 will become the
>>> >>> >> > ringback
>>> >>> >> > device for A leg it will hear the early media
>>> >>> >> > for that leg while the other legs still ring.? If some other leg
>>> >>> >> > answers the final call will still be bridged to the leg who
>>> >>> >> > answered.
>>> >>> >> >
>>> >>> >> >
>>> >>> >> > I would estimate it at $500 payable on the big paypal button on
>>> >>> >> > http://www.freeswitch.org
>>> >>> >> > but, I already added the patch to tree earlier today so I guess
>>> >>> >> > it's
>>> >>> >> > up to you to pay it or not.
>>> >>> >> >
>>> >>> >> >
>>> >>> >> >
>>> >>> >> >
>>> >>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
>>> >>> >> >
>>> >>> >> > <david.villasmil.work at gmail.com> wrote:
>>> >>> >> >> Anthony,
>>> >>> >> >>
>>> >>> >> >> I did take the "hint", don't worry. We will probably ask for a
>>> >>> >> >> bounty
>>> >>> >> >> but first we need to know:
>>> >>> >> >> 1.- whether this is possible
>>> >>> >> >> 2.- how long it would take
>>> >>> >> >> 3.- how will it exactly work
>>> >>> >> >> 4.- of course, what's the bounty (be gentle ;) )
>>> >>> >> >>
>>> >>> >> >> We would of course give this back to the community.
>>> >>> >> >>
>>> >>> >> >> in the meantime, bridge_early_media=true will discard the 2nd
>>> >>> >> >> B-leg
>>> >>> >> >> and conversely "false" it will keep on trying to connect and if
>>> >>> >> >> it
>>> >>> >> >> connects the other B-leg if will bridge to that one?
>>> >>> >> >>
>>> >>> >> >> Thanks
>>> >>> >> >>
>>> >>> >> >> David
>>> >>> >> >>
>>> >>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>>> >>> >> >>
>>> >>> >> >> <anthony.minessale at gmail.com> wrote:
>>> >>> >> >> > I added bridge_early_media=true to do the best I can do.
>>> >>> >> >> > This is the most I will do, especially for free, nobody can
>>> >>> >> >> > take a
>>> >>> >> >> > hint that
>>> >>> >> >> > you should be paying for all these custom requests so take it
>>> >>> >> >> > or
>>> >>> >> >> > leave it
>>> >>> >> >> > but this thread is done.........
>>> >>> >> >> >
>>> >>> >> >> >
>>> >>> >> >> >
>>> >>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>>> >>> >> >> > <sos at sokhapkin.dyndns.org>
>>> >>> >> >> >
>>> >>> >> >> > wrote:
>>> >>> >> >> >> No, not exactly. ignore_early_media doesn't pass early media
>>> >>> >> >> >> to
>>> >>> >> >> >> the caller
>>> >>> >> >> >> if
>>> >>> >> >> >> bypass_media is false.
>>> >>> >> >> >>
>>> >>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
>>> >>> >> >> >> > this is exactly what ignore_early_media does now.
>>> >>> >> >> >> >
>>> >>> >> >> >> > Mike
>>> >>> >> >> >> >
>>> >>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>>> >>> >> >> >> > > The issue here is when "originate" routine should return
>>> >>> >> >> >> > > and
>>> >>> >> >> >> > > set "originate_status" variable. Current behavior is to
>>> >>> >> >> >> > > return
>>> >>> >> >> >> > > on early
>>> >>> >> >> >> > > media, but what if to introduce a variable
>>> >>> >> >> >> > > "originate_wait_for_answer"
>>> >>> >> >> >> > > with default value "false" and use the variable in
>>> >>> >> >> >> > > originate
>>> >>> >> >> >> > > code to
>>> >>> >> >> >> > > decide when to return - on 18X or "200 OK"?
>>> >>> >> >> >> > >
>>> >>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>>> >>> >> >> >> > >> he wants to call 3 people at once and let the A leg
>>> >>> >> >> >> > >> hear
>>> >>> >> >> >> > >> early media
>>> >>> >> >> >> > >> from call #1 while call #2 and #3 still are progressing
>>> >>> >> >> >> > >> which
>>> >>> >> >> >> > >> is not
>>> >>> >> >> >> > >> simple to do without doing thousands of dollars in
>>> >>> >> >> >> > >> development.
>>> >>> >> >> >> > >>
>>> >>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB
>>> >>> >> >> >> > >> <djbinter at yahoo.com>
>>> >> wrote:
>>> >>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so
>>> >>> >> >> >> > >>> that your
>>> >>> >> >> >> > >>> customers
>>> >>> >> >> >> > >>> can hear recordings?
>>> >>> >> >> >> >
>>> >>> >> >> >> > _______________________________________________
>>> >>> >> >> >> > FreeSWITCH-users mailing list
>>> >>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >>> >> >> >> >
>>> >>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >> >> >> >
>>> >>> >> >> >> >
>>> >>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswit
>>> >>> >> >> >> >ch- users http://www.freeswitch.org
>>> >>> >> >> >>
>>> >>> >> >> >> _______________________________________________
>>> >>> >> >> >> FreeSWITCH-users mailing list
>>> >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >>> >> >> >>
>>> >>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >> >> >>
>>> >>> >> >> >>
>>> >>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch
>>> >>> >> >> >>-us ers http://www.freeswitch.org
>>> >>> >> >> >
>>> >>> >> >> > --
>>> >>> >> >> > Anthony Minessale II
>>> >>> >> >> >
>>> >>> >> >> > FreeSWITCH http://www.freeswitch.org/
>>> >>> >> >> > ClueCon http://www.cluecon.com/
>>> >>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >>> >> >> >
>>> >>> >> >> > AIM: anthm
>>> >>> >> >> > MSN:anthony_minessale at hotmail.com
>>> >>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> >>> >> >> > IRC: irc.freenode.net #freeswitch
>>> >>> >> >> >
>>> >>> >> >> > FreeSWITCH Developer Conference
>>> >>> >> >> > sip:888 at conference.freeswitch.org
>>> >>> >> >> > iax:guest at conference.freeswitch.org/888
>>> >>> >> >> > googletalk:conf+888 at conference.freeswitch.org
>>> >>> >> >> > pstn:+19193869900
>>> >>> >> >> >
>>> >>> >> >> > _______________________________________________
>>> >>> >> >> > FreeSWITCH-users mailing list
>>> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >> >> >
>>> >>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> >>> >> >> >use rs http://www.freeswitch.org
>>> >>> >> >>
>>> >>> >> >> _______________________________________________
>>> >>> >> >> FreeSWITCH-users mailing list
>>> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >> >>
>>> >>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>>> >>> >> >>ers http://www.freeswitch.org
>>> >>> >> >
>>> >>> >> > --
>>> >>> >> > Anthony Minessale II
>>> >>> >> >
>>> >>> >> > FreeSWITCH http://www.freeswitch.org/
>>> >>> >> > ClueCon http://www.cluecon.com/
>>> >>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >>> >> >
>>> >>> >> > AIM: anthm
>>> >>> >> > MSN:anthony_minessale at hotmail.com
>>> >>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> >>> >> > IRC: irc.freenode.net #freeswitch
>>> >>> >> >
>>> >>> >> > FreeSWITCH Developer Conference
>>> >>> >> > sip:888 at conference.freeswitch.org
>>> >>> >> > iax:guest at conference.freeswitch.org/888
>>> >>> >> > googletalk:conf+888 at conference.freeswitch.org
>>> >>> >> > pstn:+19193869900
>>> >>> >> >
>>> >>> >> > _______________________________________________
>>> >>> >> > FreeSWITCH-users mailing list
>>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >> >
>>> >>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>> >>> >> >rs http://www.freeswitch.org
>>> >>> >>
>>> >>> >> _______________________________________________
>>> >>> >> FreeSWITCH-users mailing list
>>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >>
>>> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> >> http://www.freeswitch.org
>>> >>> >
>>> >>> > _______________________________________________
>>> >>> > FreeSWITCH-users mailing list
>>> >>> > FreeSWITCH-users at lists.freeswitch.org
>>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >
>>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> > http://www.freeswitch.org
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>
>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From lawwton at gmail.com  Fri Jan 15 14:54:42 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 17:54:42 -0500
Subject: [Freeswitch-users] Issue Compiling ESL (PHP)
In-Reply-To: <5fe6fa8f1001151437x491dade6q8b316f312556ff07@mail.gmail.com>
References: <5fe6fa8f1001151437x491dade6q8b316f312556ff07@mail.gmail.com>
Message-ID: <5fe6fa8f1001151454m92d17c9le4b811c897ae8482@mail.gmail.com>

Looking over at this, perhaps it even worked.

I see now under libs/esl/php/... the following two new files:

esl_wrap.o
ESL.o

Is there a way to verify that FS has support after running make and
make phpmod for php?

Thanks in advance,

Alfredo

On Fri, Jan 15, 2010 at 5:37 PM, Alfredo Quiroga-Villamil
<lawwton at gmail.com> wrote:
> All:
>
> System:
>
> Linux usnc-rtp-01-fs 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT
> 2007 i686 i686 i386 GNU/Linux
>
> I am trying to compile ESL, following the following steps:
>
> 1- cd to my libs/esl directory as the wiki page indicates.
> 2- run make
>
> I then get right away the following:
>
> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o
> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o
> src/esl_event.o
> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o
> src/esl_threadmutex.o
> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o
> src/esl_config.o
> g++ ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
> -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o
> ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o
> src/esl_config.o src/esl_oop.o
> ranlib libesl.a
> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L.
> -L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit
> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient
> -L. -lncurses -lpthread -lesl
> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver
> -L. -lncurses -lpthread -lesl
> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes ivrd.c -o ivrd -L. -lncurses
> -lpthread -lesl
>
> 3- I try typing then:
>
> make phpmod and get the following:
>
> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
> CFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes"
> CXXFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
> -Wno-unused-variable" CXX_CFLAGS="" -C php
> make[1]: Entering directory
> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
> g++ ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
> -Wno-unused-variable -I/usr/include/php -I/usr/include/php/main
> -I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext
> -I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function
> -c esl_wrap.cpp -o esl_wrap.o
> g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/kerberos/lib
> -lcrypt -lcrypt -lncurses -laspell -lpspell -lcurl -lresolv -lm -ldl
> -lnsl -lm -ldl -lcurl -ldl -lm -lcrypt -lm -lm -lcrypt -lpthread -o
> ESL.so -L.
> make[1]: Leaving directory
> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>
> I've installed I think all the -dev dependencies listed in the wiki. Any ideas?
>
> Thanks in advance,
>
> Alfredo Q-V
>


From anthony.minessale at gmail.com  Fri Jan 15 14:54:25 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 16:54:25 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001151445s6a4dff0qf8c9e17c702b7f86@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
	<201001150726.17430.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>
	<9853f4ff1001150443t5a18cc6bxa5032f391ab6f0ed@mail.gmail.com>
	<191c3a031001150808i69b3937boe0e903057f984e96@mail.gmail.com>
	<191c3a031001151317j408794a3pa2a0a5d21d63aa62@mail.gmail.com>
	<9853f4ff1001151445s6a4dff0qf8c9e17c702b7f86@mail.gmail.com>
Message-ID: <191c3a031001151454w101ae8eh978e4ced42004fb3@mail.gmail.com>

you can email me privately at this addr.


On Fri, Jan 15, 2010 at 4:45 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Anthony,
>
>     Trying, Thanks. Is there anyway we can communicate directly?
>
>
> David
>
> On Fri, Jan 15, 2010 at 10:17 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > Try latest trunk,
> >
> > you should have exactly what you want with the same parameter, again my
> > paypal addr is cleary displayed as a big button on the website.
> >
> >
> > On Fri, Jan 15, 2010 at 10:08 AM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> one of the many reasons its a bad idea.
> >> Probably the leg with the bad audio is a different ptime.
> >> Now the amount of work I have to do escalates I would prefer you commit
> to
> >> commercial support by emailing me at consulting at freeswitch.org to
> continue
> >> with this.
> >>
> >>
> >> On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil
> >> <david.villasmil.work at gmail.com> wrote:
> >>>
> >>> I set it to "off" just in case, same thing.
> >>>
> >>> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
> >>> <david.villasmil.work at gmail.com> wrote:
> >>> > Default, haven't touched it i suppose it's off, i haven't set it
> >>> > anywhere
> >>> >
> >>> > On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
> >>> > <sos at sokhapkin.dyndns.org> wrote:
> >>> >> Is bypass_media on or off?
> >>> >>
> >>> >> On Friday 15 January 2010, David Villasmil wrote:
> >>> >>> Yeah, sorry. Early media. Audio after 200 is fine. Early media was
> ok
> >>> >>> before the change.
> >>> >>>
> >>> >>> David
> >>> >>>
> >>> >>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
> >>> >>>
> >>> >>> <sos at sokhapkin.dyndns.org> wrote:
> >>> >>> > Which audio? Early media or after 200 OK?
> >>> >>> >
> >>> >>> > On Friday 15 January 2010, David Villasmil wrote:
> >>> >>> >> Hello again Anthony,
> >>> >>> >>
> >>> >>> >> I just tested it, and although functionality does not, first
> >>> >>> >> incoming
> >>> >>> >> audio is coming in all garbled... do you know why?
> >>> >>> >>
> >>> >>> >> David
> >>> >>> >>
> >>> >>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
> >>> >>> >>
> >>> >>> >> <anthony.minessale at gmail.com> wrote:
> >>> >>> >> > {bridge_early_media=true}
> >>> >>> >> > in the dial string in place of ignore_early_media=true
> >>> >>> >> >
> >>> >>> >> > the first b leg in the list who sends 183 will become the
> >>> >>> >> > ringback
> >>> >>> >> > device for A leg it will hear the early media
> >>> >>> >> > for that leg while the other legs still ring.  If some other
> leg
> >>> >>> >> > answers the final call will still be bridged to the leg who
> >>> >>> >> > answered.
> >>> >>> >> >
> >>> >>> >> >
> >>> >>> >> > I would estimate it at $500 payable on the big paypal button
> on
> >>> >>> >> > http://www.freeswitch.org
> >>> >>> >> > but, I already added the patch to tree earlier today so I
> guess
> >>> >>> >> > it's
> >>> >>> >> > up to you to pay it or not.
> >>> >>> >> >
> >>> >>> >> >
> >>> >>> >> >
> >>> >>> >> >
> >>> >>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> >>> >>> >> >
> >>> >>> >> > <david.villasmil.work at gmail.com> wrote:
> >>> >>> >> >> Anthony,
> >>> >>> >> >>
> >>> >>> >> >> I did take the "hint", don't worry. We will probably ask for
> a
> >>> >>> >> >> bounty
> >>> >>> >> >> but first we need to know:
> >>> >>> >> >> 1.- whether this is possible
> >>> >>> >> >> 2.- how long it would take
> >>> >>> >> >> 3.- how will it exactly work
> >>> >>> >> >> 4.- of course, what's the bounty (be gentle ;) )
> >>> >>> >> >>
> >>> >>> >> >> We would of course give this back to the community.
> >>> >>> >> >>
> >>> >>> >> >> in the meantime, bridge_early_media=true will discard the 2nd
> >>> >>> >> >> B-leg
> >>> >>> >> >> and conversely "false" it will keep on trying to connect and
> if
> >>> >>> >> >> it
> >>> >>> >> >> connects the other B-leg if will bridge to that one?
> >>> >>> >> >>
> >>> >>> >> >> Thanks
> >>> >>> >> >>
> >>> >>> >> >> David
> >>> >>> >> >>
> >>> >>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> >>> >>> >> >>
> >>> >>> >> >> <anthony.minessale at gmail.com> wrote:
> >>> >>> >> >> > I added bridge_early_media=true to do the best I can do.
> >>> >>> >> >> > This is the most I will do, especially for free, nobody can
> >>> >>> >> >> > take a
> >>> >>> >> >> > hint that
> >>> >>> >> >> > you should be paying for all these custom requests so take
> it
> >>> >>> >> >> > or
> >>> >>> >> >> > leave it
> >>> >>> >> >> > but this thread is done.........
> >>> >>> >> >> >
> >>> >>> >> >> >
> >>> >>> >> >> >
> >>> >>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
> >>> >>> >> >> > <sos at sokhapkin.dyndns.org>
> >>> >>> >> >> >
> >>> >>> >> >> > wrote:
> >>> >>> >> >> >> No, not exactly. ignore_early_media doesn't pass early
> media
> >>> >>> >> >> >> to
> >>> >>> >> >> >> the caller
> >>> >>> >> >> >> if
> >>> >>> >> >> >> bypass_media is false.
> >>> >>> >> >> >>
> >>> >>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
> >>> >>> >> >> >> > this is exactly what ignore_early_media does now.
> >>> >>> >> >> >> >
> >>> >>> >> >> >> > Mike
> >>> >>> >> >> >> >
> >>> >>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> >>> >>> >> >> >> > > The issue here is when "originate" routine should
> return
> >>> >>> >> >> >> > > and
> >>> >>> >> >> >> > > set "originate_status" variable. Current behavior is
> to
> >>> >>> >> >> >> > > return
> >>> >>> >> >> >> > > on early
> >>> >>> >> >> >> > > media, but what if to introduce a variable
> >>> >>> >> >> >> > > "originate_wait_for_answer"
> >>> >>> >> >> >> > > with default value "false" and use the variable in
> >>> >>> >> >> >> > > originate
> >>> >>> >> >> >> > > code to
> >>> >>> >> >> >> > > decide when to return - on 18X or "200 OK"?
> >>> >>> >> >> >> > >
> >>> >>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> >>> >>> >> >> >> > >> he wants to call 3 people at once and let the A leg
> >>> >>> >> >> >> > >> hear
> >>> >>> >> >> >> > >> early media
> >>> >>> >> >> >> > >> from call #1 while call #2 and #3 still are
> progressing
> >>> >>> >> >> >> > >> which
> >>> >>> >> >> >> > >> is not
> >>> >>> >> >> >> > >> simple to do without doing thousands of dollars in
> >>> >>> >> >> >> > >> development.
> >>> >>> >> >> >> > >>
> >>> >>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB
> >>> >>> >> >> >> > >> <djbinter at yahoo.com>
> >>> >> wrote:
> >>> >>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so
> >>> >>> >> >> >> > >>> that your
> >>> >>> >> >> >> > >>> customers
> >>> >>> >> >> >> > >>> can hear recordings?
> >>> >>> >> >> >> >
> >>> >>> >> >> >> > _______________________________________________
> >>> >>> >> >> >> > FreeSWITCH-users mailing list
> >>> >>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >>> >> >> >> >
> >>> >>> >> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>> >> >> >> >
> >>> >>> >> >> >> >
> >>> >>> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswit
> >>> >>> >> >> >> >ch- users http://www.freeswitch.org
> >>> >>> >> >> >>
> >>> >>> >> >> >> _______________________________________________
> >>> >>> >> >> >> FreeSWITCH-users mailing list
> >>> >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >>> >> >> >>
> >>> >>> >> >> >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>> >> >> >>
> >>> >>> >> >> >>
> >>> >>> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch
> >>> >>> >> >> >>-us ers http://www.freeswitch.org
> >>> >>> >> >> >
> >>> >>> >> >> > --
> >>> >>> >> >> > Anthony Minessale II
> >>> >>> >> >> >
> >>> >>> >> >> > FreeSWITCH http://www.freeswitch.org/
> >>> >>> >> >> > ClueCon http://www.cluecon.com/
> >>> >>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>> >>> >> >> >
> >>> >>> >> >> > AIM: anthm
> >>> >>> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> >>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> >>> >> >> > IRC: irc.freenode.net #freeswitch
> >>> >>> >> >> >
> >>> >>> >> >> > FreeSWITCH Developer Conference
> >>> >>> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> >>> >> >> > iax:guest at conference.freeswitch.org/888
> >>> >>> >> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> >>> >> >> > pstn:+19193869900
> >>> >>> >> >> >
> >>> >>> >> >> > _______________________________________________
> >>> >>> >> >> > FreeSWITCH-users mailing list
> >>> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >>> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>> >> >> >
> >>> >>> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-
> >>> >>> >> >> >use rs http://www.freeswitch.org
> >>> >>> >> >>
> >>> >>> >> >> _______________________________________________
> >>> >>> >> >> FreeSWITCH-users mailing list
> >>> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >>> >> >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>> >> >>
> >>> >>> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-us
> >>> >>> >> >>ers http://www.freeswitch.org
> >>> >>> >> >
> >>> >>> >> > --
> >>> >>> >> > Anthony Minessale II
> >>> >>> >> >
> >>> >>> >> > FreeSWITCH http://www.freeswitch.org/
> >>> >>> >> > ClueCon http://www.cluecon.com/
> >>> >>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>> >>> >> >
> >>> >>> >> > AIM: anthm
> >>> >>> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> >>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> >>> >> > IRC: irc.freenode.net #freeswitch
> >>> >>> >> >
> >>> >>> >> > FreeSWITCH Developer Conference
> >>> >>> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> >>> >> > iax:guest at conference.freeswitch.org/888
> >>> >>> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> >>> >> > pstn:+19193869900
> >>> >>> >> >
> >>> >>> >> > _______________________________________________
> >>> >>> >> > FreeSWITCH-users mailing list
> >>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>> >> >
> >>> >>> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-use
> >>> >>> >> >rs http://www.freeswitch.org
> >>> >>> >>
> >>> >>> >> _______________________________________________
> >>> >>> >> FreeSWITCH-users mailing list
> >>> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>> >>
> >>> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >>> >> http://www.freeswitch.org
> >>> >>> >
> >>> >>> > _______________________________________________
> >>> >>> > FreeSWITCH-users mailing list
> >>> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>> >
> >>> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >>> > http://www.freeswitch.org
> >>> >>>
> >>> >>> _______________________________________________
> >>> >>> FreeSWITCH-users mailing list
> >>> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>>
> >>> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >>> http://www.freeswitch.org
> >>> >>
> >>> >>
> >>> >>
> >>> >> _______________________________________________
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>
> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >>
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From robert.hadley at teotech.com  Fri Jan 15 15:01:04 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 15 Jan 2010 15:01:04 -0800
Subject: [Freeswitch-users] xset warning message starting FS as daemon
In-Reply-To: <E1A9E241-30F9-4470-BEAC-869E265341D6@jerris.com>
References: <10273ADF13B842FE8527E8FF93BD6C28@greyhawk.tonecommander.com>
	<E1A9E241-30F9-4470-BEAC-869E265341D6@jerris.com>
Message-ID: <96752801704D4597954A9ED5E71E2399@greyhawk.tonecommander.com>

Hi Mike,

 

I took the one from FS build/freeswitch.init.redhat and modified it for my
paths in the /opt folder and set the FS-user to root for now (was going to
change user to freeswitch next).

 

Thanks,

Robert

 

  _____  

From: Michael Jerris [mailto:mike at jerris.com] 
Sent: Friday, January 15, 2010 1:24 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] xset warning message starting FS as daemon

 

Its something in the init script.  are you using one from the freeswitch
tree or your own?

 

Mike

 

On Jan 15, 2010, at 4:07 PM, Robert Hadley wrote:





Hi,

 

Running trunk on CentOS 5.3, I don't get any warning messages starting FS
manually with the -nc option.

 

I get this message when I start FS as a daemon.

 

[root at roberth-c53 fstrkbld]# service freeswitch start

Starting freeswitch: xset:  unable to open display ""

xset:  unable to open display ""

                                                           [  OK  ]

 

 

Is this due to something I am doing or is it FS?

 

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From anthony.minessale at gmail.com  Fri Jan 15 15:05:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 17:05:14 -0600
Subject: [Freeswitch-users] Issue Compiling ESL (PHP)
In-Reply-To: <5fe6fa8f1001151454m92d17c9le4b811c897ae8482@mail.gmail.com>
References: <5fe6fa8f1001151437x491dade6q8b316f312556ff07@mail.gmail.com>
	<5fe6fa8f1001151454m92d17c9le4b811c897ae8482@mail.gmail.com>
Message-ID: <191c3a031001151505y28e8e759sf6f3239f61d4b6b7@mail.gmail.com>

i think there is a .php and a .so you could install into your php lib dir?


On Fri, Jan 15, 2010 at 4:54 PM, Alfredo Quiroga-Villamil <lawwton at gmail.com
> wrote:

> Looking over at this, perhaps it even worked.
>
> I see now under libs/esl/php/... the following two new files:
>
> esl_wrap.o
> ESL.o
>
> Is there a way to verify that FS has support after running make and
> make phpmod for php?
>
> Thanks in advance,
>
> Alfredo
>
> On Fri, Jan 15, 2010 at 5:37 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
> > All:
> >
> > System:
> >
> > Linux usnc-rtp-01-fs 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT
> > 2007 i686 i686 i386 GNU/Linux
> >
> > I am trying to compile ESL, following the following steps:
> >
> > 1- cd to my libs/esl directory as the wiki page indicates.
> > 2- run make
> >
> > I then get right away the following:
> >
> > cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o
> > cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o
> > src/esl_event.o
> > cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o
> > src/esl_threadmutex.o
> > cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o
> > src/esl_config.o
> > g++  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
> > -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o
> > ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o
> > src/esl_config.o src/esl_oop.o
> > ranlib libesl.a
> > cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L.
> > -L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit
> > cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient
> > -L. -lncurses -lpthread -lesl
> > cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver
> > -L. -lncurses -lpthread -lesl
> > cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes ivrd.c -o ivrd -L. -lncurses
> > -lpthread -lesl
> >
> > 3- I try typing then:
> >
> > make phpmod and get the following:
> >
> > make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
> > CFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes"
> > CXXFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
> > -Wno-unused-variable" CXX_CFLAGS="" -C php
> > make[1]: Entering directory
> > `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
> > g++  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
> > -Wno-unused-variable -I/usr/include/php -I/usr/include/php/main
> > -I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext
> > -I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function
> > -c esl_wrap.cpp -o esl_wrap.o
> > g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/kerberos/lib
> > -lcrypt -lcrypt -lncurses -laspell -lpspell -lcurl -lresolv -lm -ldl
> > -lnsl -lm -ldl -lcurl -ldl -lm -lcrypt -lm -lm -lcrypt -lpthread -o
> > ESL.so -L.
> > make[1]: Leaving directory
> > `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
> >
> > I've installed I think all the -dev dependencies listed in the wiki. Any
> ideas?
> >
> > Thanks in advance,
> >
> > Alfredo Q-V
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From david.villasmil.work at gmail.com  Fri Jan 15 15:07:55 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 16 Jan 2010 00:07:55 +0100
Subject: [Freeswitch-users] Issue Compiling ESL (PHP)
In-Reply-To: <5fe6fa8f1001151454m92d17c9le4b811c897ae8482@mail.gmail.com>
References: <5fe6fa8f1001151437x491dade6q8b316f312556ff07@mail.gmail.com>
	<5fe6fa8f1001151454m92d17c9le4b811c897ae8482@mail.gmail.com>
Message-ID: <9853f4ff1001151507oa1fa8adt5101956739d8c948@mail.gmail.com>

make a script ;)

it's very easy, try it!

David

On Fri, Jan 15, 2010 at 11:54 PM, Alfredo Quiroga-Villamil
<lawwton at gmail.com> wrote:
> Looking over at this, perhaps it even worked.
>
> I see now under libs/esl/php/... the following two new files:
>
> esl_wrap.o
> ESL.o
>
> Is there a way to verify that FS has support after running make and
> make phpmod for php?
>
> Thanks in advance,
>
> Alfredo
>
> On Fri, Jan 15, 2010 at 5:37 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>> All:
>>
>> System:
>>
>> Linux usnc-rtp-01-fs 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT
>> 2007 i686 i686 i386 GNU/Linux
>>
>> I am trying to compile ESL, following the following steps:
>>
>> 1- cd to my libs/esl directory as the wiki page indicates.
>> 2- run make
>>
>> I then get right away the following:
>>
>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o
>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o
>> src/esl_event.o
>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o
>> src/esl_threadmutex.o
>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o
>> src/esl_config.o
>> g++ ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>> -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o
>> ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o
>> src/esl_config.o src/esl_oop.o
>> ranlib libesl.a
>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L.
>> -L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit
>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient
>> -L. -lncurses -lpthread -lesl
>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver
>> -L. -lncurses -lpthread -lesl
>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes ivrd.c -o ivrd -L. -lncurses
>> -lpthread -lesl
>>
>> 3- I try typing then:
>>
>> make phpmod and get the following:
>>
>> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
>> CFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes"
>> CXXFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>> -Wno-unused-variable" CXX_CFLAGS="" -C php
>> make[1]: Entering directory
>> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>> g++ ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>> -Wno-unused-variable -I/usr/include/php -I/usr/include/php/main
>> -I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext
>> -I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function
>> -c esl_wrap.cpp -o esl_wrap.o
>> g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/kerberos/lib
>> -lcrypt -lcrypt -lncurses -laspell -lpspell -lcurl -lresolv -lm -ldl
>> -lnsl -lm -ldl -lcurl -ldl -lm -lcrypt -lm -lm -lcrypt -lpthread -o
>> ESL.so -L.
>> make[1]: Leaving directory
>> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>>
>> I've installed I think all the -dev dependencies listed in the wiki. Any ideas?
>>
>> Thanks in advance,
>>
>> Alfredo Q-V
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From lawwton at gmail.com  Fri Jan 15 15:21:22 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 18:21:22 -0500
Subject: [Freeswitch-users] Issue Compiling ESL (PHP)
In-Reply-To: <9853f4ff1001151507oa1fa8adt5101956739d8c948@mail.gmail.com>
References: <5fe6fa8f1001151437x491dade6q8b316f312556ff07@mail.gmail.com>
	<5fe6fa8f1001151454m92d17c9le4b811c897ae8482@mail.gmail.com>
	<9853f4ff1001151507oa1fa8adt5101956739d8c948@mail.gmail.com>
Message-ID: <5fe6fa8f1001151521u3ef71f9g49491118bf0371ac@mail.gmail.com>

Thanks Anthony/David.

So it seems like the build worked then. I take from the previous
emails and somewhere where I think I read that I can then take:

ESL.so and ESL.php and put them on a remote system under my say for
instance 3rdParty directory and create scripts using the ESL.php
library which probably internally uses ESL.so.

Is that statement correct?

Thanks in advance,

Alfredo

On Fri, Jan 15, 2010 at 6:07 PM, David Villasmil
<david.villasmil.work at gmail.com> wrote:
> make a script ;)
>
> it's very easy, try it!
>
> David
>
> On Fri, Jan 15, 2010 at 11:54 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>> Looking over at this, perhaps it even worked.
>>
>> I see now under libs/esl/php/... the following two new files:
>>
>> esl_wrap.o
>> ESL.o
>>
>> Is there a way to verify that FS has support after running make and
>> make phpmod for php?
>>
>> Thanks in advance,
>>
>> Alfredo
>>
>> On Fri, Jan 15, 2010 at 5:37 PM, Alfredo Quiroga-Villamil
>> <lawwton at gmail.com> wrote:
>>> All:
>>>
>>> System:
>>>
>>> Linux usnc-rtp-01-fs 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT
>>> 2007 i686 i686 i386 GNU/Linux
>>>
>>> I am trying to compile ESL, following the following steps:
>>>
>>> 1- cd to my libs/esl directory as the wiki page indicates.
>>> 2- run make
>>>
>>> I then get right away the following:
>>>
>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o
>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o
>>> src/esl_event.o
>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o
>>> src/esl_threadmutex.o
>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o
>>> src/esl_config.o
>>> g++ ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>> -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o
>>> ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o
>>> src/esl_config.o src/esl_oop.o
>>> ranlib libesl.a
>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L.
>>> -L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit
>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient
>>> -L. -lncurses -lpthread -lesl
>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver
>>> -L. -lncurses -lpthread -lesl
>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes ivrd.c -o ivrd -L. -lncurses
>>> -lpthread -lesl
>>>
>>> 3- I try typing then:
>>>
>>> make phpmod and get the following:
>>>
>>> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
>>> CFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes"
>>> CXXFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>> -Wno-unused-variable" CXX_CFLAGS="" -C php
>>> make[1]: Entering directory
>>> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>>> g++ ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>> -Wno-unused-variable -I/usr/include/php -I/usr/include/php/main
>>> -I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext
>>> -I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function
>>> -c esl_wrap.cpp -o esl_wrap.o
>>> g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/kerberos/lib
>>> -lcrypt -lcrypt -lncurses -laspell -lpspell -lcurl -lresolv -lm -ldl
>>> -lnsl -lm -ldl -lcurl -ldl -lm -lcrypt -lm -lm -lcrypt -lpthread -o
>>> ESL.so -L.
>>> make[1]: Leaving directory
>>> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>>>
>>> I've installed I think all the -dev dependencies listed in the wiki. Any ideas?
>>>
>>> Thanks in advance,
>>>
>>> Alfredo Q-V
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From david.villasmil.work at gmail.com  Fri Jan 15 15:25:10 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 16 Jan 2010 00:25:10 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001151454w101ae8eh978e4ced42004fb3@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
	<201001150726.17430.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>
	<9853f4ff1001150443t5a18cc6bxa5032f391ab6f0ed@mail.gmail.com>
	<191c3a031001150808i69b3937boe0e903057f984e96@mail.gmail.com>
	<191c3a031001151317j408794a3pa2a0a5d21d63aa62@mail.gmail.com>
	<9853f4ff1001151445s6a4dff0qf8c9e17c702b7f86@mail.gmail.com>
	<191c3a031001151454w101ae8eh978e4ced42004fb3@mail.gmail.com>
Message-ID: <9853f4ff1001151525o341cf2c7k2935538cca210815@mail.gmail.com>

Anthony,

LOL, and mounting and mounting... It does work when there is answer...
but if B(2)-side rejects or times out or any other that 200 OK,
B(1)-side stays indefinitely...


On Fri, Jan 15, 2010 at 11:54 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> you can email me privately at this addr.
>
>
> On Fri, Jan 15, 2010 at 4:45 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> Anthony,
>>
>> ? ? Trying, Thanks. Is there anyway we can communicate directly?
>>
>>
>> David
>>
>> On Fri, Jan 15, 2010 at 10:17 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > Try latest trunk,
>> >
>> > you should have exactly what you want with the same parameter, again my
>> > paypal addr is cleary displayed as a big button on the website.
>> >
>> >
>> > On Fri, Jan 15, 2010 at 10:08 AM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> one of the many reasons its a bad idea.
>> >> Probably the leg with the bad audio is a different ptime.
>> >> Now the amount of work I have to do escalates I would prefer you commit
>> >> to
>> >> commercial support by emailing me at consulting at freeswitch.org to
>> >> continue
>> >> with this.
>> >>
>> >>
>> >> On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil
>> >> <david.villasmil.work at gmail.com> wrote:
>> >>>
>> >>> I set it to "off" just in case, same thing.
>> >>>
>> >>> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
>> >>> <david.villasmil.work at gmail.com> wrote:
>> >>> > Default, haven't touched it i suppose it's off, i haven't set it
>> >>> > anywhere
>> >>> >
>> >>> > On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
>> >>> > <sos at sokhapkin.dyndns.org> wrote:
>> >>> >> Is bypass_media on or off?
>> >>> >>
>> >>> >> On Friday 15 January 2010, David Villasmil wrote:
>> >>> >>> Yeah, sorry. Early media. Audio after 200 is fine. Early media was
>> >>> >>> ok
>> >>> >>> before the change.
>> >>> >>>
>> >>> >>> David
>> >>> >>>
>> >>> >>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
>> >>> >>>
>> >>> >>> <sos at sokhapkin.dyndns.org> wrote:
>> >>> >>> > Which audio? Early media or after 200 OK?
>> >>> >>> >
>> >>> >>> > On Friday 15 January 2010, David Villasmil wrote:
>> >>> >>> >> Hello again Anthony,
>> >>> >>> >>
>> >>> >>> >> I just tested it, and although functionality does not, first
>> >>> >>> >> incoming
>> >>> >>> >> audio is coming in all garbled... do you know why?
>> >>> >>> >>
>> >>> >>> >> David
>> >>> >>> >>
>> >>> >>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>> >>> >>> >>
>> >>> >>> >> <anthony.minessale at gmail.com> wrote:
>> >>> >>> >> > {bridge_early_media=true}
>> >>> >>> >> > in the dial string in place of ignore_early_media=true
>> >>> >>> >> >
>> >>> >>> >> > the first b leg in the list who sends 183 will become the
>> >>> >>> >> > ringback
>> >>> >>> >> > device for A leg it will hear the early media
>> >>> >>> >> > for that leg while the other legs still ring.? If some other
>> >>> >>> >> > leg
>> >>> >>> >> > answers the final call will still be bridged to the leg who
>> >>> >>> >> > answered.
>> >>> >>> >> >
>> >>> >>> >> >
>> >>> >>> >> > I would estimate it at $500 payable on the big paypal button
>> >>> >>> >> > on
>> >>> >>> >> > http://www.freeswitch.org
>> >>> >>> >> > but, I already added the patch to tree earlier today so I
>> >>> >>> >> > guess
>> >>> >>> >> > it's
>> >>> >>> >> > up to you to pay it or not.
>> >>> >>> >> >
>> >>> >>> >> >
>> >>> >>> >> >
>> >>> >>> >> >
>> >>> >>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
>> >>> >>> >> >
>> >>> >>> >> > <david.villasmil.work at gmail.com> wrote:
>> >>> >>> >> >> Anthony,
>> >>> >>> >> >>
>> >>> >>> >> >> I did take the "hint", don't worry. We will probably ask for
>> >>> >>> >> >> a
>> >>> >>> >> >> bounty
>> >>> >>> >> >> but first we need to know:
>> >>> >>> >> >> 1.- whether this is possible
>> >>> >>> >> >> 2.- how long it would take
>> >>> >>> >> >> 3.- how will it exactly work
>> >>> >>> >> >> 4.- of course, what's the bounty (be gentle ;) )
>> >>> >>> >> >>
>> >>> >>> >> >> We would of course give this back to the community.
>> >>> >>> >> >>
>> >>> >>> >> >> in the meantime, bridge_early_media=true will discard the
>> >>> >>> >> >> 2nd
>> >>> >>> >> >> B-leg
>> >>> >>> >> >> and conversely "false" it will keep on trying to connect and
>> >>> >>> >> >> if
>> >>> >>> >> >> it
>> >>> >>> >> >> connects the other B-leg if will bridge to that one?
>> >>> >>> >> >>
>> >>> >>> >> >> Thanks
>> >>> >>> >> >>
>> >>> >>> >> >> David
>> >>> >>> >> >>
>> >>> >>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> >>> >>> >> >>
>> >>> >>> >> >> <anthony.minessale at gmail.com> wrote:
>> >>> >>> >> >> > I added bridge_early_media=true to do the best I can do.
>> >>> >>> >> >> > This is the most I will do, especially for free, nobody
>> >>> >>> >> >> > can
>> >>> >>> >> >> > take a
>> >>> >>> >> >> > hint that
>> >>> >>> >> >> > you should be paying for all these custom requests so take
>> >>> >>> >> >> > it
>> >>> >>> >> >> > or
>> >>> >>> >> >> > leave it
>> >>> >>> >> >> > but this thread is done.........
>> >>> >>> >> >> >
>> >>> >>> >> >> >
>> >>> >>> >> >> >
>> >>> >>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>> >>> >>> >> >> > <sos at sokhapkin.dyndns.org>
>> >>> >>> >> >> >
>> >>> >>> >> >> > wrote:
>> >>> >>> >> >> >> No, not exactly. ignore_early_media doesn't pass early
>> >>> >>> >> >> >> media
>> >>> >>> >> >> >> to
>> >>> >>> >> >> >> the caller
>> >>> >>> >> >> >> if
>> >>> >>> >> >> >> bypass_media is false.
>> >>> >>> >> >> >>
>> >>> >>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
>> >>> >>> >> >> >> > this is exactly what ignore_early_media does now.
>> >>> >>> >> >> >> >
>> >>> >>> >> >> >> > Mike
>> >>> >>> >> >> >> >
>> >>> >>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> >>> >>> >> >> >> > > The issue here is when "originate" routine should
>> >>> >>> >> >> >> > > return
>> >>> >>> >> >> >> > > and
>> >>> >>> >> >> >> > > set "originate_status" variable. Current behavior is
>> >>> >>> >> >> >> > > to
>> >>> >>> >> >> >> > > return
>> >>> >>> >> >> >> > > on early
>> >>> >>> >> >> >> > > media, but what if to introduce a variable
>> >>> >>> >> >> >> > > "originate_wait_for_answer"
>> >>> >>> >> >> >> > > with default value "false" and use the variable in
>> >>> >>> >> >> >> > > originate
>> >>> >>> >> >> >> > > code to
>> >>> >>> >> >> >> > > decide when to return - on 18X or "200 OK"?
>> >>> >>> >> >> >> > >
>> >>> >>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> >>> >>> >> >> >> > >> he wants to call 3 people at once and let the A leg
>> >>> >>> >> >> >> > >> hear
>> >>> >>> >> >> >> > >> early media
>> >>> >>> >> >> >> > >> from call #1 while call #2 and #3 still are
>> >>> >>> >> >> >> > >> progressing
>> >>> >>> >> >> >> > >> which
>> >>> >>> >> >> >> > >> is not
>> >>> >>> >> >> >> > >> simple to do without doing thousands of dollars in
>> >>> >>> >> >> >> > >> development.
>> >>> >>> >> >> >> > >>
>> >>> >>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB
>> >>> >>> >> >> >> > >> <djbinter at yahoo.com>
>> >>> >> wrote:
>> >>> >>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so
>> >>> >>> >> >> >> > >>> that your
>> >>> >>> >> >> >> > >>> customers
>> >>> >>> >> >> >> > >>> can hear recordings?
>> >>> >>> >> >> >> >
>> >>> >>> >> >> >> > _______________________________________________
>> >>> >>> >> >> >> > FreeSWITCH-users mailing list
>> >>> >>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> >> >> >> >
>> >>> >>> >> >> >> >
>> >>> >>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>> >> >> >> >
>> >>> >>> >> >> >> >
>> >>> >>> >> >> >> >
>> >>> >>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswit
>> >>> >>> >> >> >> >ch- users http://www.freeswitch.org
>> >>> >>> >> >> >>
>> >>> >>> >> >> >> _______________________________________________
>> >>> >>> >> >> >> FreeSWITCH-users mailing list
>> >>> >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> >> >> >>
>> >>> >>> >> >> >>
>> >>> >>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>> >> >> >>
>> >>> >>> >> >> >>
>> >>> >>> >> >> >>
>> >>> >>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch
>> >>> >>> >> >> >>-us ers http://www.freeswitch.org
>> >>> >>> >> >> >
>> >>> >>> >> >> > --
>> >>> >>> >> >> > Anthony Minessale II
>> >>> >>> >> >> >
>> >>> >>> >> >> > FreeSWITCH http://www.freeswitch.org/
>> >>> >>> >> >> > ClueCon http://www.cluecon.com/
>> >>> >>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>> >>> >> >> >
>> >>> >>> >> >> > AIM: anthm
>> >>> >>> >> >> > MSN:anthony_minessale at hotmail.com
>> >>> >>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>> >>> >> >> > IRC: irc.freenode.net #freeswitch
>> >>> >>> >> >> >
>> >>> >>> >> >> > FreeSWITCH Developer Conference
>> >>> >>> >> >> > sip:888 at conference.freeswitch.org
>> >>> >>> >> >> > iax:guest at conference.freeswitch.org/888
>> >>> >>> >> >> > googletalk:conf+888 at conference.freeswitch.org
>> >>> >>> >> >> > pstn:+19193869900
>> >>> >>> >> >> >
>> >>> >>> >> >> > _______________________________________________
>> >>> >>> >> >> > FreeSWITCH-users mailing list
>> >>> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> >> >> >
>> >>> >>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>> >> >> >
>> >>> >>> >> >> >
>> >>> >>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >>> >>> >> >> >use rs http://www.freeswitch.org
>> >>> >>> >> >>
>> >>> >>> >> >> _______________________________________________
>> >>> >>> >> >> FreeSWITCH-users mailing list
>> >>> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> >> >>
>> >>> >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>> >> >>
>> >>> >>> >> >>
>> >>> >>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> >>> >>> >> >>ers http://www.freeswitch.org
>> >>> >>> >> >
>> >>> >>> >> > --
>> >>> >>> >> > Anthony Minessale II
>> >>> >>> >> >
>> >>> >>> >> > FreeSWITCH http://www.freeswitch.org/
>> >>> >>> >> > ClueCon http://www.cluecon.com/
>> >>> >>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>> >>> >> >
>> >>> >>> >> > AIM: anthm
>> >>> >>> >> > MSN:anthony_minessale at hotmail.com
>> >>> >>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>> >>> >> > IRC: irc.freenode.net #freeswitch
>> >>> >>> >> >
>> >>> >>> >> > FreeSWITCH Developer Conference
>> >>> >>> >> > sip:888 at conference.freeswitch.org
>> >>> >>> >> > iax:guest at conference.freeswitch.org/888
>> >>> >>> >> > googletalk:conf+888 at conference.freeswitch.org
>> >>> >>> >> > pstn:+19193869900
>> >>> >>> >> >
>> >>> >>> >> > _______________________________________________
>> >>> >>> >> > FreeSWITCH-users mailing list
>> >>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>> >> >
>> >>> >>> >> >
>> >>> >>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> >>> >>> >> >rs http://www.freeswitch.org
>> >>> >>> >>
>> >>> >>> >> _______________________________________________
>> >>> >>> >> FreeSWITCH-users mailing list
>> >>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>> >>
>> >>> >>> >>
>> >>> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >>> >> http://www.freeswitch.org
>> >>> >>> >
>> >>> >>> > _______________________________________________
>> >>> >>> > FreeSWITCH-users mailing list
>> >>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>> >
>> >>> >>> >
>> >>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >>> > http://www.freeswitch.org
>> >>> >>>
>> >>> >>> _______________________________________________
>> >>> >>> FreeSWITCH-users mailing list
>> >>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>>
>> >>> >>>
>> >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >>> http://www.freeswitch.org
>> >>> >>
>> >>> >>
>> >>> >>
>> >>> >> _______________________________________________
>> >>> >> FreeSWITCH-users mailing list
>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>
>> >>> >>
>> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> http://www.freeswitch.org
>> >>> >>
>> >>> >
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> iax:guest at conference.freeswitch.org/888
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mike at jerris.com  Fri Jan 15 15:35:49 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 Jan 2010 18:35:49 -0500
Subject: [Freeswitch-users] Issue Compiling ESL (PHP)
In-Reply-To: <5fe6fa8f1001151521u3ef71f9g49491118bf0371ac@mail.gmail.com>
References: <5fe6fa8f1001151437x491dade6q8b316f312556ff07@mail.gmail.com>
	<5fe6fa8f1001151454m92d17c9le4b811c897ae8482@mail.gmail.com>
	<9853f4ff1001151507oa1fa8adt5101956739d8c948@mail.gmail.com>
	<5fe6fa8f1001151521u3ef71f9g49491118bf0371ac@mail.gmail.com>
Message-ID: <F84AD964-A9DC-48D8-BA0D-22E97863CACE@jerris.com>

Sounds right to me.

Mike

On Jan 15, 2010, at 6:21 PM, Alfredo Quiroga-Villamil wrote:

> Thanks Anthony/David.
> 
> So it seems like the build worked then. I take from the previous
> emails and somewhere where I think I read that I can then take:
> 
> ESL.so and ESL.php and put them on a remote system under my say for
> instance 3rdParty directory and create scripts using the ESL.php
> library which probably internally uses ESL.so.
> 
> Is that statement correct?
> 
> Thanks in advance,
> 
> Alfredo
> 
> On Fri, Jan 15, 2010 at 6:07 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>> make a script ;)
>> 
>> it's very easy, try it!
>> 
>> David
>> 
>> On Fri, Jan 15, 2010 at 11:54 PM, Alfredo Quiroga-Villamil
>> <lawwton at gmail.com> wrote:
>>> Looking over at this, perhaps it even worked.
>>> 
>>> I see now under libs/esl/php/... the following two new files:
>>> 
>>> esl_wrap.o
>>> ESL.o
>>> 
>>> Is there a way to verify that FS has support after running make and
>>> make phpmod for php?
>>> 
>>> Thanks in advance,
>>> 
>>> Alfredo
>>> 
>>> On Fri, Jan 15, 2010 at 5:37 PM, Alfredo Quiroga-Villamil
>>> <lawwton at gmail.com> wrote:
>>>> All:
>>>> 
>>>> System:
>>>> 
>>>> Linux usnc-rtp-01-fs 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT
>>>> 2007 i686 i686 i386 GNU/Linux
>>>> 
>>>> I am trying to compile ESL, following the following steps:
>>>> 
>>>> 1- cd to my libs/esl directory as the wiki page indicates.
>>>> 2- run make
>>>> 
>>>> I then get right away the following:
>>>> 
>>>> cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o
>>>> cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o
>>>> src/esl_event.o
>>>> cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o
>>>> src/esl_threadmutex.o
>>>> cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o
>>>> src/esl_config.o
>>>> g++  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>>> -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o
>>>> ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o
>>>> src/esl_config.o src/esl_oop.o
>>>> ranlib libesl.a
>>>> cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L.
>>>> -L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit
>>>> cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient
>>>> -L. -lncurses -lpthread -lesl
>>>> cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver
>>>> -L. -lncurses -lpthread -lesl
>>>> cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes ivrd.c -o ivrd -L. -lncurses
>>>> -lpthread -lesl
>>>> 
>>>> 3- I try typing then:
>>>> 
>>>> make phpmod and get the following:
>>>> 
>>>> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
>>>> CFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes"
>>>> CXXFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>>> -Wno-unused-variable" CXX_CFLAGS="" -C php
>>>> make[1]: Entering directory
>>>> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>>>> g++  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>>> -Wno-unused-variable -I/usr/include/php -I/usr/include/php/main
>>>> -I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext
>>>> -I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function
>>>> -c esl_wrap.cpp -o esl_wrap.o
>>>> g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/kerberos/lib
>>>> -lcrypt -lcrypt -lncurses -laspell -lpspell -lcurl -lresolv -lm -ldl
>>>> -lnsl -lm -ldl -lcurl -ldl -lm -lcrypt -lm -lm -lcrypt -lpthread -o
>>>> ESL.so -L.
>>>> make[1]: Leaving directory
>>>> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>>>> 
>>>> I've installed I think all the -dev dependencies listed in the wiki. Any ideas?
>>>> 
>>>> Thanks in advance,
>>>> 
>>>> Alfredo Q-V
>>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From stevesteffler at shaw.ca  Fri Jan 15 15:43:12 2010
From: stevesteffler at shaw.ca (Steve Steffler)
Date: Fri, 15 Jan 2010 16:43:12 -0700
Subject: [Freeswitch-users] FIFO Originate caller ID
Message-ID: <29507085-AE04-4A43-A41A-6D279810CCEC@shaw.ca>

Hello,

I found an archived conversation on this list regarding FIFO origination caller ID, and how to modify it.  

http://old.nabble.com/FIFO-Orgination_caller_id-td26487628.html

There seems to be no easy way to customize the caller ID on originated calls from the FIFO to on-hook agents who were registered dynamically.  Anthony states a method to do it using static entries in the fifo conf file, and the general rationale is that SCREEN POPS be the preferred method, with the added nudge that good SIP phones can change the caller ID when the bridge is complete as well, which is all well and good.

The problem is for my application, all on-hook agents are using cellular phones, and they register dynamically.  Also, none of my agents are in front of a computer, so a SIP display update on the phone or screen pop on the computer in front of them is not really an option, and the only way they can identify calls from my FIFO right now is because they are the ones with NO CALLER ID (in other words, their mobile phones do not display the name, and the number is not recognized because it is set by FreeSWITCH to be "fifo+fifoname" instead of being numeric.  This is far from ideal.

I am wondering if there is anyone on the list who knows how to configure the origination_caller_id_number/name variables for dynamically registered on-hook agents so that the caller ID from the FIFO customer's incoming call is displayed to them instead of the above mangled caller ID?

I'm not disagreeing that it is an old-skewl way of thought, but in actuality it is just a way to interface with old-school telephony devices (i.e. non-Smartphone mobile phones) and I am not sure how to accomplish this.  Any help/input would be appreciated.

Regards,
Steve

From mike at jerris.com  Fri Jan 15 16:01:16 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 Jan 2010 19:01:16 -0500
Subject: [Freeswitch-users] FIFO Originate caller ID
In-Reply-To: <29507085-AE04-4A43-A41A-6D279810CCEC@shaw.ca>
References: <29507085-AE04-4A43-A41A-6D279810CCEC@shaw.ca>
Message-ID: <88D85720-C24A-43B8-A6E8-7F9179D71625@jerris.com>

At the time a call goes out to the agents, there is no specific caller they are matched too, therefore there is no way to know the caller id at this time.  When the originated call to the agent is answered, we THEN go and pick off the next caller to connect them with.  All you can do is set a caller id for the queue.

Mike

On Jan 15, 2010, at 6:43 PM, Steve Steffler wrote:

> Hello,
> 
> I found an archived conversation on this list regarding FIFO origination caller ID, and how to modify it.  
> 
> http://old.nabble.com/FIFO-Orgination_caller_id-td26487628.html
> 
> There seems to be no easy way to customize the caller ID on originated calls from the FIFO to on-hook agents who were registered dynamically.  Anthony states a method to do it using static entries in the fifo conf file, and the general rationale is that SCREEN POPS be the preferred method, with the added nudge that good SIP phones can change the caller ID when the bridge is complete as well, which is all well and good.
> 
> The problem is for my application, all on-hook agents are using cellular phones, and they register dynamically.  Also, none of my agents are in front of a computer, so a SIP display update on the phone or screen pop on the computer in front of them is not really an option, and the only way they can identify calls from my FIFO right now is because they are the ones with NO CALLER ID (in other words, their mobile phones do not display the name, and the number is not recognized because it is set by FreeSWITCH to be "fifo+fifoname" instead of being numeric.  This is far from ideal.
> 
> I am wondering if there is anyone on the list who knows how to configure the origination_caller_id_number/name variables for dynamically registered on-hook agents so that the caller ID from the FIFO customer's incoming call is displayed to them instead of the above mangled caller ID?
> 
> I'm not disagreeing that it is an old-skewl way of thought, but in actuality it is just a way to interface with old-school telephony devices (i.e. non-Smartphone mobile phones) and I am not sure how to accomplish this.  Any help/input would be appreciated.



From lawwton at gmail.com  Fri Jan 15 16:11:00 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 19:11:00 -0500
Subject: [Freeswitch-users] Issue Compiling ESL (PHP)
In-Reply-To: <F84AD964-A9DC-48D8-BA0D-22E97863CACE@jerris.com>
References: <5fe6fa8f1001151437x491dade6q8b316f312556ff07@mail.gmail.com>
	<5fe6fa8f1001151454m92d17c9le4b811c897ae8482@mail.gmail.com>
	<9853f4ff1001151507oa1fa8adt5101956739d8c948@mail.gmail.com>
	<5fe6fa8f1001151521u3ef71f9g49491118bf0371ac@mail.gmail.com>
	<F84AD964-A9DC-48D8-BA0D-22E97863CACE@jerris.com>
Message-ID: <5fe6fa8f1001151611t506c0046u708cd49fe03eb34b@mail.gmail.com>

Thanks all, appreciate the help.

On Fri, Jan 15, 2010 at 6:35 PM, Michael Jerris <mike at jerris.com> wrote:
> Sounds right to me.
>
> Mike
>
> On Jan 15, 2010, at 6:21 PM, Alfredo Quiroga-Villamil wrote:
>
>> Thanks Anthony/David.
>>
>> So it seems like the build worked then. I take from the previous
>> emails and somewhere where I think I read that I can then take:
>>
>> ESL.so and ESL.php and put them on a remote system under my say for
>> instance 3rdParty directory and create scripts using the ESL.php
>> library which probably internally uses ESL.so.
>>
>> Is that statement correct?
>>
>> Thanks in advance,
>>
>> Alfredo
>>
>> On Fri, Jan 15, 2010 at 6:07 PM, David Villasmil
>> <david.villasmil.work at gmail.com> wrote:
>>> make a script ;)
>>>
>>> it's very easy, try it!
>>>
>>> David
>>>
>>> On Fri, Jan 15, 2010 at 11:54 PM, Alfredo Quiroga-Villamil
>>> <lawwton at gmail.com> wrote:
>>>> Looking over at this, perhaps it even worked.
>>>>
>>>> I see now under libs/esl/php/... the following two new files:
>>>>
>>>> esl_wrap.o
>>>> ESL.o
>>>>
>>>> Is there a way to verify that FS has support after running make and
>>>> make phpmod for php?
>>>>
>>>> Thanks in advance,
>>>>
>>>> Alfredo
>>>>
>>>> On Fri, Jan 15, 2010 at 5:37 PM, Alfredo Quiroga-Villamil
>>>> <lawwton at gmail.com> wrote:
>>>>> All:
>>>>>
>>>>> System:
>>>>>
>>>>> Linux usnc-rtp-01-fs 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT
>>>>> 2007 i686 i686 i386 GNU/Linux
>>>>>
>>>>> I am trying to compile ESL, following the following steps:
>>>>>
>>>>> 1- cd to my libs/esl directory as the wiki page indicates.
>>>>> 2- run make
>>>>>
>>>>> I then get right away the following:
>>>>>
>>>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o
>>>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o
>>>>> src/esl_event.o
>>>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o
>>>>> src/esl_threadmutex.o
>>>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o
>>>>> src/esl_config.o
>>>>> g++ ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>>>> -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o
>>>>> ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o
>>>>> src/esl_config.o src/esl_oop.o
>>>>> ranlib libesl.a
>>>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L.
>>>>> -L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit
>>>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient
>>>>> -L. -lncurses -lpthread -lesl
>>>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver
>>>>> -L. -lncurses -lpthread -lesl
>>>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes ivrd.c -o ivrd -L. -lncurses
>>>>> -lpthread -lesl
>>>>>
>>>>> 3- I try typing then:
>>>>>
>>>>> make phpmod and get the following:
>>>>>
>>>>> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
>>>>> CFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes"
>>>>> CXXFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>>>> -Wno-unused-variable" CXX_CFLAGS="" -C php
>>>>> make[1]: Entering directory
>>>>> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>>>>> g++ ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>>>> -Wno-unused-variable -I/usr/include/php -I/usr/include/php/main
>>>>> -I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext
>>>>> -I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function
>>>>> -c esl_wrap.cpp -o esl_wrap.o
>>>>> g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/kerberos/lib
>>>>> -lcrypt -lcrypt -lncurses -laspell -lpspell -lcurl -lresolv -lm -ldl
>>>>> -lnsl -lm -ldl -lcurl -ldl -lm -lcrypt -lm -lm -lcrypt -lpthread -o
>>>>> ESL.so -L.
>>>>> make[1]: Leaving directory
>>>>> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>>>>>
>>>>> I've installed I think all the -dev dependencies listed in the wiki. Any ideas?
>>>>>
>>>>> Thanks in advance,
>>>>>
>>>>> Alfredo Q-V
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mastermind202 at gmail.com  Fri Jan 15 16:12:01 2010
From: mastermind202 at gmail.com (mm_202)
Date: Fri, 15 Jan 2010 19:12:01 -0500
Subject: [Freeswitch-users] Domains.
In-Reply-To: <2083.99622.qm@web34302.mail.mud.yahoo.com>
References: <2083.99622.qm@web34302.mail.mud.yahoo.com>
Message-ID: <63de75711001151612k514900f4w599c39f3bb2b70e4@mail.gmail.com>

On Fri, Jan 15, 2010 at 3:26 PM, FERNANDO VILLARROEL
<fvillarroel at yahoo.com>wrote:

> Dear.
>
> I installed FS FreeSWITCH Version 1.0.trunk (16144)
>
> I have a problem when i send traffic from a external gateway, the calls are
> rejected:
>
> 2010-01-05 16:52:37.254194 [DEBUG] sofia.c:5359 0 acls to check for proxy
> 2010-01-05 16:52:37.254194 [DEBUG] sofia.c:5377 network ip is a proxy [0]
> 2010-01-05 16:52:37.254194 [DEBUG] sofia.c:5405 IP XXX.XXX.XX.125 Rejected
> by acl "domains". Falling back to Digest auth.
>
>
> Anyone could me explain like i can do.
>
> Regards.
>
> Fernando
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


Edit your conf/autoload_configs/acl.conf.xml file and add that IP into the
domains list.
Then in the FS cli run 'reloadxml' and 'reloadacl'.

Read http://wiki.freeswitch.org/wiki/Acl.conf.xml for more info.

-- mm_202.
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From anthony.minessale at gmail.com  Fri Jan 15 16:44:59 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 18:44:59 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001151525o341cf2c7k2935538cca210815@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
	<201001150726.17430.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>
	<9853f4ff1001150443t5a18cc6bxa5032f391ab6f0ed@mail.gmail.com>
	<191c3a031001150808i69b3937boe0e903057f984e96@mail.gmail.com>
	<191c3a031001151317j408794a3pa2a0a5d21d63aa62@mail.gmail.com>
	<9853f4ff1001151445s6a4dff0qf8c9e17c702b7f86@mail.gmail.com>
	<191c3a031001151454w101ae8eh978e4ced42004fb3@mail.gmail.com>
	<9853f4ff1001151525o341cf2c7k2935538cca210815@mail.gmail.com>
Message-ID: <191c3a031001151644m6a838b6bp311f2a173fc65af3@mail.gmail.com>

Now we need a new feature

[leg_required=true]

set this on any legs required for the originate to proceed, if it's hungup,
the cause will be passed to any existing legs and fail the entire originate.

so use

{bridge_early_media=true}[leg_required=true]sofia/internal/real_call at foo.com
,sofia/internal/moh_call at foo.com

the leg_required will only be set on the 1st leg because of the [] vs {}
if that leg is then hungup, it will kill the other channels in the list.

please try latest trunk.



On Fri, Jan 15, 2010 at 5:25 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Anthony,
>
> LOL, and mounting and mounting... It does work when there is answer...
> but if B(2)-side rejects or times out or any other that 200 OK,
> B(1)-side stays indefinitely...
>
>
> On Fri, Jan 15, 2010 at 11:54 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > you can email me privately at this addr.
> >
> >
> > On Fri, Jan 15, 2010 at 4:45 PM, David Villasmil
> > <david.villasmil.work at gmail.com> wrote:
> >>
> >> Anthony,
> >>
> >>     Trying, Thanks. Is there anyway we can communicate directly?
> >>
> >>
> >> David
> >>
> >> On Fri, Jan 15, 2010 at 10:17 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >> > Try latest trunk,
> >> >
> >> > you should have exactly what you want with the same parameter, again
> my
> >> > paypal addr is cleary displayed as a big button on the website.
> >> >
> >> >
> >> > On Fri, Jan 15, 2010 at 10:08 AM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> one of the many reasons its a bad idea.
> >> >> Probably the leg with the bad audio is a different ptime.
> >> >> Now the amount of work I have to do escalates I would prefer you
> commit
> >> >> to
> >> >> commercial support by emailing me at consulting at freeswitch.org to
> >> >> continue
> >> >> with this.
> >> >>
> >> >>
> >> >> On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil
> >> >> <david.villasmil.work at gmail.com> wrote:
> >> >>>
> >> >>> I set it to "off" just in case, same thing.
> >> >>>
> >> >>> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
> >> >>> <david.villasmil.work at gmail.com> wrote:
> >> >>> > Default, haven't touched it i suppose it's off, i haven't set it
> >> >>> > anywhere
> >> >>> >
> >> >>> > On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
> >> >>> > <sos at sokhapkin.dyndns.org> wrote:
> >> >>> >> Is bypass_media on or off?
> >> >>> >>
> >> >>> >> On Friday 15 January 2010, David Villasmil wrote:
> >> >>> >>> Yeah, sorry. Early media. Audio after 200 is fine. Early media
> was
> >> >>> >>> ok
> >> >>> >>> before the change.
> >> >>> >>>
> >> >>> >>> David
> >> >>> >>>
> >> >>> >>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
> >> >>> >>>
> >> >>> >>> <sos at sokhapkin.dyndns.org> wrote:
> >> >>> >>> > Which audio? Early media or after 200 OK?
> >> >>> >>> >
> >> >>> >>> > On Friday 15 January 2010, David Villasmil wrote:
> >> >>> >>> >> Hello again Anthony,
> >> >>> >>> >>
> >> >>> >>> >> I just tested it, and although functionality does not, first
> >> >>> >>> >> incoming
> >> >>> >>> >> audio is coming in all garbled... do you know why?
> >> >>> >>> >>
> >> >>> >>> >> David
> >> >>> >>> >>
> >> >>> >>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
> >> >>> >>> >>
> >> >>> >>> >> <anthony.minessale at gmail.com> wrote:
> >> >>> >>> >> > {bridge_early_media=true}
> >> >>> >>> >> > in the dial string in place of ignore_early_media=true
> >> >>> >>> >> >
> >> >>> >>> >> > the first b leg in the list who sends 183 will become the
> >> >>> >>> >> > ringback
> >> >>> >>> >> > device for A leg it will hear the early media
> >> >>> >>> >> > for that leg while the other legs still ring.  If some
> other
> >> >>> >>> >> > leg
> >> >>> >>> >> > answers the final call will still be bridged to the leg who
> >> >>> >>> >> > answered.
> >> >>> >>> >> >
> >> >>> >>> >> >
> >> >>> >>> >> > I would estimate it at $500 payable on the big paypal
> button
> >> >>> >>> >> > on
> >> >>> >>> >> > http://www.freeswitch.org
> >> >>> >>> >> > but, I already added the patch to tree earlier today so I
> >> >>> >>> >> > guess
> >> >>> >>> >> > it's
> >> >>> >>> >> > up to you to pay it or not.
> >> >>> >>> >> >
> >> >>> >>> >> >
> >> >>> >>> >> >
> >> >>> >>> >> >
> >> >>> >>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> >> >>> >>> >> >
> >> >>> >>> >> > <david.villasmil.work at gmail.com> wrote:
> >> >>> >>> >> >> Anthony,
> >> >>> >>> >> >>
> >> >>> >>> >> >> I did take the "hint", don't worry. We will probably ask
> for
> >> >>> >>> >> >> a
> >> >>> >>> >> >> bounty
> >> >>> >>> >> >> but first we need to know:
> >> >>> >>> >> >> 1.- whether this is possible
> >> >>> >>> >> >> 2.- how long it would take
> >> >>> >>> >> >> 3.- how will it exactly work
> >> >>> >>> >> >> 4.- of course, what's the bounty (be gentle ;) )
> >> >>> >>> >> >>
> >> >>> >>> >> >> We would of course give this back to the community.
> >> >>> >>> >> >>
> >> >>> >>> >> >> in the meantime, bridge_early_media=true will discard the
> >> >>> >>> >> >> 2nd
> >> >>> >>> >> >> B-leg
> >> >>> >>> >> >> and conversely "false" it will keep on trying to connect
> and
> >> >>> >>> >> >> if
> >> >>> >>> >> >> it
> >> >>> >>> >> >> connects the other B-leg if will bridge to that one?
> >> >>> >>> >> >>
> >> >>> >>> >> >> Thanks
> >> >>> >>> >> >>
> >> >>> >>> >> >> David
> >> >>> >>> >> >>
> >> >>> >>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> >> >>> >>> >> >>
> >> >>> >>> >> >> <anthony.minessale at gmail.com> wrote:
> >> >>> >>> >> >> > I added bridge_early_media=true to do the best I can do.
> >> >>> >>> >> >> > This is the most I will do, especially for free, nobody
> >> >>> >>> >> >> > can
> >> >>> >>> >> >> > take a
> >> >>> >>> >> >> > hint that
> >> >>> >>> >> >> > you should be paying for all these custom requests so
> take
> >> >>> >>> >> >> > it
> >> >>> >>> >> >> > or
> >> >>> >>> >> >> > leave it
> >> >>> >>> >> >> > but this thread is done.........
> >> >>> >>> >> >> >
> >> >>> >>> >> >> >
> >> >>> >>> >> >> >
> >> >>> >>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
> >> >>> >>> >> >> > <sos at sokhapkin.dyndns.org>
> >> >>> >>> >> >> >
> >> >>> >>> >> >> > wrote:
> >> >>> >>> >> >> >> No, not exactly. ignore_early_media doesn't pass early
> >> >>> >>> >> >> >> media
> >> >>> >>> >> >> >> to
> >> >>> >>> >> >> >> the caller
> >> >>> >>> >> >> >> if
> >> >>> >>> >> >> >> bypass_media is false.
> >> >>> >>> >> >> >>
> >> >>> >>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
> >> >>> >>> >> >> >> > this is exactly what ignore_early_media does now.
> >> >>> >>> >> >> >> >
> >> >>> >>> >> >> >> > Mike
> >> >>> >>> >> >> >> >
> >> >>> >>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> >> >>> >>> >> >> >> > > The issue here is when "originate" routine should
> >> >>> >>> >> >> >> > > return
> >> >>> >>> >> >> >> > > and
> >> >>> >>> >> >> >> > > set "originate_status" variable. Current behavior
> is
> >> >>> >>> >> >> >> > > to
> >> >>> >>> >> >> >> > > return
> >> >>> >>> >> >> >> > > on early
> >> >>> >>> >> >> >> > > media, but what if to introduce a variable
> >> >>> >>> >> >> >> > > "originate_wait_for_answer"
> >> >>> >>> >> >> >> > > with default value "false" and use the variable in
> >> >>> >>> >> >> >> > > originate
> >> >>> >>> >> >> >> > > code to
> >> >>> >>> >> >> >> > > decide when to return - on 18X or "200 OK"?
> >> >>> >>> >> >> >> > >
> >> >>> >>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale
> wrote:
> >> >>> >>> >> >> >> > >> he wants to call 3 people at once and let the A
> leg
> >> >>> >>> >> >> >> > >> hear
> >> >>> >>> >> >> >> > >> early media
> >> >>> >>> >> >> >> > >> from call #1 while call #2 and #3 still are
> >> >>> >>> >> >> >> > >> progressing
> >> >>> >>> >> >> >> > >> which
> >> >>> >>> >> >> >> > >> is not
> >> >>> >>> >> >> >> > >> simple to do without doing thousands of dollars in
> >> >>> >>> >> >> >> > >> development.
> >> >>> >>> >> >> >> > >>
> >> >>> >>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB
> >> >>> >>> >> >> >> > >> <djbinter at yahoo.com>
> >> >>> >> wrote:
> >> >>> >>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK),
> so
> >> >>> >>> >> >> >> > >>> that your
> >> >>> >>> >> >> >> > >>> customers
> >> >>> >>> >> >> >> > >>> can hear recordings?
> >> >>> >>> >> >> >> >
> >> >>> >>> >> >> >> > _______________________________________________
> >> >>> >>> >> >> >> > FreeSWITCH-users mailing list
> >> >>> >>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> >> >> >> >
> >> >>> >>> >> >> >> >
> >> >>> >>> >> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>> >> >> >> >
> >> >>> >>> >> >> >> >
> >> >>> >>> >> >> >> >
> >> >>> >>> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswit
> >> >>> >>> >> >> >> >ch- users http://www.freeswitch.org
> >> >>> >>> >> >> >>
> >> >>> >>> >> >> >> _______________________________________________
> >> >>> >>> >> >> >> FreeSWITCH-users mailing list
> >> >>> >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> >> >> >>
> >> >>> >>> >> >> >>
> >> >>> >>> >> >> >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>> >> >> >>
> >> >>> >>> >> >> >>
> >> >>> >>> >> >> >>
> >> >>> >>> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch
> >> >>> >>> >> >> >>-us ers http://www.freeswitch.org
> >> >>> >>> >> >> >
> >> >>> >>> >> >> > --
> >> >>> >>> >> >> > Anthony Minessale II
> >> >>> >>> >> >> >
> >> >>> >>> >> >> > FreeSWITCH http://www.freeswitch.org/
> >> >>> >>> >> >> > ClueCon http://www.cluecon.com/
> >> >>> >>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >>> >>> >> >> >
> >> >>> >>> >> >> > AIM: anthm
> >> >>> >>> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >>> >>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >>> >>> >> >> > IRC: irc.freenode.net #freeswitch
> >> >>> >>> >> >> >
> >> >>> >>> >> >> > FreeSWITCH Developer Conference
> >> >>> >>> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >>> >>> >> >> > iax:guest at conference.freeswitch.org/888
> >> >>> >>> >> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >>> >>> >> >> > pstn:+19193869900
> >> >>> >>> >> >> >
> >> >>> >>> >> >> > _______________________________________________
> >> >>> >>> >> >> > FreeSWITCH-users mailing list
> >> >>> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> >> >> >
> >> >>> >>> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>> >> >> >
> >> >>> >>> >> >> >
> >> >>> >>> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-
> >> >>> >>> >> >> >use rs http://www.freeswitch.org
> >> >>> >>> >> >>
> >> >>> >>> >> >> _______________________________________________
> >> >>> >>> >> >> FreeSWITCH-users mailing list
> >> >>> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> >> >>
> >> >>> >>> >> >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>> >> >>
> >> >>> >>> >> >>
> >> >>> >>> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-us
> >> >>> >>> >> >>ers http://www.freeswitch.org
> >> >>> >>> >> >
> >> >>> >>> >> > --
> >> >>> >>> >> > Anthony Minessale II
> >> >>> >>> >> >
> >> >>> >>> >> > FreeSWITCH http://www.freeswitch.org/
> >> >>> >>> >> > ClueCon http://www.cluecon.com/
> >> >>> >>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >>> >>> >> >
> >> >>> >>> >> > AIM: anthm
> >> >>> >>> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >>> >>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >>> >>> >> > IRC: irc.freenode.net #freeswitch
> >> >>> >>> >> >
> >> >>> >>> >> > FreeSWITCH Developer Conference
> >> >>> >>> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >>> >>> >> > iax:guest at conference.freeswitch.org/888
> >> >>> >>> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >>> >>> >> > pstn:+19193869900
> >> >>> >>> >> >
> >> >>> >>> >> > _______________________________________________
> >> >>> >>> >> > FreeSWITCH-users mailing list
> >> >>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>> >> >
> >> >>> >>> >> >
> >> >>> >>> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-use
> >> >>> >>> >> >rs http://www.freeswitch.org
> >> >>> >>> >>
> >> >>> >>> >> _______________________________________________
> >> >>> >>> >> FreeSWITCH-users mailing list
> >> >>> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>> >>
> >> >>> >>> >>
> >> >>> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> >>> >> http://www.freeswitch.org
> >> >>> >>> >
> >> >>> >>> > _______________________________________________
> >> >>> >>> > FreeSWITCH-users mailing list
> >> >>> >>> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>> >
> >> >>> >>> >
> >> >>> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> >>> > http://www.freeswitch.org
> >> >>> >>>
> >> >>> >>> _______________________________________________
> >> >>> >>> FreeSWITCH-users mailing list
> >> >>> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>>
> >> >>> >>>
> >> >>> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> >>> http://www.freeswitch.org
> >> >>> >>
> >> >>> >>
> >> >>> >>
> >> >>> >> _______________________________________________
> >> >>> >> FreeSWITCH-users mailing list
> >> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>
> >> >>> >>
> >> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> >> http://www.freeswitch.org
> >> >>> >>
> >> >>> >
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>
> >> >>
> >> >>
> >> >> --
> >> >> Anthony Minessale II
> >> >>
> >> >> FreeSWITCH http://www.freeswitch.org/
> >> >> ClueCon http://www.cluecon.com/
> >> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >> >>
> >> >> AIM: anthm
> >> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> IRC: irc.freenode.net #freeswitch
> >> >>
> >> >> FreeSWITCH Developer Conference
> >> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> iax:guest at conference.freeswitch.org/888
> >> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> pstn:+19193869900
> >> >
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From sos at sokhapkin.dyndns.org  Fri Jan 15 17:04:25 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 15 Jan 2010 20:04:25 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001151644m6a838b6bp311f2a173fc65af3@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<9853f4ff1001151525o341cf2c7k2935538cca210815@mail.gmail.com>
	<191c3a031001151644m6a838b6bp311f2a173fc65af3@mail.gmail.com>
Message-ID: <201001152004.25949.sos@sokhapkin.dyndns.org>

Sorry for the dumb question, is there a way to find out in dialplan (some 
variable?) which one of comma separated b-legs listed in originate command 
answered the call?

originate sofia/g1/number,sofia/g2/number,sofia/g3/number

Which gateway answered? g1, g2 or g3?

On Friday 15 January 2010, Anthony Minessale wrote:
> Now we need a new feature
>
> [leg_required=true]
>
> set this on any legs required for the originate to proceed, if it's hungup,
> the cause will be passed to any existing legs and fail the entire
> originate.
>
> so use
>
> {bridge_early_media=true}[leg_required=true]sofia/internal/real_call at foo.co
>m ,sofia/internal/moh_call at foo.com
>
> the leg_required will only be set on the 1st leg because of the [] vs {}
> if that leg is then hungup, it will kill the other channels in the list.
>
> please try latest trunk.
>
>
>
> On Fri, Jan 15, 2010 at 5:25 PM, David Villasmil <
>
> david.villasmil.work at gmail.com> wrote:
> > Anthony,
> >
> > LOL, and mounting and mounting... It does work when there is answer...
> > but if B(2)-side rejects or times out or any other that 200 OK,
> > B(1)-side stays indefinitely...
> >
> >
> > On Fri, Jan 15, 2010 at 11:54 PM, Anthony Minessale
> >
> > <anthony.minessale at gmail.com> wrote:
> > > you can email me privately at this addr.
> > >
> > >
> > > On Fri, Jan 15, 2010 at 4:45 PM, David Villasmil
> > >
> > > <david.villasmil.work at gmail.com> wrote:
> > >> Anthony,
> > >>
> > >>     Trying, Thanks. Is there anyway we can communicate directly?
> > >>
> > >>
> > >> David
> > >>
> > >> On Fri, Jan 15, 2010 at 10:17 PM, Anthony Minessale
> > >>
> > >> <anthony.minessale at gmail.com> wrote:
> > >> > Try latest trunk,
> > >> >
> > >> > you should have exactly what you want with the same parameter, again
> >
> > my
> >
> > >> > paypal addr is cleary displayed as a big button on the website.
> > >> >
> > >> >
> > >> > On Fri, Jan 15, 2010 at 10:08 AM, Anthony Minessale
> > >> >
> > >> > <anthony.minessale at gmail.com> wrote:
> > >> >> one of the many reasons its a bad idea.
> > >> >> Probably the leg with the bad audio is a different ptime.
> > >> >> Now the amount of work I have to do escalates I would prefer you
> >
> > commit
> >
> > >> >> to
> > >> >> commercial support by emailing me at consulting at freeswitch.org to
> > >> >> continue
> > >> >> with this.
> > >> >>
> > >> >>
> > >> >> On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil
> > >> >>
> > >> >> <david.villasmil.work at gmail.com> wrote:
> > >> >>> I set it to "off" just in case, same thing.
> > >> >>>
> > >> >>> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
> > >> >>>
> > >> >>> <david.villasmil.work at gmail.com> wrote:
> > >> >>> > Default, haven't touched it i suppose it's off, i haven't set it
> > >> >>> > anywhere
> > >> >>> >
> > >> >>> > On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
> > >> >>> >
> > >> >>> > <sos at sokhapkin.dyndns.org> wrote:
> > >> >>> >> Is bypass_media on or off?
> > >> >>> >>
> > >> >>> >> On Friday 15 January 2010, David Villasmil wrote:
> > >> >>> >>> Yeah, sorry. Early media. Audio after 200 is fine. Early media
> >
> > was
> >
> > >> >>> >>> ok
> > >> >>> >>> before the change.
> > >> >>> >>>
> > >> >>> >>> David
> > >> >>> >>>
> > >> >>> >>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
> > >> >>> >>>
> > >> >>> >>> <sos at sokhapkin.dyndns.org> wrote:
> > >> >>> >>> > Which audio? Early media or after 200 OK?
> > >> >>> >>> >
> > >> >>> >>> > On Friday 15 January 2010, David Villasmil wrote:
> > >> >>> >>> >> Hello again Anthony,
> > >> >>> >>> >>
> > >> >>> >>> >> I just tested it, and although functionality does not,
> > >> >>> >>> >> first incoming
> > >> >>> >>> >> audio is coming in all garbled... do you know why?
> > >> >>> >>> >>
> > >> >>> >>> >> David
> > >> >>> >>> >>
> > >> >>> >>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
> > >> >>> >>> >>
> > >> >>> >>> >> <anthony.minessale at gmail.com> wrote:
> > >> >>> >>> >> > {bridge_early_media=true}
> > >> >>> >>> >> > in the dial string in place of ignore_early_media=true
> > >> >>> >>> >> >
> > >> >>> >>> >> > the first b leg in the list who sends 183 will become the
> > >> >>> >>> >> > ringback
> > >> >>> >>> >> > device for A leg it will hear the early media
> > >> >>> >>> >> > for that leg while the other legs still ring.  If some
> >
> > other
> >
> > >> >>> >>> >> > leg
> > >> >>> >>> >> > answers the final call will still be bridged to the leg
> > >> >>> >>> >> > who answered.
> > >> >>> >>> >> >
> > >> >>> >>> >> >
> > >> >>> >>> >> > I would estimate it at $500 payable on the big paypal
> >
> > button
> >
> > >> >>> >>> >> > on
> > >> >>> >>> >> > http://www.freeswitch.org
> > >> >>> >>> >> > but, I already added the patch to tree earlier today so I
> > >> >>> >>> >> > guess
> > >> >>> >>> >> > it's
> > >> >>> >>> >> > up to you to pay it or not.
> > >> >>> >>> >> >
> > >> >>> >>> >> >
> > >> >>> >>> >> >
> > >> >>> >>> >> >
> > >> >>> >>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> > >> >>> >>> >> >
> > >> >>> >>> >> > <david.villasmil.work at gmail.com> wrote:
> > >> >>> >>> >> >> Anthony,
> > >> >>> >>> >> >>
> > >> >>> >>> >> >> I did take the "hint", don't worry. We will probably ask
> >
> > for
> >
> > >> >>> >>> >> >> a
> > >> >>> >>> >> >> bounty
> > >> >>> >>> >> >> but first we need to know:
> > >> >>> >>> >> >> 1.- whether this is possible
> > >> >>> >>> >> >> 2.- how long it would take
> > >> >>> >>> >> >> 3.- how will it exactly work
> > >> >>> >>> >> >> 4.- of course, what's the bounty (be gentle ;) )
> > >> >>> >>> >> >>
> > >> >>> >>> >> >> We would of course give this back to the community.
> > >> >>> >>> >> >>
> > >> >>> >>> >> >> in the meantime, bridge_early_media=true will discard
> > >> >>> >>> >> >> the 2nd
> > >> >>> >>> >> >> B-leg
> > >> >>> >>> >> >> and conversely "false" it will keep on trying to connect
> >
> > and
> >
> > >> >>> >>> >> >> if
> > >> >>> >>> >> >> it
> > >> >>> >>> >> >> connects the other B-leg if will bridge to that one?
> > >> >>> >>> >> >>
> > >> >>> >>> >> >> Thanks
> > >> >>> >>> >> >>
> > >> >>> >>> >> >> David
> > >> >>> >>> >> >>
> > >> >>> >>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> > >> >>> >>> >> >>
> > >> >>> >>> >> >> <anthony.minessale at gmail.com> wrote:
> > >> >>> >>> >> >> > I added bridge_early_media=true to do the best I can
> > >> >>> >>> >> >> > do. This is the most I will do, especially for free,
> > >> >>> >>> >> >> > nobody can
> > >> >>> >>> >> >> > take a
> > >> >>> >>> >> >> > hint that
> > >> >>> >>> >> >> > you should be paying for all these custom requests so
> >
> > take
> >
> > >> >>> >>> >> >> > it
> > >> >>> >>> >> >> > or
> > >> >>> >>> >> >> > leave it
> > >> >>> >>> >> >> > but this thread is done.........
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
> > >> >>> >>> >> >> > <sos at sokhapkin.dyndns.org>
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> > wrote:
> > >> >>> >>> >> >> >> No, not exactly. ignore_early_media doesn't pass
> > >> >>> >>> >> >> >> early media
> > >> >>> >>> >> >> >> to
> > >> >>> >>> >> >> >> the caller
> > >> >>> >>> >> >> >> if
> > >> >>> >>> >> >> >> bypass_media is false.
> > >> >>> >>> >> >> >>
> > >> >>> >>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
> > >> >>> >>> >> >> >> > this is exactly what ignore_early_media does now.
> > >> >>> >>> >> >> >> >
> > >> >>> >>> >> >> >> > Mike
> > >> >>> >>> >> >> >> >
> > >> >>> >>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> > >> >>> >>> >> >> >> > > The issue here is when "originate" routine should
> > >> >>> >>> >> >> >> > > return
> > >> >>> >>> >> >> >> > > and
> > >> >>> >>> >> >> >> > > set "originate_status" variable. Current behavior
> >
> > is
> >
> > >> >>> >>> >> >> >> > > to
> > >> >>> >>> >> >> >> > > return
> > >> >>> >>> >> >> >> > > on early
> > >> >>> >>> >> >> >> > > media, but what if to introduce a variable
> > >> >>> >>> >> >> >> > > "originate_wait_for_answer"
> > >> >>> >>> >> >> >> > > with default value "false" and use the variable
> > >> >>> >>> >> >> >> > > in originate
> > >> >>> >>> >> >> >> > > code to
> > >> >>> >>> >> >> >> > > decide when to return - on 18X or "200 OK"?
> > >> >>> >>> >> >> >> > >
> > >> >>> >>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale
> >
> > wrote:
> > >> >>> >>> >> >> >> > >> he wants to call 3 people at once and let the A
> >
> > leg
> >
> > >> >>> >>> >> >> >> > >> hear
> > >> >>> >>> >> >> >> > >> early media
> > >> >>> >>> >> >> >> > >> from call #1 while call #2 and #3 still are
> > >> >>> >>> >> >> >> > >> progressing
> > >> >>> >>> >> >> >> > >> which
> > >> >>> >>> >> >> >> > >> is not
> > >> >>> >>> >> >> >> > >> simple to do without doing thousands of dollars
> > >> >>> >>> >> >> >> > >> in development.
> > >> >>> >>> >> >> >> > >>
> > >> >>> >>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB
> > >> >>> >>> >> >> >> > >> <djbinter at yahoo.com>
> > >> >>> >>
> > >> >>> >> wrote:
> > >> >>> >>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK),
> >
> > so
> >
> > >> >>> >>> >> >> >> > >>> that your
> > >> >>> >>> >> >> >> > >>> customers
> > >> >>> >>> >> >> >> > >>> can hear recordings?
> > >> >>> >>> >> >> >> >
> > >> >>> >>> >> >> >> > _______________________________________________
> > >> >>> >>> >> >> >> > FreeSWITCH-users mailing list
> > >> >>> >>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >> >>> >>> >> >> >> > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswit
> >
> > >> >>> >>> >> >> >> >ch- users http://www.freeswitch.org
> > >> >>> >>> >> >> >>
> > >> >>> >>> >> >> >> _______________________________________________
> > >> >>> >>> >> >> >> FreeSWITCH-users mailing list
> > >> >>> >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >> >>> >>> >> >> >> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch
> >
> > >> >>> >>> >> >> >>-us ers http://www.freeswitch.org
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> > --
> > >> >>> >>> >> >> > Anthony Minessale II
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> > FreeSWITCH http://www.freeswitch.org/
> > >> >>> >>> >> >> > ClueCon http://www.cluecon.com/
> > >> >>> >>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> > AIM: anthm
> > >> >>> >>> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_miness
> > >> >>> >>> >> >> >ale at hotmail.com>
> > >> >>> >>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL
> > >> >>> >>> >> >> >%3Aanthony.minessale at gmail.com> IRC: irc.freenode.net
> > >> >>> >>> >> >> > #freeswitch
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> > FreeSWITCH Developer Conference
> > >> >>> >>> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference
> > >> >>> >>> >> >> >.freeswitch.org>
> > >> >>> >>> >> >> > iax:guest at conference.freeswitch.org/888
> > >> >>> >>> >> >> > googletalk:conf+888 at conference.freeswitch.org<googleta
> > >> >>> >>> >> >> >lk%3Aconf%2B888 at conference.freeswitch.org>
> > >> >>> >>> >> >> > pstn:+19193869900
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> > _______________________________________________
> > >> >>> >>> >> >> > FreeSWITCH-users mailing list
> > >> >>> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >> >>> >>> >> >> > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-
> >
> > >> >>> >>> >> >> >use rs http://www.freeswitch.org
> > >> >>> >>> >> >>
> > >> >>> >>> >> >> _______________________________________________
> > >> >>> >>> >> >> FreeSWITCH-users mailing list
> > >> >>> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >> >>> >>> >> >> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-us
> >
> > >> >>> >>> >> >>ers http://www.freeswitch.org
> > >> >>> >>> >> >
> > >> >>> >>> >> > --
> > >> >>> >>> >> > Anthony Minessale II
> > >> >>> >>> >> >
> > >> >>> >>> >> > FreeSWITCH http://www.freeswitch.org/
> > >> >>> >>> >> > ClueCon http://www.cluecon.com/
> > >> >>> >>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> > >> >>> >>> >> >
> > >> >>> >>> >> > AIM: anthm
> > >> >>> >>> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale
> > >> >>> >>> >> >@hotmail.com>
> > >> >>> >>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3A
> > >> >>> >>> >> >anthony.minessale at gmail.com> IRC: irc.freenode.net
> > >> >>> >>> >> > #freeswitch
> > >> >>> >>> >> >
> > >> >>> >>> >> > FreeSWITCH Developer Conference
> > >> >>> >>> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.fr
> > >> >>> >>> >> >eeswitch.org> iax:guest at conference.freeswitch.org/888
> > >> >>> >>> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%
> > >> >>> >>> >> >3Aconf%2B888 at conference.freeswitch.org> pstn:+19193869900
> > >> >>> >>> >> >
> > >> >>> >>> >> > _______________________________________________
> > >> >>> >>> >> > FreeSWITCH-users mailing list
> > >> >>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >> >>> >>> >> > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-use
> >
> > >> >>> >>> >> >rs http://www.freeswitch.org
> > >> >>> >>> >>
> > >> >>> >>> >> _______________________________________________
> > >> >>> >>> >> FreeSWITCH-users mailing list
> > >> >>> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >> >>> >>> >> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> >>> >>> >> http://www.freeswitch.org
> > >> >>> >>> >
> > >> >>> >>> > _______________________________________________
> > >> >>> >>> > FreeSWITCH-users mailing list
> > >> >>> >>> > FreeSWITCH-users at lists.freeswitch.org
> > >> >>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-user
> > >> >>> >>> >s
> > >> >>> >>> >
> > >> >>> >>> >
> > >> >>> >>> > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> >>> >>> > http://www.freeswitch.org
> > >> >>> >>>
> > >> >>> >>> _______________________________________________
> > >> >>> >>> FreeSWITCH-users mailing list
> > >> >>> >>> FreeSWITCH-users at lists.freeswitch.org
> > >> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> >>> >>>
> > >> >>> >>>
> > >> >>> >>> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> >>> >>> http://www.freeswitch.org
> > >> >>> >>
> > >> >>> >> _______________________________________________
> > >> >>> >> FreeSWITCH-users mailing list
> > >> >>> >> FreeSWITCH-users at lists.freeswitch.org
> > >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> >>> >>
> > >> >>> >>
> > >> >>> >> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> >>> >> http://www.freeswitch.org
> > >> >>>
> > >> >>> _______________________________________________
> > >> >>> FreeSWITCH-users mailing list
> > >> >>> FreeSWITCH-users at lists.freeswitch.org
> > >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> >>>
> > >> >>> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> >>> http://www.freeswitch.org
> > >> >>
> > >> >> --
> > >> >> Anthony Minessale II
> > >> >>
> > >> >> FreeSWITCH http://www.freeswitch.org/
> > >> >> ClueCon http://www.cluecon.com/
> > >> >> Twitter: http://twitter.com/FreeSWITCH_wire
> > >> >>
> > >> >> AIM: anthm
> > >> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.c
> > >> >>om>
> > >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.mi
> > >> >>nessale at gmail.com> IRC: irc.freenode.net #freeswitch
> > >> >>
> > >> >> FreeSWITCH Developer Conference
> > >> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.o
> > >> >>rg> iax:guest at conference.freeswitch.org/888
> > >> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B8
> > >> >>88 at conference.freeswitch.org> pstn:+19193869900
> > >> >
> > >> > --
> > >> > Anthony Minessale II
> > >> >
> > >> > FreeSWITCH http://www.freeswitch.org/
> > >> > ClueCon http://www.cluecon.com/
> > >> > Twitter: http://twitter.com/FreeSWITCH_wire
> > >> >
> > >> > AIM: anthm
> > >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.co
> > >> >m>
> > >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.min
> > >> >essale at gmail.com> IRC: irc.freenode.net #freeswitch
> > >> >
> > >> > FreeSWITCH Developer Conference
> > >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.or
> > >> >g> iax:guest at conference.freeswitch.org/888
> > >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B88
> > >> >8 at conference.freeswitch.org> pstn:+19193869900
> > >> >
> > >> > _______________________________________________
> > >> > FreeSWITCH-users mailing list
> > >> > FreeSWITCH-users at lists.freeswitch.org
> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> > http://www.freeswitch.org
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> http://www.freeswitch.org
> > >
> > > --
> > > Anthony Minessale II
> > >
> > > FreeSWITCH http://www.freeswitch.org/
> > > ClueCon http://www.cluecon.com/
> > > Twitter: http://twitter.com/FreeSWITCH_wire
> > >
> > > AIM: anthm
> > > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.miness
> > >ale at gmail.com> IRC: irc.freenode.net #freeswitch
> > >
> > > FreeSWITCH Developer Conference
> > > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > > iax:guest at conference.freeswitch.org/888
> > > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at c
> > >onference.freeswitch.org> pstn:+19193869900
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From lists at redbonez.net  Fri Jan 15 17:28:22 2010
From: lists at redbonez.net (Adam Ford)
Date: Fri, 15 Jan 2010 18:28:22 -0700
Subject: [Freeswitch-users] Polycom configuration for FreeSWITCH
Message-ID: <003701ca964b$3241b100$96c51300$@net>

Does anyone who has successfully implemented Polycom IP301/501s, with
FreeSWITCH, have a configuration they can share with me as an example? I
have read the Polycom Admin Guide several times and understand what the
settings are/do, I am just not sure which FreeSWITCH supports, which it
doesn't, and which need special configuration to work with FreeSWITCH.

 

Thanks in advance,

-Adam

 

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From sos at sokhapkin.dyndns.org  Fri Jan 15 17:27:07 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 15 Jan 2010 20:27:07 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001152004.25949.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<191c3a031001151644m6a838b6bp311f2a173fc65af3@mail.gmail.com>
	<201001152004.25949.sos@sokhapkin.dyndns.org>
Message-ID: <201001152027.07209.sos@sokhapkin.dyndns.org>

Same question about 

originate aaa|bbb|ccc

syntax :-)


On Friday 15 January 2010, Sergey Okhapkin wrote:
> Sorry for the dumb question, is there a way to find out in dialplan (some
> variable?) which one of comma separated b-legs listed in originate command
> answered the call?
>
> originate sofia/g1/number,sofia/g2/number,sofia/g3/number
>
> Which gateway answered? g1, g2 or g3?
>
> On Friday 15 January 2010, Anthony Minessale wrote:
> > Now we need a new feature
> >
> > [leg_required=true]
> >
> > set this on any legs required for the originate to proceed, if it's
> > hungup, the cause will be passed to any existing legs and fail the entire
> > originate.
> >
> > so use
> >
> > {bridge_early_media=true}[leg_required=true]sofia/internal/real_call at foo.
> >co m ,sofia/internal/moh_call at foo.com
> >
> > the leg_required will only be set on the 1st leg because of the [] vs {}
> > if that leg is then hungup, it will kill the other channels in the list.
> >
> > please try latest trunk.
> >
> >
> >
> > On Fri, Jan 15, 2010 at 5:25 PM, David Villasmil <
> >
> > david.villasmil.work at gmail.com> wrote:
> > > Anthony,
> > >
> > > LOL, and mounting and mounting... It does work when there is answer...
> > > but if B(2)-side rejects or times out or any other that 200 OK,
> > > B(1)-side stays indefinitely...
> > >
> > >
> > > On Fri, Jan 15, 2010 at 11:54 PM, Anthony Minessale
> > >
> > > <anthony.minessale at gmail.com> wrote:
> > > > you can email me privately at this addr.
> > > >
> > > >
> > > > On Fri, Jan 15, 2010 at 4:45 PM, David Villasmil
> > > >
> > > > <david.villasmil.work at gmail.com> wrote:
> > > >> Anthony,
> > > >>
> > > >>     Trying, Thanks. Is there anyway we can communicate directly?
> > > >>
> > > >>
> > > >> David
> > > >>
> > > >> On Fri, Jan 15, 2010 at 10:17 PM, Anthony Minessale
> > > >>
> > > >> <anthony.minessale at gmail.com> wrote:
> > > >> > Try latest trunk,
> > > >> >
> > > >> > you should have exactly what you want with the same parameter,
> > > >> > again
> > >
> > > my
> > >
> > > >> > paypal addr is cleary displayed as a big button on the website.
> > > >> >
> > > >> >
> > > >> > On Fri, Jan 15, 2010 at 10:08 AM, Anthony Minessale
> > > >> >
> > > >> > <anthony.minessale at gmail.com> wrote:
> > > >> >> one of the many reasons its a bad idea.
> > > >> >> Probably the leg with the bad audio is a different ptime.
> > > >> >> Now the amount of work I have to do escalates I would prefer you
> > >
> > > commit
> > >
> > > >> >> to
> > > >> >> commercial support by emailing me at consulting at freeswitch.org to
> > > >> >> continue
> > > >> >> with this.
> > > >> >>
> > > >> >>
> > > >> >> On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil
> > > >> >>
> > > >> >> <david.villasmil.work at gmail.com> wrote:
> > > >> >>> I set it to "off" just in case, same thing.
> > > >> >>>
> > > >> >>> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
> > > >> >>>
> > > >> >>> <david.villasmil.work at gmail.com> wrote:
> > > >> >>> > Default, haven't touched it i suppose it's off, i haven't set
> > > >> >>> > it anywhere
> > > >> >>> >
> > > >> >>> > On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
> > > >> >>> >
> > > >> >>> > <sos at sokhapkin.dyndns.org> wrote:
> > > >> >>> >> Is bypass_media on or off?
> > > >> >>> >>
> > > >> >>> >> On Friday 15 January 2010, David Villasmil wrote:
> > > >> >>> >>> Yeah, sorry. Early media. Audio after 200 is fine. Early
> > > >> >>> >>> media
> > >
> > > was
> > >
> > > >> >>> >>> ok
> > > >> >>> >>> before the change.
> > > >> >>> >>>
> > > >> >>> >>> David
> > > >> >>> >>>
> > > >> >>> >>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
> > > >> >>> >>>
> > > >> >>> >>> <sos at sokhapkin.dyndns.org> wrote:
> > > >> >>> >>> > Which audio? Early media or after 200 OK?
> > > >> >>> >>> >
> > > >> >>> >>> > On Friday 15 January 2010, David Villasmil wrote:
> > > >> >>> >>> >> Hello again Anthony,
> > > >> >>> >>> >>
> > > >> >>> >>> >> I just tested it, and although functionality does not,
> > > >> >>> >>> >> first incoming
> > > >> >>> >>> >> audio is coming in all garbled... do you know why?
> > > >> >>> >>> >>
> > > >> >>> >>> >> David
> > > >> >>> >>> >>
> > > >> >>> >>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
> > > >> >>> >>> >>
> > > >> >>> >>> >> <anthony.minessale at gmail.com> wrote:
> > > >> >>> >>> >> > {bridge_early_media=true}
> > > >> >>> >>> >> > in the dial string in place of ignore_early_media=true
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > the first b leg in the list who sends 183 will become
> > > >> >>> >>> >> > the ringback
> > > >> >>> >>> >> > device for A leg it will hear the early media
> > > >> >>> >>> >> > for that leg while the other legs still ring.  If some
> > >
> > > other
> > >
> > > >> >>> >>> >> > leg
> > > >> >>> >>> >> > answers the final call will still be bridged to the leg
> > > >> >>> >>> >> > who answered.
> > > >> >>> >>> >> >
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > I would estimate it at $500 payable on the big paypal
> > >
> > > button
> > >
> > > >> >>> >>> >> > on
> > > >> >>> >>> >> > http://www.freeswitch.org
> > > >> >>> >>> >> > but, I already added the patch to tree earlier today so
> > > >> >>> >>> >> > I guess
> > > >> >>> >>> >> > it's
> > > >> >>> >>> >> > up to you to pay it or not.
> > > >> >>> >>> >> >
> > > >> >>> >>> >> >
> > > >> >>> >>> >> >
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > <david.villasmil.work at gmail.com> wrote:
> > > >> >>> >>> >> >> Anthony,
> > > >> >>> >>> >> >>
> > > >> >>> >>> >> >> I did take the "hint", don't worry. We will probably
> > > >> >>> >>> >> >> ask
> > >
> > > for
> > >
> > > >> >>> >>> >> >> a
> > > >> >>> >>> >> >> bounty
> > > >> >>> >>> >> >> but first we need to know:
> > > >> >>> >>> >> >> 1.- whether this is possible
> > > >> >>> >>> >> >> 2.- how long it would take
> > > >> >>> >>> >> >> 3.- how will it exactly work
> > > >> >>> >>> >> >> 4.- of course, what's the bounty (be gentle ;) )
> > > >> >>> >>> >> >>
> > > >> >>> >>> >> >> We would of course give this back to the community.
> > > >> >>> >>> >> >>
> > > >> >>> >>> >> >> in the meantime, bridge_early_media=true will discard
> > > >> >>> >>> >> >> the 2nd
> > > >> >>> >>> >> >> B-leg
> > > >> >>> >>> >> >> and conversely "false" it will keep on trying to
> > > >> >>> >>> >> >> connect
> > >
> > > and
> > >
> > > >> >>> >>> >> >> if
> > > >> >>> >>> >> >> it
> > > >> >>> >>> >> >> connects the other B-leg if will bridge to that one?
> > > >> >>> >>> >> >>
> > > >> >>> >>> >> >> Thanks
> > > >> >>> >>> >> >>
> > > >> >>> >>> >> >> David
> > > >> >>> >>> >> >>
> > > >> >>> >>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> > > >> >>> >>> >> >>
> > > >> >>> >>> >> >> <anthony.minessale at gmail.com> wrote:
> > > >> >>> >>> >> >> > I added bridge_early_media=true to do the best I can
> > > >> >>> >>> >> >> > do. This is the most I will do, especially for free,
> > > >> >>> >>> >> >> > nobody can
> > > >> >>> >>> >> >> > take a
> > > >> >>> >>> >> >> > hint that
> > > >> >>> >>> >> >> > you should be paying for all these custom requests
> > > >> >>> >>> >> >> > so
> > >
> > > take
> > >
> > > >> >>> >>> >> >> > it
> > > >> >>> >>> >> >> > or
> > > >> >>> >>> >> >> > leave it
> > > >> >>> >>> >> >> > but this thread is done.........
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
> > > >> >>> >>> >> >> > <sos at sokhapkin.dyndns.org>
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> > wrote:
> > > >> >>> >>> >> >> >> No, not exactly. ignore_early_media doesn't pass
> > > >> >>> >>> >> >> >> early media
> > > >> >>> >>> >> >> >> to
> > > >> >>> >>> >> >> >> the caller
> > > >> >>> >>> >> >> >> if
> > > >> >>> >>> >> >> >> bypass_media is false.
> > > >> >>> >>> >> >> >>
> > > >> >>> >>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
> > > >> >>> >>> >> >> >> > this is exactly what ignore_early_media does now.
> > > >> >>> >>> >> >> >> >
> > > >> >>> >>> >> >> >> > Mike
> > > >> >>> >>> >> >> >> >
> > > >> >>> >>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin 
wrote:
> > > >> >>> >>> >> >> >> > > The issue here is when "originate" routine
> > > >> >>> >>> >> >> >> > > should return
> > > >> >>> >>> >> >> >> > > and
> > > >> >>> >>> >> >> >> > > set "originate_status" variable. Current
> > > >> >>> >>> >> >> >> > > behavior
> > >
> > > is
> > >
> > > >> >>> >>> >> >> >> > > to
> > > >> >>> >>> >> >> >> > > return
> > > >> >>> >>> >> >> >> > > on early
> > > >> >>> >>> >> >> >> > > media, but what if to introduce a variable
> > > >> >>> >>> >> >> >> > > "originate_wait_for_answer"
> > > >> >>> >>> >> >> >> > > with default value "false" and use the variable
> > > >> >>> >>> >> >> >> > > in originate
> > > >> >>> >>> >> >> >> > > code to
> > > >> >>> >>> >> >> >> > > decide when to return - on 18X or "200 OK"?
> > > >> >>> >>> >> >> >> > >
> > > >> >>> >>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale
> > >
> > > wrote:
> > > >> >>> >>> >> >> >> > >> he wants to call 3 people at once and let the
> > > >> >>> >>> >> >> >> > >> A
> > >
> > > leg
> > >
> > > >> >>> >>> >> >> >> > >> hear
> > > >> >>> >>> >> >> >> > >> early media
> > > >> >>> >>> >> >> >> > >> from call #1 while call #2 and #3 still are
> > > >> >>> >>> >> >> >> > >> progressing
> > > >> >>> >>> >> >> >> > >> which
> > > >> >>> >>> >> >> >> > >> is not
> > > >> >>> >>> >> >> >> > >> simple to do without doing thousands of
> > > >> >>> >>> >> >> >> > >> dollars in development.
> > > >> >>> >>> >> >> >> > >>
> > > >> >>> >>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB
> > > >> >>> >>> >> >> >> > >> <djbinter at yahoo.com>
> > > >> >>> >>
> > > >> >>> >> wrote:
> > > >> >>> >>> >> >> >> > >>> What about sending Sip 183 with SDP (no
> > > >> >>> >>> >> >> >> > >>> 200OK),
> > >
> > > so
> > >
> > > >> >>> >>> >> >> >> > >>> that your
> > > >> >>> >>> >> >> >> > >>> customers
> > > >> >>> >>> >> >> >> > >>> can hear recordings?
> > > >> >>> >>> >> >> >> >
> > > >> >>> >>> >> >> >> > _______________________________________________
> > > >> >>> >>> >> >> >> > FreeSWITCH-users mailing list
> > > >> >>> >>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >> >>> >>> >> >> >> > UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswit
> > >
> > > >> >>> >>> >> >> >> >ch- users http://www.freeswitch.org
> > > >> >>> >>> >> >> >>
> > > >> >>> >>> >> >> >> _______________________________________________
> > > >> >>> >>> >> >> >> FreeSWITCH-users mailing list
> > > >> >>> >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >> >>> >>> >> >> >> UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch
> > >
> > > >> >>> >>> >> >> >>-us ers http://www.freeswitch.org
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> > --
> > > >> >>> >>> >> >> > Anthony Minessale II
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> > FreeSWITCH http://www.freeswitch.org/
> > > >> >>> >>> >> >> > ClueCon http://www.cluecon.com/
> > > >> >>> >>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> > AIM: anthm
> > > >> >>> >>> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_mine
> > > >> >>> >>> >> >> >ss ale at hotmail.com>
> > > >> >>> >>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYP
> > > >> >>> >>> >> >> >AL %3Aanthony.minessale at gmail.com> IRC:
> > > >> >>> >>> >> >> > irc.freenode.net #freeswitch
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> > FreeSWITCH Developer Conference
> > > >> >>> >>> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conferen
> > > >> >>> >>> >> >> >ce .freeswitch.org>
> > > >> >>> >>> >> >> > iax:guest at conference.freeswitch.org/888
> > > >> >>> >>> >> >> > googletalk:conf+888 at conference.freeswitch.org<google
> > > >> >>> >>> >> >> >ta lk%3Aconf%2B888 at conference.freeswitch.org>
> > > >> >>> >>> >> >> > pstn:+19193869900
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> > _______________________________________________
> > > >> >>> >>> >> >> > FreeSWITCH-users mailing list
> > > >> >>> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
> > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >> >>> >>> >> >> > UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-
> > >
> > > >> >>> >>> >> >> >use rs http://www.freeswitch.org
> > > >> >>> >>> >> >>
> > > >> >>> >>> >> >> _______________________________________________
> > > >> >>> >>> >> >> FreeSWITCH-users mailing list
> > > >> >>> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
> > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >> >>> >>> >> >> UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-us
> > >
> > > >> >>> >>> >> >>ers http://www.freeswitch.org
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > --
> > > >> >>> >>> >> > Anthony Minessale II
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > FreeSWITCH http://www.freeswitch.org/
> > > >> >>> >>> >> > ClueCon http://www.cluecon.com/
> > > >> >>> >>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > AIM: anthm
> > > >> >>> >>> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessa
> > > >> >>> >>> >> >le @hotmail.com>
> > > >> >>> >>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%
> > > >> >>> >>> >> >3A anthony.minessale at gmail.com> IRC: irc.freenode.net
> > > >> >>> >>> >> > #freeswitch
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > FreeSWITCH Developer Conference
> > > >> >>> >>> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.
> > > >> >>> >>> >> >fr eeswitch.org> iax:guest at conference.freeswitch.org/888
> > > >> >>> >>> >> > googletalk:conf+888 at conference.freeswitch.org<googletal
> > > >> >>> >>> >> >k% 3Aconf%2B888 at conference.freeswitch.org>
> > > >> >>> >>> >> > pstn:+19193869900
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > _______________________________________________
> > > >> >>> >>> >> > FreeSWITCH-users mailing list
> > > >> >>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >> >>> >>> >> > UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-use
> > >
> > > >> >>> >>> >> >rs http://www.freeswitch.org
> > > >> >>> >>> >>
> > > >> >>> >>> >> _______________________________________________
> > > >> >>> >>> >> FreeSWITCH-users mailing list
> > > >> >>> >>> >> FreeSWITCH-users at lists.freeswitch.org
> > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >> >>> >>> >> UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> >>> >>> >> http://www.freeswitch.org
> > > >> >>> >>> >
> > > >> >>> >>> > _______________________________________________
> > > >> >>> >>> > FreeSWITCH-users mailing list
> > > >> >>> >>> > FreeSWITCH-users at lists.freeswitch.org
> > > >> >>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-us
> > > >> >>> >>> >er s
> > > >> >>> >>> >
> > > >> >>> >>> >
> > > >> >>> >>> > UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> >>> >>> > http://www.freeswitch.org
> > > >> >>> >>>
> > > >> >>> >>> _______________________________________________
> > > >> >>> >>> FreeSWITCH-users mailing list
> > > >> >>> >>> FreeSWITCH-users at lists.freeswitch.org
> > > >> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-user
> > > >> >>> >>>s
> > > >> >>> >>>
> > > >> >>> >>>
> > > >> >>> >>> UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> >>> >>> http://www.freeswitch.org
> > > >> >>> >>
> > > >> >>> >> _______________________________________________
> > > >> >>> >> FreeSWITCH-users mailing list
> > > >> >>> >> FreeSWITCH-users at lists.freeswitch.org
> > > >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >> >>> >>
> > > >> >>> >>
> > > >> >>> >> UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> >>> >> http://www.freeswitch.org
> > > >> >>>
> > > >> >>> _______________________________________________
> > > >> >>> FreeSWITCH-users mailing list
> > > >> >>> FreeSWITCH-users at lists.freeswitch.org
> > > >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >> >>>
> > > >> >>> UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> >>> http://www.freeswitch.org
> > > >> >>
> > > >> >> --
> > > >> >> Anthony Minessale II
> > > >> >>
> > > >> >> FreeSWITCH http://www.freeswitch.org/
> > > >> >> ClueCon http://www.cluecon.com/
> > > >> >> Twitter: http://twitter.com/FreeSWITCH_wire
> > > >> >>
> > > >> >> AIM: anthm
> > > >> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail
> > > >> >>.c om>
> > > >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.
> > > >> >>mi nessale at gmail.com> IRC: irc.freenode.net #freeswitch
> > > >> >>
> > > >> >> FreeSWITCH Developer Conference
> > > >> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch
> > > >> >>.o rg> iax:guest at conference.freeswitch.org/888
> > > >> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2
> > > >> >>B8 88 at conference.freeswitch.org> pstn:+19193869900
> > > >> >
> > > >> > --
> > > >> > Anthony Minessale II
> > > >> >
> > > >> > FreeSWITCH http://www.freeswitch.org/
> > > >> > ClueCon http://www.cluecon.com/
> > > >> > Twitter: http://twitter.com/FreeSWITCH_wire
> > > >> >
> > > >> > AIM: anthm
> > > >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.
> > > >> >co m>
> > > >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.m
> > > >> >in essale at gmail.com> IRC: irc.freenode.net #freeswitch
> > > >> >
> > > >> > FreeSWITCH Developer Conference
> > > >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.
> > > >> >or g> iax:guest at conference.freeswitch.org/888
> > > >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B
> > > >> >88 8 at conference.freeswitch.org> pstn:+19193869900
> > > >> >
> > > >> > _______________________________________________
> > > >> > FreeSWITCH-users mailing list
> > > >> > FreeSWITCH-users at lists.freeswitch.org
> > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >> > UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> > http://www.freeswitch.org
> > > >>
> > > >> _______________________________________________
> > > >> FreeSWITCH-users mailing list
> > > >> FreeSWITCH-users at lists.freeswitch.org
> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >> UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> http://www.freeswitch.org
> > > >
> > > > --
> > > > Anthony Minessale II
> > > >
> > > > FreeSWITCH http://www.freeswitch.org/
> > > > ClueCon http://www.cluecon.com/
> > > > Twitter: http://twitter.com/FreeSWITCH_wire
> > > >
> > > > AIM: anthm
> > > > MSN:anthony_minessale at hotmail.com
> > > > <MSN%3Aanthony_minessale at hotmail.com>
> > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.mine
> > > >ss ale at gmail.com> IRC: irc.freenode.net #freeswitch
> > > >
> > > > FreeSWITCH Developer Conference
> > > > sip:888 at conference.freeswitch.org
> > > > <sip%3A888 at conference.freeswitch.org>
> > > > iax:guest at conference.freeswitch.org/888
> > > > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888
> > > >@c onference.freeswitch.org> pstn:+19193869900
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
> > > >er s http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From stevesteffler at shaw.ca  Fri Jan 15 20:36:03 2010
From: stevesteffler at shaw.ca (Steve Steffler)
Date: Fri, 15 Jan 2010 21:36:03 -0700
Subject: [Freeswitch-users] FIFO Originate caller ID
In-Reply-To: <88D85720-C24A-43B8-A6E8-7F9179D71625@jerris.com>
References: <29507085-AE04-4A43-A41A-6D279810CCEC@shaw.ca>
	<88D85720-C24A-43B8-A6E8-7F9179D71625@jerris.com>
Message-ID: <D85EDE43-C1D0-438C-A273-0016856CA723@shaw.ca>


Mike,

That explanation makes sense - thanks.

Perhaps a n00b question, but where can I configure the caller ID for the outbound queue calls?

Thanks,
Steve

On Jan 15, 2010, at 5:01 PM, Michael Jerris wrote:

> At the time a call goes out to the agents, there is no specific caller they are matched too, therefore there is no way to know the caller id at this time.  When the originated call to the agent is answered, we THEN go and pick off the next caller to connect them with.  All you can do is set a caller id for the queue.
> 
> Mike
> 
> On Jan 15, 2010, at 6:43 PM, Steve Steffler wrote:
> 
>> Hello,
>> 
>> I found an archived conversation on this list regarding FIFO origination caller ID, and how to modify it.  
>> 
>> http://old.nabble.com/FIFO-Orgination_caller_id-td26487628.html
>> 
>> There seems to be no easy way to customize the caller ID on originated calls from the FIFO to on-hook agents who were registered dynamically.  Anthony states a method to do it using static entries in the fifo conf file, and the general rationale is that SCREEN POPS be the preferred method, with the added nudge that good SIP phones can change the caller ID when the bridge is complete as well, which is all well and good.
>> 
>> The problem is for my application, all on-hook agents are using cellular phones, and they register dynamically.  Also, none of my agents are in front of a computer, so a SIP display update on the phone or screen pop on the computer in front of them is not really an option, and the only way they can identify calls from my FIFO right now is because they are the ones with NO CALLER ID (in other words, their mobile phones do not display the name, and the number is not recognized because it is set by FreeSWITCH to be "fifo+fifoname" instead of being numeric.  This is far from ideal.
>> 
>> I am wondering if there is anyone on the list who knows how to configure the origination_caller_id_number/name variables for dynamically registered on-hook agents so that the caller ID from the FIFO customer's incoming call is displayed to them instead of the above mangled caller ID?
>> 
>> I'm not disagreeing that it is an old-skewl way of thought, but in actuality it is just a way to interface with old-school telephony devices (i.e. non-Smartphone mobile phones) and I am not sure how to accomplish this.  Any help/input would be appreciated.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Fri Jan 15 20:56:10 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 Jan 2010 23:56:10 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001152027.07209.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<191c3a031001151644m6a838b6bp311f2a173fc65af3@mail.gmail.com>
	<201001152004.25949.sos@sokhapkin.dyndns.org>
	<201001152027.07209.sos@sokhapkin.dyndns.org>
Message-ID: <84089307-723F-4910-9874-CFE09147B13E@jerris.com>

Well, in the dialplan, typically your not going to get past the bridge  
line in your dialplan because the call is done when the bridge is  
over.  But as Anthony noted you can set vars in [. ] that only apply  
to the b legs.  These you can get in the cdr or in an api hangup hook  
or otherwise in the reporting state.

Mike

On Jan 15, 2010, at 8:27 PM, Sergey Okhapkin  
<sos at sokhapkin.dyndns.org> wrote:

> Same question about
>
> originate aaa|bbb|ccc
>
> syntax :-)
>
>
> On Friday 15 January 2010, Sergey Okhapkin wrote:
>> Sorry for the dumb question, is there a way to find out in dialplan  
>> (some
>> variable?) which one of comma separated b-legs listed in originate  
>> command
>> answered the call?
>>
>> originate sofia/g1/number,sofia/g2/number,sofia/g3/number
>>
>> Which gateway answered? g1, g2 or g3?
>>
>> On Friday 15 January 2010, Anthony Minessale wrote:
>>> Now we need a new feature
>>>
>>> [leg_required=true]
>>>
>>> set this on any legs required for the originate to proceed, if it's
>>> hungup, the cause will be passed to any existing legs and fail the  
>>> entire
>>> originate.
>>>
>>> so use
>>>
>>> {bridge_early_media=true}[leg_required=true]sofia/internal/ 
>>> real_call at foo.
>>> co m ,sofia/internal/moh_call at foo.com
>>>
>>> the leg_required will only be set on the 1st leg because of the []  
>>> vs {}
>>> if that leg is then hungup, it will kill the other channels in the  
>>> list.
>>>
>>> please try latest trunk.
>>>
>>>
>>>
>>> On Fri, Jan 15, 2010 at 5:25 PM, David Villasmil <
>>>
>>> david.villasmil.work at gmail.com> wrote:
>>>> Anthony,
>>>>
>>>> LOL, and mounting and mounting... It does work when there is  
>>>> answer...
>>>> but if B(2)-side rejects or times out or any other that 200 OK,
>>>> B(1)-side stays indefinitely...
>>>>
>>>>
>>>> On Fri, Jan 15, 2010 at 11:54 PM, Anthony Minessale
>>>>
>>>> <anthony.minessale at gmail.com> wrote:
>>>>> you can email me privately at this addr.
>>>>>
>>>>>
>>>>> On Fri, Jan 15, 2010 at 4:45 PM, David Villasmil
>>>>>
>>>>> <david.villasmil.work at gmail.com> wrote:
>>>>>> Anthony,
>>>>>>
>>>>>>    Trying, Thanks. Is there anyway we can communicate directly?
>>>>>>
>>>>>>
>>>>>> David
>>>>>>
>>>>>> On Fri, Jan 15, 2010 at 10:17 PM, Anthony Minessale
>>>>>>
>>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>>> Try latest trunk,
>>>>>>>
>>>>>>> you should have exactly what you want with the same parameter,
>>>>>>> again
>>>>
>>>> my
>>>>
>>>>>>> paypal addr is cleary displayed as a big button on the website.
>>>>>>>
>>>>>>>
>>>>>>> On Fri, Jan 15, 2010 at 10:08 AM, Anthony Minessale
>>>>>>>
>>>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>>>> one of the many reasons its a bad idea.
>>>>>>>> Probably the leg with the bad audio is a different ptime.
>>>>>>>> Now the amount of work I have to do escalates I would prefer  
>>>>>>>> you
>>>>
>>>> commit
>>>>
>>>>>>>> to
>>>>>>>> commercial support by emailing me at  
>>>>>>>> consulting at freeswitch.org to
>>>>>>>> continue
>>>>>>>> with this.
>>>>>>>>
>>>>>>>>
>>>>>>>> On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil
>>>>>>>>
>>>>>>>> <david.villasmil.work at gmail.com> wrote:
>>>>>>>>> I set it to "off" just in case, same thing.
>>>>>>>>>
>>>>>>>>> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
>>>>>>>>>
>>>>>>>>> <david.villasmil.work at gmail.com> wrote:
>>>>>>>>>> Default, haven't touched it i suppose it's off, i haven't set
>>>>>>>>>> it anywhere
>>>>>>>>>>
>>>>>>>>>> On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
>>>>>>>>>>
>>>>>>>>>> <sos at sokhapkin.dyndns.org> wrote:
>>>>>>>>>>> Is bypass_media on or off?
>>>>>>>>>>>
>>>>>>>>>>> On Friday 15 January 2010, David Villasmil wrote:
>>>>>>>>>>>> Yeah, sorry. Early media. Audio after 200 is fine. Early
>>>>>>>>>>>> media
>>>>
>>>> was
>>>>
>>>>>>>>>>>> ok
>>>>>>>>>>>> before the change.
>>>>>>>>>>>>
>>>>>>>>>>>> David
>>>>>>>>>>>>
>>>>>>>>>>>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
>>>>>>>>>>>>
>>>>>>>>>>>> <sos at sokhapkin.dyndns.org> wrote:
>>>>>>>>>>>>> Which audio? Early media or after 200 OK?
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Friday 15 January 2010, David Villasmil wrote:
>>>>>>>>>>>>>> Hello again Anthony,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I just tested it, and although functionality does not,
>>>>>>>>>>>>>> first incoming
>>>>>>>>>>>>>> audio is coming in all garbled... do you know why?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> David
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>>>>>>>>>>> {bridge_early_media=true}
>>>>>>>>>>>>>>> in the dial string in place of ignore_early_media=true
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> the first b leg in the list who sends 183 will become
>>>>>>>>>>>>>>> the ringback
>>>>>>>>>>>>>>> device for A leg it will hear the early media
>>>>>>>>>>>>>>> for that leg while the other legs still ring.  If some
>>>>
>>>> other
>>>>
>>>>>>>>>>>>>>> leg
>>>>>>>>>>>>>>> answers the final call will still be bridged to the leg
>>>>>>>>>>>>>>> who answered.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> I would estimate it at $500 payable on the big paypal
>>>>
>>>> button
>>>>
>>>>>>>>>>>>>>> on
>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>> but, I already added the patch to tree earlier today so
>>>>>>>>>>>>>>> I guess
>>>>>>>>>>>>>>> it's
>>>>>>>>>>>>>>> up to you to pay it or not.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> <david.villasmil.work at gmail.com> wrote:
>>>>>>>>>>>>>>>> Anthony,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> I did take the "hint", don't worry. We will probably
>>>>>>>>>>>>>>>> ask
>>>>
>>>> for
>>>>
>>>>>>>>>>>>>>>> a
>>>>>>>>>>>>>>>> bounty
>>>>>>>>>>>>>>>> but first we need to know:
>>>>>>>>>>>>>>>> 1.- whether this is possible
>>>>>>>>>>>>>>>> 2.- how long it would take
>>>>>>>>>>>>>>>> 3.- how will it exactly work
>>>>>>>>>>>>>>>> 4.- of course, what's the bounty (be gentle ;) )
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> We would of course give this back to the community.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> in the meantime, bridge_early_media=true will discard
>>>>>>>>>>>>>>>> the 2nd
>>>>>>>>>>>>>>>> B-leg
>>>>>>>>>>>>>>>> and conversely "false" it will keep on trying to
>>>>>>>>>>>>>>>> connect
>>>>
>>>> and
>>>>
>>>>>>>>>>>>>>>> if
>>>>>>>>>>>>>>>> it
>>>>>>>>>>>>>>>> connects the other B-leg if will bridge to that one?
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Thanks
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> David
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>>>>>>>>>>>>> I added bridge_early_media=true to do the best I can
>>>>>>>>>>>>>>>>> do. This is the most I will do, especially for free,
>>>>>>>>>>>>>>>>> nobody can
>>>>>>>>>>>>>>>>> take a
>>>>>>>>>>>>>>>>> hint that
>>>>>>>>>>>>>>>>> you should be paying for all these custom requests
>>>>>>>>>>>>>>>>> so
>>>>
>>>> take
>>>>
>>>>>>>>>>>>>>>>> it
>>>>>>>>>>>>>>>>> or
>>>>>>>>>>>>>>>>> leave it
>>>>>>>>>>>>>>>>> but this thread is done.........
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>>>>>>>>>>>>>>>>> <sos at sokhapkin.dyndns.org>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> wrote:
>>>>>>>>>>>>>>>>>> No, not exactly. ignore_early_media doesn't pass
>>>>>>>>>>>>>>>>>> early media
>>>>>>>>>>>>>>>>>> to
>>>>>>>>>>>>>>>>>> the caller
>>>>>>>>>>>>>>>>>> if
>>>>>>>>>>>>>>>>>> bypass_media is false.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> On Thursday 14 January 2010, Michael Jerris wrote:
>>>>>>>>>>>>>>>>>>> this is exactly what ignore_early_media does now.
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> Mike
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin
> wrote:
>>>>>>>>>>>>>>>>>>>> The issue here is when "originate" routine
>>>>>>>>>>>>>>>>>>>> should return
>>>>>>>>>>>>>>>>>>>> and
>>>>>>>>>>>>>>>>>>>> set "originate_status" variable. Current
>>>>>>>>>>>>>>>>>>>> behavior
>>>>
>>>> is
>>>>
>>>>>>>>>>>>>>>>>>>> to
>>>>>>>>>>>>>>>>>>>> return
>>>>>>>>>>>>>>>>>>>> on early
>>>>>>>>>>>>>>>>>>>> media, but what if to introduce a variable
>>>>>>>>>>>>>>>>>>>> "originate_wait_for_answer"
>>>>>>>>>>>>>>>>>>>> with default value "false" and use the variable
>>>>>>>>>>>>>>>>>>>> in originate
>>>>>>>>>>>>>>>>>>>> code to
>>>>>>>>>>>>>>>>>>>> decide when to return - on 18X or "200 OK"?
>>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>> On Thursday 14 January 2010, Anthony Minessale
>>>>
>>>> wrote:
>>>>>>>>>>>>>>>>>>>>> he wants to call 3 people at once and let the
>>>>>>>>>>>>>>>>>>>>> A
>>>>
>>>> leg
>>>>
>>>>>>>>>>>>>>>>>>>>> hear
>>>>>>>>>>>>>>>>>>>>> early media
>>>>>>>>>>>>>>>>>>>>> from call #1 while call #2 and #3 still are
>>>>>>>>>>>>>>>>>>>>> progressing
>>>>>>>>>>>>>>>>>>>>> which
>>>>>>>>>>>>>>>>>>>>> is not
>>>>>>>>>>>>>>>>>>>>> simple to do without doing thousands of
>>>>>>>>>>>>>>>>>>>>> dollars in development.
>>>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>>> On Thu, Jan 14, 2010 at 11:39 AM, DJB
>>>>>>>>>>>>>>>>>>>>> <djbinter at yahoo.com>
>>>>>>>>>>>
>>>>>>>>>>> wrote:
>>>>>>>>>>>>>>>>>>>>>> What about sending Sip 183 with SDP (no
>>>>>>>>>>>>>>>>>>>>>> 200OK),
>>>>
>>>> so
>>>>
>>>>>>>>>>>>>>>>>>>>>> that your
>>>>>>>>>>>>>>>>>>>>>> customers
>>>>>>>>>>>>>>>>>>>>>> can hear recordings?
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>>>>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswit
>>>>
>>>>>>>>>>>>>>>>>>> ch- users http://www.freeswitch.org
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>>>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch
>>>>
>>>>>>>>>>>>>>>>>> -us ers http://www.freeswitch.org
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>>>> Anthony Minessale II
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>>>>>>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>>>>>>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> AIM: anthm
>>>>>>>>>>>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_mine
>>>>>>>>>>>>>>>>> ss ale at hotmail.com>
>>>>>>>>>>>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYP
>>>>>>>>>>>>>>>>> AL %3Aanthony.minessale at gmail.com> IRC:
>>>>>>>>>>>>>>>>> irc.freenode.net #freeswitch
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> FreeSWITCH Developer Conference
>>>>>>>>>>>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conferen
>>>>>>>>>>>>>>>>> ce .freeswitch.org>
>>>>>>>>>>>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>>>>>>>>>>>> googletalk:conf+888 at conference.freeswitch.org<google
>>>>>>>>>>>>>>>>> ta lk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>>>>>>>>>>> pstn:+19193869900
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>
>>>>>>>>>>>>>>>>> use rs http://www.freeswitch.org
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-us
>>>>
>>>>>>>>>>>>>>>> ers http://www.freeswitch.org
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>> Anthony Minessale II
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>>>>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>>>>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> AIM: anthm
>>>>>>>>>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessa
>>>>>>>>>>>>>>> le @hotmail.com>
>>>>>>>>>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%
>>>>>>>>>>>>>>> 3A anthony.minessale at gmail.com> IRC: irc.freenode.net
>>>>>>>>>>>>>>> #freeswitch
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> FreeSWITCH Developer Conference
>>>>>>>>>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.
>>>>>>>>>>>>>>> fr eeswitch.org> iax:guest at conference.freeswitch.org/888
>>>>>>>>>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletal
>>>>>>>>>>>>>>> k% 3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>>>>>>>>> pstn:+19193869900
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-use
>>>>
>>>>>>>>>>>>>>> rs http://www.freeswitch.org
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-us
>>>>>>>>>>>>> er s
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch- 
>>>>>>>>>>>> user
>>>>>>>>>>>> s
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch- 
>>>>>>>>>>> users
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>
>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>> --
>>>>>>>> Anthony Minessale II
>>>>>>>>
>>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>>
>>>>>>>> AIM: anthm
>>>>>>>> MSN:anthony_minessale at hotmail.com<MSN 
>>>>>>>> %3Aanthony_minessale at hotmail
>>>>>>>> .c om>
>>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL 
>>>>>>>> %3Aanthony.
>>>>>>>> mi nessale at gmail.com> IRC: irc.freenode.net #freeswitch
>>>>>>>>
>>>>>>>> FreeSWITCH Developer Conference
>>>>>>>> sip:888 at conference.freeswitch.org<sip 
>>>>>>>> %3A888 at conference.freeswitch
>>>>>>>> .o rg> iax:guest at conference.freeswitch.org/888
>>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk 
>>>>>>>> %3Aconf%2
>>>>>>>> B8 88 at conference.freeswitch.org> pstn:+19193869900
>>>>>>>
>>>>>>> --
>>>>>>> Anthony Minessale II
>>>>>>>
>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>
>>>>>>> AIM: anthm
>>>>>>> MSN:anthony_minessale at hotmail.com<MSN 
>>>>>>> %3Aanthony_minessale at hotmail.
>>>>>>> co m>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL 
>>>>>>> %3Aanthony.m
>>>>>>> in essale at gmail.com> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>> sip:888 at conference.freeswitch.org<sip 
>>>>>>> %3A888 at conference.freeswitch.
>>>>>>> or g> iax:guest at conference.freeswitch.org/888
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf 
>>>>>>> %2B
>>>>>>> 88 8 at conference.freeswitch.org> pstn:+19193869900
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>
>>>>>> http://www.freeswitch.org
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>> <MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL 
>>>>> %3Aanthony.mine
>>>>> ss ale at gmail.com> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> <sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf 
>>>>> %2B888
>>>>> @c onference.freeswitch.org> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>>> freeswitch-us
>>>>> er s http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>> freeswitch-user
>>>> s http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From mike at jerris.com  Fri Jan 15 21:10:08 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 16 Jan 2010 00:10:08 -0500
Subject: [Freeswitch-users] FIFO Originate caller ID
In-Reply-To: <D85EDE43-C1D0-438C-A273-0016856CA723@shaw.ca>
References: <29507085-AE04-4A43-A41A-6D279810CCEC@shaw.ca>
	<88D85720-C24A-43B8-A6E8-7F9179D71625@jerris.com>
	<D85EDE43-C1D0-438C-A273-0016856CA723@shaw.ca>
Message-ID: <2CCD724A-65E4-4EB6-A813-50F26CC0642A@jerris.com>

You can put vars for manipulating callerid right in the originate  
string either in config or using the add for the on hook agents.

Mike

On Jan 15, 2010, at 11:36 PM, Steve Steffler <stevesteffler at shaw.ca>  
wrote:

>
> Mike,
>
> That explanation makes sense - thanks.
>
> Perhaps a n00b question, but where can I configure the caller ID for  
> the outbound queue calls?
>
> Thanks,
> Steve
>
> On Jan 15, 2010, at 5:01 PM, Michael Jerris wrote:
>
>> At the time a call goes out to the agents, there is no specific  
>> caller they are matched too, therefore there is no way to know the  
>> caller id at this time.  When the originated call to the agent is  
>> answered, we THEN go and pick off the next caller to connect them  
>> with.  All you can do is set a caller id for the queue.
>>
>> Mike
>>
>> On Jan 15, 2010, at 6:43 PM, Steve Steffler wrote:
>>
>>> Hello,
>>>
>>> I found an archived conversation on this list regarding FIFO  
>>> origination caller ID, and how to modify it.
>>>
>>> http://old.nabble.com/FIFO-Orgination_caller_id-td26487628.html
>>>
>>> There seems to be no easy way to customize the caller ID on  
>>> originated calls from the FIFO to on-hook agents who were  
>>> registered dynamically.  Anthony states a method to do it using  
>>> static entries in the fifo conf file, and the general rationale is  
>>> that SCREEN POPS be the preferred method, with the added nudge  
>>> that good SIP phones can change the caller ID when the bridge is  
>>> complete as well, which is all well and good.
>>>
>>> The problem is for my application, all on-hook agents are using  
>>> cellular phones, and they register dynamically.  Also, none of my  
>>> agents are in front of a computer, so a SIP display update on the  
>>> phone or screen pop on the computer in front of them is not really  
>>> an option, and the only way they can identify calls from my FIFO  
>>> right now is because they are the ones with NO CALLER ID (in other  
>>> words, their mobile phones do not display the name, and the number  
>>> is not recognized because it is set by FreeSWITCH to be "fifo 
>>> +fifoname" instead of being numeric.  This is far from ideal.
>>>
>>> I am wondering if there is anyone on the list who knows how to  
>>> configure the origination_caller_id_number/name variables for  
>>> dynamically registered on-hook agents so that the caller ID from  
>>> the FIFO customer's incoming call is displayed to them instead of  
>>> the above mangled caller ID?
>>>
>>> I'm not disagreeing that it is an old-skewl way of thought, but in  
>>> actuality it is just a way to interface with old-school telephony  
>>> devices (i.e. non-Smartphone mobile phones) and I am not sure how  
>>> to accomplish this.  Any help/input would be appreciated.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From stevesteffler at shaw.ca  Fri Jan 15 21:17:29 2010
From: stevesteffler at shaw.ca (Steve Steffler)
Date: Fri, 15 Jan 2010 22:17:29 -0700
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
Message-ID: <776BA929-D7EC-4366-AF49-348F0E11488B@shaw.ca>


Costa,

I wrote this script to handle fax2email (but not email2fax).  It uses variables you set in the dialplan in advance for the email address for that fax DID.

http://steffler.info/wp-content/uploads/2009/06/process-rxfax.py

Regards,
Steve

On Jan 14, 2010, at 7:54 AM, Costa Zikalala wrote:

> Hi All
> 
> Has anyone worked on a email2fax script for mod_fax? 
> If not how much would it cost for some genius here to quickly whip-up one?
> 
> Ideally both email2fax and fax2email should come standard with mod_fax or is this not possible?
> 
> Thanks 
> Costa
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From yehavi.bourvine at gmail.com  Fri Jan 15 21:57:07 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 16 Jan 2010 07:57:07 +0200
Subject: [Freeswitch-users] Polycom configuration for FreeSWITCH
In-Reply-To: <003701ca964b$3241b100$96c51300$@net>
References: <003701ca964b$3241b100$96c51300$@net>
Message-ID: <d361ca7b1001152157x45b8d8c4l29625c763b24157f@mail.gmail.com>

I am working with various Polycom phones; I'll send you sample configuration
files next week (I am at home now). In the meantime, please send me your
requirenents so I may incorporate some of them into the files.

Have you managed to boot them from your TFTP/FTP./HTTP server? As long as
you did not provision them through a server you can do that through the
phone's WEB interface, but it is very limited and lacks a lot of
configuration options. I do the provisioning via a TFTP server.

                    Regards, __Yehavi:

 2010/1/16 Adam Ford <lists at redbonez.net>

>   Does anyone who has successfully implemented Polycom IP301/501s, with
> FreeSWITCH, have a configuration they can share with me as an example? I
> have read the Polycom Admin Guide several times and understand what the
> settings are/do, I am just not sure which FreeSWITCH supports, which it
> doesn?t, and which need special configuration to work with FreeSWITCH.
>
>
>
> Thanks in advance,
>
> -Adam
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Fri Jan 15 22:16:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 16 Jan 2010 01:16:18 -0500
Subject: [Freeswitch-users] Polycom configuration for FreeSWITCH
In-Reply-To: <d361ca7b1001152157x45b8d8c4l29625c763b24157f@mail.gmail.com>
References: <003701ca964b$3241b100$96c51300$@net>
	<d361ca7b1001152157x45b8d8c4l29625c763b24157f@mail.gmail.com>
Message-ID: <84F040CA-3FA1-4A5D-A417-B4C4F1B21E43@jerris.com>

If people have phone config examples, a good place to share them is on  
the wiki.

Mike

On Jan 16, 2010, at 12:57 AM, Yehavi Bourvine  
<yehavi.bourvine at gmail.com> wrote:

> I am working with various Polycom phones; I'll send you sample  
> configuration files next week (I am at home now). In the meantime,  
> please send me your requirenents so I may incorporate some of them  
> into the files.
>
> Have you managed to boot them from your TFTP/FTP./HTTP server? As  
> long as you did not provision them through a server you can do that  
> through the phone's WEB interface, but it is very limited and lacks  
> a lot of configuration options. I do the provisioning via a TFTP  
> server.
>
>                     Regards, __Yehavi:
>
> 2010/1/16 Adam Ford <lists at redbonez.net>
> Does anyone who has successfully implemented Polycom IP301/501s,  
> with FreeSWITCH, have a configuration they can share with me as an  
> example? I have read the Polycom Admin Guide several times and  
> understand what the settings are/do, I am just not sure which  
> FreeSWITCH supports, which it doesn?t, and which need special config 
> uration to work with FreeSWITCH.
>
>
>
> Thanks in advance,
>
> -Adam
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From kawarod at laposte.net  Fri Jan 15 22:22:53 2010
From: kawarod at laposte.net (rod)
Date: Sat, 16 Jan 2010 10:22:53 +0400
Subject: [Freeswitch-users] Eavesdrop in LUA
In-Reply-To: <007201ca95bb$ba673770$2f35a650$@com>
References: <4B4ED32E.30706@laposte.net> <4B4F33C7.6020403@laposte.net>
	<007201ca95bb$ba673770$2f35a650$@com>
Message-ID: <4B515B3D.9020309@laposte.net>

Hi Pete,

to get the beginning of the communication, I find this:
use the pre-answer/ringback command before bridging the call, this will 
issue a 183 with ringback in RTP.  Doing this, eavesdrop application  
can listen  the  caller  A talking  in the phone (and the ringback tone) 
even if the call is not connected to B :o

in the dialplan I did this:
<action application="set" data="ringback=$${fr-ring}"/>
<action application="pre_answer"/>
<action application="sleep" data="500"/>
<actionapplication="bridge"data="{ecoute=${caller_id_number}}sofia/gateway/${caller_id_number}/${destination_numbe\
r}"/>

A colleague wrote a perl script using mod_event that looks for when a 
call that should be eavesdrop is connecting and originate a call to C 
using eavesdrop on A leg. I will ask my colleague/boss if he's okay to 
share his script on the wiki. I think, he'll be okay, but prefer asking 
before.

Give a try to pre-anwer with lua, and let me know if you could eavesdrop 
to a call before the call is exchanging media.
Are you ok to share your lua script, so that we could document the 
eavesdrop page ?

regards,
rod

Pete Mueller a ?crit :
> I had a similar problem.  I solved it by first making bridging the call
> between A and B.
> Then originate C with a LUA script, the last line of which is:
>
> session:execute("eavesdrop", uuid_of_a_leg)
>
> The down side here is that A and B can talk while C is ringing, but in my
> case that is not a problem.
> -p
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod
> Sent: Thursday, January 14, 2010 8:10 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Eavesdrop in LUA
>
> Hi all,
>
> I have an incomplete solution for those interested.
>
> I did it like this in dialplan:
>         <action application="set" data="api_on_answer=originate 
> sofia/internal/2000 at 10.10.10.10 &eavesdrop(${uuid})"/>-->
>         <action application="bridge" 
> data="{ecoute=${caller_id_number}}sofia/gateway/${caller_id_number}/${destin
> ation_number}"/>
>
> so when a call is setup, FS initiate a new call to 2000 and eavesdrop 
> the call.
> But I have a small problem, the callee receives no sound until the 
> eavesdropper send a SIP reply, so there is a 2-3 seconds delay before 
> caller and callee can talk each other.
>
> rod
>
>
> rod a ?crit :
>   
>> Hi all,
>>
>> I'm trying to do this in LUA:
>>      A call B
>>
>> and I'd like to setup a new call to C with eavesdrop of A conversation 
>> with B.
>>
>> I have no idea how to do this if someone can help.
>> I switched to LUA cause I see no way to achieve this with dialplan 
>> (snippets are welcome).
>>
>> regards,
>> rod
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>   
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   


From woodydickson at gmail.com  Fri Jan 15 22:36:32 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 16 Jan 2010 14:36:32 +0800
Subject: [Freeswitch-users] ODBC Not Available!
Message-ID: <a44688a1001152236y57f8fe8bk5aced53cfd41e425@mail.gmail.com>

Hello,

I am trying to get ODBC within the voicemail module to work, but I am
getting the following error:

2010-01-16 14:32:32.584124 [CRIT] switch_core_sqldb.c:306 Failure! OBDC NOT
AVAILABLE!
2010-01-16 14:32:32.584124 [ERR] mod_voicemail.c:214 Error Opening DB


[root at e-d freeswitch]# isql my_odbc
+---------------------------------------+
| Connected!                            |
|                                       |
| sql-statement                         |
| help [tablename]                      |
| quit                                  |
|                                       |
+---------------------------------------+

In my voicemail.conf.xml, I have:
      <param name="odbc-dsn" value="my_odbc:root:123456"/>


Does anyone know what may be wrong with my config?

Thanks for your help.

woody
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From mike at jerris.com  Fri Jan 15 23:15:49 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 16 Jan 2010 02:15:49 -0500
Subject: [Freeswitch-users] ODBC Not Available!
In-Reply-To: <a44688a1001152236y57f8fe8bk5aced53cfd41e425@mail.gmail.com>
References: <a44688a1001152236y57f8fe8bk5aced53cfd41e425@mail.gmail.com>
Message-ID: <6461720E-23F5-4C6B-9D77-3042A853F14D@jerris.com>

It looks like freeswitch was built without odbc support.  Install odbc  
libs and dev packages, before you run configure to build with odbc  
support.

Mike

On Jan 16, 2010, at 1:36 AM, Woody Dickson <woodydickson at gmail.com>  
wrote:

> Hello,
>
> I am trying to get ODBC within the voicemail module to work, but I  
> am getting the following error:
>
> 2010-01-16 14:32:32.584124 [CRIT] switch_core_sqldb.c:306 Failure!  
> OBDC NOT AVAILABLE!
> 2010-01-16 14:32:32.584124 [ERR] mod_voicemail.c:214 Error Opening DB
>
>
> [root at e-d freeswitch]# isql my_odbc
> +---------------------------------------+
> | Connected!                            |
> |                                       |
> | sql-statement                         |
> | help [tablename]                      |
> | quit                                  |
> |                                       |
> +---------------------------------------+
>
> In my voicemail.conf.xml, I have:
>       <param name="odbc-dsn" value="my_odbc:root:123456"/>
>
>
> Does anyone know what may be wrong with my config?
>
> Thanks for your help.
>
> woody
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From sharad at coraltele.com  Fri Jan 15 23:27:40 2010
From: sharad at coraltele.com (Sharad)
Date: Fri, 15 Jan 2010 23:27:40 -0800 (PST)
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
Message-ID: <1263626860891-4403364.post@n2.nabble.com>


We also would like to work on the same. Plz let me know all the possibilities
-

1.  Email body to be sent as fax body.
2.  What about attached document to email ?
3.  Covering page ?

Plz let us know the possibilities, so that me a7 my team can start the work.

regards
sharad




Aza1 wrote:
> 
> Hi All
> 
> Has anyone worked on a email2fax script for mod_fax?
> If not how much would it cost for some genius here to quickly whip-up one?
> 
> Ideally both email2fax and fax2email should come standard with mod_fax or
> is
> this not possible?
> 
> Thanks
> Costa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Mod-Fax-tp4393083p4403364.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sharad at coraltele.com  Fri Jan 15 23:36:37 2010
From: sharad at coraltele.com (Sharad)
Date: Fri, 15 Jan 2010 23:36:37 -0800 (PST)
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <1263626860891-4403364.post@n2.nabble.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
	<1263626860891-4403364.post@n2.nabble.com>
Message-ID: <1263627397511-4403377.post@n2.nabble.com>


Just adding one more thing that first we are getting this done from a web
page...



Sharad wrote:
> 
> We also would like to work on the same. Plz let me know all the
> possibilities -
> 
> 1.  Email body to be sent as fax body.
> 2.  What about attached document to email ?
> 3.  Covering page ?
> 
> Plz let us know the possibilities, so that me a7 my team can start the
> work.
> 
> regards
> sharad
> 
> 
> 
> 
> Aza1 wrote:
>> 
>> Hi All
>> 
>> Has anyone worked on a email2fax script for mod_fax?
>> If not how much would it cost for some genius here to quickly whip-up
>> one?
>> 
>> Ideally both email2fax and fax2email should come standard with mod_fax or
>> is
>> this not possible?
>> 
>> Thanks
>> Costa
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 

-- 
View this message in context: http://n2.nabble.com/Mod-Fax-tp4393083p4403377.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From peder at networkoblivion.com  Sat Jan 16 05:08:35 2010
From: peder at networkoblivion.com (Peder)
Date: Sat, 16 Jan 2010 07:08:35 -0600
Subject: [Freeswitch-users] Polycom configuration for FreeSWITCH
In-Reply-To: <003701ca964b$3241b100$96c51300$@net>
References: <003701ca964b$3241b100$96c51300$@net>
Message-ID: <052c01ca96ad$02d7b4c0$08871e40$@com>

It is usually better if you ask a specific question, rather than just a
general "how do I configure it".  Freeswitch supports standard registration,
shared line, SCA ( thanks to a lot of work by the dev team recently), one
touch voicemail, mwi, conference, transfer, blind transfer, hold, DND, etc.


 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam
Ford
Sent: Friday, January 15, 2010 7:28 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Polycom configuration for FreeSWITCH

 

Does anyone who has successfully implemented Polycom IP301/501s, with
FreeSWITCH, have a configuration they can share with me as an example? I
have read the Polycom Admin Guide several times and understand what the
settings are/do, I am just not sure which FreeSWITCH supports, which it
doesn't, and which need special configuration to work with FreeSWITCH.

 

Thanks in advance,

-Adam

 

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From math.parent at gmail.com  Sat Jan 16 05:40:59 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Sat, 16 Jan 2010 14:40:59 +0100
Subject: [Freeswitch-users] Email2Pdf
In-Reply-To: <59daa2cd1001141603w462cdc84pb8fa721f5924cabc@mail.gmail.com>
References: <59daa2cd1001141603w462cdc84pb8fa721f5924cabc@mail.gmail.com>
Message-ID: <960738411001160540j197dfc5ay86f5e2e5990550be@mail.gmail.com>

On Fri, Jan 15, 2010 at 1:03 AM, Costa Zikalala
<costa.zikalala at gmail.com> wrote:
> Hi Mathieu
>
> Thank you for your very wonderful script.

You are talking about
http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/sathieu/email2pdf/

> I'm just trying understand it a bit and am not too good with Perl.
> Kindly confirm the folowing for me:
> - Only attachments and not the whole email is converted to PDF?

No, the message body should also be converted (in the script, this is
call $header ;).

> - Is the PDF output saved in the variable $file_out?
Yes, or to stdout, if $file_out is "-". more info with:
./email2pdf --man

> - Will you be completing your script as Email2Fax for Freeswitch?
The remaining thing will be added to FS wiki (don't have time now), as
this is site-specific:
- postfix config (see http://hylafax.sourceforge.net/howto/faxing.php#ss5.4)
- script email2fax:
 + guess From: fax number from the email
 + run email2pdf
 + convert to tiff
 + call txfax within FS

Cheers

Mathieu Parent

> Thanks again,
> Costa
>
>


From mailinglist at fribert.dk  Sat Jan 16 05:50:45 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Sat, 16 Jan 2010 14:50:45 +0100
Subject: [Freeswitch-users] How do I carry dial-in number to extension?
Message-ID: <4B51D245020000E1000003C0@mail.fribert.dk>

I have two DID's registered at my SIP.
This works nicely, I've created groups for holding the local extensions, and one of my phones can subscribe to two SIP accounts, so I can distinguish between if it's one or the other DID (it will show which SIP account is called in the display).
 
Problem is, another one of the phones can only subscribe to one SIP, and has no display (linksys sipura 901), any idea how to accomplish that I can distinnguish between the DID's on that one?
 
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From mike at jerris.com  Sat Jan 16 07:53:29 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 16 Jan 2010 10:53:29 -0500
Subject: [Freeswitch-users] How do I carry dial-in number to extension?
In-Reply-To: <4B51D245020000E1000003C0@mail.fribert.dk>
References: <4B51D245020000E1000003C0@mail.fribert.dk>
Message-ID: <54E291D4-6954-49DD-91FC-50F927C89532@jerris.com>



On Jan 16, 2010, at 8:50 AM, "mailinglist" <mailinglist at fribert.dk>  
wrote:

> I have two DID's registered at my SIP.

At your sip what?

> This works nicely, I've created groups for holding the local  
> extensions, and one of my phones can subscribe to two SIP accounts,  
> so I can distinguish between if it's one or the other DID (it will  
> show which SIP account is called in the display).
>

Are you talking about a phone registering or subscribing to freeswitch  
here?

> Problem is, another one of the phones can only subscribe to one SIP,  
> and has no display (linksys sipura 901), any idea how to accomplish  
> that I can distinnguish between the DID's on that one?
>

So if I understand you want to make a single line no display phone and  
send it multiple dids to it and be able to tell what did called?   
Really, the phone has no way to display this, but you could play a  
sound when they answer the phone.

Mike

>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From dome at tel.co.th  Sat Jan 16 09:17:02 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 17 Jan 2010 00:17:02 +0700
Subject: [Freeswitch-users] How to register from FS to Huawei SoftX3000
Message-ID: <8ccbff061001160917m22edfed8n291ea8a07518e4f7@mail.gmail.com>

Dear sir,
          I found some provider use Huawei SoftX3000 and can limit use
call from they softphone only. (use eyeball SDK).
They can limit some account can register and call by sip server like
an FS and Asterisk. but some account can't. (register and call by
softphone). and i don't know how they can do that.
So i try to use wireshark to debug sip headeder when use softphone
with both account type. it's nothing diferent.
          I want to use both account work by FS register to  Huawei
SoftX3000. Can someone help me. i can give you softphone and both
account type for test.


Best Regards.

Dome C.


From jmesquita at freeswitch.org  Sat Jan 16 09:17:09 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 16 Jan 2010 14:17:09 -0300
Subject: [Freeswitch-users] RTCP information
In-Reply-To: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE09AC568494@SBS2008SERVER.consiglia.local>
References: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE09AC568494@SBS2008SERVER.consiglia.local>
Message-ID: <ebbc2d551001160917n7b0278dcv4b4c0bf30dac3a91@mail.gmail.com>

No, there isn't..

Jo?o Mesquita


On Fri, Jan 15, 2010 at 2:02 PM, Jon Bruel <jbr at consiglia.dk> wrote:

>  In a real setup with 5-20 VoIP calls a day, every now and then there are
> some problems with sound quality, and I need some tools to investigate the
> cause.
>
> The phones support RTCP, and I would like to hear if I can get the FS to
> relay those packets to some kind of logger, including the signalling
> information?
>
> /Jon
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sat Jan 16 10:27:28 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 16 Jan 2010 13:27:28 -0500
Subject: [Freeswitch-users] How to register from FS to Huawei SoftX3000
In-Reply-To: <8ccbff061001160917m22edfed8n291ea8a07518e4f7@mail.gmail.com>
References: <8ccbff061001160917m22edfed8n291ea8a07518e4f7@mail.gmail.com>
Message-ID: <77F7CA9E-060E-4645-83D9-034B46C1C843@jerris.com>

Try changing the user agent, thats the only thing I can think they would be using.

Mike

On Jan 16, 2010, at 12:17 PM, Dome Charoenyost wrote:

> Dear sir,
>          I found some provider use Huawei SoftX3000 and can limit use
> call from they softphone only. (use eyeball SDK).
> They can limit some account can register and call by sip server like
> an FS and Asterisk. but some account can't. (register and call by
> softphone). and i don't know how they can do that.
> So i try to use wireshark to debug sip headeder when use softphone
> with both account type. it's nothing diferent.
>          I want to use both account work by FS register to  Huawei
> SoftX3000. Can someone help me. i can give you softphone and both
> account type for test.
> 



From lawwton at gmail.com  Sat Jan 16 11:58:44 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 16 Jan 2010 14:58:44 -0500
Subject: [Freeswitch-users] reloadxml question
Message-ID: <5fe6fa8f1001161158t54300ae5r62845f3baa1f97f5@mail.gmail.com>

All:

I am starting to play a bit with FS to get more familiar with it. I am
trying the following:

1- sofia status  // command for example to see the list of profiles.

2- I then went ahead and renamed the example.com.xml under
conf/directory/default to example.com.xml.orig

3- I ran then reloadxml using the fs_cli

4- sofia status still displays the same info as before for
example.conf as a gateway.

I then went ahead and stopped FW and re-started it. After I did this I
re-ran sofia status and this time the example.conf gateway that was
previously listed went away as it was supposed to.

Am I missing something? Is reloadxml the proper way to reload
configuration changes. If it is why is it not working for this case?

Thanks in advance,

Alfredo


From costa.zikalala at gmail.com  Sat Jan 16 11:58:47 2010
From: costa.zikalala at gmail.com (Costa Zikalala)
Date: Sat, 16 Jan 2010 21:58:47 +0200
Subject: [Freeswitch-users] Email2Pdf
In-Reply-To: <960738411001160540j197dfc5ay86f5e2e5990550be@mail.gmail.com>
References: <59daa2cd1001141603w462cdc84pb8fa721f5924cabc@mail.gmail.com> 
	<960738411001160540j197dfc5ay86f5e2e5990550be@mail.gmail.com>
Message-ID: <59daa2cd1001161158h51629821u80afa1168d7be796@mail.gmail.com>

This will be a very valuable addition to the Mod_Fax Wiki.

Thanks



2010/1/16 Mathieu Parent <math.parent at gmail.com>

> On Fri, Jan 15, 2010 at 1:03 AM, Costa Zikalala
> <costa.zikalala at gmail.com> wrote:
> > Hi Mathieu
> >
> > Thank you for your very wonderful script.
>
> You are talking about
> http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/sathieu/email2pdf/
>
> > I'm just trying understand it a bit and am not too good with Perl.
> > Kindly confirm the folowing for me:
> > - Only attachments and not the whole email is converted to PDF?
>
> No, the message body should also be converted (in the script, this is
> call $header ;).
>
> > - Is the PDF output saved in the variable $file_out?
> Yes, or to stdout, if $file_out is "-". more info with:
> ./email2pdf --man
>
> > - Will you be completing your script as Email2Fax for Freeswitch?
> The remaining thing will be added to FS wiki (don't have time now), as
> this is site-specific:
> - postfix config (see
> http://hylafax.sourceforge.net/howto/faxing.php#ss5.4)
> - script email2fax:
>  + guess From: fax number from the email
>  + run email2pdf
>  + convert to tiff
>  + call txfax within FS
>
> Cheers
>
> Mathieu Parent
>
> > Thanks again,
> > Costa
> >
> >
>
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From costa.zikalala at gmail.com  Sat Jan 16 12:05:22 2010
From: costa.zikalala at gmail.com (Costa Zikalala)
Date: Sat, 16 Jan 2010 22:05:22 +0200
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <776BA929-D7EC-4366-AF49-348F0E11488B@shaw.ca>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com> 
	<776BA929-D7EC-4366-AF49-348F0E11488B@shaw.ca>
Message-ID: <59daa2cd1001161205h3f6feeddla0be8e1fa9311666@mail.gmail.com>

Yes Steve, I'm already using that for fax2email.
I'm now trying to do things in the opposite direction.

*A realy great script by the way*

Thanks
Costa


2010/1/16 Steve Steffler <stevesteffler at shaw.ca>

>
> Costa,
>
> I wrote this script to handle fax2email (but not email2fax).  It uses
> variables you set in the dialplan in advance for the email address for that
> fax DID.
>
> http://steffler.info/wp-content/uploads/2009/06/process-rxfax.py
>
> Regards,
> Steve
>
> On Jan 14, 2010, at 7:54 AM, Costa Zikalala wrote:
>
> Hi All
>
> Has anyone worked on a email2fax script for mod_fax?
> If not how much would it cost for some genius here to quickly whip-up one?
>
> Ideally both email2fax and fax2email should come standard with mod_fax or
> is this not possible?
>
> Thanks
> Costa
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Sat Jan 16 12:16:18 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sat, 16 Jan 2010 15:16:18 -0500
Subject: [Freeswitch-users] reloadxml question
In-Reply-To: <5fe6fa8f1001161158t54300ae5r62845f3baa1f97f5@mail.gmail.com>
References: <5fe6fa8f1001161158t54300ae5r62845f3baa1f97f5@mail.gmail.com>
Message-ID: <3E8FB576-F49D-49A1-B3E2-8C6F5D3A6059@avgs.ca>

reloadxml preprocesses the xml config again, it does not tell modules  
to reload their configuration from that file.

In order to delete a gateway, you do: sofia profile <profilename>  
killgw <gwname>

If you make changes to a gateway and want to reload it, you first  
delete it with that command and then do:
	sofia profile <profilename> rescan

If you don't have any active calls on the box, restarting the profile  
will do the equivalent of killgw+rescan.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 16-Jan-10, at 2:58 PM, Alfredo Quiroga-Villamil wrote:

> All:
>
> I am starting to play a bit with FS to get more familiar with it. I am
> trying the following:
>
> 1- sofia status  // command for example to see the list of profiles.
>
> 2- I then went ahead and renamed the example.com.xml under
> conf/directory/default to example.com.xml.orig
>
> 3- I ran then reloadxml using the fs_cli
>
> 4- sofia status still displays the same info as before for
> example.conf as a gateway.
>
> I then went ahead and stopped FW and re-started it. After I did this I
> re-ran sofia status and this time the example.conf gateway that was
> previously listed went away as it was supposed to.
>
> Am I missing something? Is reloadxml the proper way to reload
> configuration changes. If it is why is it not working for this case?
>
> Thanks in advance,
>
> Alfredo
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mohamedezz.fci at gmail.com  Sat Jan 16 13:06:42 2010
From: mohamedezz.fci at gmail.com (Mohamed Hassan)
Date: Sat, 16 Jan 2010 23:06:42 +0200
Subject: [Freeswitch-users] No media after Originate
Message-ID: <f97004c31001161306o10517d1y9e0cf0229bce3589@mail.gmail.com>

Hi Freeswitch users
thank you in advance

as i am new to freeswitch iam facing a problem while writing a Callback service
i tried to bridge the two legs through one sip provider the call
estaplished successfully but without media passing
and after some searching i found that there is a bug with sofia
loopback so i shoulkd handle the two legs with different sip gateways
and after trying i found the same error


using cli :
originate {ignore_early_media=true}sofia/gateway/prov/00xxxxxxxxxxx
9999 XML callback

and here the 9999 context:
<context name="callback">
<extension name="9999">
<condition field="destination_number" expression="9999">
<action application="sleep" data="5000"/>
<action application="answer"/>
<action application="set" data="ringback=${us-ring}"/>
<action application="bridge"
data="[ignore_early_media=false]sofia/gateway/prov/00xxxxxxxxxx"/>
</condition>
</extension>
</context>


From mailinglist at fribert.dk  Sat Jan 16 13:33:30 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Sat, 16 Jan 2010 22:33:30 +0100
Subject: [Freeswitch-users] Svar: Re: How do I carry dial-in number to	
	extension?
In-Reply-To: <54E291D4-6954-49DD-91FC-50F927C89532@jerris.com>
References: <4B51D245020000E1000003C0@mail.fribert.dk>
	<54E291D4-6954-49DD-91FC-50F927C89532@jerris.com>
Message-ID: <4B523EBA020000E1000003C5@mail.fribert.dk>

>> I have two DID's registered at my SIP.
> At your sip what?
 
provider
 
>> This works nicely, I've created groups for holding the local  
>> extensions, and one of my phones can subscribe to two SIP accounts,  
>> so I can distinguish between if it's one or the other DID (it will  
>> show which SIP account is called in the display).
> Are you talking about a phone registering or subscribing to freeswitch  
> here?
 
FS is subscribing to two accounts at my provider.
I have a couple of phones subscribing to the FS.
Two phones has the ability of several sipaccounts, the last has one SIP subscription.
 
>> Problem is, another one of the phones can only subscribe to one SIP,  
>> and has no display (linksys sipura 901), any idea how to accomplish  
>> that I can distinnguish between the DID's on that one?
> So if I understand you want to make a single line no display phone and  
> send it multiple dids to it and be able to tell what did called?   
> Really, the phone has no way to display this, but you could play a  
> sound when they answer the phone.
 
Ahh, yes that sounds interesting, I thought of something like 'distinctive ringing' the phone seams to support that, but I have no idea how to use that.
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From lawwton at gmail.com  Sat Jan 16 13:39:41 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 16 Jan 2010 16:39:41 -0500
Subject: [Freeswitch-users] reloadxml question
In-Reply-To: <3E8FB576-F49D-49A1-B3E2-8C6F5D3A6059@avgs.ca>
References: <5fe6fa8f1001161158t54300ae5r62845f3baa1f97f5@mail.gmail.com>
	<3E8FB576-F49D-49A1-B3E2-8C6F5D3A6059@avgs.ca>
Message-ID: <5fe6fa8f1001161339m38c4c44ej71d7a30be4e3fb71@mail.gmail.com>

Thanks Mathieu, appreciate the help.

Alfredo

On Sat, Jan 16, 2010 at 3:16 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> reloadxml preprocesses the xml config again, it does not tell modules
> to reload their configuration from that file.
>
> In order to delete a gateway, you do: sofia profile <profilename>
> killgw <gwname>
>
> If you make changes to a gateway and want to reload it, you first
> delete it with that command and then do:
> ? ? ? ?sofia profile <profilename> rescan
>
> If you don't have any active calls on the box, restarting the profile
> will do the equivalent of killgw+rescan.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 16-Jan-10, at 2:58 PM, Alfredo Quiroga-Villamil wrote:
>
>> All:
>>
>> I am starting to play a bit with FS to get more familiar with it. I am
>> trying the following:
>>
>> 1- sofia status ?// command for example to see the list of profiles.
>>
>> 2- I then went ahead and renamed the example.com.xml under
>> conf/directory/default to example.com.xml.orig
>>
>> 3- I ran then reloadxml using the fs_cli
>>
>> 4- sofia status still displays the same info as before for
>> example.conf as a gateway.
>>
>> I then went ahead and stopped FW and re-started it. After I did this I
>> re-ran sofia status and this time the example.conf gateway that was
>> previously listed went away as it was supposed to.
>>
>> Am I missing something? Is reloadxml the proper way to reload
>> configuration changes. If it is why is it not working for this case?
>>
>> Thanks in advance,
>>
>> Alfredo
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Sat Jan 16 13:44:03 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 16 Jan 2010 15:44:03 -0600
Subject: [Freeswitch-users] No media after Originate
In-Reply-To: <f97004c31001161306o10517d1y9e0cf0229bce3589@mail.gmail.com>
References: <f97004c31001161306o10517d1y9e0cf0229bce3589@mail.gmail.com>
Message-ID: <3081C466-F735-47F1-BAEC-2B497F8F08D8@freeswitch.org>

Is nat involved?

/b

On Jan 16, 2010, at 3:06 PM, Mohamed Hassan wrote:

> estaplished successfully but without media passing



From brian at freeswitch.org  Sat Jan 16 13:46:00 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 16 Jan 2010 15:46:00 -0600
Subject: [Freeswitch-users] reloadxml question
In-Reply-To: <3E8FB576-F49D-49A1-B3E2-8C6F5D3A6059@avgs.ca>
References: <5fe6fa8f1001161158t54300ae5r62845f3baa1f97f5@mail.gmail.com>
	<3E8FB576-F49D-49A1-B3E2-8C6F5D3A6059@avgs.ca>
Message-ID: <E9FF7C20-8D77-4DA0-9B2D-DD75091C1457@freeswitch.org>

Also noted killing a gateway and reinstalling it doesn't touch currently active calls that happened to have used that gateway.
/b

On Jan 16, 2010, at 2:16 PM, Mathieu Rene wrote:

> If you don't have any active calls on the box, restarting the profile  
> will do the equivalent of killgw+rescan.



From mike at jerris.com  Sat Jan 16 14:24:49 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 16 Jan 2010 17:24:49 -0500
Subject: [Freeswitch-users] Svar: Re: How do I carry dial-in number to
	extension?
In-Reply-To: <4B523EBA020000E1000003C5@mail.fribert.dk>
References: <4B51D245020000E1000003C0@mail.fribert.dk>
	<54E291D4-6954-49DD-91FC-50F927C89532@jerris.com>
	<4B523EBA020000E1000003C5@mail.fribert.dk>
Message-ID: <F65EBF8D-0194-47BF-897F-FF8C2CE68C98@jerris.com>


On Jan 16, 2010, at 4:33 PM, mailinglist wrote:

> >> I have two DID's registered at my SIP.
> > At your sip what?
>  
> provider
>  
> >> This works nicely, I've created groups for holding the local  
> >> extensions, and one of my phones can subscribe to two SIP accounts,  
> >> so I can distinguish between if it's one or the other DID (it will  
> >> show which SIP account is called in the display).
> > Are you talking about a phone registering or subscribing to freeswitch  
> > here?
>  
> FS is subscribing to two accounts at my provider.
> I have a couple of phones subscribing to the FS.
> Two phones has the ability of several sipaccounts, the last has one SIP subscription.

I think you are trying to say sip registration, not subscription.  subscription is totally different but still a concept in sip.

> >> Problem is, another one of the phones can only subscribe to one SIP,  
> >> and has no display (linksys sipura 901), any idea how to accomplish  
> >> that I can distinnguish between the DID's on that one?
> > So if I understand you want to make a single line no display phone and  
> > send it multiple dids to it and be able to tell what did called?   
> > Really, the phone has no way to display this, but you could play a  
> > sound when they answer the phone.
>  
> Ahh, yes that sounds interesting, I thought of something like 'distinctive ringing' the phone seams to support that, but I have no idea how to use that.

Its quite possible this phone has other features for this, if so, read up on what they are and how they work and provide back some details and we can help with how to do this in freeswitch.

Mike

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From mohamedezz.fci at gmail.com  Sat Jan 16 15:22:54 2010
From: mohamedezz.fci at gmail.com (Mohamed Hassan)
Date: Sun, 17 Jan 2010 01:22:54 +0200
Subject: [Freeswitch-users] No media after Originate
In-Reply-To: <3081C466-F735-47F1-BAEC-2B497F8F08D8@freeswitch.org>
References: <f97004c31001161306o10517d1y9e0cf0229bce3589@mail.gmail.com>
	<3081C466-F735-47F1-BAEC-2B497F8F08D8@freeswitch.org>
Message-ID: <f97004c31001161522g65a0da7agec1eabffa321692d@mail.gmail.com>

There is no nat my server has public ip and not nated
and my sip provider too  as i can make regular calls through the same
provider without originate

On Sat, Jan 16, 2010 at 11:44 PM, Brian West <brian at freeswitch.org> wrote:
> Is nat involved?
>
> /b
>
> On Jan 16, 2010, at 3:06 PM, Mohamed Hassan wrote:
>
>> estaplished successfully but without media passing
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From chris.chen2004 at gmail.com  Sat Jan 16 17:37:46 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Sat, 16 Jan 2010 20:37:46 -0500
Subject: [Freeswitch-users] Originate_timeout not working with latest SVN
	trunk
Message-ID: <507898381001161737x17435f9ds33d9585aa83418bc@mail.gmail.com>

Hi there, if anybody has the similar issue like me regarding the
originate_timeout variable.

I have the following dialplan:

<extension name="Support_Extension">
      <condition field="destination_number" expression="^(support)$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="bind_meta_app" data="1 b s
execute_extension::dx XML features_nln"/>
        <action application="bind_meta_app" data="2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftim$
        <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features_nln"/>
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="call_timeout=10"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="db"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
        <action application="db"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
        <action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}$
        <action application="db"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
        <action application="bridge" data="user/1000 at My_FS_Public_IP
,user/1001 at My_FS_Public_IP"/>
        <action application="set" data="originate_timeout=15"/>
        <action application="bridge"
data="sofia/gateway/unlimitel/6475552222"/>
        <action application="bridge"
data="sofia/gateway/unlimitel/4165552222"/>
         <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="voicemail" data="default_nln ${domain_name}
1001"/>
      </condition>
    </extension>


in the past the call_timeout and originate_timeout all working as expected,
but with latest SVN trunk since 16318, the originate_timeout not working,
once the call forwarded to the cell phone by
  <action application="bridge" data="sofia/gateway/unlimitel/6475552222"/>
        <action application="bridge"
data="sofia/gateway/unlimitel/4165552222"/>
it never timed out
so the voicemail never got executed.
 Any helps are greatly appreciated
Thanks,

Chris
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From lawwton at gmail.com  Sat Jan 16 17:38:22 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 16 Jan 2010 20:38:22 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
Message-ID: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>

All:

Any ideas why there is an alias here? What does that exactly mean? Do
I need to have that? How do I remove that?

freeswitch at internal> sofia status
                     Name          Type
Data      State
=================================================================================================
                 external       profile
sip:mod_sofia at 192.168.1.148:5080      RUNNING (0)
                 internal       profile
sip:mod_sofia at 192.168.1.148:5060      RUNNING (0)
            192.168.1.148         alias
internal      ALIASED
=================================================================================================
2 profiles 1 alias

Lots of questions there all trying to figure out why it's showing up there.

Thanks in advance,

Alfredo Q-V


From lawwton at gmail.com  Sat Jan 16 17:49:26 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 16 Jan 2010 20:49:26 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
Message-ID: <5fe6fa8f1001161749m1c60fb77i579c7e7ecfe98e98@mail.gmail.com>

All:

I was able to eliminate the alias by setting the parameter in the
internal.xml file under sip_profiles to false. However to actually
load the change I had to restart FS. Is there a way to reload this
kind of change without restarting FS (not dropping calls).

Thanks in advance,

Alfredo

On Sat, Jan 16, 2010 at 8:38 PM, Alfredo Quiroga-Villamil
<lawwton at gmail.com> wrote:
> All:
>
> Any ideas why there is an alias here? What does that exactly mean? Do
> I need to have that? How do I remove that?
>
> freeswitch at internal> sofia status
> ? ? ? ? ? ? ? ? ? ? Name ? ? ? ? ?Type
> Data ? ? ?State
> =================================================================================================
> ? ? ? ? ? ? ? ? external ? ? ? profile
> sip:mod_sofia at 192.168.1.148:5080 ? ? ?RUNNING (0)
> ? ? ? ? ? ? ? ? internal ? ? ? profile
> sip:mod_sofia at 192.168.1.148:5060 ? ? ?RUNNING (0)
> ? ? ? ? ? ?192.168.1.148 ? ? ? ? alias
> internal ? ? ?ALIASED
> =================================================================================================
> 2 profiles 1 alias
>
> Lots of questions there all trying to figure out why it's showing up there.
>
> Thanks in advance,
>
> Alfredo Q-V
>

From mike at jerris.com  Sat Jan 16 17:58:40 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 16 Jan 2010 20:58:40 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
Message-ID: <9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>

it lets you use your ip as the profile name as well.  There are some things in the default configs that take advantage of and assume that the profile name is the domain name.  In the case of the default configs, we use the detected ip address for this.  If you remove it, things will probably break unless you have devices that all work right, dns setup right, and all your devices dns.  

Mike

On Jan 16, 2010, at 8:38 PM, Alfredo Quiroga-Villamil wrote:

> All:
> 
> Any ideas why there is an alias here? What does that exactly mean? Do
> I need to have that? How do I remove that?
> 
> freeswitch at internal> sofia status
>                     Name          Type
> Data      State
> =================================================================================================
>                 external       profile
> sip:mod_sofia at 192.168.1.148:5080      RUNNING (0)
>                 internal       profile
> sip:mod_sofia at 192.168.1.148:5060      RUNNING (0)
>            192.168.1.148         alias
> internal      ALIASED
> =================================================================================================
> 2 profiles 1 alias
> 
> Lots of questions there all trying to figure out why it's showing up there.
> 
> Thanks in advance,
> 
> Alfredo Q-V



From lawwton at gmail.com  Sat Jan 16 18:09:55 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 16 Jan 2010 21:09:55 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>
Message-ID: <5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>

Appreciate the response Mike.

That's a little different than everything I've seen in other systems.
I don't want to be forced to display 3 different profiles when in
reality one is just an alias name. It says alias and all; but I find
that a bit repetitive, specially if I remove it and it breaks things.

When a change is made on one of the sip_profiles, take internal.xml
for example where I changed a parameter from true to false. What
command needs to be ran to reload the changes (non-disruptive).

Thanks in advance,

Alfredo

On Sat, Jan 16, 2010 at 8:58 PM, Michael Jerris <mike at jerris.com> wrote:
> it lets you use your ip as the profile name as well. ?There are some things in the default configs that take advantage of and assume that the profile name is the domain name. ?In the case of the default configs, we use the detected ip address for this. ?If you remove it, things will probably break unless you have devices that all work right, dns setup right, and all your devices dns.
>
> Mike
>
> On Jan 16, 2010, at 8:38 PM, Alfredo Quiroga-Villamil wrote:
>
>> All:
>>
>> Any ideas why there is an alias here? What does that exactly mean? Do
>> I need to have that? How do I remove that?
>>
>> freeswitch at internal> sofia status
>> ? ? ? ? ? ? ? ? ? ? Name ? ? ? ? ?Type
>> Data ? ? ?State
>> =================================================================================================
>> ? ? ? ? ? ? ? ? external ? ? ? profile
>> sip:mod_sofia at 192.168.1.148:5080 ? ? ?RUNNING (0)
>> ? ? ? ? ? ? ? ? internal ? ? ? profile
>> sip:mod_sofia at 192.168.1.148:5060 ? ? ?RUNNING (0)
>> ? ? ? ? ? ?192.168.1.148 ? ? ? ? alias
>> internal ? ? ?ALIASED
>> =================================================================================================
>> 2 profiles 1 alias
>>
>> Lots of questions there all trying to figure out why it's showing up there.
>>
>> Thanks in advance,
>>
>> Alfredo Q-V
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mastermind202 at gmail.com  Sat Jan 16 18:39:26 2010
From: mastermind202 at gmail.com (mm_202)
Date: Sat, 16 Jan 2010 21:39:26 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>
	<5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>
Message-ID: <63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>

On Sat, Jan 16, 2010 at 9:09 PM, Alfredo Quiroga-Villamil <lawwton at gmail.com
> wrote:

> Appreciate the response Mike.
>
> That's a little different than everything I've seen in other systems.
> I don't want to be forced to display 3 different profiles when in
> reality one is just an alias name. It says alias and all; but I find
> that a bit repetitive, specially if I remove it and it breaks things.
>
> When a change is made on one of the sip_profiles, take internal.xml
> for example where I changed a parameter from true to false. What
> command needs to be ran to reload the changes (non-disruptive).
>
> Thanks in advance,
>
> Alfredo
>
> On Sat, Jan 16, 2010 at 8:58 PM, Michael Jerris <mike at jerris.com> wrote:
> > it lets you use your ip as the profile name as well.  There are some
> things in the default configs that take advantage of and assume that the
> profile name is the domain name.  In the case of the default configs, we use
> the detected ip address for this.  If you remove it, things will probably
> break unless you have devices that all work right, dns setup right, and all
> your devices dns.
> >
> > Mike
> >
> > On Jan 16, 2010, at 8:38 PM, Alfredo Quiroga-Villamil wrote:
> >
> >> All:
> >>
> >> Any ideas why there is an alias here? What does that exactly mean? Do
> >> I need to have that? How do I remove that?
> >>
> >> freeswitch at internal> sofia status
> >>                     Name          Type
> >> Data      State
> >>
> =================================================================================================
> >>                 external       profile
> >> sip:mod_sofia at 192.168.1.148:5080      RUNNING (0)
> >>                 internal       profile
> >> sip:mod_sofia at 192.168.1.148:5060      RUNNING (0)
> >>            192.168.1.148         alias
> >> internal      ALIASED
> >>
> =================================================================================================
> >> 2 profiles 1 alias
> >>
> >> Lots of questions there all trying to figure out why it's showing up
> there.
> >>
> >> Thanks in advance,
> >>
> >> Alfredo Q-V
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


Alfredo,
Use 'reloadxml' and then for mod_sofia to actually use the changes, you'll
have to run 'sofia profile internal restart', but that will drop any calls
that are on that profile.

-- mm_202.
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From lawwton at gmail.com  Sat Jan 16 19:00:08 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 16 Jan 2010 22:00:08 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>
	<5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>
	<63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>
Message-ID: <5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>

mm_202:

Appreciate the response. So, essentially what that means is that there
is no way to reload changes unless we do a restart which will drop
calls? I apologize before hand for saying this; but that's really
really bad when you have a production server to which changes are made
sometimes through out the day, new turnups/disconnects, etc...

I actually just did the following test:

a) Made sure I removed the alias from my internal profile. Verified it
was gone with "sofia status"

b) Edit the internal.xml file again and reset the flag to true for the alias.

c) Ran: sofia profile internal rescan  reloadxml

I received the expected message about the alias being added:

freeswitch at internal> 2010-01-16 22:17:49.778272 [INFO]
switch_time.c:812 Timezone reloaded 530 definitions
2010-01-16 22:17:49.778272 [DEBUG] sofia.c:2252 Duration out of
bounds, using default of 2000!
2010-01-16 22:17:49.778272 [NOTICE] sofia.c:1804 Adding Alias
[192.168.1.148] for profile [internal]

d) I then edited the file again and set the flag again to false

e) I then re-ran: sofia profile internal rescan  reloadxml again to
find out that the alias is not removed this time around. In other
words, the alias is only added with a re-scan after the flag is set to
true. It's not removed with rescan if the flag is set to false.

So, two big concerns now:

1- If there is no way to reload things dynamically without disrupting
service, is this by design? Specially the sip_profiles part, that's
really important.

2- Why are the a-e) steps above partially working, it might seem at
first glance that sofia profile internal rescan  reloadxml should do
what it says, to re-scan for new changes and load them for that
profile. Is this a bug or this is by design?

All:

Feel free to keep me honest here and let me know if I am doing
something wrong since I just started playing with FS today.

Appreciate the help,

Alfredo

That command

On Sat, Jan 16, 2010 at 9:39 PM, mm_202 <mastermind202 at gmail.com> wrote:
>
> On Sat, Jan 16, 2010 at 9:09 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> Appreciate the response Mike.
>>
>> That's a little different than everything I've seen in other systems.
>> I don't want to be forced to display 3 different profiles when in
>> reality one is just an alias name. It says alias and all; but I find
>> that a bit repetitive, specially if I remove it and it breaks things.
>>
>> When a change is made on one of the sip_profiles, take internal.xml
>> for example where I changed a parameter from true to false. What
>> command needs to be ran to reload the changes (non-disruptive).
>>
>> Thanks in advance,
>>
>> Alfredo
>>
>> On Sat, Jan 16, 2010 at 8:58 PM, Michael Jerris <mike at jerris.com> wrote:
>> > it lets you use your ip as the profile name as well. ?There are some
>> > things in the default configs that take advantage of and assume that the
>> > profile name is the domain name. ?In the case of the default configs, we use
>> > the detected ip address for this. ?If you remove it, things will probably
>> > break unless you have devices that all work right, dns setup right, and all
>> > your devices dns.
>> >
>> > Mike
>> >
>> > On Jan 16, 2010, at 8:38 PM, Alfredo Quiroga-Villamil wrote:
>> >
>> >> All:
>> >>
>> >> Any ideas why there is an alias here? What does that exactly mean? Do
>> >> I need to have that? How do I remove that?
>> >>
>> >> freeswitch at internal> sofia status
>> >> ? ? ? ? ? ? ? ? ? ? Name ? ? ? ? ?Type
>> >> Data ? ? ?State
>> >>
>> >> =================================================================================================
>> >> ? ? ? ? ? ? ? ? external ? ? ? profile
>> >> sip:mod_sofia at 192.168.1.148:5080 ? ? ?RUNNING (0)
>> >> ? ? ? ? ? ? ? ? internal ? ? ? profile
>> >> sip:mod_sofia at 192.168.1.148:5060 ? ? ?RUNNING (0)
>> >> ? ? ? ? ? ?192.168.1.148 ? ? ? ? alias
>> >> internal ? ? ?ALIASED
>> >>
>> >> =================================================================================================
>> >> 2 profiles 1 alias
>> >>
>> >> Lots of questions there all trying to figure out why it's showing up
>> >> there.
>> >>
>> >> Thanks in advance,
>> >>
>> >> Alfredo Q-V
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> Alfredo,
> Use 'reloadxml' and then for mod_sofia to actually use the changes, you'll
> have to run 'sofia profile internal restart', but that will drop any calls
> that are on that profile.
>
> -- mm_202.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mastermind202 at gmail.com  Sat Jan 16 19:35:28 2010
From: mastermind202 at gmail.com (mm_202)
Date: Sat, 16 Jan 2010 22:35:28 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>
	<5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>
	<63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>
	<5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>
Message-ID: <63de75711001161935w2d1e204ek81ebe1143afb1497@mail.gmail.com>

On Sat, Jan 16, 2010 at 10:00 PM, Alfredo Quiroga-Villamil <
lawwton at gmail.com> wrote:

> mm_202:
>
> Appreciate the response. So, essentially what that means is that there
> is no way to reload changes unless we do a restart which will drop
> calls? I apologize before hand for saying this; but that's really
> really bad when you have a production server to which changes are made
> sometimes through out the day, new turnups/disconnects, etc...
>
> I actually just did the following test:
>
> a) Made sure I removed the alias from my internal profile. Verified it
> was gone with "sofia status"
>
> b) Edit the internal.xml file again and reset the flag to true for the
> alias.
>
> c) Ran: sofia profile internal rescan  reloadxml
>
> I received the expected message about the alias being added:
>
> freeswitch at internal> 2010-01-16 22:17:49.778272 [INFO]
> switch_time.c:812 Timezone reloaded 530 definitions
> 2010-01-16 22:17:49.778272 [DEBUG] sofia.c:2252 Duration out of
> bounds, using default of 2000!
> 2010-01-16 22:17:49.778272 [NOTICE] sofia.c:1804 Adding Alias
> [192.168.1.148] for profile [internal]
>
> d) I then edited the file again and set the flag again to false
>
> e) I then re-ran: sofia profile internal rescan  reloadxml again to
> find out that the alias is not removed this time around. In other
> words, the alias is only added with a re-scan after the flag is set to
> true. It's not removed with rescan if the flag is set to false.
>
> So, two big concerns now:
>
> 1- If there is no way to reload things dynamically without disrupting
> service, is this by design? Specially the sip_profiles part, that's
> really important.
>
> 2- Why are the a-e) steps above partially working, it might seem at
> first glance that sofia profile internal rescan  reloadxml should do
> what it says, to re-scan for new changes and load them for that
> profile. Is this a bug or this is by design?
>
> All:
>
> Feel free to keep me honest here and let me know if I am doing
> something wrong since I just started playing with FS today.
>
> Appreciate the help,
>
> Alfredo
>
> That command
>
> On Sat, Jan 16, 2010 at 9:39 PM, mm_202 <mastermind202 at gmail.com> wrote:
> >
> > On Sat, Jan 16, 2010 at 9:09 PM, Alfredo Quiroga-Villamil
> > <lawwton at gmail.com> wrote:
> >>
> >> Appreciate the response Mike.
> >>
> >> That's a little different than everything I've seen in other systems.
> >> I don't want to be forced to display 3 different profiles when in
> >> reality one is just an alias name. It says alias and all; but I find
> >> that a bit repetitive, specially if I remove it and it breaks things.
> >>
> >> When a change is made on one of the sip_profiles, take internal.xml
> >> for example where I changed a parameter from true to false. What
> >> command needs to be ran to reload the changes (non-disruptive).
> >>
> >> Thanks in advance,
> >>
> >> Alfredo
> >>
> >> On Sat, Jan 16, 2010 at 8:58 PM, Michael Jerris <mike at jerris.com>
> wrote:
> >> > it lets you use your ip as the profile name as well.  There are some
> >> > things in the default configs that take advantage of and assume that
> the
> >> > profile name is the domain name.  In the case of the default configs,
> we use
> >> > the detected ip address for this.  If you remove it, things will
> probably
> >> > break unless you have devices that all work right, dns setup right,
> and all
> >> > your devices dns.
> >> >
> >> > Mike
> >> >
> >> > On Jan 16, 2010, at 8:38 PM, Alfredo Quiroga-Villamil wrote:
> >> >
> >> >> All:
> >> >>
> >> >> Any ideas why there is an alias here? What does that exactly mean? Do
> >> >> I need to have that? How do I remove that?
> >> >>
> >> >> freeswitch at internal> sofia status
> >> >>                     Name          Type
> >> >> Data      State
> >> >>
> >> >>
> =================================================================================================
> >> >>                 external       profile
> >> >> sip:mod_sofia at 192.168.1.148:5080      RUNNING (0)
> >> >>                 internal       profile
> >> >> sip:mod_sofia at 192.168.1.148:5060      RUNNING (0)
> >> >>            192.168.1.148         alias
> >> >> internal      ALIASED
> >> >>
> >> >>
> =================================================================================================
> >> >> 2 profiles 1 alias
> >> >>
> >> >> Lots of questions there all trying to figure out why it's showing up
> >> >> there.
> >> >>
> >> >> Thanks in advance,
> >> >>
> >> >> Alfredo Q-V
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > Alfredo,
> > Use 'reloadxml' and then for mod_sofia to actually use the changes,
> you'll
> > have to run 'sofia profile internal restart', but that will drop any
> calls
> > that are on that profile.
> >
> > -- mm_202.
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

Alfredo,

In a production system, you *shouldnt* have to change the main profiles at
all.
When adding / removing gateways, you can use 'sofia profile [profilename]
rescan',
that will not drop calls, only if you run 'reload'.

As far as why the rescan adds the alias but doesnt remove it, I would say
that is by design.
That way you cant 'break' anything if your dialplan is using that alias.
But someone more
experienced than me may have a better / more accurate answer.

-- mm_202.
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From dujinfang at gmail.com  Sat Jan 16 19:37:31 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 17 Jan 2010 11:37:31 +0800
Subject: [Freeswitch-users] Questions about mod_fifo
In-Reply-To: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
References: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
Message-ID: <23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>

2010/1/15, Jingwei Yang <jingwei.yang at gmail.com>:
> Hi Guys,
>
> I'm implementing an ACD system using ESL and mod_fifo. Based on what Anthony
> suggested in this post:
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html
>
> *You can make an event socket application that listens for FIFO events and
> keeps track of what FIFOs are currently busy and when there are people
> waiting you can have that script generate a call to a group of SIP phones so
> when the first one answers, it sends them in as an agent where they can
> field the calls.
> *
>
> 1. How should I handle the concurrent issues? If I bridge a user to two
> agents and both of them answers, how does FS take care of this situation?
> Will a slower agent get a busy tone automatically?
>

I think it just follow the standard originate dialstring rules.

> 2. If the socket application is brought up after some users have called in,
> what command should I use to check the busy queues? fifo list?
>
Yes. Perhaps you can also check the fifo db, either sqlite or ODBC.

> 3. Am I using fifo list and fifo count correctly?
>
> here's the testing dialplan:
>
>     <extension name="in_fifo">
>       <condition field="destination_number" expression="^779$">
>         <action application="set" data="fifo_music=$${hold_music}"/>
>         <action application="answer"/>
>         <action application="fifo" data="*myq *in"/>
>       </condition>
>     </extension>
>
> when a call comes in and gets queued, these are the results of some commands
> I tried.
>
> freeswitch at localhost.localdomain> fifo list
> API CALL [fifo(list)] output:
> <fifo_report>
>   <fifo name="cool_fifo at 192.168.1.104" consumer_count="0" caller_count="0"
> waiting_count="0" importance="0">
>     <callers></callers>
>     <consumers></consumers>
>   </fifo>
> </fifo_report>
>
>
> freeswitch at localhost.localdomain> fifo list myq
> API CALL [fifo(list myq)] output:
> <fifo_report></fifo_report>
>
>
> freeswitch at localhost.localdomain> fifo count myq
> API CALL [fifo(count myq)] output:
> none
>
> It seems *myq* doesn't get created at all? Please enlighten.
>
> Thanks and best regards,
> -Jingwei
>
AFAIK, thant means the channel didn't queued in. Did you see any error
logs? I think you need to remove the stars in   <action
application="fifo" data="*myq *in"/>.


From lawwton at gmail.com  Sat Jan 16 21:00:21 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sun, 17 Jan 2010 00:00:21 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <63de75711001161935w2d1e204ek81ebe1143afb1497@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>
	<5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>
	<63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>
	<5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>
	<63de75711001161935w2d1e204ek81ebe1143afb1497@mail.gmail.com>
Message-ID: <5fe6fa8f1001162100n699077faw8ac1e8cef4e4ea5@mail.gmail.com>

mm_202:

Thanks for the reply.

I don't agree with your statement: "In a production system, you
*shouldnt* have to change the main profiles at all."

I think it's always good to be able to reload things dynamically in a
non-disruptive way.

The alias is a bit unusual. To be honest I had never seen that before
in any of the Telecommunication Systems I've worked on. Not that is
not a good idea, just one that at first glance doesn't seem too clear
for some reason. Specially when you see two profiles both representing
the same object listed under "sofia status". Being forced to have the
alias doesn't seem like an appealing option, should be optional I
think. I would like if possible a more detailed explanation on what
would break if not present.

In any case, I am sure there is a reason for these things. I am trying
to understand how they all work together. Things are not as apparent
in FS at first glance when compared to other systems; but the platform
seems to be built with the purpose of offering a lot of flexibility
which is really good.

Appreciate the feedback.

Alfredo Q-V

Appreciate the help,

Alfredo



On Sat, Jan 16, 2010 at 10:35 PM, mm_202 <mastermind202 at gmail.com> wrote:
>
> On Sat, Jan 16, 2010 at 10:00 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> mm_202:
>>
>> Appreciate the response. So, essentially what that means is that there
>> is no way to reload changes unless we do a restart which will drop
>> calls? I apologize before hand for saying this; but that's really
>> really bad when you have a production server to which changes are made
>> sometimes through out the day, new turnups/disconnects, etc...
>>
>> I actually just did the following test:
>>
>> a) Made sure I removed the alias from my internal profile. Verified it
>> was gone with "sofia status"
>>
>> b) Edit the internal.xml file again and reset the flag to true for the
>> alias.
>>
>> c) Ran: sofia profile internal rescan ?reloadxml
>>
>> I received the expected message about the alias being added:
>>
>> freeswitch at internal> 2010-01-16 22:17:49.778272 [INFO]
>> switch_time.c:812 Timezone reloaded 530 definitions
>> 2010-01-16 22:17:49.778272 [DEBUG] sofia.c:2252 Duration out of
>> bounds, using default of 2000!
>> 2010-01-16 22:17:49.778272 [NOTICE] sofia.c:1804 Adding Alias
>> [192.168.1.148] for profile [internal]
>>
>> d) I then edited the file again and set the flag again to false
>>
>> e) I then re-ran: sofia profile internal rescan ?reloadxml again to
>> find out that the alias is not removed this time around. In other
>> words, the alias is only added with a re-scan after the flag is set to
>> true. It's not removed with rescan if the flag is set to false.
>>
>> So, two big concerns now:
>>
>> 1- If there is no way to reload things dynamically without disrupting
>> service, is this by design? Specially the sip_profiles part, that's
>> really important.
>>
>> 2- Why are the a-e) steps above partially working, it might seem at
>> first glance that sofia profile internal rescan ?reloadxml should do
>> what it says, to re-scan for new changes and load them for that
>> profile. Is this a bug or this is by design?
>>
>> All:
>>
>> Feel free to keep me honest here and let me know if I am doing
>> something wrong since I just started playing with FS today.
>>
>> Appreciate the help,
>>
>> Alfredo
>>
>> That command
>>
>> On Sat, Jan 16, 2010 at 9:39 PM, mm_202 <mastermind202 at gmail.com> wrote:
>> >
>> > On Sat, Jan 16, 2010 at 9:09 PM, Alfredo Quiroga-Villamil
>> > <lawwton at gmail.com> wrote:
>> >>
>> >> Appreciate the response Mike.
>> >>
>> >> That's a little different than everything I've seen in other systems.
>> >> I don't want to be forced to display 3 different profiles when in
>> >> reality one is just an alias name. It says alias and all; but I find
>> >> that a bit repetitive, specially if I remove it and it breaks things.
>> >>
>> >> When a change is made on one of the sip_profiles, take internal.xml
>> >> for example where I changed a parameter from true to false. What
>> >> command needs to be ran to reload the changes (non-disruptive).
>> >>
>> >> Thanks in advance,
>> >>
>> >> Alfredo
>> >>
>> >> On Sat, Jan 16, 2010 at 8:58 PM, Michael Jerris <mike at jerris.com>
>> >> wrote:
>> >> > it lets you use your ip as the profile name as well. ?There are some
>> >> > things in the default configs that take advantage of and assume that
>> >> > the
>> >> > profile name is the domain name. ?In the case of the default configs,
>> >> > we use
>> >> > the detected ip address for this. ?If you remove it, things will
>> >> > probably
>> >> > break unless you have devices that all work right, dns setup right,
>> >> > and all
>> >> > your devices dns.
>> >> >
>> >> > Mike
>> >> >
>> >> > On Jan 16, 2010, at 8:38 PM, Alfredo Quiroga-Villamil wrote:
>> >> >
>> >> >> All:
>> >> >>
>> >> >> Any ideas why there is an alias here? What does that exactly mean?
>> >> >> Do
>> >> >> I need to have that? How do I remove that?
>> >> >>
>> >> >> freeswitch at internal> sofia status
>> >> >> ? ? ? ? ? ? ? ? ? ? Name ? ? ? ? ?Type
>> >> >> Data ? ? ?State
>> >> >>
>> >> >>
>> >> >> =================================================================================================
>> >> >> ? ? ? ? ? ? ? ? external ? ? ? profile
>> >> >> sip:mod_sofia at 192.168.1.148:5080 ? ? ?RUNNING (0)
>> >> >> ? ? ? ? ? ? ? ? internal ? ? ? profile
>> >> >> sip:mod_sofia at 192.168.1.148:5060 ? ? ?RUNNING (0)
>> >> >> ? ? ? ? ? ?192.168.1.148 ? ? ? ? alias
>> >> >> internal ? ? ?ALIASED
>> >> >>
>> >> >>
>> >> >> =================================================================================================
>> >> >> 2 profiles 1 alias
>> >> >>
>> >> >> Lots of questions there all trying to figure out why it's showing up
>> >> >> there.
>> >> >>
>> >> >> Thanks in advance,
>> >> >>
>> >> >> Alfredo Q-V
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > Alfredo,
>> > Use 'reloadxml' and then for mod_sofia to actually use the changes,
>> > you'll
>> > have to run 'sofia profile internal restart', but that will drop any
>> > calls
>> > that are on that profile.
>> >
>> > -- mm_202.
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> Alfredo,
>
> In a production system, you *shouldnt* have to change the main profiles at
> all.
> When adding / removing gateways, you can use 'sofia profile [profilename]
> rescan',
> that will not drop calls, only if you run 'reload'.
>
> As far as why the rescan adds the alias but doesnt remove it, I would say
> that is by design.
> That way you cant 'break' anything if your dialplan is using that alias.
> But someone more
> experienced than me may have a better / more accurate answer.
>
> -- mm_202.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Sat Jan 16 21:20:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 16 Jan 2010 23:20:56 -0600
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <191c3a031001162120k50fcca08l667fd3b842d5ebbd@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>
	<5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>
	<63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>
	<5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>
	<63de75711001161935w2d1e204ek81ebe1143afb1497@mail.gmail.com>
	<5fe6fa8f1001162100n699077faw8ac1e8cef4e4ea5@mail.gmail.com>
	<191c3a031001162117u413f0d5bi1903c69e519c3632@mail.gmail.com>
	<191c3a031001162120k50fcca08l667fd3b842d5ebbd@mail.gmail.com>
Message-ID: <191c3a031001162120j499f0e9xe359bb9a97607415@mail.gmail.com>

We do offer a triple your money back guarentee!

Many params can change without restarting and many can't ip/port fo
example.  FYI, You don't sound sincere when you use the term appriciate when
you are arguing with these guys voulenteering to help explain it to you.
You should make sure you make the most of their assistance as they could
have just said rtfm

On Jan 16, 2010 11:07 PM, "Alfredo Quiroga-Villamil" <lawwton at gmail.com>
wrote:

mm_202:

Thanks for the reply.

I don't agree with your statement: "In a production system, you

*shouldnt* have to change the main profiles at all."
I think it's always good to be able to reload things dynamically in a
non-disruptive way.

The alias is a bit unusual. To be honest I had never seen that before
in any of the Telecommunication Systems I've worked on. Not that is
not a good idea, just one that at first glance doesn't seem too clear
for some reason. Specially when you see two profiles both representing
the same object listed under "sofia status". Being forced to have the
alias doesn't seem like an appealing option, should be optional I
think. I would like if possible a more detailed explanation on what
would break if not present.

In any case, I am sure there is a reason for these things. I am trying
to understand how they all work together. Things are not as apparent
in FS at first glance when compared to other systems; but the platform
seems to be built with the purpose of offering a lot of flexibility
which is really good.

Appreciate the feedback.

Alfredo Q-V

Appreciate the help,

Alfredo

On Sat, Jan 16, 2010 at 10:35 PM, mm_202 <mastermind202 at gmail.com> wrote: >
> On Sat, Jan 16, 201...
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From lawwton at gmail.com  Sun Jan 17 06:07:25 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sun, 17 Jan 2010 09:07:25 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <191c3a031001162120j499f0e9xe359bb9a97607415@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>
	<5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>
	<63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>
	<5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>
	<63de75711001161935w2d1e204ek81ebe1143afb1497@mail.gmail.com>
	<5fe6fa8f1001162100n699077faw8ac1e8cef4e4ea5@mail.gmail.com>
	<191c3a031001162117u413f0d5bi1903c69e519c3632@mail.gmail.com>
	<191c3a031001162120k50fcca08l667fd3b842d5ebbd@mail.gmail.com>
	<191c3a031001162120j499f0e9xe359bb9a97607415@mail.gmail.com>
Message-ID: <5fe6fa8f1001170607v52c0516egc6f69a60a934722f@mail.gmail.com>

I am pretty sincere when I say "I appreciate the help", otherwise I
wouldn't say it. I don't think I've argued in any of my previous
statements and I've tried to be very polite when raising questions or
concerns. Disagreeing in a polite way doesn't necessarily mean that I
am 100% rejecting the idea and by the way it's not a bad thing to
disagree. No one said "You are wrong"; but "I disagree". There is a
difference there.

If you expect to put a system out there and new users that have used
other ones not to ask why this or that I think you'll keep running
into this with other people. No one is criticizing or putting down the
system here, on the contrary I am trying to simply find out how it
works so I can use it. But things like reloading configurations and
specific system details that are quite different when compared to
others (alias for instance which I am still not sure what would break
if removed) do require some asking if after reading the online
documentation and previous archived messages the reason is still not
clear.

So I am not sure how you expect people to ask questions or if this is
more a "this is what it is" take it or leave it dictatorship approach.
Not the paradigm followed in open source projects where collaboration,
asking and questioning is usually the  way to go.

Interestingly enough in this one case I DON'T APPRECIATE YOUR
RESPONSE. Now tell me if that doesn't sound sincere to you. In any
case I will continue to play with the system and possibly ask
questions as well.

Alfredo

On Sun, Jan 17, 2010 at 12:20 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> We do offer a triple your money back guarentee!
>
> Many params can change without restarting and many can't ip/port fo
> example.? FYI, You don't sound sincere when you use the term appriciate when
> you are arguing with these guys voulenteering to help explain it to you.
> You should make sure you make the most of their assistance as they could
> have just said rtfm
>
> On Jan 16, 2010 11:07 PM, "Alfredo Quiroga-Villamil" <lawwton at gmail.com>
> wrote:
>
> mm_202:
>
> Thanks for the reply.
>
> I don't agree with your statement: "In a production system, you
>
> *shouldnt* have to change the main profiles at all."
>
> I think it's always good to be able to reload things dynamically in a
> non-disruptive way.
>
> The alias is a bit unusual. To be honest I had never seen that before
> in any of the Telecommunication Systems I've worked on. Not that is
> not a good idea, just one that at first glance doesn't seem too clear
> for some reason. Specially when you see two profiles both representing
> the same object listed under "sofia status". Being forced to have the
> alias doesn't seem like an appealing option, should be optional I
> think. I would like if possible a more detailed explanation on what
> would break if not present.
>
> In any case, I am sure there is a reason for these things. I am trying
> to understand how they all work together. Things are not as apparent
> in FS at first glance when compared to other systems; but the platform
> seems to be built with the purpose of offering a lot of flexibility
> which is really good.
>
> Appreciate the feedback.
>
> Alfredo Q-V
>
> Appreciate the help,
>
> Alfredo
>
> On Sat, Jan 16, 2010 at 10:35 PM, mm_202 <mastermind202 at gmail.com> wrote: >
>> On Sat, Jan 16, 201...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From max.bridgewater at gmail.com  Sun Jan 17 10:55:34 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 17 Jan 2010 13:55:34 -0500
Subject: [Freeswitch-users] Trouble with Originate/Bridge from ESL
Message-ID: <c55d46951001171055p41f8e01dje6a5427fa2f3aeed@mail.gmail.com>

Hi,

The following command works great from the command line:
originate
{ringback=\'%(2000,4000,440.0,480.0)\'}sofia/internal/1004%74.63.243.52
&bridge(sofia/internal/1005%74.63.243.52)

But this one isn't working from the ESL:
api originate
{ringback=\'%(2000,4000,440.0,480.0)\'}sofia/internal/1004%74.63.243.52
&bridge(sofia/internal/1005%74.63.243.52)

The observed behavior is following: The A leg is dialed, then B leg is also
dialed but immediately followed by a hangup. That is, the B user doesn't
even has time to answer.

Any idea what I'm doing wrong?

Thanks,
Max.
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From devel at thom.fr.eu.org  Sun Jan 17 11:45:38 2010
From: devel at thom.fr.eu.org (devel at thom.fr.eu.org)
Date: Sun, 17 Jan 2010 20:45:38 +0100
Subject: [Freeswitch-users] No hangup tone after zap channel closed,
	tones 	in general
In-Reply-To: <87f2f3b91001151124r596418abta126ebdaae10465@mail.gmail.com>
References: <8b18ffe229516c90285de8a54a00e845@thom.fr.eu.org>	<87f2f3b91001141723u4b0b12e1rd0f922db493492cd@mail.gmail.com>	<001201ca95d1$045eb6e0$0d1c24a0$@fr.eu.org>
	<87f2f3b91001151124r596418abta126ebdaae10465@mail.gmail.com>
Message-ID: <002701ca97ad$a523e590$ef6bb0b0$@fr.eu.org>

It?s not exactly that. I expected an FXS port would by itself generate a
busy tone after a call (initiated or not by this port) is terminated by the
other end. Doing this permit the phone connected to this port to detect the
end of communication and hang up automatically.

 

That said, I guess the dialplan example would work when the FXS port is the
A leg, but what should I do when the FXS port is the B leg ?

 

Thanks

 

Fran?ois

 

De : freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael
Collins
Envoy? : vendredi 15 janvier 2010 20:24
? : freeswitch-users at lists.freeswitch.org
Objet : Re: [Freeswitch-users] No hangup tone after zap channel closed,
tones in general

 

 

On Fri, Jan 15, 2010 at 2:53 AM, <devel at thom.fr.eu.org> wrote:

Thank you for the link. I googled through but could not find anything
relevant.

 

So then with my FXS port, do I have to, when a call is over, bridge the
channel (which is either A or B leg depending on the cases) to an extension
with for instance

<action application="playback" data="tone_stream://%(500,500,440)"/>

if you're just trying to manually send out that tone then yes, you can just
add the line in your dialplan. You can then hangup after playing the tone.
The other end will have to decide what to do on its own.
-MC

 

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From max.bridgewater at gmail.com  Sun Jan 17 12:51:12 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 17 Jan 2010 15:51:12 -0500
Subject: [Freeswitch-users] Trouble with Originate/Bridge from ESL
In-Reply-To: <c55d46951001171055p41f8e01dje6a5427fa2f3aeed@mail.gmail.com>
References: <c55d46951001171055p41f8e01dje6a5427fa2f3aeed@mail.gmail.com>
Message-ID: <c55d46951001171251l42dee174kee35eeadaf1ef40@mail.gmail.com>

One more piece of information: the call is being terminated by Freeswitch
with the event:

Event-Name: CHANNEL_HANGUP
Hangup-Cause: NO_ANSWER

Which is strange because B leg doesn't even have the time to answer.


On Sun, Jan 17, 2010 at 1:55 PM, Max Bridgewater
<max.bridgewater at gmail.com>wrote:

> Hi,
>
> The following command works great from the command line:
> originate
> {ringback=\'%(2000,4000,440.0,480.0)\'}sofia/internal/1004%74.63.243.52
> &bridge(sofia/internal/1005%74.63.243.52)
>
> But this one isn't working from the ESL:
> api originate
> {ringback=\'%(2000,4000,440.0,480.0)\'}sofia/internal/1004%74.63.243.52
> &bridge(sofia/internal/1005%74.63.243.52)
>
> The observed behavior is following: The A leg is dialed, then B leg is also
> dialed but immediately followed by a hangup. That is, the B user doesn't
> even has time to answer.
>
> Any idea what I'm doing wrong?
>
> Thanks,
> Max.
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From mastermind202 at gmail.com  Sun Jan 17 13:53:52 2010
From: mastermind202 at gmail.com (mm_202)
Date: Sun, 17 Jan 2010 16:53:52 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <5fe6fa8f1001170607v52c0516egc6f69a60a934722f@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>
	<63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>
	<5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>
	<63de75711001161935w2d1e204ek81ebe1143afb1497@mail.gmail.com>
	<5fe6fa8f1001162100n699077faw8ac1e8cef4e4ea5@mail.gmail.com>
	<191c3a031001162117u413f0d5bi1903c69e519c3632@mail.gmail.com>
	<191c3a031001162120k50fcca08l667fd3b842d5ebbd@mail.gmail.com>
	<191c3a031001162120j499f0e9xe359bb9a97607415@mail.gmail.com>
	<5fe6fa8f1001170607v52c0516egc6f69a60a934722f@mail.gmail.com>
Message-ID: <63de75711001171353hca2a330he29cf2760f355a3d@mail.gmail.com>

On Sun, Jan 17, 2010 at 9:07 AM, Alfredo Quiroga-Villamil <lawwton at gmail.com
> wrote:

> I am pretty sincere when I say "I appreciate the help", otherwise I
> wouldn't say it. I don't think I've argued in any of my previous
> statements and I've tried to be very polite when raising questions or
> concerns. Disagreeing in a polite way doesn't necessarily mean that I
> am 100% rejecting the idea and by the way it's not a bad thing to
> disagree. No one said "You are wrong"; but "I disagree". There is a
> difference there.
>
> If you expect to put a system out there and new users that have used
> other ones not to ask why this or that I think you'll keep running
> into this with other people. No one is criticizing or putting down the
> system here, on the contrary I am trying to simply find out how it
> works so I can use it. But things like reloading configurations and
> specific system details that are quite different when compared to
> others (alias for instance which I am still not sure what would break
> if removed) do require some asking if after reading the online
> documentation and previous archived messages the reason is still not
> clear.
>
> So I am not sure how you expect people to ask questions or if this is
> more a "this is what it is" take it or leave it dictatorship approach.
> Not the paradigm followed in open source projects where collaboration,
> asking and questioning is usually the  way to go.
>
> Interestingly enough in this one case I DON'T APPRECIATE YOUR
> RESPONSE. Now tell me if that doesn't sound sincere to you. In any
> case I will continue to play with the system and possibly ask
> questions as well.
>
> Alfredo
>
> On Sun, Jan 17, 2010 at 12:20 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > We do offer a triple your money back guarentee!
> >
> > Many params can change without restarting and many can't ip/port fo
> > example.  FYI, You don't sound sincere when you use the term appriciate
> when
> > you are arguing with these guys voulenteering to help explain it to you.
> > You should make sure you make the most of their assistance as they could
> > have just said rtfm
> >
> > On Jan 16, 2010 11:07 PM, "Alfredo Quiroga-Villamil" <lawwton at gmail.com>
> > wrote:
> >
> > mm_202:
> >
> > Thanks for the reply.
> >
> > I don't agree with your statement: "In a production system, you
> >
> > *shouldnt* have to change the main profiles at all."
> >
> > I think it's always good to be able to reload things dynamically in a
> > non-disruptive way.
> >
> > The alias is a bit unusual. To be honest I had never seen that before
> > in any of the Telecommunication Systems I've worked on. Not that is
> > not a good idea, just one that at first glance doesn't seem too clear
> > for some reason. Specially when you see two profiles both representing
> > the same object listed under "sofia status". Being forced to have the
> > alias doesn't seem like an appealing option, should be optional I
> > think. I would like if possible a more detailed explanation on what
> > would break if not present.
> >
> > In any case, I am sure there is a reason for these things. I am trying
> > to understand how they all work together. Things are not as apparent
> > in FS at first glance when compared to other systems; but the platform
> > seems to be built with the purpose of offering a lot of flexibility
> > which is really good.
> >
> > Appreciate the feedback.
> >
> > Alfredo Q-V
> >
> > Appreciate the help,
> >
> > Alfredo
> >
> > On Sat, Jan 16, 2010 at 10:35 PM, mm_202 <mastermind202 at gmail.com>
> wrote: >
> >> On Sat, Jan 16, 201...
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

Alfredo,

What I meant by "In a production system, you *shouldnt* have to change the
main profiles at all", I meant core things like IP address / port; once they
are setup, they shouldnt change. FreeSWITCH is extremely dynamic and almost
everything can be changed on the fly without impacting production.

When I first started using FS, I was also a bit curious why there were
profile aliases, but I assure you that they are NOT a bad thing.  Just play
with FS some more and get a feel for it, you may even run into a situation
when you actually need the aliases :)

As for Anthony Minessale (anthm), he is the main creator of FreeSWITCH.  He
constantly deals with new users critizing FS before even using it and/or
reading the docs, hence he can come off quite direct sometimes. I would
guess that there was just a misunderstanding between you two.

Regardless, hopefully you'll continue to play with FS and see what myself
and so many other people did, -- a kickass system!

-- mm_202.
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From max.bridgewater at gmail.com  Sun Jan 17 14:02:21 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 17 Jan 2010 17:02:21 -0500
Subject: [Freeswitch-users] Trouble with Originate/Bridge from ESL
	[SOLVED]
Message-ID: <c55d46951001171402i6f906972p896c5b8c584bf919@mail.gmail.com>

Given that I'm using Java, I had to escape the ringback value's quote twice:

originate
{ringback=\\'%(2000,4000,440.0,480.0)\\'}sofia/internal/1004%74.63.243.52
&bridge(sofia/internal/1005%74.63.243.52)


On Sun, Jan 17, 2010 at 3:51 PM, Max Bridgewater
<max.bridgewater at gmail.com>wrote:

> One more piece of information: the call is being terminated by Freeswitch
> with the event:
>
> Event-Name: CHANNEL_HANGUP
> Hangup-Cause: NO_ANSWER
>
> Which is strange because B leg doesn't even have the time to answer.
>
>
>
> On Sun, Jan 17, 2010 at 1:55 PM, Max Bridgewater <
> max.bridgewater at gmail.com> wrote:
>
>> Hi,
>>
>> The following command works great from the command line:
>> originate
>> {ringback=\'%(2000,4000,440.0,480.0)\'}sofia/internal/1004%74.63.243.52
>> &bridge(sofia/internal/1005%74.63.243.52)
>>
>> But this one isn't working from the ESL:
>> api originate
>> {ringback=\'%(2000,4000,440.0,480.0)\'}sofia/internal/1004%74.63.243.52
>> &bridge(sofia/internal/1005%74.63.243.52)
>>
>> The observed behavior is following: The A leg is dialed, then B leg is
>> also dialed but immediately followed by a hangup. That is, the B user
>> doesn't even has time to answer.
>>
>> Any idea what I'm doing wrong?
>>
>> Thanks,
>> Max.
>
>
>
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From lists at redbonez.net  Sun Jan 17 14:20:58 2010
From: lists at redbonez.net (Adam Ford)
Date: Sun, 17 Jan 2010 15:20:58 -0700
Subject: [Freeswitch-users] Polycom configuration for FreeSWITCH
In-Reply-To: <d361ca7b1001152157x45b8d8c4l29625c763b24157f@mail.gmail.com>
References: <003701ca964b$3241b100$96c51300$@net>
	<d361ca7b1001152157x45b8d8c4l29625c763b24157f@mail.gmail.com>
Message-ID: <26AF402E733A4882B088F846AC2AE3CD@redbonez>

Thank you very much.  Mike is right, if you would be willing to post a
breakdown of the configurations that relate to FreeSWITCH on the wiki that
would be great.  I understand it would be easier to answer a specific
question, but I am not really looking for anything specific just a
comparison.  Yes, I have all the phones booting from an FTP server, they are
running the latest bootrom and SIP software that they support (4.1.4 &
3.1.4), and they register with FreeSWITCH ok.  I am mostly just looking for
a working config as an example to make sure I am not missing something, or
maybe not using the best settings for FreeSWITCH. 

 

I greatly appreciate the help,

 

-Adam

 

  

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yehavi
Bourvine
Sent: Friday, January 15, 2010 10:57 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom configuration for FreeSWITCH

 

I am working with various Polycom phones; I'll send you sample configuration
files next week (I am at home now). In the meantime, please send me your
requirenents so I may incorporate some of them into the files.

 

Have you managed to boot them from your TFTP/FTP./HTTP server? As long as
you did not provision them through a server you can do that through the
phone's WEB interface, but it is very limited and lacks a lot of
configuration options. I do the provisioning via a TFTP server.

 

                    Regards, __Yehavi:

2010/1/16 Adam Ford <lists at redbonez.net>

Does anyone who has successfully implemented Polycom IP301/501s, with
FreeSWITCH, have a configuration they can share with me as an example? I
have read the Polycom Admin Guide several times and understand what the
settings are/do, I am just not sure which FreeSWITCH supports, which it
doesn't, and which need special configuration to work with FreeSWITCH.

 

Thanks in advance,

-Adam

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org <http://www.freeswitch.org/> 

 

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From lawwton at gmail.com  Sun Jan 17 14:54:41 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sun, 17 Jan 2010 17:54:41 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <63de75711001171353hca2a330he29cf2760f355a3d@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>
	<5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>
	<63de75711001161935w2d1e204ek81ebe1143afb1497@mail.gmail.com>
	<5fe6fa8f1001162100n699077faw8ac1e8cef4e4ea5@mail.gmail.com>
	<191c3a031001162117u413f0d5bi1903c69e519c3632@mail.gmail.com>
	<191c3a031001162120k50fcca08l667fd3b842d5ebbd@mail.gmail.com>
	<191c3a031001162120j499f0e9xe359bb9a97607415@mail.gmail.com>
	<5fe6fa8f1001170607v52c0516egc6f69a60a934722f@mail.gmail.com>
	<63de75711001171353hca2a330he29cf2760f355a3d@mail.gmail.com>
Message-ID: <5fe6fa8f1001171454g6bfad0dy7a9d87238be5d3a0@mail.gmail.com>

Thanks mm_202.

Yeah, I figured he was having a bad day or something, don't think my
reply to you was offensive or contained criticism in any way. So I
said that I really appreciated your help which I of course meant. To
see his response was a bit of a shocker since I made sure I stated I
understood most of the time things are done in a way for a reason. I
had also spent a bit of time reading up documentation and trying
things before I posted the question.

So far I've been reading and trying to come up to speed and I really
like what I see. I will definitely continue to explore it and
hopefully use it in a production system one day.

Regards,

Alfredo

On Sun, Jan 17, 2010 at 4:53 PM, mm_202 <mastermind202 at gmail.com> wrote:
>
>
> On Sun, Jan 17, 2010 at 9:07 AM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> I am pretty sincere when I say "I appreciate the help", otherwise I
>> wouldn't say it. I don't think I've argued in any of my previous
>> statements and I've tried to be very polite when raising questions or
>> concerns. Disagreeing in a polite way doesn't necessarily mean that I
>> am 100% rejecting the idea and by the way it's not a bad thing to
>> disagree. No one said "You are wrong"; but "I disagree". There is a
>> difference there.
>>
>> If you expect to put a system out there and new users that have used
>> other ones not to ask why this or that I think you'll keep running
>> into this with other people. No one is criticizing or putting down the
>> system here, on the contrary I am trying to simply find out how it
>> works so I can use it. But things like reloading configurations and
>> specific system details that are quite different when compared to
>> others (alias for instance which I am still not sure what would break
>> if removed) do require some asking if after reading the online
>> documentation and previous archived messages the reason is still not
>> clear.
>>
>> So I am not sure how you expect people to ask questions or if this is
>> more a "this is what it is" take it or leave it dictatorship approach.
>> Not the paradigm followed in open source projects where collaboration,
>> asking and questioning is usually the ?way to go.
>>
>> Interestingly enough in this one case I DON'T APPRECIATE YOUR
>> RESPONSE. Now tell me if that doesn't sound sincere to you. In any
>> case I will continue to play with the system and possibly ask
>> questions as well.
>>
>> Alfredo
>>
>> On Sun, Jan 17, 2010 at 12:20 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > We do offer a triple your money back guarentee!
>> >
>> > Many params can change without restarting and many can't ip/port fo
>> > example.? FYI, You don't sound sincere when you use the term appriciate
>> > when
>> > you are arguing with these guys voulenteering to help explain it to you.
>> > You should make sure you make the most of their assistance as they could
>> > have just said rtfm
>> >
>> > On Jan 16, 2010 11:07 PM, "Alfredo Quiroga-Villamil" <lawwton at gmail.com>
>> > wrote:
>> >
>> > mm_202:
>> >
>> > Thanks for the reply.
>> >
>> > I don't agree with your statement: "In a production system, you
>> >
>> > *shouldnt* have to change the main profiles at all."
>> >
>> > I think it's always good to be able to reload things dynamically in a
>> > non-disruptive way.
>> >
>> > The alias is a bit unusual. To be honest I had never seen that before
>> > in any of the Telecommunication Systems I've worked on. Not that is
>> > not a good idea, just one that at first glance doesn't seem too clear
>> > for some reason. Specially when you see two profiles both representing
>> > the same object listed under "sofia status". Being forced to have the
>> > alias doesn't seem like an appealing option, should be optional I
>> > think. I would like if possible a more detailed explanation on what
>> > would break if not present.
>> >
>> > In any case, I am sure there is a reason for these things. I am trying
>> > to understand how they all work together. Things are not as apparent
>> > in FS at first glance when compared to other systems; but the platform
>> > seems to be built with the purpose of offering a lot of flexibility
>> > which is really good.
>> >
>> > Appreciate the feedback.
>> >
>> > Alfredo Q-V
>> >
>> > Appreciate the help,
>> >
>> > Alfredo
>> >
>> > On Sat, Jan 16, 2010 at 10:35 PM, mm_202 <mastermind202 at gmail.com>
>> > wrote: >
>> >> On Sat, Jan 16, 201...
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> Alfredo,
>
> What I meant by "In a production system, you *shouldnt* have to change the
> main profiles at all", I meant core things like IP address / port; once they
> are setup, they shouldnt change. FreeSWITCH is extremely dynamic and almost
> everything can be changed on the fly without impacting production.
>
> When I first started using FS, I was also a bit curious why there were
> profile aliases, but I assure you that they are NOT a bad thing.? Just play
> with FS some more and get a feel for it, you may even run into a situation
> when you actually need the aliases :)
>
> As for Anthony Minessale (anthm), he is the main creator of FreeSWITCH.? He
> constantly deals with new users critizing FS before even using it and/or
> reading the docs, hence he can come off quite direct sometimes. I would
> guess that there was just a misunderstanding between you two.
>
> Regardless, hopefully you'll continue to play with FS and see what myself
> and so many other people did, -- a kickass system!
>
> -- mm_202.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mastermind202 at gmail.com  Sun Jan 17 15:09:14 2010
From: mastermind202 at gmail.com (mm_202)
Date: Sun, 17 Jan 2010 18:09:14 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <5fe6fa8f1001171454g6bfad0dy7a9d87238be5d3a0@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>
	<63de75711001161935w2d1e204ek81ebe1143afb1497@mail.gmail.com>
	<5fe6fa8f1001162100n699077faw8ac1e8cef4e4ea5@mail.gmail.com>
	<191c3a031001162117u413f0d5bi1903c69e519c3632@mail.gmail.com>
	<191c3a031001162120k50fcca08l667fd3b842d5ebbd@mail.gmail.com>
	<191c3a031001162120j499f0e9xe359bb9a97607415@mail.gmail.com>
	<5fe6fa8f1001170607v52c0516egc6f69a60a934722f@mail.gmail.com>
	<63de75711001171353hca2a330he29cf2760f355a3d@mail.gmail.com>
	<5fe6fa8f1001171454g6bfad0dy7a9d87238be5d3a0@mail.gmail.com>
Message-ID: <63de75711001171509q67ca6b0ap747fa992f3cd97e2@mail.gmail.com>

On Sun, Jan 17, 2010 at 5:54 PM, Alfredo Quiroga-Villamil <lawwton at gmail.com
> wrote:

> Thanks mm_202.
>
> Yeah, I figured he was having a bad day or something, don't think my
> reply to you was offensive or contained criticism in any way. So I
> said that I really appreciated your help which I of course meant. To
> see his response was a bit of a shocker since I made sure I stated I
> understood most of the time things are done in a way for a reason. I
> had also spent a bit of time reading up documentation and trying
> things before I posted the question.
>
> So far I've been reading and trying to come up to speed and I really
> like what I see. I will definitely continue to explore it and
> hopefully use it in a production system one day.
>
> Regards,
>
> Alfredo
>
> On Sun, Jan 17, 2010 at 4:53 PM, mm_202 <mastermind202 at gmail.com> wrote:
> >
> >
> > On Sun, Jan 17, 2010 at 9:07 AM, Alfredo Quiroga-Villamil
> > <lawwton at gmail.com> wrote:
> >>
> >> I am pretty sincere when I say "I appreciate the help", otherwise I
> >> wouldn't say it. I don't think I've argued in any of my previous
> >> statements and I've tried to be very polite when raising questions or
> >> concerns. Disagreeing in a polite way doesn't necessarily mean that I
> >> am 100% rejecting the idea and by the way it's not a bad thing to
> >> disagree. No one said "You are wrong"; but "I disagree". There is a
> >> difference there.
> >>
> >> If you expect to put a system out there and new users that have used
> >> other ones not to ask why this or that I think you'll keep running
> >> into this with other people. No one is criticizing or putting down the
> >> system here, on the contrary I am trying to simply find out how it
> >> works so I can use it. But things like reloading configurations and
> >> specific system details that are quite different when compared to
> >> others (alias for instance which I am still not sure what would break
> >> if removed) do require some asking if after reading the online
> >> documentation and previous archived messages the reason is still not
> >> clear.
> >>
> >> So I am not sure how you expect people to ask questions or if this is
> >> more a "this is what it is" take it or leave it dictatorship approach.
> >> Not the paradigm followed in open source projects where collaboration,
> >> asking and questioning is usually the  way to go.
> >>
> >> Interestingly enough in this one case I DON'T APPRECIATE YOUR
> >> RESPONSE. Now tell me if that doesn't sound sincere to you. In any
> >> case I will continue to play with the system and possibly ask
> >> questions as well.
> >>
> >> Alfredo
> >>
> >> On Sun, Jan 17, 2010 at 12:20 AM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >> > We do offer a triple your money back guarentee!
> >> >
> >> > Many params can change without restarting and many can't ip/port fo
> >> > example.  FYI, You don't sound sincere when you use the term
> appriciate
> >> > when
> >> > you are arguing with these guys voulenteering to help explain it to
> you.
> >> > You should make sure you make the most of their assistance as they
> could
> >> > have just said rtfm
> >> >
> >> > On Jan 16, 2010 11:07 PM, "Alfredo Quiroga-Villamil" <
> lawwton at gmail.com>
> >> > wrote:
> >> >
> >> > mm_202:
> >> >
> >> > Thanks for the reply.
> >> >
> >> > I don't agree with your statement: "In a production system, you
> >> >
> >> > *shouldnt* have to change the main profiles at all."
> >> >
> >> > I think it's always good to be able to reload things dynamically in a
> >> > non-disruptive way.
> >> >
> >> > The alias is a bit unusual. To be honest I had never seen that before
> >> > in any of the Telecommunication Systems I've worked on. Not that is
> >> > not a good idea, just one that at first glance doesn't seem too clear
> >> > for some reason. Specially when you see two profiles both representing
> >> > the same object listed under "sofia status". Being forced to have the
> >> > alias doesn't seem like an appealing option, should be optional I
> >> > think. I would like if possible a more detailed explanation on what
> >> > would break if not present.
> >> >
> >> > In any case, I am sure there is a reason for these things. I am trying
> >> > to understand how they all work together. Things are not as apparent
> >> > in FS at first glance when compared to other systems; but the platform
> >> > seems to be built with the purpose of offering a lot of flexibility
> >> > which is really good.
> >> >
> >> > Appreciate the feedback.
> >> >
> >> > Alfredo Q-V
> >> >
> >> > Appreciate the help,
> >> >
> >> > Alfredo
> >> >
> >> > On Sat, Jan 16, 2010 at 10:35 PM, mm_202 <mastermind202 at gmail.com>
> >> > wrote: >
> >> >> On Sat, Jan 16, 201...
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > Alfredo,
> >
> > What I meant by "In a production system, you *shouldnt* have to change
> the
> > main profiles at all", I meant core things like IP address / port; once
> they
> > are setup, they shouldnt change. FreeSWITCH is extremely dynamic and
> almost
> > everything can be changed on the fly without impacting production.
> >
> > When I first started using FS, I was also a bit curious why there were
> > profile aliases, but I assure you that they are NOT a bad thing.  Just
> play
> > with FS some more and get a feel for it, you may even run into a
> situation
> > when you actually need the aliases :)
> >
> > As for Anthony Minessale (anthm), he is the main creator of FreeSWITCH.
> He
> > constantly deals with new users critizing FS before even using it and/or
> > reading the docs, hence he can come off quite direct sometimes. I would
> > guess that there was just a misunderstanding between you two.
> >
> > Regardless, hopefully you'll continue to play with FS and see what myself
> > and so many other people did, -- a kickass system!
> >
> > -- mm_202.
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


Excellent to hear!  If you have any more questions or even curiosities,
don't hesitate to ask.
The FS community is great in answering even the most perplexing questions.

You should also (if you havent yet) join us on IRC in #freeswitch (
irc.freenode.net), you'll get answers much
quicker and see more of the 'community'.

-- mm_202.
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From jingwei.yang at gmail.com  Sun Jan 17 17:55:09 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 18 Jan 2010 09:55:09 +0800
Subject: [Freeswitch-users] Questions about mod_fifo
In-Reply-To: <23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>
References: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
	<23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>
Message-ID: <13529f9d1001171755v229c2137u4fa5cfe2f30d518@mail.gmail.com>

Thanks for replying. This is my dialplan

    <extension name="in_fifo">
      <condition field="destination_number" expression="^779$">
        <action application="set" data="fifo_music=$${hold_music}"/>
        <action application="answer"/>
        <action application="fifo" data="myq in undef undef"/>
      </condition>
    </extension>

And I created a queue in fifo.conf.xml like this

    <fifo name="myq" importance="0">
      <!--<member timeout="60" simo="1"
lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
    </fifo>

However, I'm still not able to see the incoming call get queued.

freeswitch at localhost.localdomain> fifo list myq
API CALL [fifo(list myq)] output:
<fifo_report>
  <fifo name="myq" consumer_count="0" caller_count="0" waiting_count="0"
importance="0">
    <callers></callers>
    <consumers></consumers>
  </fifo>
</fifo_report>

I tried both mod_skypiax and mod_dingaling, but with the same result.

Regards,
-Jingwei

On Sun, Jan 17, 2010 at 11:37 AM, Seven Du <dujinfang at gmail.com> wrote:

> 2010/1/15, Jingwei Yang <jingwei.yang at gmail.com>:
> > Hi Guys,
> >
> > I'm implementing an ACD system using ESL and mod_fifo. Based on what
> Anthony
> > suggested in this post:
> >
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html
> >
> > *You can make an event socket application that listens for FIFO events
> and
> > keeps track of what FIFOs are currently busy and when there are people
> > waiting you can have that script generate a call to a group of SIP phones
> so
> > when the first one answers, it sends them in as an agent where they can
> > field the calls.
> > *
> >
> > 1. How should I handle the concurrent issues? If I bridge a user to two
> > agents and both of them answers, how does FS take care of this situation?
> > Will a slower agent get a busy tone automatically?
> >
>
> I think it just follow the standard originate dialstring rules.
>
> > 2. If the socket application is brought up after some users have called
> in,
> > what command should I use to check the busy queues? fifo list?
> >
> Yes. Perhaps you can also check the fifo db, either sqlite or ODBC.
>
> > 3. Am I using fifo list and fifo count correctly?
> >
> > here's the testing dialplan:
> >
> >     <extension name="in_fifo">
> >       <condition field="destination_number" expression="^779$">
> >         <action application="set" data="fifo_music=$${hold_music}"/>
> >         <action application="answer"/>
> >         <action application="fifo" data="*myq *in"/>
> >       </condition>
> >     </extension>
> >
> > when a call comes in and gets queued, these are the results of some
> commands
> > I tried.
> >
> > freeswitch at localhost.localdomain> fifo list
> > API CALL [fifo(list)] output:
> > <fifo_report>
> >   <fifo name="cool_fifo at 192.168.1.104" consumer_count="0"
> caller_count="0"
> > waiting_count="0" importance="0">
> >     <callers></callers>
> >     <consumers></consumers>
> >   </fifo>
> > </fifo_report>
> >
> >
> > freeswitch at localhost.localdomain> fifo list myq
> > API CALL [fifo(list myq)] output:
> > <fifo_report></fifo_report>
> >
> >
> > freeswitch at localhost.localdomain> fifo count myq
> > API CALL [fifo(count myq)] output:
> > none
> >
> > It seems *myq* doesn't get created at all? Please enlighten.
> >
> > Thanks and best regards,
> > -Jingwei
> >
> AFAIK, thant means the channel didn't queued in. Did you see any error
> logs? I think you need to remove the stars in   <action
> application="fifo" data="*myq *in"/>.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From darklion11 at yahoo.com  Sun Jan 17 18:13:32 2010
From: darklion11 at yahoo.com (Edmar Cruz)
Date: Sun, 17 Jan 2010 18:13:32 -0800 (PST)
Subject: [Freeswitch-users]  Change Ip to Domain Name
Message-ID: <27104680.post@talk.nabble.com>


Dear All,

   How can i change the domain of my freeswitch 52.236.125.12 to
sip.grandminister.com to be able to detect the presence of the user whos
online or not...

Thanks
Edmar
-- 
View this message in context: http://old.nabble.com/Change-Ip-to-Domain-Name-tp27104680p27104680.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From dujinfang at gmail.com  Sun Jan 17 19:42:28 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 18 Jan 2010 11:42:28 +0800
Subject: [Freeswitch-users] Questions about mod_fifo
In-Reply-To: <13529f9d1001171755v229c2137u4fa5cfe2f30d518@mail.gmail.com>
References: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
	<23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>
	<13529f9d1001171755v229c2137u4fa5cfe2f30d518@mail.gmail.com>
Message-ID: <23f91031001171942h42e655c9vbc5ecb4165b7aca9@mail.gmail.com>

better to pastebin your log.

2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> Thanks for replying. This is my dialplan
>
> ??? <extension name="in_fifo">
> ????? <condition field="destination_number" expression="^779$">
> ??????? <action application="set" data="fifo_music=$${hold_music}"/>
> ??????? <action application="answer"/>
> ??????? <action application="fifo" data="myq in undef undef"/>
> ????? </condition>
> ??? </extension>
>
> And I created a queue in fifo.conf.xml like this
>
> ??? <fifo name="myq" importance="0">
> ????? <!--<member timeout="60" simo="1"
> lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
> ??? </fifo>
>
> However, I'm still not able to see the incoming call get queued.
>
> freeswitch at localhost.localdomain> fifo list myq
> API CALL [fifo(list myq)] output:
> <fifo_report>
> ? <fifo name="myq" consumer_count="0" caller_count="0" waiting_count="0"
> importance="0">
> ??? <callers></callers>
> ??? <consumers></consumers>
> ? </fifo>
> </fifo_report>
>
> I tried both mod_skypiax and mod_dingaling, but with the same result.
>
> Regards,
> -Jingwei
>
> On Sun, Jan 17, 2010 at 11:37 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> 2010/1/15, Jingwei Yang <jingwei.yang at gmail.com>:
>> > Hi Guys,
>> >
>> > I'm implementing an ACD system using ESL and mod_fifo. Based on what
>> > Anthony
>> > suggested in this post:
>> >
>> > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html
>> >
>> > *You can make an event socket application that listens for FIFO events
>> > and
>> > keeps track of what FIFOs are currently busy and when there are people
>> > waiting you can have that script generate a call to a group of SIP
>> > phones so
>> > when the first one answers, it sends them in as an agent where they can
>> > field the calls.
>> > *
>> >
>> > 1. How should I handle the concurrent issues? If I bridge a user to two
>> > agents and both of them answers, how does FS take care of this
>> > situation?
>> > Will a slower agent get a busy tone automatically?
>> >
>>
>> I think it just follow the standard originate dialstring rules.
>>
>> > 2. If the socket application is brought up after some users have called
>> > in,
>> > what command should I use to check the busy queues? fifo list?
>> >
>> Yes. Perhaps you can also check the fifo db, either sqlite or ODBC.
>>
>> > 3. Am I using fifo list and fifo count correctly?
>> >
>> > here's the testing dialplan:
>> >
>> > ? ? <extension name="in_fifo">
>> > ? ? ? <condition field="destination_number" expression="^779$">
>> > ? ? ? ? <action application="set" data="fifo_music=$${hold_music}"/>
>> > ? ? ? ? <action application="answer"/>
>> > ? ? ? ? <action application="fifo" data="*myq *in"/>
>> > ? ? ? </condition>
>> > ? ? </extension>
>> >
>> > when a call comes in and gets queued, these are the results of some
>> > commands
>> > I tried.
>> >
>> > freeswitch at localhost.localdomain> fifo list
>> > API CALL [fifo(list)] output:
>> > <fifo_report>
>> > ? <fifo name="cool_fifo at 192.168.1.104" consumer_count="0"
>> > caller_count="0"
>> > waiting_count="0" importance="0">
>> > ? ? <callers></callers>
>> > ? ? <consumers></consumers>
>> > ? </fifo>
>> > </fifo_report>
>> >
>> >
>> > freeswitch at localhost.localdomain> fifo list myq
>> > API CALL [fifo(list myq)] output:
>> > <fifo_report></fifo_report>
>> >
>> >
>> > freeswitch at localhost.localdomain> fifo count myq
>> > API CALL [fifo(count myq)] output:
>> > none
>> >
>> > It seems *myq* doesn't get created at all? Please enlighten.
>> >
>> > Thanks and best regards,
>> > -Jingwei
>> >
>> AFAIK, thant means the channel didn't queued in. Did you see any error
>> logs? I think you need to remove the stars in ? <action
>> application="fifo" data="*myq *in"/>.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Sun Jan 17 20:13:20 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 17 Jan 2010 22:13:20 -0600
Subject: [Freeswitch-users] Change Ip to Domain Name
In-Reply-To: <27104680.post@talk.nabble.com>
References: <27104680.post@talk.nabble.com>
Message-ID: <FF1ABD07-18BA-412B-9F3B-D0178D878883@freeswitch.org>

Change it in the config and setup proper DNS.

/b

On Jan 17, 2010, at 8:13 PM, Edmar Cruz wrote:

> 
> Dear All,
> 
>   How can i change the domain of my freeswitch 52.236.125.12 to
> sip.grandminister.com to be able to detect the presence of the user whos
> online or not...
> 
> Thanks
> Edmar

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From brian at freeswitch.org  Sun Jan 17 20:15:33 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 17 Jan 2010 22:15:33 -0600
Subject: [Freeswitch-users] Trouble with Originate/Bridge from ESL
	[SOLVED]
In-Reply-To: <c55d46951001171402i6f906972p896c5b8c584bf919@mail.gmail.com>
References: <c55d46951001171402i6f906972p896c5b8c584bf919@mail.gmail.com>
Message-ID: <062F43CC-5C35-4415-B202-81551BD71E1C@freeswitch.org>

Its not just java!  :P 

/b

On Jan 17, 2010, at 4:02 PM, Max Bridgewater wrote:

> Given that I'm using Java, I had to escape the ringback value's quote twice:
> 
> originate {ringback=\\'%(2000,4000,440.0,480.0)\\'}sofia/internal/1004%74.63.243.52 &bridge(sofia/internal/1005%74.63.243.52)



From brian at freeswitch.org  Sun Jan 17 20:15:53 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 17 Jan 2010 22:15:53 -0600
Subject: [Freeswitch-users] Trouble with Originate/Bridge from ESL
	[SOLVED]
In-Reply-To: <c55d46951001171402i6f906972p896c5b8c584bf919@mail.gmail.com>
References: <c55d46951001171402i6f906972p896c5b8c584bf919@mail.gmail.com>
Message-ID: <5696EFC4-8DC0-42B0-A909-02CA03000E84@freeswitch.org>

Oh btw happen to wanna write me some sample ESL java samples to put in tree.. see the examples in the perl folder.

/b

On Jan 17, 2010, at 4:02 PM, Max Bridgewater wrote:

> Given that I'm using Java, I had to escape the ringback value's quote twice:
> 
> originate {ringback=\\'%(2000,4000,440.0,480.0)\\'}sofia/internal/1004%74.63.243.52 &bridge(sofia/internal/1005%74.63.243.52)



From brian at freeswitch.org  Sun Jan 17 20:16:36 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 17 Jan 2010 22:16:36 -0600
Subject: [Freeswitch-users] No media after Originate
In-Reply-To: <f97004c31001161522g65a0da7agec1eabffa321692d@mail.gmail.com>
References: <f97004c31001161306o10517d1y9e0cf0229bce3589@mail.gmail.com>
	<3081C466-F735-47F1-BAEC-2B497F8F08D8@freeswitch.org>
	<f97004c31001161522g65a0da7agec1eabffa321692d@mail.gmail.com>
Message-ID: <D44E6C4F-58D0-4925-B025-2C34FB6E9D53@freeswitch.org>

Are you trying to do bypass media?

/b

On Jan 16, 2010, at 5:22 PM, Mohamed Hassan wrote:

> There is no nat my server has public ip and not nated
> and my sip provider too  as i can make regular calls through the same
> provider without originate



From max.bridgewater at gmail.com  Sun Jan 17 20:23:27 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 17 Jan 2010 23:23:27 -0500
Subject: [Freeswitch-users] Trouble with Originate/Bridge from ESL
	[SOLVED]
In-Reply-To: <5696EFC4-8DC0-42B0-A909-02CA03000E84@freeswitch.org>
References: <c55d46951001171402i6f906972p896c5b8c584bf919@mail.gmail.com>
	<5696EFC4-8DC0-42B0-A909-02CA03000E84@freeswitch.org>
Message-ID: <c55d46951001172023w1b2e5d4fm22f23d7d58d079a4@mail.gmail.com>

I might. Stay tuned. Self imposed deadline: Feb 14th ;).

On Sun, Jan 17, 2010 at 11:15 PM, Brian West <brian at freeswitch.org> wrote:

> Oh btw happen to wanna write me some sample ESL java samples to put in
> tree.. see the examples in the perl folder.
>
> /b
>
> On Jan 17, 2010, at 4:02 PM, Max Bridgewater wrote:
>
> > Given that I'm using Java, I had to escape the ringback value's quote
> twice:
> >
> > originate
> {ringback=\\'%(2000,4000,440.0,480.0)\\'}sofia/internal/1004%74.63.243.52
> &bridge(sofia/internal/1005%74.63.243.52)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jingwei.yang at gmail.com  Sun Jan 17 22:48:51 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 18 Jan 2010 14:48:51 +0800
Subject: [Freeswitch-users] Questions about mod_fifo
In-Reply-To: <23f91031001171942h42e655c9vbc5ecb4165b7aca9@mail.gmail.com>
References: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
	<23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>
	<13529f9d1001171755v229c2137u4fa5cfe2f30d518@mail.gmail.com>
	<23f91031001171942h42e655c9vbc5ecb4165b7aca9@mail.gmail.com>
Message-ID: <13529f9d1001172248x1a1ddc8arb7f4aba2e71ffb0f@mail.gmail.com>

Logs submitted: http://pastebin.freeswitch.org/11836

I was trying to check whether the call had been added into the queue via
telnet, but failed to find the fifo events. Here's my simplified dialplan:

<extension name="in_fifo">
   <condition field="destination_number" expression="^779$">
     <action application="answer"/>
     <action application="fifo" data="myq in undef undef"/>
  </condition>
</extension>

Please advise where went wrong.

Thanks and best regards,
-Jingwei


On Mon, Jan 18, 2010 at 11:42 AM, Seven Du <dujinfang at gmail.com> wrote:

> better to pastebin your log.
>
> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> > Thanks for replying. This is my dialplan
> >
> >     <extension name="in_fifo">
> >       <condition field="destination_number" expression="^779$">
> >         <action application="set" data="fifo_music=$${hold_music}"/>
> >         <action application="answer"/>
> >         <action application="fifo" data="myq in undef undef"/>
> >       </condition>
> >     </extension>
> >
> > And I created a queue in fifo.conf.xml like this
> >
> >     <fifo name="myq" importance="0">
> >       <!--<member timeout="60" simo="1"
> > lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
> >     </fifo>
> >
> > However, I'm still not able to see the incoming call get queued.
> >
> > freeswitch at localhost.localdomain> fifo list myq
> > API CALL [fifo(list myq)] output:
> > <fifo_report>
> >   <fifo name="myq" consumer_count="0" caller_count="0" waiting_count="0"
> > importance="0">
> >     <callers></callers>
> >     <consumers></consumers>
> >   </fifo>
> > </fifo_report>
> >
> > I tried both mod_skypiax and mod_dingaling, but with the same result.
> >
> > Regards,
> > -Jingwei
> >
> > On Sun, Jan 17, 2010 at 11:37 AM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> 2010/1/15, Jingwei Yang <jingwei.yang at gmail.com>:
> >> > Hi Guys,
> >> >
> >> > I'm implementing an ACD system using ESL and mod_fifo. Based on what
> >> > Anthony
> >> > suggested in this post:
> >> >
> >> >
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html
> >> >
> >> > *You can make an event socket application that listens for FIFO events
> >> > and
> >> > keeps track of what FIFOs are currently busy and when there are people
> >> > waiting you can have that script generate a call to a group of SIP
> >> > phones so
> >> > when the first one answers, it sends them in as an agent where they
> can
> >> > field the calls.
> >> > *
> >> >
> >> > 1. How should I handle the concurrent issues? If I bridge a user to
> two
> >> > agents and both of them answers, how does FS take care of this
> >> > situation?
> >> > Will a slower agent get a busy tone automatically?
> >> >
> >>
> >> I think it just follow the standard originate dialstring rules.
> >>
> >> > 2. If the socket application is brought up after some users have
> called
> >> > in,
> >> > what command should I use to check the busy queues? fifo list?
> >> >
> >> Yes. Perhaps you can also check the fifo db, either sqlite or ODBC.
> >>
> >> > 3. Am I using fifo list and fifo count correctly?
> >> >
> >> > here's the testing dialplan:
> >> >
> >> >     <extension name="in_fifo">
> >> >       <condition field="destination_number" expression="^779$">
> >> >         <action application="set" data="fifo_music=$${hold_music}"/>
> >> >         <action application="answer"/>
> >> >         <action application="fifo" data="*myq *in"/>
> >> >       </condition>
> >> >     </extension>
> >> >
> >> > when a call comes in and gets queued, these are the results of some
> >> > commands
> >> > I tried.
> >> >
> >> > freeswitch at localhost.localdomain> fifo list
> >> > API CALL [fifo(list)] output:
> >> > <fifo_report>
> >> >   <fifo name="cool_fifo at 192.168.1.104" consumer_count="0"
> >> > caller_count="0"
> >> > waiting_count="0" importance="0">
> >> >     <callers></callers>
> >> >     <consumers></consumers>
> >> >   </fifo>
> >> > </fifo_report>
> >> >
> >> >
> >> > freeswitch at localhost.localdomain> fifo list myq
> >> > API CALL [fifo(list myq)] output:
> >> > <fifo_report></fifo_report>
> >> >
> >> >
> >> > freeswitch at localhost.localdomain> fifo count myq
> >> > API CALL [fifo(count myq)] output:
> >> > none
> >> >
> >> > It seems *myq* doesn't get created at all? Please enlighten.
> >> >
> >> > Thanks and best regards,
> >> > -Jingwei
> >> >
> >> AFAIK, thant means the channel didn't queued in. Did you see any error
> >> logs? I think you need to remove the stars in   <action
> >> application="fifo" data="*myq *in"/>.
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From mailinglist at fribert.dk  Sun Jan 17 22:50:30 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Mon, 18 Jan 2010 07:50:30 +0100
Subject: [Freeswitch-users] How do I invite group to join existing call?
Message-ID: <4B5412C6020000E1000003D6@mail.fribert.dk>

Hi All

I would like to be able to invite a group / global to join an existing call, but how do I accomplish this, can it be done?


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From dujinfang at gmail.com  Sun Jan 17 23:25:23 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 18 Jan 2010 15:25:23 +0800
Subject: [Freeswitch-users] Questions about mod_fifo
In-Reply-To: <13529f9d1001172248x1a1ddc8arb7f4aba2e71ffb0f@mail.gmail.com>
References: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
	<23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>
	<13529f9d1001171755v229c2137u4fa5cfe2f30d518@mail.gmail.com>
	<23f91031001171942h42e655c9vbc5ecb4165b7aca9@mail.gmail.com>
	<13529f9d1001172248x1a1ddc8arb7f4aba2e71ffb0f@mail.gmail.com>
Message-ID: <23f91031001172325n2dd0f462ga202abec3a79d019@mail.gmail.com>

did you happened to run "show channels" ?

clearly it's a dialplan problem other than a fifo one.


Dialplan: skypiax/interface8 Regex (PASS) [eavesdrop]
destination_number(779) =~ /^779$/ break=on-false
Dialplan: skypiax/interface8 Action answer()
Dialplan: skypiax/interface8 Action
set(eavesdrop_indicate_failed=tone_stream://%(500, 0, 320))
Dialplan: skypiax/interface8 Action
set(eavesdrop_indicate_new=tone_stream://%(500, 0, 620))
Dialplan: skypiax/interface8 Action
set(eavesdrop_indicate_idle=tone_stream://%(250, 0, 920))
Dialplan: skypiax/interface8 Action eavesdrop(all)



2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> Logs submitted: http://pastebin.freeswitch.org/11836
>
> I was trying to check whether the call had been added into the queue via
> telnet, but failed to find the fifo events. Here's my simplified dialplan:
>
> <extension name="in_fifo">
> ?? <condition field="destination_number" expression="^779$">
> ???? <action application="answer"/>
> ???? <action application="fifo" data="myq in undef undef"/>
> ? </condition>
> </extension>
>
> Please advise where went wrong.
>
> Thanks and best regards,
> -Jingwei
>
>
> On Mon, Jan 18, 2010 at 11:42 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> better to pastebin your log.
>>
>> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
>> > Thanks for replying. This is my dialplan
>> >
>> > ??? <extension name="in_fifo">
>> > ????? <condition field="destination_number" expression="^779$">
>> > ??????? <action application="set" data="fifo_music=$${hold_music}"/>
>> > ??????? <action application="answer"/>
>> > ??????? <action application="fifo" data="myq in undef undef"/>
>> > ????? </condition>
>> > ??? </extension>
>> >
>> > And I created a queue in fifo.conf.xml like this
>> >
>> > ??? <fifo name="myq" importance="0">
>> > ????? <!--<member timeout="60" simo="1"
>> > lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
>> > ??? </fifo>
>> >
>> > However, I'm still not able to see the incoming call get queued.
>> >
>> > freeswitch at localhost.localdomain> fifo list myq
>> > API CALL [fifo(list myq)] output:
>> > <fifo_report>
>> > ? <fifo name="myq" consumer_count="0" caller_count="0" waiting_count="0"
>> > importance="0">
>> > ??? <callers></callers>
>> > ??? <consumers></consumers>
>> > ? </fifo>
>> > </fifo_report>
>> >
>> > I tried both mod_skypiax and mod_dingaling, but with the same result.
>> >
>> > Regards,
>> > -Jingwei
>> >
>> > On Sun, Jan 17, 2010 at 11:37 AM, Seven Du <dujinfang at gmail.com> wrote:
>> >>
>> >> 2010/1/15, Jingwei Yang <jingwei.yang at gmail.com>:
>> >> > Hi Guys,
>> >> >
>> >> > I'm implementing an ACD system using ESL and mod_fifo. Based on what
>> >> > Anthony
>> >> > suggested in this post:
>> >> >
>> >> >
>> >> > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html
>> >> >
>> >> > *You can make an event socket application that listens for FIFO
>> >> > events
>> >> > and
>> >> > keeps track of what FIFOs are currently busy and when there are
>> >> > people
>> >> > waiting you can have that script generate a call to a group of SIP
>> >> > phones so
>> >> > when the first one answers, it sends them in as an agent where they
>> >> > can
>> >> > field the calls.
>> >> > *
>> >> >
>> >> > 1. How should I handle the concurrent issues? If I bridge a user to
>> >> > two
>> >> > agents and both of them answers, how does FS take care of this
>> >> > situation?
>> >> > Will a slower agent get a busy tone automatically?
>> >> >
>> >>
>> >> I think it just follow the standard originate dialstring rules.
>> >>
>> >> > 2. If the socket application is brought up after some users have
>> >> > called
>> >> > in,
>> >> > what command should I use to check the busy queues? fifo list?
>> >> >
>> >> Yes. Perhaps you can also check the fifo db, either sqlite or ODBC.
>> >>
>> >> > 3. Am I using fifo list and fifo count correctly?
>> >> >
>> >> > here's the testing dialplan:
>> >> >
>> >> > ? ? <extension name="in_fifo">
>> >> > ? ? ? <condition field="destination_number" expression="^779$">
>> >> > ? ? ? ? <action application="set" data="fifo_music=$${hold_music}"/>
>> >> > ? ? ? ? <action application="answer"/>
>> >> > ? ? ? ? <action application="fifo" data="*myq *in"/>
>> >> > ? ? ? </condition>
>> >> > ? ? </extension>
>> >> >
>> >> > when a call comes in and gets queued, these are the results of some
>> >> > commands
>> >> > I tried.
>> >> >
>> >> > freeswitch at localhost.localdomain> fifo list
>> >> > API CALL [fifo(list)] output:
>> >> > <fifo_report>
>> >> > ? <fifo name="cool_fifo at 192.168.1.104" consumer_count="0"
>> >> > caller_count="0"
>> >> > waiting_count="0" importance="0">
>> >> > ? ? <callers></callers>
>> >> > ? ? <consumers></consumers>
>> >> > ? </fifo>
>> >> > </fifo_report>
>> >> >
>> >> >
>> >> > freeswitch at localhost.localdomain> fifo list myq
>> >> > API CALL [fifo(list myq)] output:
>> >> > <fifo_report></fifo_report>
>> >> >
>> >> >
>> >> > freeswitch at localhost.localdomain> fifo count myq
>> >> > API CALL [fifo(count myq)] output:
>> >> > none
>> >> >
>> >> > It seems *myq* doesn't get created at all? Please enlighten.
>> >> >
>> >> > Thanks and best regards,
>> >> > -Jingwei
>> >> >
>> >> AFAIK, thant means the channel didn't queued in. Did you see any error
>> >> logs? I think you need to remove the stars in ? <action
>> >> application="fifo" data="*myq *in"/>.
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From kond at nstel.ru  Sun Jan 17 23:26:56 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Mon, 18 Jan 2010 10:26:56 +0300
Subject: [Freeswitch-users] eavesdrop problem?
In-Reply-To: <191c3a031001151445n51ba1514rb387179bb837c558@mail.gmail.com>
Message-ID: <20100118072655.F29E011F68@mail.nstel.ru>

Anthony,

Inserting 

<action application="hash"
data="insert/${domain_name}-spymap/${destination_number}/${uuid}"/>

into <extension name="global" continue="true"> in the dialplan

looks to work ok. I can now eavesdrop a colee.

By the way, should I do something to remove a uuid from the database when
the call is ended? Or will it be removed automatically?

Thanks and regards,

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Saturday, January 16, 2010 1:46 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] eavesdrop problem?

 

yes, exactly.
That is a demo of how you could possibly store a uuid by inserting them into
the db keyed from your user extension in the caller id.

if you do show channel and you see the uuid for each leg that is the
argument eavesdrop takes.
you can also do "all" in place of a uuid so you can cycle all the calls with
DTMF




On Fri, Jan 15, 2010 at 11:41 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Anthony,

Thanks for the reply.

Can you please point me to the document where I could read about it?

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop does not say
anything about it.

But let me guess: I should add 

<action application="hash"
data="insert/${domain_name}-spymap/${destination_number}/${uuid}"/>

Into <extension name="global" continue="true"> in the dialplan.

Am I close?

Thanks and regards,

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Friday, January 15, 2010 7:05 PM


To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] eavesdrop problem?

 

don't bother,



only inbound legs are added to the db that is used to lookup for eavesdrop
because the action is in the dialplan.
The extensions to eavesdrop you are using are just a demo to show you how to
work it.
you need to know the uuid of the channel you are trying to eavesdrop on
before you can do what you want.




On Fri, Jan 15, 2010 at 8:33 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Hi all,

 

I want to use eavesdrop application.

Playing with it I found that when one tries to eavesdrop caller the feature
works ok.

But when trying to eavesdrop callee eavesdrop attempt failes.

I just updated to the latest version from http://latest.freeswitch.org
<http://latest.freeswitch.org/>  

[freeswitch at freeswitch log]$ fs_cli -x version

FreeSWITCH Version 1.0.5-20100115-0400 (16318M)

 

My setup is as following:

I have 3 internal extensions: 2853, 2840, 2879 (all are xlite).

1. 2879 calls 2840. Then 2853 dials 882879 - eavesdrop worked as expected.

2. 2840 calls 2879. Then 2853 dials 882879 - eavesdrop failed. 

 

I attached logs for both cases.

 

I don't believe it's intended behavior. 

 

Can anybody please advise if it is a configuration or a software problem?

 

Thanks and regards,

Nikolay.


_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From jingwei.yang at gmail.com  Mon Jan 18 01:01:31 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 18 Jan 2010 17:01:31 +0800
Subject: [Freeswitch-users] Questions about mod_fifo
In-Reply-To: <23f91031001172325n2dd0f462ga202abec3a79d019@mail.gmail.com>
References: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
	<23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>
	<13529f9d1001171755v229c2137u4fa5cfe2f30d518@mail.gmail.com>
	<23f91031001171942h42e655c9vbc5ecb4165b7aca9@mail.gmail.com>
	<13529f9d1001172248x1a1ddc8arb7f4aba2e71ffb0f@mail.gmail.com>
	<23f91031001172325n2dd0f462ga202abec3a79d019@mail.gmail.com>
Message-ID: <13529f9d1001180101y6d345297y4d152628d6d6113e@mail.gmail.com>

Yes, I'm able to see the inbound channel created:

freeswitch at localhost.localdomain> show channels
API CALL [show(channels)] output:
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure,hostname,presence_id,presence_data
6d347eaa-aea8-47e3-b915-8cfdf0f8fd9c,inbound,2010-01-19
17:04:22,1263891862,skypiax/interface5,CS_EXECUTE,Jingwei,jingwei.yang,,779,eavesdrop,all,XML,default,L16,16000,L16,16000,,localhost.localdomain,,

Hmmm, may I know how you could tell it's a dialplan problem?

Regards,
-Jingwei

On Mon, Jan 18, 2010 at 3:25 PM, Seven Du <dujinfang at gmail.com> wrote:

> did you happened to run "show channels" ?
>
> clearly it's a dialplan problem other than a fifo one.
>
>
> Dialplan: skypiax/interface8 Regex (PASS) [eavesdrop]
> destination_number(779) =~ /^779$/ break=on-false
> Dialplan: skypiax/interface8 Action answer()
> Dialplan: skypiax/interface8 Action
> set(eavesdrop_indicate_failed=tone_stream://%(500, 0, 320))
> Dialplan: skypiax/interface8 Action
> set(eavesdrop_indicate_new=tone_stream://%(500, 0, 620))
> Dialplan: skypiax/interface8 Action
> set(eavesdrop_indicate_idle=tone_stream://%(250, 0, 920))
> Dialplan: skypiax/interface8 Action eavesdrop(all)
>
>
>
> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> > Logs submitted: http://pastebin.freeswitch.org/11836
> >
> > I was trying to check whether the call had been added into the queue via
> > telnet, but failed to find the fifo events. Here's my simplified
> dialplan:
> >
> > <extension name="in_fifo">
> >    <condition field="destination_number" expression="^779$">
> >      <action application="answer"/>
> >      <action application="fifo" data="myq in undef undef"/>
> >   </condition>
> > </extension>
> >
> > Please advise where went wrong.
> >
> > Thanks and best regards,
> > -Jingwei
> >
> >
> > On Mon, Jan 18, 2010 at 11:42 AM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> better to pastebin your log.
> >>
> >> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> >> > Thanks for replying. This is my dialplan
> >> >
> >> >     <extension name="in_fifo">
> >> >       <condition field="destination_number" expression="^779$">
> >> >         <action application="set" data="fifo_music=$${hold_music}"/>
> >> >         <action application="answer"/>
> >> >         <action application="fifo" data="myq in undef undef"/>
> >> >       </condition>
> >> >     </extension>
> >> >
> >> > And I created a queue in fifo.conf.xml like this
> >> >
> >> >     <fifo name="myq" importance="0">
> >> >       <!--<member timeout="60" simo="1"
> >> > lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
> >> >     </fifo>
> >> >
> >> > However, I'm still not able to see the incoming call get queued.
> >> >
> >> > freeswitch at localhost.localdomain> fifo list myq
> >> > API CALL [fifo(list myq)] output:
> >> > <fifo_report>
> >> >   <fifo name="myq" consumer_count="0" caller_count="0"
> waiting_count="0"
> >> > importance="0">
> >> >     <callers></callers>
> >> >     <consumers></consumers>
> >> >   </fifo>
> >> > </fifo_report>
> >> >
> >> > I tried both mod_skypiax and mod_dingaling, but with the same result.
> >> >
> >> > Regards,
> >> > -Jingwei
> >> >
> >> > On Sun, Jan 17, 2010 at 11:37 AM, Seven Du <dujinfang at gmail.com>
> wrote:
> >> >>
> >> >> 2010/1/15, Jingwei Yang <jingwei.yang at gmail.com>:
> >> >> > Hi Guys,
> >> >> >
> >> >> > I'm implementing an ACD system using ESL and mod_fifo. Based on
> what
> >> >> > Anthony
> >> >> > suggested in this post:
> >> >> >
> >> >> >
> >> >> >
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html
> >> >> >
> >> >> > *You can make an event socket application that listens for FIFO
> >> >> > events
> >> >> > and
> >> >> > keeps track of what FIFOs are currently busy and when there are
> >> >> > people
> >> >> > waiting you can have that script generate a call to a group of SIP
> >> >> > phones so
> >> >> > when the first one answers, it sends them in as an agent where they
> >> >> > can
> >> >> > field the calls.
> >> >> > *
> >> >> >
> >> >> > 1. How should I handle the concurrent issues? If I bridge a user to
> >> >> > two
> >> >> > agents and both of them answers, how does FS take care of this
> >> >> > situation?
> >> >> > Will a slower agent get a busy tone automatically?
> >> >> >
> >> >>
> >> >> I think it just follow the standard originate dialstring rules.
> >> >>
> >> >> > 2. If the socket application is brought up after some users have
> >> >> > called
> >> >> > in,
> >> >> > what command should I use to check the busy queues? fifo list?
> >> >> >
> >> >> Yes. Perhaps you can also check the fifo db, either sqlite or ODBC.
> >> >>
> >> >> > 3. Am I using fifo list and fifo count correctly?
> >> >> >
> >> >> > here's the testing dialplan:
> >> >> >
> >> >> >     <extension name="in_fifo">
> >> >> >       <condition field="destination_number" expression="^779$">
> >> >> >         <action application="set"
> data="fifo_music=$${hold_music}"/>
> >> >> >         <action application="answer"/>
> >> >> >         <action application="fifo" data="*myq *in"/>
> >> >> >       </condition>
> >> >> >     </extension>
> >> >> >
> >> >> > when a call comes in and gets queued, these are the results of some
> >> >> > commands
> >> >> > I tried.
> >> >> >
> >> >> > freeswitch at localhost.localdomain> fifo list
> >> >> > API CALL [fifo(list)] output:
> >> >> > <fifo_report>
> >> >> >   <fifo name="cool_fifo at 192.168.1.104" consumer_count="0"
> >> >> > caller_count="0"
> >> >> > waiting_count="0" importance="0">
> >> >> >     <callers></callers>
> >> >> >     <consumers></consumers>
> >> >> >   </fifo>
> >> >> > </fifo_report>
> >> >> >
> >> >> >
> >> >> > freeswitch at localhost.localdomain> fifo list myq
> >> >> > API CALL [fifo(list myq)] output:
> >> >> > <fifo_report></fifo_report>
> >> >> >
> >> >> >
> >> >> > freeswitch at localhost.localdomain> fifo count myq
> >> >> > API CALL [fifo(count myq)] output:
> >> >> > none
> >> >> >
> >> >> > It seems *myq* doesn't get created at all? Please enlighten.
> >> >> >
> >> >> > Thanks and best regards,
> >> >> > -Jingwei
> >> >> >
> >> >> AFAIK, thant means the channel didn't queued in. Did you see any
> error
> >> >> logs? I think you need to remove the stars in   <action
> >> >> application="fifo" data="*myq *in"/>.
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Mon Jan 18 01:15:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 Jan 2010 04:15:52 -0500
Subject: [Freeswitch-users] IMPORTANT -- Call for bug updates
Message-ID: <7E869910-75D2-40C3-8FA3-849FE1854F4D@jerris.com>

To all FreeSWITCH users-

	If you have bugs open on http://jira.freeswitch.org please login today and post a status update on these bugs (even if they appear to be awaiting comment by the development team).  If you have a patch, please update these patches to svn trunk so that they may be reviewed. I know many patches have been sitting for quite some time but I will make a strong effort to review and merge patches that are ready to go in.  If you have a bug, please update to svn trunk and comment if the issue still exists or is now fixed in trunk.  If you bug has a comment requesting more information, please provide it.  If you don't have any open bugs, and are not currently using recent svn trunk, I would appreciate it if you could carve a little bit of time out of your days and test out trunk.  Feel free to look through jira and find a bug you are interested in and test it as well.  We are working towards having the most stable and feature rich release of FreeSWITCH yet and we need your support and assistance to do so.  As always, if you have a new bug to file, please do so as soon as possible and try to get as much information as possible on the bug.  There are some good guidelines for reporting at http://wiki.freeswitch.org/wiki/Reporting_Bugs  As always, many thanks to the community for all your hard work and support.

Mike

From dujinfang at gmail.com  Mon Jan 18 02:19:07 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 18 Jan 2010 18:19:07 +0800
Subject: [Freeswitch-users] Questions about mod_fifo
In-Reply-To: <13529f9d1001180101y6d345297y4d152628d6d6113e@mail.gmail.com>
References: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
	<23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>
	<13529f9d1001171755v229c2137u4fa5cfe2f30d518@mail.gmail.com>
	<23f91031001171942h42e655c9vbc5ecb4165b7aca9@mail.gmail.com>
	<13529f9d1001172248x1a1ddc8arb7f4aba2e71ffb0f@mail.gmail.com>
	<23f91031001172325n2dd0f462ga202abec3a79d019@mail.gmail.com>
	<13529f9d1001180101y6d345297y4d152628d6d6113e@mail.gmail.com>
Message-ID: <23f91031001180219x3a46111bo8f081c59a78a3e0@mail.gmail.com>

from the log you can see that it routed to eavesdrop 779 but not
fifo_in 779 as you expected. In other words, either there are two 779
entries in your dialplan or your fifo_in 779 dialplan not been loaded
into freeswitch. Did you run reloadxml after you edited the dialplan
file?

2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> Yes, I'm able to see the inbound channel created:
>
> freeswitch at localhost.localdomain> show channels
> API CALL [show(channels)] output:
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure,hostname,presence_id,presence_data
> 6d347eaa-aea8-47e3-b915-8cfdf0f8fd9c,inbound,2010-01-19
> 17:04:22,1263891862,skypiax/interface5,CS_EXECUTE,Jingwei,jingwei.yang,,779,eavesdrop,all,XML,default,L16,16000,L16,16000,,localhost.localdomain,,
>
> Hmmm, may I know how you could tell it's a dialplan problem?
>
> Regards,
> -Jingwei
>
> On Mon, Jan 18, 2010 at 3:25 PM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> did you happened to run "show channels" ?
>>
>> clearly it's a dialplan problem other than a fifo one.
>>
>>
>> Dialplan: skypiax/interface8 Regex (PASS) [eavesdrop]
>> destination_number(779) =~ /^779$/ break=on-false
>> Dialplan: skypiax/interface8 Action answer()
>> Dialplan: skypiax/interface8 Action
>> set(eavesdrop_indicate_failed=tone_stream://%(500, 0, 320))
>> Dialplan: skypiax/interface8 Action
>> set(eavesdrop_indicate_new=tone_stream://%(500, 0, 620))
>> Dialplan: skypiax/interface8 Action
>> set(eavesdrop_indicate_idle=tone_stream://%(250, 0, 920))
>> Dialplan: skypiax/interface8 Action eavesdrop(all)
>>
>>
>>
>> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
>> > Logs submitted: http://pastebin.freeswitch.org/11836
>> >
>> > I was trying to check whether the call had been added into the queue via
>> > telnet, but failed to find the fifo events. Here's my simplified
>> > dialplan:
>> >
>> > <extension name="in_fifo">
>> > ?? <condition field="destination_number" expression="^779$">
>> > ???? <action application="answer"/>
>> > ???? <action application="fifo" data="myq in undef undef"/>
>> > ? </condition>
>> > </extension>
>> >
>> > Please advise where went wrong.
>> >
>> > Thanks and best regards,
>> > -Jingwei
>> >
>> >
>> > On Mon, Jan 18, 2010 at 11:42 AM, Seven Du <dujinfang at gmail.com> wrote:
>> >>
>> >> better to pastebin your log.
>> >>
>> >> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
>> >> > Thanks for replying. This is my dialplan
>> >> >
>> >> > ??? <extension name="in_fifo">
>> >> > ????? <condition field="destination_number" expression="^779$">
>> >> > ??????? <action application="set" data="fifo_music=$${hold_music}"/>
>> >> > ??????? <action application="answer"/>
>> >> > ??????? <action application="fifo" data="myq in undef undef"/>
>> >> > ????? </condition>
>> >> > ??? </extension>
>> >> >
>> >> > And I created a queue in fifo.conf.xml like this
>> >> >
>> >> > ??? <fifo name="myq" importance="0">
>> >> > ????? <!--<member timeout="60" simo="1"
>> >> > lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
>> >> > ??? </fifo>
>> >> >
>> >> > However, I'm still not able to see the incoming call get queued.
>> >> >
>> >> > freeswitch at localhost.localdomain> fifo list myq
>> >> > API CALL [fifo(list myq)] output:
>> >> > <fifo_report>
>> >> > ? <fifo name="myq" consumer_count="0" caller_count="0"
>> >> > waiting_count="0"
>> >> > importance="0">
>> >> > ??? <callers></callers>
>> >> > ??? <consumers></consumers>
>> >> > ? </fifo>
>> >> > </fifo_report>
>> >> >
>> >> > I tried both mod_skypiax and mod_dingaling, but with the same result.
>> >> >
>> >> > Regards,
>> >> > -Jingwei
>> >> >
>> >> > On Sun, Jan 17, 2010 at 11:37 AM, Seven Du <dujinfang at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> 2010/1/15, Jingwei Yang <jingwei.yang at gmail.com>:
>> >> >> > Hi Guys,
>> >> >> >
>> >> >> > I'm implementing an ACD system using ESL and mod_fifo. Based on
>> >> >> > what
>> >> >> > Anthony
>> >> >> > suggested in this post:
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html
>> >> >> >
>> >> >> > *You can make an event socket application that listens for FIFO
>> >> >> > events
>> >> >> > and
>> >> >> > keeps track of what FIFOs are currently busy and when there are
>> >> >> > people
>> >> >> > waiting you can have that script generate a call to a group of SIP
>> >> >> > phones so
>> >> >> > when the first one answers, it sends them in as an agent where
>> >> >> > they
>> >> >> > can
>> >> >> > field the calls.
>> >> >> > *
>> >> >> >
>> >> >> > 1. How should I handle the concurrent issues? If I bridge a user
>> >> >> > to
>> >> >> > two
>> >> >> > agents and both of them answers, how does FS take care of this
>> >> >> > situation?
>> >> >> > Will a slower agent get a busy tone automatically?
>> >> >> >
>> >> >>
>> >> >> I think it just follow the standard originate dialstring rules.
>> >> >>
>> >> >> > 2. If the socket application is brought up after some users have
>> >> >> > called
>> >> >> > in,
>> >> >> > what command should I use to check the busy queues? fifo list?
>> >> >> >
>> >> >> Yes. Perhaps you can also check the fifo db, either sqlite or ODBC.
>> >> >>
>> >> >> > 3. Am I using fifo list and fifo count correctly?
>> >> >> >
>> >> >> > here's the testing dialplan:
>> >> >> >
>> >> >> > ? ? <extension name="in_fifo">
>> >> >> > ? ? ? <condition field="destination_number" expression="^779$">
>> >> >> > ? ? ? ? <action application="set"
>> >> >> > data="fifo_music=$${hold_music}"/>
>> >> >> > ? ? ? ? <action application="answer"/>
>> >> >> > ? ? ? ? <action application="fifo" data="*myq *in"/>
>> >> >> > ? ? ? </condition>
>> >> >> > ? ? </extension>
>> >> >> >
>> >> >> > when a call comes in and gets queued, these are the results of
>> >> >> > some
>> >> >> > commands
>> >> >> > I tried.
>> >> >> >
>> >> >> > freeswitch at localhost.localdomain> fifo list
>> >> >> > API CALL [fifo(list)] output:
>> >> >> > <fifo_report>
>> >> >> > ? <fifo name="cool_fifo at 192.168.1.104" consumer_count="0"
>> >> >> > caller_count="0"
>> >> >> > waiting_count="0" importance="0">
>> >> >> > ? ? <callers></callers>
>> >> >> > ? ? <consumers></consumers>
>> >> >> > ? </fifo>
>> >> >> > </fifo_report>
>> >> >> >
>> >> >> >
>> >> >> > freeswitch at localhost.localdomain> fifo list myq
>> >> >> > API CALL [fifo(list myq)] output:
>> >> >> > <fifo_report></fifo_report>
>> >> >> >
>> >> >> >
>> >> >> > freeswitch at localhost.localdomain> fifo count myq
>> >> >> > API CALL [fifo(count myq)] output:
>> >> >> > none
>> >> >> >
>> >> >> > It seems *myq* doesn't get created at all? Please enlighten.
>> >> >> >
>> >> >> > Thanks and best regards,
>> >> >> > -Jingwei
>> >> >> >
>> >> >> AFAIK, thant means the channel didn't queued in. Did you see any
>> >> >> error
>> >> >> logs? I think you need to remove the stars in ? <action
>> >> >> application="fifo" data="*myq *in"/>.
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jingwei.yang at gmail.com  Mon Jan 18 02:39:54 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 18 Jan 2010 18:39:54 +0800
Subject: [Freeswitch-users] Questions about mod_fifo
In-Reply-To: <23f91031001180219x3a46111bo8f081c59a78a3e0@mail.gmail.com>
References: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
	<23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>
	<13529f9d1001171755v229c2137u4fa5cfe2f30d518@mail.gmail.com>
	<23f91031001171942h42e655c9vbc5ecb4165b7aca9@mail.gmail.com>
	<13529f9d1001172248x1a1ddc8arb7f4aba2e71ffb0f@mail.gmail.com>
	<23f91031001172325n2dd0f462ga202abec3a79d019@mail.gmail.com>
	<13529f9d1001180101y6d345297y4d152628d6d6113e@mail.gmail.com>
	<23f91031001180219x3a46111bo8f081c59a78a3e0@mail.gmail.com>
Message-ID: <13529f9d1001180239j70a00cefo334539af9304725c@mail.gmail.com>

Yes, you're right!! There's a eavesdrop 779 matched first. Thank you so
much!

On Mon, Jan 18, 2010 at 6:19 PM, Seven Du <dujinfang at gmail.com> wrote:

> from the log you can see that it routed to eavesdrop 779 but not
> fifo_in 779 as you expected. In other words, either there are two 779
> entries in your dialplan or your fifo_in 779 dialplan not been loaded
> into freeswitch. Did you run reloadxml after you edited the dialplan
> file?
>
> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> > Yes, I'm able to see the inbound channel created:
> >
> > freeswitch at localhost.localdomain> show channels
> > API CALL [show(channels)] output:
> >
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure,hostname,presence_id,presence_data
> > 6d347eaa-aea8-47e3-b915-8cfdf0f8fd9c,inbound,2010-01-19
> >
> 17:04:22,1263891862,skypiax/interface5,CS_EXECUTE,Jingwei,jingwei.yang,,779,eavesdrop,all,XML,default,L16,16000,L16,16000,,localhost.localdomain,,
> >
> > Hmmm, may I know how you could tell it's a dialplan problem?
> >
> > Regards,
> > -Jingwei
> >
> > On Mon, Jan 18, 2010 at 3:25 PM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> did you happened to run "show channels" ?
> >>
> >> clearly it's a dialplan problem other than a fifo one.
> >>
> >>
> >> Dialplan: skypiax/interface8 Regex (PASS) [eavesdrop]
> >> destination_number(779) =~ /^779$/ break=on-false
> >> Dialplan: skypiax/interface8 Action answer()
> >> Dialplan: skypiax/interface8 Action
> >> set(eavesdrop_indicate_failed=tone_stream://%(500, 0, 320))
> >> Dialplan: skypiax/interface8 Action
> >> set(eavesdrop_indicate_new=tone_stream://%(500, 0, 620))
> >> Dialplan: skypiax/interface8 Action
> >> set(eavesdrop_indicate_idle=tone_stream://%(250, 0, 920))
> >> Dialplan: skypiax/interface8 Action eavesdrop(all)
> >>
> >>
> >>
> >> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> >> > Logs submitted: http://pastebin.freeswitch.org/11836
> >> >
> >> > I was trying to check whether the call had been added into the queue
> via
> >> > telnet, but failed to find the fifo events. Here's my simplified
> >> > dialplan:
> >> >
> >> > <extension name="in_fifo">
> >> >    <condition field="destination_number" expression="^779$">
> >> >      <action application="answer"/>
> >> >      <action application="fifo" data="myq in undef undef"/>
> >> >   </condition>
> >> > </extension>
> >> >
> >> > Please advise where went wrong.
> >> >
> >> > Thanks and best regards,
> >> > -Jingwei
> >> >
> >> >
> >> > On Mon, Jan 18, 2010 at 11:42 AM, Seven Du <dujinfang at gmail.com>
> wrote:
> >> >>
> >> >> better to pastebin your log.
> >> >>
> >> >> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> >> >> > Thanks for replying. This is my dialplan
> >> >> >
> >> >> >     <extension name="in_fifo">
> >> >> >       <condition field="destination_number" expression="^779$">
> >> >> >         <action application="set"
> data="fifo_music=$${hold_music}"/>
> >> >> >         <action application="answer"/>
> >> >> >         <action application="fifo" data="myq in undef undef"/>
> >> >> >       </condition>
> >> >> >     </extension>
> >> >> >
> >> >> > And I created a queue in fifo.conf.xml like this
> >> >> >
> >> >> >     <fifo name="myq" importance="0">
> >> >> >       <!--<member timeout="60" simo="1"
> >> >> > lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
> >> >> >     </fifo>
> >> >> >
> >> >> > However, I'm still not able to see the incoming call get queued.
> >> >> >
> >> >> > freeswitch at localhost.localdomain> fifo list myq
> >> >> > API CALL [fifo(list myq)] output:
> >> >> > <fifo_report>
> >> >> >   <fifo name="myq" consumer_count="0" caller_count="0"
> >> >> > waiting_count="0"
> >> >> > importance="0">
> >> >> >     <callers></callers>
> >> >> >     <consumers></consumers>
> >> >> >   </fifo>
> >> >> > </fifo_report>
> >> >> >
> >> >> > I tried both mod_skypiax and mod_dingaling, but with the same
> result.
> >> >> >
> >> >> > Regards,
> >> >> > -Jingwei
> >> >> >
> >> >> > On Sun, Jan 17, 2010 at 11:37 AM, Seven Du <dujinfang at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >> 2010/1/15, Jingwei Yang <jingwei.yang at gmail.com>:
> >> >> >> > Hi Guys,
> >> >> >> >
> >> >> >> > I'm implementing an ACD system using ESL and mod_fifo. Based on
> >> >> >> > what
> >> >> >> > Anthony
> >> >> >> > suggested in this post:
> >> >> >> >
> >> >> >> >
> >> >> >> >
> >> >> >> >
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html
> >> >> >> >
> >> >> >> > *You can make an event socket application that listens for FIFO
> >> >> >> > events
> >> >> >> > and
> >> >> >> > keeps track of what FIFOs are currently busy and when there are
> >> >> >> > people
> >> >> >> > waiting you can have that script generate a call to a group of
> SIP
> >> >> >> > phones so
> >> >> >> > when the first one answers, it sends them in as an agent where
> >> >> >> > they
> >> >> >> > can
> >> >> >> > field the calls.
> >> >> >> > *
> >> >> >> >
> >> >> >> > 1. How should I handle the concurrent issues? If I bridge a user
> >> >> >> > to
> >> >> >> > two
> >> >> >> > agents and both of them answers, how does FS take care of this
> >> >> >> > situation?
> >> >> >> > Will a slower agent get a busy tone automatically?
> >> >> >> >
> >> >> >>
> >> >> >> I think it just follow the standard originate dialstring rules.
> >> >> >>
> >> >> >> > 2. If the socket application is brought up after some users have
> >> >> >> > called
> >> >> >> > in,
> >> >> >> > what command should I use to check the busy queues? fifo list?
> >> >> >> >
> >> >> >> Yes. Perhaps you can also check the fifo db, either sqlite or
> ODBC.
> >> >> >>
> >> >> >> > 3. Am I using fifo list and fifo count correctly?
> >> >> >> >
> >> >> >> > here's the testing dialplan:
> >> >> >> >
> >> >> >> >     <extension name="in_fifo">
> >> >> >> >       <condition field="destination_number" expression="^779$">
> >> >> >> >         <action application="set"
> >> >> >> > data="fifo_music=$${hold_music}"/>
> >> >> >> >         <action application="answer"/>
> >> >> >> >         <action application="fifo" data="*myq *in"/>
> >> >> >> >       </condition>
> >> >> >> >     </extension>
> >> >> >> >
> >> >> >> > when a call comes in and gets queued, these are the results of
> >> >> >> > some
> >> >> >> > commands
> >> >> >> > I tried.
> >> >> >> >
> >> >> >> > freeswitch at localhost.localdomain> fifo list
> >> >> >> > API CALL [fifo(list)] output:
> >> >> >> > <fifo_report>
> >> >> >> >   <fifo name="cool_fifo at 192.168.1.104" consumer_count="0"
> >> >> >> > caller_count="0"
> >> >> >> > waiting_count="0" importance="0">
> >> >> >> >     <callers></callers>
> >> >> >> >     <consumers></consumers>
> >> >> >> >   </fifo>
> >> >> >> > </fifo_report>
> >> >> >> >
> >> >> >> >
> >> >> >> > freeswitch at localhost.localdomain> fifo list myq
> >> >> >> > API CALL [fifo(list myq)] output:
> >> >> >> > <fifo_report></fifo_report>
> >> >> >> >
> >> >> >> >
> >> >> >> > freeswitch at localhost.localdomain> fifo count myq
> >> >> >> > API CALL [fifo(count myq)] output:
> >> >> >> > none
> >> >> >> >
> >> >> >> > It seems *myq* doesn't get created at all? Please enlighten.
> >> >> >> >
> >> >> >> > Thanks and best regards,
> >> >> >> > -Jingwei
> >> >> >> >
> >> >> >> AFAIK, thant means the channel didn't queued in. Did you see any
> >> >> >> error
> >> >> >> logs? I think you need to remove the stars in   <action
> >> >> >> application="fifo" data="*myq *in"/>.
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From n.geordzhev at gmail.com  Mon Jan 18 03:00:03 2010
From: n.geordzhev at gmail.com (Nikolai Geordzhev)
Date: Mon, 18 Jan 2010 13:00:03 +0200
Subject: [Freeswitch-users] database disk image is malformed ISSUE
Message-ID: <c8df2a131001180300m25419dd8y65ff80ce64824cef@mail.gmail.com>

Hi Guys,

I have an issue playing with FS as a registrar server. I have made some
tests with 3000 subscribers registering every 600 seconds running for days
and everything went fine.
Then I have tried with 6000 subscribers registering every 3600 seconds and
after some time ( between 1 and 2 days) I received  [ERR]
switch_core_sqldb.c:662 SQL ERR [database disk image is malformed] message
in the freeswitch.log file. The only solution I have found is deleting the
db folder ( mounted in tmpfs) and restarting the application.
When I measure the packets/sec rates of both setups i see 13 pack/s for the
first setup and 7 pack/s for the second one. Can somebody advise what can
cause this Error and if there is some kind of a solution.

Thanks and best regards
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From lakindia89 at gmail.com  Mon Jan 18 03:22:20 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 18 Jan 2010 16:52:20 +0530
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <7d79b3931001132357i36bbb482jdf2bbdd3aea2a583@mail.gmail.com>
References: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
	<191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>
	<7d79b3931001132357i36bbb482jdf2bbdd3aea2a583@mail.gmail.com>
Message-ID: <7d79b3931001180322p74a1227qe0c2199a77cbfe2@mail.gmail.com>

Here is the result

Program:
require ESL;
use IO::Socket::INET;
use Data::Dumper;

my $ip = "192.168.1.222";
my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
Proto => 'tcp',  Listen => 1,  Reuse => 1 );
die "Could not create socket: $!\n" unless $sock;


for(;;) {
    my $new_sock = $sock->accept();
    next if (not defined ($new_sock));
    my $pid = fork();
    if ($pid) {
        close($new_sock);
        next;
    }
    print "CHILD PID: $$\n";
    &register_Signals_Child();
    sub register_Signals_Child() {
        foreach ( keys %SIG ) {
            $SIG{$_} = 'Handler';
        }
    }

    sub Handler() {
        my $handle=$_[0];
        if($handle eq "INT") {
            print "CHILD $$: SIGNAL SIG$handle is generated\n";
        }
        else
        {
            print "CHILD $$: Received $handle\n";
        }
    }
    my $host = $new_sock->sockhost();
    my $fd = fileno($new_sock);

    my $con = new ESL::ESLconnection($fd);
    my $info = $con->getInfo();

    my $uuid = $info->getHeader("unique-id");

    printf "Connected call %s, from %s\n", $uuid,
$info->getHeader("caller-caller-id-number");
    my $r=$con->execute("answer");
    print Dumper $r;
    $con->events("plain","all");
    my
$re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
    while($con->connected()) {
        my $e = $con->recvEvent();

        if ($e) {
            my $name = $e->getHeader("event-name");
            print "EVENT [$name]\n";
            if ($name eq "DTMF") {
                my $digit = $e->getHeader("dtmf-digit");
                print "$digit\n";
            }
        }
    }
    close($new_sock);
}

I executed the program and the following things were printed

CHILD PID: 6778
Connected call e0d1001a-03f4-11df-b002-db488337e0ea, from 1001
$VAR1 = 0;
EVENT [CHANNEL_EXECUTE]
EVENT [CHANNEL_ANSWER]
EVENT [CHANNEL_EXECUTE_COMPLETE]
EVENT [COMMAND]
EVENT [CHANNEL_EXECUTE]
EVENT [HEARTBEAT]
EVENT [RE_SCHEDULE]
EVENT [CHANNEL_EXECUTE_COMPLETE]

Then from another shell I executed kill -2 6778, the result is follows
CHILD 6778: SIGNAL SIGINT is generated
EVENT [SERVER_DISCONNECTED]

But the child process is still running as expected.
But  I don't know why I received SERVER_DISCONNECTED from freeswitch???





On Thu, Jan 14, 2010 at 1:27 PM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> I taught the signal handler will be inherited by the child process. It also
> does like that.
> After making a call, If I press ctrl + c, the above program printed
> PARENT PID: Signal SIGINT is generated
> CHILD PID: Signal SIGINT is generated.
>
> So I think the sigal handlers will be inherited to the child.
> Anyway I'll also try registering signal handlers in child also, and then
> I'll come back with that result.
>
> Thanks....
> On Wed, Jan 13, 2010 at 9:48 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> you would have to register signals in your child process too
>>
>>   On Wed, Jan 13, 2010 at 3:13 AM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>>  Hi all,
>>>
>>> I've done a sample program (In perl ESL) , which play a file to the
>>> caller and then it will call recvEvent() and print the event name. I've
>>> handled signals also.
>>>
>>> When I send SIGINT to my program (Perl), the signal handler is called and
>>> I can see the print output. But in the same time,  I received
>>> SERVER_DISCONNECTED from freeswitch as event.
>>>
>>> I don't know why I received SERVER_DISCONNECTED from freeswitch. Is it
>>> because, the recvEvent() from perl internally  calls the recvevent function
>>> in the Esl.c and when it waits to receive the information from socket, the
>>> signal occurred???
>>>
>>> Please clarify me!!
>>>
>>> Here is my program
>>> require ESL;
>>> use IO::Socket::INET;
>>> use Data::Dumper;
>>>
>>> my $ip = "192.168.1.222";
>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>> die "Could not create socket: $!\n" unless $sock;
>>> &register_Signals();
>>>
>>> for(;;) {
>>>     my $new_sock = $sock->accept();
>>>     next if (not defined ($new_sock));
>>>     my $pid = fork();
>>>     if ($pid) {
>>>         close($new_sock);
>>>         next;
>>>     }
>>>     print "CHILD PID: $$\n";
>>>     my $host = $new_sock->sockhost();
>>>     my $fd = fileno($new_sock);
>>>
>>>     my $con = new ESL::ESLconnection($fd);
>>>     my $info = $con->getInfo();
>>>
>>>     my $uuid = $info->getHeader("unique-id");
>>>
>>>     printf "Connected call %s, from %s\n", $uuid,
>>> $info->getHeader("caller-caller-id-number");
>>>     my $r=$con->execute("answer");
>>>     print Dumper $r;
>>>     $con->events("plain","all");
>>>     my
>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>
>>>     while($con->connected()) {
>>>         my $e = $con->recvEvent();
>>>
>>>         if ($e) {
>>>             my $name = $e->getHeader("event-name");
>>>             print "EVENT [$name]\n";
>>>             if ($name eq "DTMF") {
>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>                 print "$digit\n";
>>>             }
>>>         }
>>>     }
>>>     close($new_sock);
>>> }
>>> sub register_Signals() {
>>>     foreach ( keys %SIG ) {
>>>         $SIG{$_} = 'sig_Handler';
>>>     }
>>> }
>>>
>>> sub sig_Handler() {
>>>     my $handle=$_[0];
>>>     if($handle eq "INT") {
>>>         print "$$: SIGNAL SIG$handle is generated\n";
>>>     }
>>> }
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From mike at jerris.com  Mon Jan 18 03:28:04 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 Jan 2010 06:28:04 -0500
Subject: [Freeswitch-users] database disk image is malformed ISSUE
In-Reply-To: <c8df2a131001180300m25419dd8y65ff80ce64824cef@mail.gmail.com>
References: <c8df2a131001180300m25419dd8y65ff80ce64824cef@mail.gmail.com>
Message-ID: <206AA031-94DA-4AAB-8761-90A06A80E9F5@jerris.com>

sqlite doesn't scale.  If you want to do anything serious, use odbc and a real database instead.

Mike

On Jan 18, 2010, at 6:00 AM, Nikolai Geordzhev wrote:

> Hi Guys,
> 
> I have an issue playing with FS as a registrar server. I have made some tests with 3000 subscribers registering every 600 seconds running for days and everything went fine. 
> Then I have tried with 6000 subscribers registering every 3600 seconds and after some time ( between 1 and 2 days) I received  [ERR] switch_core_sqldb.c:662 SQL ERR [database disk image is malformed] message in the freeswitch.log file. The only solution I have found is deleting the db folder ( mounted in tmpfs) and restarting the application.
> When I measure the packets/sec rates of both setups i see 13 pack/s for the first setup and 7 pack/s for the second one. Can somebody advise what can cause this Error and if there is some kind of a solution. 



From Prometheus001 at gmx.net  Mon Jan 18 03:56:36 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 18 Jan 2010 12:56:36 +0100
Subject: [Freeswitch-users] Multiple registrations: Only calling
 the	first element in the list in this mode
In-Reply-To: <4B50C757.3050901@gmx.net>
References: <4B4DADD5.3010507@gmx.net>
	<b39273531001130450k16af27d2v209a300f441d39d8@mail.gmail.com>
	<4B50C757.3050901@gmx.net>
Message-ID: <4B544C74.1000301@gmx.net>

I just found out that this method unfortunately had a side effect,

when I use
    ":_:" the caller does not receive a dialtone
If I change to
    "," the dialtone is there

Any clue how I can work around this?

Best regards
Peter

Peter P GMX schrieb:
> Thanks Rupa,
>
> this worked. I have documented this in the wiki:
>     http://wiki.freeswitch.org/wiki/Ring_group
>
> Best regards
> Peter
>
> Rupa Schomaker schrieb:
>   
>> Try:
>>
>> bridge user/100 at domain:_:user/101 at domain:_:user/102 at domain
>>
>> Then document it up if it works.
>>
>> On Wed, Jan 13, 2010 at 5:26 AM, Peter P GMX <Prometheus001 at gmx.net
>> <mailto:Prometheus001 at gmx.net>> wrote:
>>
>>     Hello,
>>
>>     I habe the following behaviour
>>
>>     when I call a user which is registered twice with 2 phones via
>>        bridge user/100 at domain
>>     both phones are ringing. This is correct as I allow multiple
>>     registrations in a profile
>>
>>     However when I call multiple endpoints via
>>        bridge user/100 at domain,user/101 at domain,user/102 at domain
>>     only one phone with number100 is ringing.
>>
>>     Console log shows "Only calling the first element in the list in this
>>     mode.":
>>     2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734
>>     variable
>>     string 0 = [presence_id=100 at domain]
>>     2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734
>>     variable
>>     string 1 = [transfer_fallback_extension=100]
>>     2010-01-12 19:52:18.236361 [WARNING] switch_ivr_originate.c:2048 Only
>>     calling the first element in the list in this mode.
>>     2010-01-12 19:52:18.236361 [NOTICE] switch_channel.c:613 New Channel
>>     sofia/internal/sip:100 at 10.11.12.203:2048
>>     <http://sip:100 at 10.11.12.203:2048>
>>     [9b95fcdc-ffab-11de-9ba2-13b2daa7ce61]
>>
>>     Is there any way to work around this? I need all phones to be
>>     ringing in
>>     this scenario.
>>
>>     Best regards
>>     Peter
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> -Rupa
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>   
>>     
>
>   


From rupa at rupa.com  Mon Jan 18 04:33:34 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 18 Jan 2010 06:33:34 -0600
Subject: [Freeswitch-users] Multiple registrations: Only calling the
	first element in the list in this mode
In-Reply-To: <4B544C74.1000301@gmx.net>
References: <4B4DADD5.3010507@gmx.net>
	<b39273531001130450k16af27d2v209a300f441d39d8@mail.gmail.com>
	<4B50C757.3050901@gmx.net> <4B544C74.1000301@gmx.net>
Message-ID: <b39273531001180433m79bced04ja60225615588577c@mail.gmail.com>

Can you open a ticket on jira for that?

By dialtone, do you mean ringback (call progress)?

On Mon, Jan 18, 2010 at 5:56 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> I just found out that this method unfortunately had a side effect,
>
> when I use
>    ":_:" the caller does not receive a dialtone
> If I change to
>    "," the dialtone is there
>
> Any clue how I can work around this?
>
> Best regards
> Peter
>
> Peter P GMX schrieb:
> > Thanks Rupa,
> >
> > this worked. I have documented this in the wiki:
> >     http://wiki.freeswitch.org/wiki/Ring_group
> >
> > Best regards
> > Peter
> >
> > Rupa Schomaker schrieb:
> >
> >> Try:
> >>
> >> bridge user/100 at domain:_:user/101 at domain:_:user/102 at domain
> >>
> >> Then document it up if it works.
> >>
> >> On Wed, Jan 13, 2010 at 5:26 AM, Peter P GMX <Prometheus001 at gmx.net
> >> <mailto:Prometheus001 at gmx.net>> wrote:
> >>
> >>     Hello,
> >>
> >>     I habe the following behaviour
> >>
> >>     when I call a user which is registered twice with 2 phones via
> >>        bridge user/100 at domain
> >>     both phones are ringing. This is correct as I allow multiple
> >>     registrations in a profile
> >>
> >>     However when I call multiple endpoints via
> >>        bridge user/100 at domain,user/101 at domain,user/102 at domain
> >>     only one phone with number100 is ringing.
> >>
> >>     Console log shows "Only calling the first element in the list in
> this
> >>     mode.":
> >>     2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734
> >>     variable
> >>     string 0 = [presence_id=100 at domain]
> >>     2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734
> >>     variable
> >>     string 1 = [transfer_fallback_extension=100]
> >>     2010-01-12 19:52:18.236361 [WARNING] switch_ivr_originate.c:2048
> Only
> >>     calling the first element in the list in this mode.
> >>     2010-01-12 19:52:18.236361 [NOTICE] switch_channel.c:613 New Channel
> >>     sofia/internal/sip:100 at 10.11.12.203:2048
> >>     <http://sip:100 at 10.11.12.203:2048>
> >>     [9b95fcdc-ffab-11de-9ba2-13b2daa7ce61]
> >>
> >>     Is there any way to work around this? I need all phones to be
> >>     ringing in
> >>     this scenario.
> >>
> >>     Best regards
> >>     Peter
> >>
> >>     _______________________________________________
> >>     FreeSWITCH-users mailing list
> >>     FreeSWITCH-users at lists.freeswitch.org
> >>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>     http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >> --
> >> -Rupa
> >> ------------------------------------------------------------------------
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From Prometheus001 at gmx.net  Mon Jan 18 05:48:31 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 18 Jan 2010 14:48:31 +0100
Subject: [Freeswitch-users] Multiple registrations: Only calling
 the	first element in the list in this mode
In-Reply-To: <b39273531001180433m79bced04ja60225615588577c@mail.gmail.com>
References: <4B4DADD5.3010507@gmx.net>	<b39273531001130450k16af27d2v209a300f441d39d8@mail.gmail.com>	<4B50C757.3050901@gmx.net>
	<4B544C74.1000301@gmx.net>
	<b39273531001180433m79bced04ja60225615588577c@mail.gmail.com>
Message-ID: <4B5466AF.2060102@gmx.net>

Hello Rupa,

I've opened a JIRA for this. And yes, I mean ringback.

Best regards
Peter

Rupa Schomaker schrieb:
> Can you open a ticket on jira for that?  
>
> By dialtone, do you mean ringback (call progress)?
>
> On Mon, Jan 18, 2010 at 5:56 AM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     I just found out that this method unfortunately had a side effect,
>
>     when I use
>        ":_:" the caller does not receive a dialtone
>     If I change to
>        "," the dialtone is there
>
>     Any clue how I can work around this?
>
>     Best regards
>     Peter
>
>     Peter P GMX schrieb:
>     > Thanks Rupa,
>     >
>     > this worked. I have documented this in the wiki:
>     >     http://wiki.freeswitch.org/wiki/Ring_group
>     >
>     > Best regards
>     > Peter
>     >
>     > Rupa Schomaker schrieb:
>     >
>     >> Try:
>     >>
>     >> bridge user/100 at domain:_:user/101 at domain:_:user/102 at domain
>     >>
>     >> Then document it up if it works.
>     >>
>     >> On Wed, Jan 13, 2010 at 5:26 AM, Peter P GMX
>     <Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>
>     >> <mailto:Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>>>
>     wrote:
>     >>
>     >>     Hello,
>     >>
>     >>     I habe the following behaviour
>     >>
>     >>     when I call a user which is registered twice with 2 phones via
>     >>        bridge user/100 at domain
>     >>     both phones are ringing. This is correct as I allow multiple
>     >>     registrations in a profile
>     >>
>     >>     However when I call multiple endpoints via
>     >>        bridge user/100 at domain,user/101 at domain,user/102 at domain
>     >>     only one phone with number100 is ringing.
>     >>
>     >>     Console log shows "Only calling the first element in the
>     list in this
>     >>     mode.":
>     >>     2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734
>     >>     variable
>     >>     string 0 = [presence_id=100 at domain]
>     >>     2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734
>     >>     variable
>     >>     string 1 = [transfer_fallback_extension=100]
>     >>     2010-01-12 19:52:18.236361 [WARNING]
>     switch_ivr_originate.c:2048 Only
>     >>     calling the first element in the list in this mode.
>     >>     2010-01-12 19:52:18.236361 [NOTICE] switch_channel.c:613
>     New Channel
>     >>     sofia/internal/sip:100 at 10.11.12.203:2048
>     <http://sip:100 at 10.11.12.203:2048>
>     >>     <http://sip:100 at 10.11.12.203:2048>
>     >>     [9b95fcdc-ffab-11de-9ba2-13b2daa7ce61]
>     >>
>     >>     Is there any way to work around this? I need all phones to be
>     >>     ringing in
>     >>     this scenario.
>     >>
>     >>     Best regards
>     >>     Peter
>     >>
>     >>     _______________________________________________
>     >>     FreeSWITCH-users mailing list
>     >>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >>     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>    
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >>     http://www.freeswitch.org
>     >>
>     >>
>     >>
>     >>
>     >> --
>     >> -Rupa
>     >>
>     ------------------------------------------------------------------------
>     >>
>     >> _______________________________________________
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >>
>     >>
>     >
>     >
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> -Rupa
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From brian at freeswitch.org  Mon Jan 18 06:22:18 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 Jan 2010 08:22:18 -0600
Subject: [Freeswitch-users] eavesdrop problem?
In-Reply-To: <20100118072655.F29E011F68@mail.nstel.ru>
References: <20100118072655.F29E011F68@mail.nstel.ru>
Message-ID: <9C65429E-482D-4438-89CE-2CE2E5D73355@freeswitch.org>

It'll overwrite on the next call... or reboot.

You can use the api_hangup_hook to remove it if you wish... see variables page on wiki.

/b

On Jan 18, 2010, at 1:26 AM, Nikolay Kondratyev wrote:

> By the way, should I do something to remove a uuid from the database when the call is ended? Or will it be removed automatically?
> Thanks and regards,
> Nikolay.

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From brian at freeswitch.org  Mon Jan 18 06:27:17 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 Jan 2010 08:27:17 -0600
Subject: [Freeswitch-users] Multiple registrations: Only calling
	the	first element in the list in this mode
In-Reply-To: <4B5466AF.2060102@gmx.net>
References: <4B4DADD5.3010507@gmx.net>	<b39273531001130450k16af27d2v209a300f441d39d8@mail.gmail.com>	<4B50C757.3050901@gmx.net>
	<4B544C74.1000301@gmx.net>
	<b39273531001180433m79bced04ja60225615588577c@mail.gmail.com>
	<4B5466AF.2060102@gmx.net>
Message-ID: <B7AD5B44-7FC6-4703-ADD3-4F13851C9CBF@freeswitch.org>

In this mode you'll have to set the ringback variable.. did you do that?

/b

On Jan 18, 2010, at 7:48 AM, Peter P GMX wrote:

> Hello Rupa,
> 
> I've opened a JIRA for this. And yes, I mean ringback.
> 
> Best regards
> Peter



From lart2150 at gmail.com  Mon Jan 18 08:00:52 2010
From: lart2150 at gmail.com (Brian Engert)
Date: Mon, 18 Jan 2010 10:00:52 -0600
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <59daa2cd1001161205h3f6feeddla0be8e1fa9311666@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
	<776BA929-D7EC-4366-AF49-348F0E11488B@shaw.ca>
	<59daa2cd1001161205h3f6feeddla0be8e1fa9311666@mail.gmail.com>
Message-ID: <a88e84c21001180800p24e7a182m61ff689827278882@mail.gmail.com>

on a related subject I'm working on a web->fax solution and would be
willing to share my code when I'm done but I'm stuck on getting the
fax status for api_hangup_hook.  I wish I could do something like this

originate {fax_ident=312-123-4567,fax_header='Soliant Consulting -
Brian',api_hangup_hook='system /usr/bin/php
/usr/local/freeswitch/scripts/sentFax.php bob at smith.com
${fax_result_code} ${fax_result_text}
${fax_document_total_pages}'}sofia/gateway/outbound.fax/1004
&txfax(/tmp/fax.tiff)

However freeswitch does not seem to like channel variables inside
api_hangup_hook.  I've thought about using python or lua to send the
fax instead of originate but I don't know how to do the txfax call.

On Sat, Jan 16, 2010 at 2:05 PM, Costa Zikalala
<costa.zikalala at gmail.com> wrote:
> Yes Steve, I'm already using that for fax2email.
> I'm now trying to do things in the opposite direction.
>
> *A realy great script by the way*
>
> Thanks
> Costa
>
>
> 2010/1/16 Steve Steffler <stevesteffler at shaw.ca>
>>
>> Costa,
>> I wrote this script to handle fax2email (but not email2fax). ?It uses
>> variables you set in the dialplan in advance for the email address for that
>> fax DID.
>> http://steffler.info/wp-content/uploads/2009/06/process-rxfax.py
>> Regards,
>> Steve
>> On Jan 14, 2010, at 7:54 AM, Costa Zikalala wrote:
>>
>> Hi All
>>
>> Has anyone worked on a email2fax script for mod_fax?
>> If not how much would it cost for some genius here to quickly whip-up one?
>>
>> Ideally both email2fax and fax2email should come standard with mod_fax or
>> is this not possible?
>>
>> Thanks
>> Costa
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From devel at thom.fr.eu.org  Mon Jan 18 08:03:47 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Mon, 18 Jan 2010 17:03:47 +0100
Subject: [Freeswitch-users] voicemail->email
Message-ID: <79fa7b97c59bbfd35a68a88ce667f82c@thom.fr.eu.org>



Sorry to come back on this topic, but I could not manage to fix this,
and I don't know what to do. 

voicemail to email was working on FS 1.0.3,
but not anymore since upgrade to 1.0.4 (now running1.0.5-20100112-0400
(hacked)) 

When FS sneds the message, sendmail segfaults. Running FS as
root or standard user does not change this. Modifying FS config to run a
script like 

exec tee -a /tmp/fsmail.log | /usr/sbin/sendmail -O
LogLevel=7 -t >> /tmp/fsmail.log 2>&1 

make sendmail segfaults too. 

then
doing su -l freeeswitch then cat /tmp/fsmail.log | /usr/sbin/sendmail -f
freeswitch at mydomain.com -t calle at mydomain.com succeeds. 

Don't know what
to look for now. 

The system was not (except freeswitch) updated between
running with FS 1.0.3 and 1.0.4. 

Anybody can help ? 

Fran?ois 

 
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From anthony.minessale at gmail.com  Mon Jan 18 08:24:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 18 Jan 2010 10:24:00 -0600
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <7d79b3931001180322p74a1227qe0c2199a77cbfe2@mail.gmail.com>
References: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
	<191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>
	<7d79b3931001132357i36bbb482jdf2bbdd3aea2a583@mail.gmail.com>
	<7d79b3931001180322p74a1227qe0c2199a77cbfe2@mail.gmail.com>
Message-ID: <191c3a031001180824q6e364c72g3f789892597e9469@mail.gmail.com>

try a less famous signal like SIGUSR1 it's possible something in perl still
reacts to SIGINT


On Mon, Jan 18, 2010 at 5:22 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Here is the result
>
> Program:
>
> require ESL;
> use IO::Socket::INET;
> use Data::Dumper;
>
> my $ip = "192.168.1.222";
> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
> Proto => 'tcp',  Listen => 1,  Reuse => 1 );
> die "Could not create socket: $!\n" unless $sock;
>
>
> for(;;) {
>     my $new_sock = $sock->accept();
>     next if (not defined ($new_sock));
>     my $pid = fork();
>     if ($pid) {
>         close($new_sock);
>         next;
>     }
>     print "CHILD PID: $$\n";
>     &register_Signals_Child();
>     sub register_Signals_Child() {
>         foreach ( keys %SIG ) {
>             $SIG{$_} = 'Handler';
>         }
>     }
>
>     sub Handler() {
>
>         my $handle=$_[0];
>         if($handle eq "INT") {
>             print "CHILD $$: SIGNAL SIG$handle is generated\n";
>         }
>         else
>         {
>             print "CHILD $$: Received $handle\n";
>
>         }
>     }
>     my $host = $new_sock->sockhost();
>     my $fd = fileno($new_sock);
>
>     my $con = new ESL::ESLconnection($fd);
>     my $info = $con->getInfo();
>
>     my $uuid = $info->getHeader("unique-id");
>
>     printf "Connected call %s, from %s\n", $uuid,
> $info->getHeader("caller-caller-id-number");
>     my $r=$con->execute("answer");
>     print Dumper $r;
>     $con->events("plain","all");
>     my
> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>     while($con->connected()) {
>         my $e = $con->recvEvent();
>
>         if ($e) {
>             my $name = $e->getHeader("event-name");
>             print "EVENT [$name]\n";
>             if ($name eq "DTMF") {
>                 my $digit = $e->getHeader("dtmf-digit");
>                 print "$digit\n";
>             }
>         }
>     }
>     close($new_sock);
> }
>
> I executed the program and the following things were printed
>
> CHILD PID: 6778
> Connected call e0d1001a-03f4-11df-b002-db488337e0ea, from 1001
> $VAR1 = 0;
> EVENT [CHANNEL_EXECUTE]
> EVENT [CHANNEL_ANSWER]
> EVENT [CHANNEL_EXECUTE_COMPLETE]
> EVENT [COMMAND]
> EVENT [CHANNEL_EXECUTE]
> EVENT [HEARTBEAT]
> EVENT [RE_SCHEDULE]
> EVENT [CHANNEL_EXECUTE_COMPLETE]
>
> Then from another shell I executed kill -2 6778, the result is follows
> CHILD 6778: SIGNAL SIGINT is generated
> EVENT [SERVER_DISCONNECTED]
>
> But the child process is still running as expected.
> But  I don't know why I received SERVER_DISCONNECTED from freeswitch???
>
>
>
>
>
>
> On Thu, Jan 14, 2010 at 1:27 PM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> I taught the signal handler will be inherited by the child process. It
>> also does like that.
>> After making a call, If I press ctrl + c, the above program printed
>> PARENT PID: Signal SIGINT is generated
>> CHILD PID: Signal SIGINT is generated.
>>
>> So I think the sigal handlers will be inherited to the child.
>> Anyway I'll also try registering signal handlers in child also, and then
>> I'll come back with that result.
>>
>> Thanks....
>> On Wed, Jan 13, 2010 at 9:48 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> you would have to register signals in your child process too
>>>
>>>   On Wed, Jan 13, 2010 at 3:13 AM, lakshmanan ganapathy <
>>> lakindia89 at gmail.com> wrote:
>>>
>>>>  Hi all,
>>>>
>>>> I've done a sample program (In perl ESL) , which play a file to the
>>>> caller and then it will call recvEvent() and print the event name. I've
>>>> handled signals also.
>>>>
>>>> When I send SIGINT to my program (Perl), the signal handler is called
>>>> and I can see the print output. But in the same time,  I received
>>>> SERVER_DISCONNECTED from freeswitch as event.
>>>>
>>>> I don't know why I received SERVER_DISCONNECTED from freeswitch. Is it
>>>> because, the recvEvent() from perl internally  calls the recvevent function
>>>> in the Esl.c and when it waits to receive the information from socket, the
>>>> signal occurred???
>>>>
>>>> Please clarify me!!
>>>>
>>>> Here is my program
>>>> require ESL;
>>>> use IO::Socket::INET;
>>>> use Data::Dumper;
>>>>
>>>> my $ip = "192.168.1.222";
>>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>>> die "Could not create socket: $!\n" unless $sock;
>>>> &register_Signals();
>>>>
>>>> for(;;) {
>>>>     my $new_sock = $sock->accept();
>>>>     next if (not defined ($new_sock));
>>>>     my $pid = fork();
>>>>     if ($pid) {
>>>>         close($new_sock);
>>>>         next;
>>>>     }
>>>>     print "CHILD PID: $$\n";
>>>>     my $host = $new_sock->sockhost();
>>>>     my $fd = fileno($new_sock);
>>>>
>>>>     my $con = new ESL::ESLconnection($fd);
>>>>     my $info = $con->getInfo();
>>>>
>>>>     my $uuid = $info->getHeader("unique-id");
>>>>
>>>>     printf "Connected call %s, from %s\n", $uuid,
>>>> $info->getHeader("caller-caller-id-number");
>>>>     my $r=$con->execute("answer");
>>>>     print Dumper $r;
>>>>     $con->events("plain","all");
>>>>     my
>>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>>
>>>>     while($con->connected()) {
>>>>         my $e = $con->recvEvent();
>>>>
>>>>         if ($e) {
>>>>             my $name = $e->getHeader("event-name");
>>>>             print "EVENT [$name]\n";
>>>>             if ($name eq "DTMF") {
>>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>>                 print "$digit\n";
>>>>             }
>>>>         }
>>>>     }
>>>>     close($new_sock);
>>>> }
>>>> sub register_Signals() {
>>>>     foreach ( keys %SIG ) {
>>>>         $SIG{$_} = 'sig_Handler';
>>>>     }
>>>> }
>>>>
>>>> sub sig_Handler() {
>>>>     my $handle=$_[0];
>>>>     if($handle eq "INT") {
>>>>         print "$$: SIGNAL SIG$handle is generated\n";
>>>>     }
>>>> }
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From Russell.Mosemann at cune.org  Mon Jan 18 08:47:38 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 18 Jan 2010 16:47:38 -0000
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <191c3a031001180824q6e364c72g3f789892597e9469@mail.gmail.com>
Message-ID: <20100118164738.9814D21DD50@cuneorg-email.cune.pri>

> try a less famous signal like SIGUSR1 it's possible something in perl still
> reacts to SIGINT

Such as an interrupted system call, perhaps? Depending on the operating
system, some system calls restart and some do not. There is not enough
debugging information in the original message to know what is happening
in the program when it is interrupted. If the interrupted system call is
not restarted, it probably returns an "interrupted" error code and needs
to be called, again.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From help at pdscc.com  Mon Jan 18 09:52:49 2010
From: help at pdscc.com (Harondel J. Sibble)
Date: Mon, 18 Jan 2010 09:52:49 -0800
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4 (fixed)
In-Reply-To: <8C20E0B1-0E56-4892-9D36-4D6DF3331244@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20100105065356.AEE0612F5@sinclaire.sibble.net>,
	<8C20E0B1-0E56-4892-9D36-4D6DF3331244@freeswitch.org>
Message-ID: <20100118175251.531A21DB501@sinclaire.sibble.net>

Thanks, I'll do that this week and report back.

When you say the latest lib and client, are you refering to developer only 
versions?  All I have access to are the official releases on the zfone site.

On 7 Jan 2010 at 17:30, Brian West wrote:

> Harondel,
> 	Please update your FreeSWITCH source rev 16204 fixes the SAS passing in all
> cases from my testing.  You'll need the latest ZRTP Lib and zfone
> application to make this work... I'm not too sure Tiviphone does this yet as
> I don't have one to test with.  This also fixes the issue when both sides
> are enrolled.  Next we will fix the video portion so both video and audio
> will go thru zrtp.
> 
> Please try it and let me know.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From freeswitch at aastral.net  Mon Jan 18 10:04:20 2010
From: freeswitch at aastral.net (Bill W)
Date: Mon, 18 Jan 2010 13:04:20 -0500
Subject: [Freeswitch-users] database disk image is malformed ISSUE
In-Reply-To: <206AA031-94DA-4AAB-8761-90A06A80E9F5@jerris.com>
References: <c8df2a131001180300m25419dd8y65ff80ce64824cef@mail.gmail.com>
	<206AA031-94DA-4AAB-8761-90A06A80E9F5@jerris.com>
Message-ID: <4B54A2A4.2040601@aastral.net>

Hey Michael,

Is there a way to get the core.db and fifo.db into ODBC?  I didn't see 
anything on the wiki about that.


Thanks,
Bill W


Michael Jerris wrote:
> sqlite doesn't scale.  If you want to do anything serious, use odbc and a real database instead.
> 


From troy at tlainvestments.com  Mon Jan 18 11:11:31 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Mon, 18 Jan 2010 12:11:31 -0700
Subject: [Freeswitch-users] More playing with sessions in lua
Message-ID: <095F1264-AB44-45A8-A84B-2C528E61E771@tlainvestments.com>

I think there may be other ways to do this, but if I could understand how to do it from lua, I think it would help me understand more about what's going on and make the mod_lua even more valuable to me and anyone else struggling with this kind of issue.

I am trying to answer a call in lua, exchange a bit of information with the caller, then I want to originate a call to another endpoint, and depending on what their response to some audio questions is, connect the original caller to them, or send the original caller away.  The problem is that I want the original caller to her ring tones or music on hold while they're waiting.  In addition to the way presented here, I've tried parking the caller using uuid_park, but still can't figure out how to play music/ringback for them.

session:answer();
session:sleep(1000);
session:execute("playback","pleasehold.wav")

local targetEndpoint = "1100 at default" -- or wherever

-- ringback works only AFTER the bridge line, below.  How do I get it to start immediately?
local destSession = freeswitch.Session("{ringback='myringback.wav'}".. targetEndpoint)	-- this "originates" a call to targetEndpoint

local digit destSession:playAndGetDigits(1,1,3,3000,"#","instructions.wav","[1-3]")

if (digit == "1") then
	freeswitch.bridge(session,destSession)
else
	session:execute("playback","goodbye.wav")
	session:hangup()
	destSession:hangup()
end


I appreciate any help on this.

-Troy


From john at acsol.net  Mon Jan 18 11:24:12 2010
From: john at acsol.net (John)
Date: Mon, 18 Jan 2010 12:24:12 -0700
Subject: [Freeswitch-users] Call Manager
Message-ID: <4B54B55C.3090402@acsol.net>

Is there an active project for a client call control software for 
Freeswitch, such as ShoreTel's or Cisco's Call Manager? Looking for the 
ability for a Windows or Mac user to be able to transfer calls, See 
presence, conference calls etc.

Thanks



From nicolas at medularis.com  Mon Jan 18 11:38:39 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 18 Jan 2010 16:38:39 -0300
Subject: [Freeswitch-users] Takes 5 minutes for FS to start (weird
	Test-Average-Step messages)
Message-ID: <1b46b4e81001181138g37157922sfb4f5ddf5f91dae8@mail.gmail.com>

I got a server on Rackspace Cloud (www.rackspacecloud.com). I compiled
FreeSWITCH from svn trunk and once it starts it works fine, no problem at
all. But, it takes FreeSWITCH about 5 minutes to start. It prints,
literally, hundreds of messages like the ones below, until the value of
"Test:" reaches 0. I looked for info about this on the wiki and the mailing
list but couldn't find anything. Is there a way to suppress this? Why does
it happen? Is it because the server is a virtual machine? Thanks!


2010-01-18 19:12:00.373937 [CONSOLE] switch_time.c:188 Test: 611 Average:
10000 Step: 13
2010-01-18 19:12:01.373937 [CONSOLE] switch_time.c:188 Test: 598 Average:
10000 Step: 12
2010-01-18 19:12:02.373937 [CONSOLE] switch_time.c:188 Test: 586 Average:
10000 Step: 11
2010-01-18 19:12:03.373937 [CONSOLE] switch_time.c:188 Test: 575 Average:
10000 Step: 10
2010-01-18 19:12:04.373937 [CONSOLE] switch_time.c:188 Test: 565 Average:
10000 Step: 9
2010-01-18 19:12:05.373937 [CONSOLE] switch_time.c:188 Test: 556 Average:
10000 Step: 8
2010-01-18 19:12:06.373937 [CONSOLE] switch_time.c:188 Test: 548 Average:
10000 Step: 7
2010-01-18 19:12:07.373937 [CONSOLE] switch_time.c:188 Test: 541 Average:
10000 Step: 6
2010-01-18 19:12:08.373937 [CONSOLE] switch_time.c:188 Test: 535 Average:
10000 Step: 5
2010-01-18 19:12:09.373937 [CONSOLE] switch_time.c:188 Test: 530 Average:
10000 Step: 4
2010-01-18 19:12:10.373937 [CONSOLE] switch_time.c:188 Test: 526 Average:
10000 Step: 3
2010-01-18 19:12:11.373937 [CONSOLE] switch_time.c:188 Test: 523 Average:
10000 Step: 2
2010-01-18 19:12:12.373937 [CONSOLE] switch_time.c:188 Test: 521 Average:
10000 Step: 1
2010-01-18 19:12:13.373937 [CONSOLE] switch_time.c:188 Test: 520 Average:
10000 Step: 1
2010-01-18 19:12:14.373937 [CONSOLE] switch_time.c:188 Test: 519 Average:
10000 Step: 1
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From anthony.minessale at gmail.com  Mon Jan 18 11:44:44 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 18 Jan 2010 13:44:44 -0600
Subject: [Freeswitch-users] More playing with sessions in lua
In-Reply-To: <095F1264-AB44-45A8-A84B-2C528E61E771@tlainvestments.com>
References: <095F1264-AB44-45A8-A84B-2C528E61E771@tlainvestments.com>
Message-ID: <191c3a031001181144g200e0a41i23dfe6813865965b@mail.gmail.com>

as soon as you play please hold there is no longer a chance for signaling
based ringback
you could have used the application ring_ready to send your phone an 180
ringing if you did not play the file
once you played the file you are responsible for sending audio to the
channel, think of it like a gui where the runtime loop takes input and you
must not do anything blocking in that loop and you can't play the file and
do something else at the same time without more threads which is not easy to
do from an embedded script.



On Mon, Jan 18, 2010 at 1:11 PM, Troy Anderson <troy at tlainvestments.com>wrote:

> I think there may be other ways to do this, but if I could understand how
> to do it from lua, I think it would help me understand more about what's
> going on and make the mod_lua even more valuable to me and anyone else
> struggling with this kind of issue.
>
> I am trying to answer a call in lua, exchange a bit of information with the
> caller, then I want to originate a call to another endpoint, and depending
> on what their response to some audio questions is, connect the original
> caller to them, or send the original caller away.  The problem is that I
> want the original caller to her ring tones or music on hold while they're
> waiting.  In addition to the way presented here, I've tried parking the
> caller using uuid_park, but still can't figure out how to play
> music/ringback for them.
>
> session:answer();
> session:sleep(1000);
> session:execute("playback","pleasehold.wav")
>
> local targetEndpoint = "1100 at default" -- or wherever
>
> -- ringback works only AFTER the bridge line, below.  How do I get it to
> start immediately?
> local destSession = freeswitch.Session("{ringback='myringback.wav'}"..
> targetEndpoint)  -- this "originates" a call to targetEndpoint
>
> local digit
> destSession:playAndGetDigits(1,1,3,3000,"#","instructions.wav","[1-3]")
>
> if (digit == "1") then
>        freeswitch.bridge(session,destSession)
> else
>        session:execute("playback","goodbye.wav")
>        session:hangup()
>        destSession:hangup()
> end
>
>
> I appreciate any help on this.
>
> -Troy
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Mon Jan 18 11:45:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 18 Jan 2010 13:45:51 -0600
Subject: [Freeswitch-users] Takes 5 minutes for FS to start (weird
	Test-Average-Step messages)
In-Reply-To: <1b46b4e81001181138g37157922sfb4f5ddf5f91dae8@mail.gmail.com>
References: <1b46b4e81001181138g37157922sfb4f5ddf5f91dae8@mail.gmail.com>
Message-ID: <191c3a031001181145v34600633uacdee6c9ebc12276@mail.gmail.com>

I dont think you have latest trunk based on what you are reporting.
you should try to get the up to the minuted latest and try again.


On Mon, Jan 18, 2010 at 1:38 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> I got a server on Rackspace Cloud (www.rackspacecloud.com). I compiled
> FreeSWITCH from svn trunk and once it starts it works fine, no problem at
> all. But, it takes FreeSWITCH about 5 minutes to start. It prints,
> literally, hundreds of messages like the ones below, until the value of
> "Test:" reaches 0. I looked for info about this on the wiki and the mailing
> list but couldn't find anything. Is there a way to suppress this? Why does
> it happen? Is it because the server is a virtual machine? Thanks!
>
>
> 2010-01-18 19:12:00.373937 [CONSOLE] switch_time.c:188 Test: 611 Average:
> 10000 Step: 13
> 2010-01-18 19:12:01.373937 [CONSOLE] switch_time.c:188 Test: 598 Average:
> 10000 Step: 12
> 2010-01-18 19:12:02.373937 [CONSOLE] switch_time.c:188 Test: 586 Average:
> 10000 Step: 11
> 2010-01-18 19:12:03.373937 [CONSOLE] switch_time.c:188 Test: 575 Average:
> 10000 Step: 10
> 2010-01-18 19:12:04.373937 [CONSOLE] switch_time.c:188 Test: 565 Average:
> 10000 Step: 9
> 2010-01-18 19:12:05.373937 [CONSOLE] switch_time.c:188 Test: 556 Average:
> 10000 Step: 8
> 2010-01-18 19:12:06.373937 [CONSOLE] switch_time.c:188 Test: 548 Average:
> 10000 Step: 7
> 2010-01-18 19:12:07.373937 [CONSOLE] switch_time.c:188 Test: 541 Average:
> 10000 Step: 6
> 2010-01-18 19:12:08.373937 [CONSOLE] switch_time.c:188 Test: 535 Average:
> 10000 Step: 5
> 2010-01-18 19:12:09.373937 [CONSOLE] switch_time.c:188 Test: 530 Average:
> 10000 Step: 4
> 2010-01-18 19:12:10.373937 [CONSOLE] switch_time.c:188 Test: 526 Average:
> 10000 Step: 3
> 2010-01-18 19:12:11.373937 [CONSOLE] switch_time.c:188 Test: 523 Average:
> 10000 Step: 2
> 2010-01-18 19:12:12.373937 [CONSOLE] switch_time.c:188 Test: 521 Average:
> 10000 Step: 1
> 2010-01-18 19:12:13.373937 [CONSOLE] switch_time.c:188 Test: 520 Average:
> 10000 Step: 1
> 2010-01-18 19:12:14.373937 [CONSOLE] switch_time.c:188 Test: 519 Average:
> 10000 Step: 1
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jmesquita at freeswitch.org  Mon Jan 18 11:51:12 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 18 Jan 2010 17:51:12 -0200
Subject: [Freeswitch-users] Call Manager
In-Reply-To: <4B54B55C.3090402@acsol.net>
References: <4B54B55C.3090402@acsol.net>
Message-ID: <ebbc2d551001181151j4a41bb34s4ba7771d0a874eb7@mail.gmail.com>

FSComm will have the ability to do some of it when the plugins infraestrucre
is developed. It shouldn't take too long for that to happen and of course
that the first big plugin would be ESL goodies.

Regards,
Jo?o Mesquita
FSComm Developer


On Mon, Jan 18, 2010 at 5:24 PM, John <john at acsol.net> wrote:

> Is there an active project for a client call control software for
> Freeswitch, such as ShoreTel's or Cisco's Call Manager? Looking for the
> ability for a Windows or Mac user to be able to transfer calls, See
> presence, conference calls etc.
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Mon Jan 18 11:55:27 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 18 Jan 2010 20:55:27 +0100
Subject: [Freeswitch-users] How to register from FS to Huawei SoftX3000
In-Reply-To: <8ccbff061001160917m22edfed8n291ea8a07518e4f7@mail.gmail.com>
References: <8ccbff061001160917m22edfed8n291ea8a07518e4f7@mail.gmail.com>
Message-ID: <65d96fc81001181155y631d03bdr697360e0bff28ba@mail.gmail.com>

hi Dome,

well,  SOFTX3000 is a great switch :)


anyhow... there are 2 ways you can provide services via SIP to your
customers:


1. MSBR ( Multimedia subscriber)
2. SIP trunk



with MSBR you actually create an account (MMTE - multimedia terminal) and
bind a phone number to that account... When using MSBR you MUST register in
order to place/receive calls and of course you are limited to only 1
simultaneous calls on that account. From my oppinion this is not to be used
with FS because of the limultaneous calls limit per account!

with SIP trunk ... it is simple, this is just number analisys. The important
thing is that SoftX3000 doesn't support registration on SIP Trunk! You must
not register. Also, it uses SIP OPTIONS for keepalive and you should respond
to that messages as well... otherwise it will bring the trunk down.


regarding call limitation ... well, this is something realy unlikelly...
there is no such feature on SX to filter by user-agent.


Anyhow, did you try to register with any softphone? ... I've tested x-lite,
jsphone, ekiga, 3cx, ... and everyone is working!



can you send us the wireshark capture ?

or you can send me the accounts to check that out... of course off the list
:)


T.

On Sat, Jan 16, 2010 at 6:17 PM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear sir,
>          I found some provider use Huawei SoftX3000 and can limit use
> call from they softphone only. (use eyeball SDK).
> They can limit some account can register and call by sip server like
> an FS and Asterisk. but some account can't. (register and call by
> softphone). and i don't know how they can do that.
> So i try to use wireshark to debug sip headeder when use softphone
> with both account type. it's nothing diferent.
>          I want to use both account work by FS register to  Huawei
> SoftX3000. Can someone help me. i can give you softphone and both
> account type for test.
>
>
> Best Regards.
>
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Jan 18 12:09:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Jan 2010 12:09:26 -0800
Subject: [Freeswitch-users] How do I invite group to join existing call?
In-Reply-To: <4B5412C6020000E1000003D6@mail.fribert.dk>
References: <4B5412C6020000E1000003D6@mail.fribert.dk>
Message-ID: <87f2f3b91001181209y7a0aa68fs8a580712484c7a11@mail.gmail.com>

On Sun, Jan 17, 2010 at 10:50 PM, mailinglist <mailinglist at fribert.dk>wrote:

> Hi All
>
> I would like to be able to invite a group / global to join an existing
> call, but how do I accomplish this, can it be done?
>

Malfunction! Need Input!

Could you give us a few more details on what you're trying to accomplish?
What's the big picture? I'm curious what problem you're trying to solve. I'm
sure the gang here will have thoughts to pass along. :)

-MC
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From pete at privateconnect.com  Mon Jan 18 12:22:53 2010
From: pete at privateconnect.com (Pete Mueller)
Date: Mon, 18 Jan 2010 13:22:53 -0700
Subject: [Freeswitch-users] More playing with sessions in lua
Message-ID: <20100118132253.2ad02225396a31c9de30536f2e338977.df3ad2f537.wbe@email04.secureserver.net>

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From nicolas at medularis.com  Mon Jan 18 12:25:53 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 18 Jan 2010 17:25:53 -0300
Subject: [Freeswitch-users] Takes 5 minutes for FS to start (weird
	Test-Average-Step messages)
In-Reply-To: <191c3a031001181145v34600633uacdee6c9ebc12276@mail.gmail.com>
References: <1b46b4e81001181138g37157922sfb4f5ddf5f91dae8@mail.gmail.com>
	<191c3a031001181145v34600633uacdee6c9ebc12276@mail.gmail.com>
Message-ID: <1b46b4e81001181225j39c61895y3360d01685963973@mail.gmail.com>

You are right, I got the source code a few days ago. I just updated and
compiled again. Now I'm not getting those messages, only this:

2010-01-18 20:23:50.386188 [CONSOLE] switch_time.c:959 Calibrating timer,
please wait...
2010-01-18 20:23:50.386188 [WARNING] switch_time.c:190 Timer resolution of
10000 microseconds detected!
Do you have your kernel timer set to higher than 1 kHz? You may experience
audio problems.
2010-01-18 20:23:55.396188 [CONSOLE] switch_loadable_module.c:890
Successfully Loaded [CORE_SOFTTIMER_MODULE]
2010-01-18 20:23:55.396188 [NOTICE] switch_loadable_module.c:229 Adding
Timer 'soft'


Thanks!



On Mon, Jan 18, 2010 at 4:45 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I dont think you have latest trunk based on what you are reporting.
> you should try to get the up to the minuted latest and try again.
>
>
> On Mon, Jan 18, 2010 at 1:38 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> I got a server on Rackspace Cloud (www.rackspacecloud.com). I compiled
>> FreeSWITCH from svn trunk and once it starts it works fine, no problem at
>> all. But, it takes FreeSWITCH about 5 minutes to start. It prints,
>> literally, hundreds of messages like the ones below, until the value of
>> "Test:" reaches 0. I looked for info about this on the wiki and the mailing
>> list but couldn't find anything. Is there a way to suppress this? Why does
>> it happen? Is it because the server is a virtual machine? Thanks!
>>
>>
>> 2010-01-18 19:12:00.373937 [CONSOLE] switch_time.c:188 Test: 611 Average:
>> 10000 Step: 13
>> 2010-01-18 19:12:01.373937 [CONSOLE] switch_time.c:188 Test: 598 Average:
>> 10000 Step: 12
>> 2010-01-18 19:12:02.373937 [CONSOLE] switch_time.c:188 Test: 586 Average:
>> 10000 Step: 11
>> 2010-01-18 19:12:03.373937 [CONSOLE] switch_time.c:188 Test: 575 Average:
>> 10000 Step: 10
>> 2010-01-18 19:12:04.373937 [CONSOLE] switch_time.c:188 Test: 565 Average:
>> 10000 Step: 9
>> 2010-01-18 19:12:05.373937 [CONSOLE] switch_time.c:188 Test: 556 Average:
>> 10000 Step: 8
>> 2010-01-18 19:12:06.373937 [CONSOLE] switch_time.c:188 Test: 548 Average:
>> 10000 Step: 7
>> 2010-01-18 19:12:07.373937 [CONSOLE] switch_time.c:188 Test: 541 Average:
>> 10000 Step: 6
>> 2010-01-18 19:12:08.373937 [CONSOLE] switch_time.c:188 Test: 535 Average:
>> 10000 Step: 5
>> 2010-01-18 19:12:09.373937 [CONSOLE] switch_time.c:188 Test: 530 Average:
>> 10000 Step: 4
>> 2010-01-18 19:12:10.373937 [CONSOLE] switch_time.c:188 Test: 526 Average:
>> 10000 Step: 3
>> 2010-01-18 19:12:11.373937 [CONSOLE] switch_time.c:188 Test: 523 Average:
>> 10000 Step: 2
>> 2010-01-18 19:12:12.373937 [CONSOLE] switch_time.c:188 Test: 521 Average:
>> 10000 Step: 1
>> 2010-01-18 19:12:13.373937 [CONSOLE] switch_time.c:188 Test: 520 Average:
>> 10000 Step: 1
>> 2010-01-18 19:12:14.373937 [CONSOLE] switch_time.c:188 Test: 519 Average:
>> 10000 Step: 1
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jan 18 13:18:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Jan 2010 13:18:50 -0800
Subject: [Freeswitch-users] Call for help - content ideas for freeswitch.org
Message-ID: <87f2f3b91001181318n98af6beuf6e50ebbcbd4ba5c@mail.gmail.com>

Hello all,

I'd like to ask everyone to think about things that we can put up on
freeswitch.org. We are interested in anything related to FreeSWITCH,
certainly, but also any VoIP or telecom news/articles on other sites that
are of interest to FreeSWITCH users. (See
http://www.freeswitch.org/node/229as an example.)

Please email me off list if you have ideas about content that would be
appropriate for our main page. Likewise, if you would like to be the author
of stories and/or blog posts on the main page then definitely let me know.
We would love to have some fresh perspectives represented.

Thanks!
-Michael
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From djbinter at gmail.com  Mon Jan 18 13:27:49 2010
From: djbinter at gmail.com (DJB INTERNATIONAL)
Date: Mon, 18 Jan 2010 13:27:49 -0800
Subject: [Freeswitch-users] Dial Plan bridge did not return a correct
	variable from reqular expression
Message-ID: <94f7dfb11001181327p5a807b45ld3d44a7238d2f3ee@mail.gmail.com>

Does anyone experience this problem?

SVN:   16395

Dialplan:

   <extension name="tollfree">
     <condition field="${toll_allow}" expression="tollfree"/>
     <condition field="destination_number"
expression="^(\+1|1)?(8(00|88|77|66)[2-9]\d{6})$">
      <action application="set" data="bypass_media=true"/>
       <action application="bridge" data="sofia/external/1$
2 at tollfreetollfree.com"/>
     </condition>
   </extension>

Possible Error:
Dialplan: sofia/internal/2132345567 at 204.110.15.190 Regex (PASS) [tollfree]
destination_number(18005551212) =~ /^(\+1|1)?(8(00|88|77|66)[2-9]\d{6})$/
break=on-false
Dialplan: sofia/internal/2132345567 at 204.110.15.190 Action
set(bypass_media=true)
Dialplan: sofia/internal/2132345567 at 204.110.15.190 Action
bridge(sofia/external/1 at tollfreetollfree.com)   ******  (It should return $2
as 8005551212)


Thank you,
Dorn B. (djbinter)
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From nicolas at medularis.com  Mon Jan 18 13:58:59 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 18 Jan 2010 18:58:59 -0300
Subject: [Freeswitch-users] Call for help - content ideas for
	freeswitch.org
In-Reply-To: <87f2f3b91001181318n98af6beuf6e50ebbcbd4ba5c@mail.gmail.com>
References: <87f2f3b91001181318n98af6beuf6e50ebbcbd4ba5c@mail.gmail.com>
Message-ID: <1b46b4e81001181358m3ebb56eeke827a64ea33debf6@mail.gmail.com>

Here's a little post on getting started with FreeSWITCH to create a small
click to call app:

- http://www.guayal.com/how-to-bridge-two-calls-with-freeswitch

At least 2 more parts to come.


On Mon, Jan 18, 2010 at 6:18 PM, Michael Collins <msc at freeswitch.org> wrote:

> Hello all,
>
> I'd like to ask everyone to think about things that we can put up on
> freeswitch.org. We are interested in anything related to FreeSWITCH,
> certainly, but also any VoIP or telecom news/articles on other sites that
> are of interest to FreeSWITCH users. (See
> http://www.freeswitch.org/node/229 as an example.)
>
> Please email me off list if you have ideas about content that would be
> appropriate for our main page. Likewise, if you would like to be the author
> of stories and/or blog posts on the main page then definitely let me know.
> We would love to have some fresh perspectives represented.
>
> Thanks!
> -Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lists at redbonez.net  Mon Jan 18 14:43:23 2010
From: lists at redbonez.net (Adam Ford)
Date: Mon, 18 Jan 2010 15:43:23 -0700
Subject: [Freeswitch-users] Bria softphone registration issue (NAT/rport
	issue?)
Message-ID: <00f301ca988f$a54eca70$efec5f50$@net>

I don't know if any of you would be able to help with this, but I figured I
would ask.  

 

After a few hours of inactivity, it seems my Bria softphones lose connection
with FreeSWITCH.  They are able to call out, but when calling in I just get
silence until it goes to voicemail.  It sounds like a classic NAT issue,
except the NAT is completely handled by a Soincwall that has VoIP features
to support SIP transformations (SonicOS Enhanced 5.x).  

 

It seems as though after a period of time the port in which the Sonicwall is
assigning for the SIP UA and the port that FreeSWITCH has registered get out
of sync, despite the Bria re-registering every 5 minutes.  Could this be
caused by 'Use rport' being enabled by default in the Bria softphone?

 

Setup -

FreeSWITCH 1.0.4

Bria 2.5.4

Sonicwall E5500 with SIP transformations and Consistent NAT enabled

 

 

Thanks in advance for any help,

 

-Adam

 

 

 

 

 

 

 

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From brian at freeswitch.org  Mon Jan 18 14:55:11 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 Jan 2010 16:55:11 -0600
Subject: [Freeswitch-users] Bria softphone registration issue (NAT/rport
	issue?)
In-Reply-To: <00f301ca988f$a54eca70$efec5f50$@net>
References: <00f301ca988f$a54eca70$efec5f50$@net>
Message-ID: <A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>

1.0.4 is not supported anymore please update to latest.  http://latest.freeswitch.org

/b

On Jan 18, 2010, at 4:43 PM, Adam Ford wrote:

> FreeSWITCH 1.0.4

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From lists at redbonez.net  Mon Jan 18 16:14:30 2010
From: lists at redbonez.net (Adam Ford)
Date: Mon, 18 Jan 2010 17:14:30 -0700
Subject: [Freeswitch-users] Bria softphone registration issue
	(NAT/rport	issue?)
In-Reply-To: <A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>
References: <00f301ca988f$a54eca70$efec5f50$@net>
	<A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>
Message-ID: <010101ca989c$5fb5f1c0$1f21d540$@net>

This is more of a Bria configuration question than FreeSWITCH.  I wouldn't
think the version of FreeSWITCH would matter much, but thanks for the
suggestion.

 

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, January 18, 2010 3:55 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bria softphone registration issue (NAT/rport
issue?)

 

1.0.4 is not supported anymore please update to latest.
http://latest.freeswitch.org

 

/b

 

On Jan 18, 2010, at 4:43 PM, Adam Ford wrote:





FreeSWITCH 1.0.4

 

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From brian at freeswitch.org  Mon Jan 18 16:17:17 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 Jan 2010 18:17:17 -0600
Subject: [Freeswitch-users] Bria softphone registration issue (NAT/rport
	issue?)
In-Reply-To: <010101ca989c$5fb5f1c0$1f21d540$@net>
References: <00f301ca988f$a54eca70$efec5f50$@net>
	<A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>
	<010101ca989c$5fb5f1c0$1f21d540$@net>
Message-ID: <C64D1DDC-2846-4A9F-BB82-F45D59CA99FF@freeswitch.org>

Is the phone behind the nat with FreeSWITCH?

/b

On Jan 18, 2010, at 6:14 PM, Adam Ford wrote:

> This is more of a Bria configuration question than FreeSWITCH.  I wouldn?t think the version of FreeSWITCH would matter much, but thanks for the suggestion.
>  
> -Adam
>  

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From lists at redbonez.net  Mon Jan 18 16:28:16 2010
From: lists at redbonez.net (Adam Ford)
Date: Mon, 18 Jan 2010 17:28:16 -0700
Subject: [Freeswitch-users] Bria softphone registration issue
	(NAT/rport	issue?)
In-Reply-To: <C64D1DDC-2846-4A9F-BB82-F45D59CA99FF@freeswitch.org>
References: <00f301ca988f$a54eca70$efec5f50$@net>	<A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>	<010101ca989c$5fb5f1c0$1f21d540$@net>
	<C64D1DDC-2846-4A9F-BB82-F45D59CA99FF@freeswitch.org>
Message-ID: <012101ca989e$4c6877d0$e5396770$@net>

Yes, but the nat is handled by the Sonicwall, which works perfectly for the
Polycom desk phones.   It is only the Bria softphones that are behaving
strangely.  I think it is just caused by them attempting to use rport, which
is sort of doing to work of the Sonicwall twice, causing them to lose sync
with the FreeSWITCH registration.  I just wanted the opinion of some other
professionals if this sound likely, before I consider the matter resolved.

 

Thank you,

 

-Adam

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, January 18, 2010 5:17 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bria softphone registration issue (NAT/rport
issue?)

 

Is the phone behind the nat with FreeSWITCH?

 

/b

 

On Jan 18, 2010, at 6:14 PM, Adam Ford wrote:





This is more of a Bria configuration question than FreeSWITCH.  I wouldn't
think the version of FreeSWITCH would matter much, but thanks for the
suggestion.

 

-Adam

 

 

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From brian at freeswitch.org  Mon Jan 18 16:47:05 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 Jan 2010 18:47:05 -0600
Subject: [Freeswitch-users] Bria softphone registration issue (NAT/rport
	issue?)
In-Reply-To: <012101ca989e$4c6877d0$e5396770$@net>
References: <00f301ca988f$a54eca70$efec5f50$@net>	<A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>	<010101ca989c$5fb5f1c0$1f21d540$@net>
	<C64D1DDC-2846-4A9F-BB82-F45D59CA99FF@freeswitch.org>
	<012101ca989e$4c6877d0$e5396770$@net>
Message-ID: <3FBC790C-EDC4-4323-935E-F5CA957F3BD7@freeswitch.org>

Well if the Bria Phone is behind nat with FS why is the sonic wall even involved? (Btw... I just realized they can't spell Brian... Bria... who calls a phone Bria?)

/b

On Jan 18, 2010, at 6:28 PM, Adam Ford wrote:

> Yes, but the nat is handled by the Sonicwall, which works perfectly for the Polycom desk phones.   It is only the Bria softphones that are behaving strangely.  I think it is just caused by them attempting to use rport, which is sort of doing to work of the Sonicwall twice, causing them to lose sync with the FreeSWITCH registration.  I just wanted the opinion of some other professionals if this sound likely, before I consider the matter resolved.
>  
> Thank you,
>  
> -Adam

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From lists at redbonez.net  Mon Jan 18 17:08:17 2010
From: lists at redbonez.net (Adam Ford)
Date: Mon, 18 Jan 2010 18:08:17 -0700
Subject: [Freeswitch-users] Bria softphone registration issue
	(NAT/rport	issue?)
In-Reply-To: <3FBC790C-EDC4-4323-935E-F5CA957F3BD7@freeswitch.org>
References: <00f301ca988f$a54eca70$efec5f50$@net>	<A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>	<010101ca989c$5fb5f1c0$1f21d540$@net>	<C64D1DDC-2846-4A9F-BB82-F45D59CA99FF@freeswitch.org>	<012101ca989e$4c6877d0$e5396770$@net>
	<3FBC790C-EDC4-4323-935E-F5CA957F3BD7@freeswitch.org>
Message-ID: <013201ca98a3$e3cca010$ab65e030$@net>

I guess I misunderstood your question.  The phones aren't behind a nat WITH
the FreeSWITCH.  FreeSWITCH is in collocation site, the phones are in my
office, and connecting to FreeSWITCH through the nat.

 

Maybe they wanted to call it Brian but then someone complained that it would
be to common ;)

 

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, January 18, 2010 5:47 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bria softphone registration issue (NAT/rport
issue?)

 

Well if the Bria Phone is behind nat with FS why is the sonic wall even
involved? (Btw... I just realized they can't spell Brian... Bria... who
calls a phone Bria?)

 

/b

 

On Jan 18, 2010, at 6:28 PM, Adam Ford wrote:





Yes, but the nat is handled by the Sonicwall, which works perfectly for the
Polycom desk phones.   It is only the Bria softphones that are behaving
strangely.  I think it is just caused by them attempting to use rport, which
is sort of doing to work of the Sonicwall twice, causing them to lose sync
with the FreeSWITCH registration.  I just wanted the opinion of some other
professionals if this sound likely, before I consider the matter resolved.

 

Thank you,

 

-Adam

 

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From brian at freeswitch.org  Mon Jan 18 17:12:40 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 Jan 2010 19:12:40 -0600
Subject: [Freeswitch-users] Bria softphone registration issue (NAT/rport
	issue?)
In-Reply-To: <013201ca98a3$e3cca010$ab65e030$@net>
References: <00f301ca988f$a54eca70$efec5f50$@net>	<A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>	<010101ca989c$5fb5f1c0$1f21d540$@net>	<C64D1DDC-2846-4A9F-BB82-F45D59CA99FF@freeswitch.org>	<012101ca989e$4c6877d0$e5396770$@net>
	<3FBC790C-EDC4-4323-935E-F5CA957F3BD7@freeswitch.org>
	<013201ca98a3$e3cca010$ab65e030$@net>
Message-ID: <AF55B055-5A95-446F-B700-19A81F2496ED@freeswitch.org>

Try 1.0.5

/b

On Jan 18, 2010, at 7:08 PM, Adam Ford wrote:

> I guess I misunderstood your question.  The phones aren?t behind a nat WITH the FreeSWITCH.  FreeSWITCH is in collocation site, the phones are in my office, and connecting to FreeSWITCH through the nat.
>  
> Maybe they wanted to call it Brian but then someone complained that it would be to common ;)
>  
> -Adam

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From mcampbellsmith at gmail.com  Mon Jan 18 20:52:05 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 19 Jan 2010 15:52:05 +1100
Subject: [Freeswitch-users] Bria softphone registration issue (NAT/rport
	issue?)
In-Reply-To: <AF55B055-5A95-446F-B700-19A81F2496ED@freeswitch.org>
References: <00f301ca988f$a54eca70$efec5f50$@net>
	<A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>
	<010101ca989c$5fb5f1c0$1f21d540$@net>
	<C64D1DDC-2846-4A9F-BB82-F45D59CA99FF@freeswitch.org>
	<012101ca989e$4c6877d0$e5396770$@net>
	<3FBC790C-EDC4-4323-935E-F5CA957F3BD7@freeswitch.org>
	<013201ca98a3$e3cca010$ab65e030$@net>
	<AF55B055-5A95-446F-B700-19A81F2496ED@freeswitch.org>
Message-ID: <33c87fa31001182052n18602400jafe5090e5189da8d@mail.gmail.com>

Is there a NAT keep alive option in Bria?  Look under SIP Account
Properties, Advanced.

On Tue, Jan 19, 2010 at 12:12 PM, Brian West <brian at freeswitch.org> wrote:
> Try 1.0.5
> /b
> On Jan 18, 2010, at 7:08 PM, Adam Ford wrote:
>
> I guess I misunderstood your question.? The phones aren?t behind a nat WITH
> the FreeSWITCH.? FreeSWITCH is in collocation site, the phones are in my
> office, and connecting to FreeSWITCH through the nat.
>
> Maybe they wanted to call it Brian but then someone complained that it would
> be to common ;)
>
> -Adam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From magesh.freeswitch at gmail.com  Mon Jan 18 23:27:35 2010
From: magesh.freeswitch at gmail.com (Magesh R)
Date: Tue, 19 Jan 2010 02:27:35 -0500
Subject: [Freeswitch-users] Error in finding OpenZAP span id
Message-ID: <369c72d81001182327u38772291pa42ec950c11a1055@mail.gmail.com>

Dear All,
   I have installed Sangoma PRI card and installed wanpipe drivers. The
wanrouter process started sucessfully.
    I had the following configurations,


openzap.conf:
==========
[span wanpipe PRI_1]
name => OpenZAP
number => 1
trunk_type => e1
b-channel => 1:1-15
d-channel => 1:16
b-channel => 1:17-31


[span wanpipe PRI_2]
name => OpenZAP
number => 2
trunk_type => e1
b-channel => 2:1-15
d-channel => 2:16
b-channel => 2:17-31

openzap.conf.xml:
===========
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="1"/>
  </settings>
   <libpri_spans>
     <span name="PRI_1">
        <param name="node" value="cpe"/>
        <param name="switch" value="ni2"/>
        <param name="dialplan" value="XML"/>
        <param name="context" value="public"/>
      </span>
   </libpri_spans>
</configuration>

 When I started the freeswitch I have received the following error,

2010-01-19 12:41:22.693212 [ERR] mod_openzap.c:2039 Error finding OpenZAP
span id: name:PRI_1

Any one please tell me a way to solve this problem...

Thanks,
Mag.
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From lists at redbonez.net  Mon Jan 18 23:33:12 2010
From: lists at redbonez.net (Adam Ford)
Date: Tue, 19 Jan 2010 00:33:12 -0700
Subject: [Freeswitch-users] Bria softphone registration issue
	(NAT/rportissue?)
In-Reply-To: <33c87fa31001182052n18602400jafe5090e5189da8d@mail.gmail.com>
References: <00f301ca988f$a54eca70$efec5f50$@net><A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org><010101ca989c$5fb5f1c0$1f21d540$@net><C64D1DDC-2846-4A9F-BB82-F45D59CA99FF@freeswitch.org><012101ca989e$4c6877d0$e5396770$@net><3FBC790C-EDC4-4323-935E-F5CA957F3BD7@freeswitch.org><013201ca98a3$e3cca010$ab65e030$@net><AF55B055-5A95-446F-B700-19A81F2496ED@freeswitch.org>
	<33c87fa31001182052n18602400jafe5090e5189da8d@mail.gmail.com>
Message-ID: <C9E790B9434246C199F31D9F79218097@redbonez>

Yes, that is enabled as well.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark
Campbell-Smith
Sent: Monday, January 18, 2010 9:52 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bria softphone registration issue
(NAT/rportissue?)

Is there a NAT keep alive option in Bria?  Look under SIP Account
Properties, Advanced.

On Tue, Jan 19, 2010 at 12:12 PM, Brian West <brian at freeswitch.org> wrote:
> Try 1.0.5
> /b
> On Jan 18, 2010, at 7:08 PM, Adam Ford wrote:
>
> I guess I misunderstood your question.? The phones aren?t behind a nat
WITH
> the FreeSWITCH.? FreeSWITCH is in collocation site, the phones are in my
> office, and connecting to FreeSWITCH through the nat.
>
> Maybe they wanted to call it Brian but then someone complained that it
would
> be to common ;)
>
> -Adam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From scottferri09 at gmail.com  Tue Jan 19 01:07:06 2010
From: scottferri09 at gmail.com (Scott Fernandez)
Date: Tue, 19 Jan 2010 14:37:06 +0530
Subject: [Freeswitch-users] Establishing a Call from .Net based application
Message-ID: <df97b9c51001190107x39d94eaai7400635f0ac03f86@mail.gmail.com>

Hi,

Is there any API modules available for me to initiate a call from .Net based
application?.

The idea is to include the API modules if any with the .NET base classes so
that the API commands will be made available on it. I know it is doable when
I use socket programming in .NET in which Telnet session is created.
However, this would potentially hamper the performance of the application
because of multiple sessions that will be created for each call.

Other than that, Is there any Freeswitch API modules (like plug-ins)
available in order to include it into the .Net classes and start building
the customized application?

Any help from any one is highly appreciated.

Thanks,
Scott
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From lakindia89 at gmail.com  Tue Jan 19 01:07:34 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 19 Jan 2010 14:37:34 +0530
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <191c3a031001180824q6e364c72g3f789892597e9469@mail.gmail.com>
References: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
	<191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>
	<7d79b3931001132357i36bbb482jdf2bbdd3aea2a583@mail.gmail.com>
	<7d79b3931001180322p74a1227qe0c2199a77cbfe2@mail.gmail.com>
	<191c3a031001180824q6e364c72g3f789892597e9469@mail.gmail.com>
Message-ID: <7d79b3931001190107o200ec04dredd76689dd235588@mail.gmail.com>

I tried with SIGUSR1, but no progress. I got SERVER_DISCONNECTED.
Output:

CHILD 3814: Received USR1
EVENT [SERVER_DISCONNECTED]

In esl.c, in esl_recv_event() function, line no: 824
if (rrval < 0) {
                        strerror_r(handle->errnum, handle->err,
sizeof(handle->err));
                        goto fail;
}
When the program is blocked under receive, I passed the signal. So recv
returns -1, and in fail: it call esl_disconnect(handle).

Is it because of this??? If so, whether it should be fixed or not???



On Mon, Jan 18, 2010 at 9:54 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try a less famous signal like SIGUSR1 it's possible something in perl still
> reacts to SIGINT
>
>
>
> On Mon, Jan 18, 2010 at 5:22 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Here is the result
>>
>> Program:
>>
>> require ESL;
>> use IO::Socket::INET;
>> use Data::Dumper;
>>
>> my $ip = "192.168.1.222";
>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
>> Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>> die "Could not create socket: $!\n" unless $sock;
>>
>>
>> for(;;) {
>>     my $new_sock = $sock->accept();
>>     next if (not defined ($new_sock));
>>     my $pid = fork();
>>     if ($pid) {
>>         close($new_sock);
>>         next;
>>     }
>>     print "CHILD PID: $$\n";
>>     &register_Signals_Child();
>>     sub register_Signals_Child() {
>>         foreach ( keys %SIG ) {
>>             $SIG{$_} = 'Handler';
>>         }
>>     }
>>
>>     sub Handler() {
>>
>>         my $handle=$_[0];
>>         if($handle eq "INT") {
>>             print "CHILD $$: SIGNAL SIG$handle is generated\n";
>>         }
>>         else
>>         {
>>             print "CHILD $$: Received $handle\n";
>>
>>         }
>>     }
>>     my $host = $new_sock->sockhost();
>>     my $fd = fileno($new_sock);
>>
>>     my $con = new ESL::ESLconnection($fd);
>>     my $info = $con->getInfo();
>>
>>     my $uuid = $info->getHeader("unique-id");
>>
>>     printf "Connected call %s, from %s\n", $uuid,
>> $info->getHeader("caller-caller-id-number");
>>     my $r=$con->execute("answer");
>>     print Dumper $r;
>>     $con->events("plain","all");
>>     my
>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>     while($con->connected()) {
>>         my $e = $con->recvEvent();
>>
>>         if ($e) {
>>             my $name = $e->getHeader("event-name");
>>             print "EVENT [$name]\n";
>>             if ($name eq "DTMF") {
>>                 my $digit = $e->getHeader("dtmf-digit");
>>                 print "$digit\n";
>>             }
>>         }
>>     }
>>     close($new_sock);
>> }
>>
>> I executed the program and the following things were printed
>>
>> CHILD PID: 6778
>> Connected call e0d1001a-03f4-11df-b002-db488337e0ea, from 1001
>> $VAR1 = 0;
>> EVENT [CHANNEL_EXECUTE]
>> EVENT [CHANNEL_ANSWER]
>> EVENT [CHANNEL_EXECUTE_COMPLETE]
>> EVENT [COMMAND]
>> EVENT [CHANNEL_EXECUTE]
>> EVENT [HEARTBEAT]
>> EVENT [RE_SCHEDULE]
>> EVENT [CHANNEL_EXECUTE_COMPLETE]
>>
>> Then from another shell I executed kill -2 6778, the result is follows
>> CHILD 6778: SIGNAL SIGINT is generated
>> EVENT [SERVER_DISCONNECTED]
>>
>> But the child process is still running as expected.
>> But  I don't know why I received SERVER_DISCONNECTED from freeswitch???
>>
>>
>>
>>
>>
>>
>> On Thu, Jan 14, 2010 at 1:27 PM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>> I taught the signal handler will be inherited by the child process. It
>>> also does like that.
>>> After making a call, If I press ctrl + c, the above program printed
>>> PARENT PID: Signal SIGINT is generated
>>> CHILD PID: Signal SIGINT is generated.
>>>
>>> So I think the sigal handlers will be inherited to the child.
>>> Anyway I'll also try registering signal handlers in child also, and then
>>> I'll come back with that result.
>>>
>>> Thanks....
>>> On Wed, Jan 13, 2010 at 9:48 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> you would have to register signals in your child process too
>>>>
>>>>   On Wed, Jan 13, 2010 at 3:13 AM, lakshmanan ganapathy <
>>>> lakindia89 at gmail.com> wrote:
>>>>
>>>>>  Hi all,
>>>>>
>>>>> I've done a sample program (In perl ESL) , which play a file to the
>>>>> caller and then it will call recvEvent() and print the event name. I've
>>>>> handled signals also.
>>>>>
>>>>> When I send SIGINT to my program (Perl), the signal handler is called
>>>>> and I can see the print output. But in the same time,  I received
>>>>> SERVER_DISCONNECTED from freeswitch as event.
>>>>>
>>>>> I don't know why I received SERVER_DISCONNECTED from freeswitch. Is it
>>>>> because, the recvEvent() from perl internally  calls the recvevent function
>>>>> in the Esl.c and when it waits to receive the information from socket, the
>>>>> signal occurred???
>>>>>
>>>>> Please clarify me!!
>>>>>
>>>>> Here is my program
>>>>> require ESL;
>>>>> use IO::Socket::INET;
>>>>> use Data::Dumper;
>>>>>
>>>>> my $ip = "192.168.1.222";
>>>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>>>> die "Could not create socket: $!\n" unless $sock;
>>>>> &register_Signals();
>>>>>
>>>>> for(;;) {
>>>>>     my $new_sock = $sock->accept();
>>>>>     next if (not defined ($new_sock));
>>>>>     my $pid = fork();
>>>>>     if ($pid) {
>>>>>         close($new_sock);
>>>>>         next;
>>>>>     }
>>>>>     print "CHILD PID: $$\n";
>>>>>     my $host = $new_sock->sockhost();
>>>>>     my $fd = fileno($new_sock);
>>>>>
>>>>>     my $con = new ESL::ESLconnection($fd);
>>>>>     my $info = $con->getInfo();
>>>>>
>>>>>     my $uuid = $info->getHeader("unique-id");
>>>>>
>>>>>     printf "Connected call %s, from %s\n", $uuid,
>>>>> $info->getHeader("caller-caller-id-number");
>>>>>     my $r=$con->execute("answer");
>>>>>     print Dumper $r;
>>>>>     $con->events("plain","all");
>>>>>     my
>>>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>>>
>>>>>     while($con->connected()) {
>>>>>         my $e = $con->recvEvent();
>>>>>
>>>>>         if ($e) {
>>>>>             my $name = $e->getHeader("event-name");
>>>>>             print "EVENT [$name]\n";
>>>>>             if ($name eq "DTMF") {
>>>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>>>                 print "$digit\n";
>>>>>             }
>>>>>         }
>>>>>     }
>>>>>     close($new_sock);
>>>>> }
>>>>> sub register_Signals() {
>>>>>     foreach ( keys %SIG ) {
>>>>>         $SIG{$_} = 'sig_Handler';
>>>>>     }
>>>>> }
>>>>>
>>>>> sub sig_Handler() {
>>>>>     my $handle=$_[0];
>>>>>     if($handle eq "INT") {
>>>>>         print "$$: SIGNAL SIG$handle is generated\n";
>>>>>     }
>>>>> }
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yehavi.bourvine at gmail.com  Tue Jan 19 02:26:38 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 19 Jan 2010 12:26:38 +0200
Subject: [Freeswitch-users] Polycom configuration for FreeSWITCH
In-Reply-To: <003701ca964b$3241b100$96c51300$@net>
References: <003701ca964b$3241b100$96c51300$@net>
Message-ID: <d361ca7b1001190226y70f6dfa2o3cbf4bcffff6a887@mail.gmail.com>

I've documented my Polycom's setup in the wiki under "Polycom
configuration".

                  Regards, __Yehavi:


 2010/1/16 Adam Ford <lists at redbonez.net>

>   Does anyone who has successfully implemented Polycom IP301/501s, with
> FreeSWITCH, have a configuration they can share with me as an example? I
> have read the Polycom Admin Guide several times and understand what the
> settings are/do, I am just not sure which FreeSWITCH supports, which it
> doesn?t, and which need special configuration to work with FreeSWITCH.
>
>
>
> Thanks in advance,
>
> -Adam
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From linux4michelle at tamay-dogan.net  Tue Jan 19 02:28:54 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Tue, 19 Jan 2010 11:28:54 +0100
Subject: [Freeswitch-users] Which port I have to open?
Message-ID: <20100119102854.GQ4767@tamay-dogan.net>

Hello *,

my FreeSWITCH was in a DMZ but now I have installed it  on  a  dedicated
server (TI OMAP3530) and nothing as working... :-/

Please can someone tell me which ports I have to forward from my  router
fo Freeswitch and reverse?

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From linux4michelle at tamay-dogan.net  Tue Jan 19 02:31:39 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Tue, 19 Jan 2010 11:31:39 +0100
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
Message-ID: <20100119103139.GR4767@tamay-dogan.net>

Hi *,

I loss my last nerv, compiling all the time FreeSWITCH from source...

Is there someone providing a Debian Package from a repository?

Also it would be nice if FreeSWITCH go into the Debian distribution.

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From Russell.Mosemann at cune.org  Tue Jan 19 04:14:56 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Tue, 19 Jan 2010 06:14:56 -0600
Subject: [Freeswitch-users] Error in finding OpenZAP span id
In-Reply-To: <369c72d81001182327u38772291pa42ec950c11a1055@mail.gmail.com>
References: <369c72d81001182327u38772291pa42ec950c11a1055@mail.gmail.com>
Message-ID: <56DA8BCA12C2488A9CD829A12C87C81E@cune.pri>

>  When I started the freeswitch I have received the following error,
> 
> 2010-01-19 12:41:22.693212 [ERR] mod_openzap.c:2039 Error finding OpenZAP
> span id: name:PRI_1

You defined PRI_1 inside libpri_spans. Did you compile OpenZAP with libpri?

--
Russell Mosemann



From kond at nstel.ru  Tue Jan 19 05:22:12 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Tue, 19 Jan 2010 16:22:12 +0300
Subject: [Freeswitch-users] Which port I have to open?
In-Reply-To: <20100119102854.GQ4767@tamay-dogan.net>
Message-ID: <20100119132212.D20B011F53@mail.nstel.ru>

Did you take a look at
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Firewall?


> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Michelle Konzack
> Sent: Tuesday, January 19, 2010 1:29 PM
> To: FreeSWITCH Users
> Subject: [Freeswitch-users] Which port I have to open?
> 
> Hello *,
> 
> my FreeSWITCH was in a DMZ but now I have installed it  on  a  dedicated
> server (TI OMAP3530) and nothing as working... :-/
> 
> Please can someone tell me which ports I have to forward from my  router
> fo Freeswitch and reverse?
> 
> Thanks, Greetings and nice Day/Evening
>     Michelle Konzack
>     Systemadministrator
>     Electronic Engineer
>     Tamay Dogan Network
>     Debian GNU/Linux Consultant
> 
> --
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
> ##################### Debian GNU/Linux Consultant #####################
> <http://www.tamay-dogan.net/>                 Michelle Konzack
> <http://www.can4linux.org/>                   Apt. 917
> <http://www.flexray4linux.org/>               50, rue de Soultz
> Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
> IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
> ICQ    #328449886                             Tel. FR: +33  6  61925193



From stevendt at primrosebank.net  Tue Jan 19 05:31:42 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Tue, 19 Jan 2010 13:31:42 -0000
Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors
Message-ID: <2CCD94BEC92D4DE493B6FA65A554F286@bp1.ad.bp.com>

Hi,

I have just upgraded to the latest SVN (16400) and see a couple of errors when FS starts, although the basic functionality seems to be working.
Some DLLs have not been built, should I be worried about these errors and/or how do I correct them ?

2010-01-19 12:51:33.280718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_cluechoo.dll
**dll open error [126l]

2010-01-19 12:51:33.905718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_file_string.dll
**dll open error [126l]

2010-01-19 12:51:34.108843 [ERR] mod_spidermonkey.c:930 Error Loading module C:\FreeSWITCH\Debug\mod\mod_spidermonkey_core_db.dll
**dll open error [126l]


There is nothing in the Build log on the first two, but there is an error against the last one

88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.pdb'.
88>Make sure that the file is not open by another process and is not write-protected.
88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.idb'.
88>Make sure that the file is not open by another process and is not write-protected

regards
Dave

WindowsXP
VC++ Express 2008
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From linux4michelle at tamay-dogan.net  Tue Jan 19 05:59:20 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Tue, 19 Jan 2010 14:59:20 +0100
Subject: [Freeswitch-users] Which port I have to open?
In-Reply-To: <20100119132212.D20B011F53@mail.nstel.ru>
References: <20100119102854.GQ4767@tamay-dogan.net>
	<20100119132212.D20B011F53@mail.nstel.ru>
Message-ID: <20100119135919.GU4767@tamay-dogan.net>

Hello,

Am 2010-01-19 16:22:12, schrieb Nikolay Kondratyev:
> Did you take a look at
> http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Firewall?

Hmmm, -- do I have to forward incoming 16384-32768 ports to
my FreeSwitch box?  This would kick off my entired network.

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From rupa at rupa.com  Tue Jan 19 06:49:56 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Jan 2010 08:49:56 -0600
Subject: [Freeswitch-users] Which port I have to open?
In-Reply-To: <20100119135919.GU4767@tamay-dogan.net>
References: <20100119102854.GQ4767@tamay-dogan.net>
	<20100119132212.D20B011F53@mail.nstel.ru>
	<20100119135919.GU4767@tamay-dogan.net>
Message-ID: <b39273531001190649j4a510c6fu4cc6dcf3826c8d59@mail.gmail.com>

Does your router support NAT-PMP or UPNP?  If so, FS will do all the port
forwarding configuration automatically.

On Tue, Jan 19, 2010 at 7:59 AM, Michelle Konzack <
linux4michelle at tamay-dogan.net> wrote:

> Hello,
>
> Am 2010-01-19 16:22:12, schrieb Nikolay Kondratyev:
> > Did you take a look at
> > http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Firewall?
>
> Hmmm, -- do I have to forward incoming 16384-32768 ports to
> my FreeSwitch box?  This would kick off my entired network.
>
> Thanks, Greetings and nice Day/Evening
>    Michelle Konzack
>    Systemadministrator
>    Electronic Engineer
>    Tamay Dogan Network
>    Debian GNU/Linux Consultant
>
> --
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
> ##################### Debian GNU/Linux Consultant #####################
> <http://www.tamay-dogan.net/>                 Michelle Konzack
> <http://www.can4linux.org/>                   Apt. 917
> <http://www.flexray4linux.org/>               50, rue de Soultz
> Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
> IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
> ICQ    #328449886                             Tel. FR: +33  6  61925193
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From linux4michelle at tamay-dogan.net  Tue Jan 19 07:39:40 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Tue, 19 Jan 2010 16:39:40 +0100
Subject: [Freeswitch-users] Which port I have to open?
In-Reply-To: <b39273531001190649j4a510c6fu4cc6dcf3826c8d59@mail.gmail.com>
References: <20100119102854.GQ4767@tamay-dogan.net>
	<20100119132212.D20B011F53@mail.nstel.ru>
	<20100119135919.GU4767@tamay-dogan.net>
	<b39273531001190649j4a510c6fu4cc6dcf3826c8d59@mail.gmail.com>
Message-ID: <20100119153940.GW4767@tamay-dogan.net>

Hello,

Am 2010-01-19 08:49:56, schrieb Rupa Schomaker:
> Does your router support NAT-PMP or UPNP?  If so, FS will do all the port
> forwarding configuration automatically.

My router is the Freebox from my ISP <http://www.free.fr>  and  it  does
UPNP.  Also I have complete config datd to connect FreeSWITCH to my  SIP
account from <free.fr> and to my SIP-Povider <sipgate.de>

Since I have a bunch of problems with the Freebox, I  like  to  put  the
FreeBox into bridge mode and use a "Debian GNU/Linux Lenny/Squeeze"  box
to do the routing.

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From help at pdscc.com  Tue Jan 19 07:41:30 2010
From: help at pdscc.com (Harondel J. Sibble)
Date: Tue, 19 Jan 2010 07:41:30 -0800
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4 (fixed)
In-Reply-To: <8C20E0B1-0E56-4892-9D36-4D6DF3331244@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20100105065356.AEE0612F5@sinclaire.sibble.net>,
	<8C20E0B1-0E56-4892-9D36-4D6DF3331244@freeswitch.org>
Message-ID: <20100119154135.E51471DB501@sinclaire.sibble.net>

Okay, all I've done so far is make current from SVN and now have latest 
installed

2010-01-19 07:36:12.591283 [CONSOLE] switch_core.c:1565
FreeSWITCH Version 1.0.trunk (16400) Started.

Do I need to rebuild libzrtp for Freeswitch? Or just try it as is?


On 7 Jan 2010 at 17:30, Brian West wrote:

> Harondel,
> 	Please update your FreeSWITCH source rev 16204 fixes the SAS passing in all
> cases from my testing.  You'll need the latest ZRTP Lib and zfone
> application to make this work... I'm not too sure Tiviphone does this yet as
> I don't have one to test with.  This also fixes the issue when both sides
> are enrolled.  Next we will fix the video portion so both video and audio
> will go thru zrtp.
> 
> Please try it and let me know.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From jeff at jefflenk.com  Tue Jan 19 07:44:41 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 19 Jan 2010 09:44:41 -0600
Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors
In-Reply-To: <2CCD94BEC92D4DE493B6FA65A554F286@bp1.ad.bp.com>
References: <2CCD94BEC92D4DE493B6FA65A554F286@bp1.ad.bp.com>
Message-ID: <COL117-W6424736CC849832D6D9DBFB9650@phx.gbl>


Hi Dave,

 

mod_cluechoo is not built for Windows and is not needed.

The other files should build fine. Do they exist in your Debug folder?

 

- Jeff
 


From: stevendt at primrosebank.net
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 19 Jan 2010 13:31:42 +0000
Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors




Hi,
 
I have just upgraded to the latest SVN (16400) and see a couple of errors when FS starts, although the basic functionality seems to be working.
Some DLLs have not been built, should I be worried about these errors and/or how do I correct them ?
 
2010-01-19 12:51:33.280718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_cluechoo.dll
**dll open error [126l]

2010-01-19 12:51:33.905718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_file_string.dll
**dll open error [126l]

2010-01-19 12:51:34.108843 [ERR] mod_spidermonkey.c:930 Error Loading module C:\FreeSWITCH\Debug\mod\mod_spidermonkey_core_db.dll
**dll open error [126l]

 
There is nothing in the Build log on the first two, but there is an error against the last one
 
88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.pdb'.
88>Make sure that the file is not open by another process and is not write-protected.
88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.idb'.
88>Make sure that the file is not open by another process and is not write-protected
 
regards
Dave
 
WindowsXP
VC++ Express 2008 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with powerful SPAM protection.
http://clk.atdmt.com/GBL/go/196390707/direct/01/
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From john at acsol.net  Tue Jan 19 08:02:30 2010
From: john at acsol.net (John)
Date: Tue, 19 Jan 2010 09:02:30 -0700
Subject: [Freeswitch-users] Voicemail to email problems
Message-ID: <4B55D796.6070001@acsol.net>

Voicemail to email gets a broken pipe error. I believe this is happening 
while writing the file to /tmp? The mail.xxxxxxxxxxxx file is never 
written. Any ideas? Thanks




From anthony.minessale at gmail.com  Tue Jan 19 08:04:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Jan 2010 10:04:45 -0600
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <7d79b3931001190107o200ec04dredd76689dd235588@mail.gmail.com>
References: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
	<191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>
	<7d79b3931001132357i36bbb482jdf2bbdd3aea2a583@mail.gmail.com>
	<7d79b3931001180322p74a1227qe0c2199a77cbfe2@mail.gmail.com>
	<191c3a031001180824q6e364c72g3f789892597e9469@mail.gmail.com>
	<7d79b3931001190107o200ec04dredd76689dd235588@mail.gmail.com>
Message-ID: <191c3a031001190804m692f63acsaf852b8809db09d1@mail.gmail.com>

Its nothing we can fix, that is what you must do on a failed read syscall.
you can do a non blocking read instead and take your chances.


On Tue, Jan 19, 2010 at 3:07 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> I tried with SIGUSR1, but no progress. I got SERVER_DISCONNECTED.
> Output:
>
> CHILD 3814: Received USR1
> EVENT [SERVER_DISCONNECTED]
>
> In esl.c, in esl_recv_event() function, line no: 824
> if (rrval < 0) {
>                         strerror_r(handle->errnum, handle->err,
> sizeof(handle->err));
>                         goto fail;
> }
> When the program is blocked under receive, I passed the signal. So recv
> returns -1, and in fail: it call esl_disconnect(handle).
>
> Is it because of this??? If so, whether it should be fixed or not???
>
>
>
>
> On Mon, Jan 18, 2010 at 9:54 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> try a less famous signal like SIGUSR1 it's possible something in perl
>> still reacts to SIGINT
>>
>>
>>
>> On Mon, Jan 18, 2010 at 5:22 AM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>> Here is the result
>>>
>>> Program:
>>>
>>> require ESL;
>>> use IO::Socket::INET;
>>> use Data::Dumper;
>>>
>>> my $ip = "192.168.1.222";
>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>> die "Could not create socket: $!\n" unless $sock;
>>>
>>>
>>> for(;;) {
>>>     my $new_sock = $sock->accept();
>>>     next if (not defined ($new_sock));
>>>     my $pid = fork();
>>>     if ($pid) {
>>>         close($new_sock);
>>>         next;
>>>     }
>>>     print "CHILD PID: $$\n";
>>>     &register_Signals_Child();
>>>     sub register_Signals_Child() {
>>>         foreach ( keys %SIG ) {
>>>             $SIG{$_} = 'Handler';
>>>         }
>>>     }
>>>
>>>     sub Handler() {
>>>
>>>         my $handle=$_[0];
>>>         if($handle eq "INT") {
>>>             print "CHILD $$: SIGNAL SIG$handle is generated\n";
>>>         }
>>>         else
>>>         {
>>>             print "CHILD $$: Received $handle\n";
>>>
>>>         }
>>>     }
>>>     my $host = $new_sock->sockhost();
>>>     my $fd = fileno($new_sock);
>>>
>>>     my $con = new ESL::ESLconnection($fd);
>>>     my $info = $con->getInfo();
>>>
>>>     my $uuid = $info->getHeader("unique-id");
>>>
>>>     printf "Connected call %s, from %s\n", $uuid,
>>> $info->getHeader("caller-caller-id-number");
>>>     my $r=$con->execute("answer");
>>>     print Dumper $r;
>>>     $con->events("plain","all");
>>>     my
>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>     while($con->connected()) {
>>>         my $e = $con->recvEvent();
>>>
>>>         if ($e) {
>>>             my $name = $e->getHeader("event-name");
>>>             print "EVENT [$name]\n";
>>>             if ($name eq "DTMF") {
>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>                 print "$digit\n";
>>>             }
>>>         }
>>>     }
>>>     close($new_sock);
>>> }
>>>
>>> I executed the program and the following things were printed
>>>
>>> CHILD PID: 6778
>>> Connected call e0d1001a-03f4-11df-b002-db488337e0ea, from 1001
>>> $VAR1 = 0;
>>> EVENT [CHANNEL_EXECUTE]
>>> EVENT [CHANNEL_ANSWER]
>>> EVENT [CHANNEL_EXECUTE_COMPLETE]
>>> EVENT [COMMAND]
>>> EVENT [CHANNEL_EXECUTE]
>>> EVENT [HEARTBEAT]
>>> EVENT [RE_SCHEDULE]
>>> EVENT [CHANNEL_EXECUTE_COMPLETE]
>>>
>>> Then from another shell I executed kill -2 6778, the result is follows
>>> CHILD 6778: SIGNAL SIGINT is generated
>>> EVENT [SERVER_DISCONNECTED]
>>>
>>> But the child process is still running as expected.
>>> But  I don't know why I received SERVER_DISCONNECTED from freeswitch???
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Thu, Jan 14, 2010 at 1:27 PM, lakshmanan ganapathy <
>>> lakindia89 at gmail.com> wrote:
>>>
>>>> I taught the signal handler will be inherited by the child process. It
>>>> also does like that.
>>>> After making a call, If I press ctrl + c, the above program printed
>>>> PARENT PID: Signal SIGINT is generated
>>>> CHILD PID: Signal SIGINT is generated.
>>>>
>>>> So I think the sigal handlers will be inherited to the child.
>>>> Anyway I'll also try registering signal handlers in child also, and then
>>>> I'll come back with that result.
>>>>
>>>> Thanks....
>>>> On Wed, Jan 13, 2010 at 9:48 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> you would have to register signals in your child process too
>>>>>
>>>>>   On Wed, Jan 13, 2010 at 3:13 AM, lakshmanan ganapathy <
>>>>> lakindia89 at gmail.com> wrote:
>>>>>
>>>>>>  Hi all,
>>>>>>
>>>>>> I've done a sample program (In perl ESL) , which play a file to the
>>>>>> caller and then it will call recvEvent() and print the event name. I've
>>>>>> handled signals also.
>>>>>>
>>>>>> When I send SIGINT to my program (Perl), the signal handler is called
>>>>>> and I can see the print output. But in the same time,  I received
>>>>>> SERVER_DISCONNECTED from freeswitch as event.
>>>>>>
>>>>>> I don't know why I received SERVER_DISCONNECTED from freeswitch. Is it
>>>>>> because, the recvEvent() from perl internally  calls the recvevent function
>>>>>> in the Esl.c and when it waits to receive the information from socket, the
>>>>>> signal occurred???
>>>>>>
>>>>>> Please clarify me!!
>>>>>>
>>>>>> Here is my program
>>>>>> require ESL;
>>>>>> use IO::Socket::INET;
>>>>>> use Data::Dumper;
>>>>>>
>>>>>> my $ip = "192.168.1.222";
>>>>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>>>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>>>>> die "Could not create socket: $!\n" unless $sock;
>>>>>> &register_Signals();
>>>>>>
>>>>>> for(;;) {
>>>>>>     my $new_sock = $sock->accept();
>>>>>>     next if (not defined ($new_sock));
>>>>>>     my $pid = fork();
>>>>>>     if ($pid) {
>>>>>>         close($new_sock);
>>>>>>         next;
>>>>>>     }
>>>>>>     print "CHILD PID: $$\n";
>>>>>>     my $host = $new_sock->sockhost();
>>>>>>     my $fd = fileno($new_sock);
>>>>>>
>>>>>>     my $con = new ESL::ESLconnection($fd);
>>>>>>     my $info = $con->getInfo();
>>>>>>
>>>>>>     my $uuid = $info->getHeader("unique-id");
>>>>>>
>>>>>>     printf "Connected call %s, from %s\n", $uuid,
>>>>>> $info->getHeader("caller-caller-id-number");
>>>>>>     my $r=$con->execute("answer");
>>>>>>     print Dumper $r;
>>>>>>     $con->events("plain","all");
>>>>>>     my
>>>>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>>>>
>>>>>>     while($con->connected()) {
>>>>>>         my $e = $con->recvEvent();
>>>>>>
>>>>>>         if ($e) {
>>>>>>             my $name = $e->getHeader("event-name");
>>>>>>             print "EVENT [$name]\n";
>>>>>>             if ($name eq "DTMF") {
>>>>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>>>>                 print "$digit\n";
>>>>>>             }
>>>>>>         }
>>>>>>     }
>>>>>>     close($new_sock);
>>>>>> }
>>>>>> sub register_Signals() {
>>>>>>     foreach ( keys %SIG ) {
>>>>>>         $SIG{$_} = 'sig_Handler';
>>>>>>     }
>>>>>> }
>>>>>>
>>>>>> sub sig_Handler() {
>>>>>>     my $handle=$_[0];
>>>>>>     if($handle eq "INT") {
>>>>>>         print "$$: SIGNAL SIG$handle is generated\n";
>>>>>>     }
>>>>>> }
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From stevendt at primrosebank.net  Tue Jan 19 08:13:44 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Tue, 19 Jan 2010 16:13:44 -0000
Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors
References: <2CCD94BEC92D4DE493B6FA65A554F286@bp1.ad.bp.com>
	<COL117-W6424736CC849832D6D9DBFB9650@phx.gbl>
Message-ID: <E5A67EA005F34E68B5501C860A9B96BC@bp1.ad.bp.com>

Hi Jeff,

the DLLs are not in the Debug folders pointed to by the error messages ?

regards
Dave

  ----- Original Message ----- 
  From: Jeff Lenk 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, January 19, 2010 3:44 PM
  Subject: Re: [Freeswitch-users] SVN 16400 - Windows Build Errors


  Hi Dave,
   
  mod_cluechoo is not built for Windows and is not needed.
  The other files should build fine. Do they exist in your Debug folder?
   
  - Jeff
   

------------------------------------------------------------------------------
  From: stevendt at primrosebank.net
  To: freeswitch-users at lists.freeswitch.org
  Date: Tue, 19 Jan 2010 13:31:42 +0000
  Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors


  Hi,

  I have just upgraded to the latest SVN (16400) and see a couple of errors when FS starts, although the basic functionality seems to be working.
  Some DLLs have not been built, should I be worried about these errors and/or how do I correct them ?

  2010-01-19 12:51:33.280718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_cluechoo.dll
  **dll open error [126l]

  2010-01-19 12:51:33.905718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_file_string.dll
  **dll open error [126l]

  2010-01-19 12:51:34.108843 [ERR] mod_spidermonkey.c:930 Error Loading module C:\FreeSWITCH\Debug\mod\mod_spidermonkey_core_db.dll
  **dll open error [126l]


  There is nothing in the Build log on the first two, but there is an error against the last one

  88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.pdb'.
  88>Make sure that the file is not open by another process and is not write-protected.
  88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.idb'.
  88>Make sure that the file is not open by another process and is not write-protected

  regards
  Dave

  WindowsXP
  VC++ Express 2008


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From devel at thom.fr.eu.org  Tue Jan 19 08:31:45 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Tue, 19 Jan 2010 17:31:45 +0100
Subject: [Freeswitch-users] Voicemail to email problems
Message-ID: <ab161f4749d2cf203ec197497f2ce22e@thom.fr.eu.org>

This is not the error I get.

I modified my config so that the command executed to send the email is cat
/tmp/mail.xxxxxxx | /usr/local/freeswitch/email.sh
with email.sh teeing the stdin to a file and to sendmail.

syslog reports sendmail segfault.

Fran?ois

On Tue, 19 Jan 2010 09:02:30 -0700, John <john at acsol.net> wrote:
> Voicemail to email gets a broken pipe error. I believe this is happening

> while writing the file to /tmp? The mail.xxxxxxxxxxxx file is never 
> written. Any ideas? Thanks
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From a.alalousi at gmail.com  Tue Jan 19 08:31:45 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Tue, 19 Jan 2010 16:31:45 +0000
Subject: [Freeswitch-users] G729 coded issues
Message-ID: <cd93b3701001190831u4147214cn9832c6f469c6d17e@mail.gmail.com>

Hi everyone,

I have the following scenario and a major customer-affecting issue thereof.

Here is the scenario: customer traffic encoded as G.729 from a cisco gateway
-> our FS (G729 passthrough) -> remote end gw (G729)

Calls were failing at an alarming rate, so I looked at the debug logs. It
transpired that the Cisco is offering G729 annex b, while the remote end can
only do G729a.

Besides changing source or destination preferences, is there a way to ensure
that G729a is used end-end ?

Thanks in advance.



-- 
Ahmed Naji
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From brian at freeswitch.org  Tue Jan 19 08:36:32 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Jan 2010 10:36:32 -0600
Subject: [Freeswitch-users] G729 coded issues
In-Reply-To: <cd93b3701001190831u4147214cn9832c6f469c6d17e@mail.gmail.com>
References: <cd93b3701001190831u4147214cn9832c6f469c6d17e@mail.gmail.com>
Message-ID: <6D67C7A9-42CD-4F5F-AD18-E9568098FB71@freeswitch.org>

g729a is 100% INVALID in the sdp fix the param in your cisco SPA or your Linksys SPA phone and it will stop doing that.  Hopefully they'll fix this "bug" soon in the cisco phones to not include the a in the sdp.  The fmtp is the proper way to specify annex a or any other options for g729.

/b

On Jan 19, 2010, at 10:31 AM, Ahmed Naji wrote:

> Hi everyone,
> 
> I have the following scenario and a major customer-affecting issue thereof.
> 
> Here is the scenario: customer traffic encoded as G.729 from a cisco gateway
> -> our FS (G729 passthrough) -> remote end gw (G729)
> 
> Calls were failing at an alarming rate, so I looked at the debug logs. It
> transpired that the Cisco is offering G729 annex b, while the remote end can
> only do G729a.
> 
> Besides changing source or destination preferences, is there a way to ensure
> that G729a is used end-end ?
> 
> Thanks in advance.



From mailinglist at fribert.dk  Tue Jan 19 08:56:01 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Tue, 19 Jan 2010 17:56:01 +0100
Subject: [Freeswitch-users] Svar: Re: How do I invite group to join	
	existing call?
In-Reply-To: <87f2f3b91001181209y7a0aa68fs8a580712484c7a11@mail.gmail.com>
References: <4B5412C6020000E1000003D6@mail.fribert.dk>
	<87f2f3b91001181209y7a0aa68fs8a580712484c7a11@mail.gmail.com>
Message-ID: <4B55F231020000E1000003DB@mail.fribert.dk>

GRIN, ok, I'll see if I can punder a bit more on the subject.
 
I have a home setup, I have a couple of phones attached, as well as a couple of computers.
The private line sends a call to the private group.
The phones attached to the private group rings
My wife picks it up
She wants me to join the conversation, so she presses *11 or something :-)
Now the phones in the private groups rings again, and I pick it up, and we have a conference with a calling party, and two local phones.
 
Any ideas from you guru's?
 
BR
Fribse

>>> 18-01-2010 kl. 21:09 skrev Michael Collins <msc at freeswitch.org> i meddelelsen <87f2f3b91001181209y7a0aa68fs8a580712484c7a11 at mail.gmail.com>:



On Sun, Jan 17, 2010 at 10:50 PM, mailinglist <mailinglist at fribert.dk> wrote:


Hi All

I would like to be able to invite a group / global to join an existing call, but how do I accomplish this, can it be done?


Malfunction! Need Input!

Could you give us a few more details on what you're trying to accomplish? What's the big picture? I'm curious what problem you're trying to solve. I'm sure the gang here will have thoughts to pass along. :)

-MC

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From lists at redbonez.net  Tue Jan 19 08:59:37 2010
From: lists at redbonez.net (Adam Ford)
Date: Tue, 19 Jan 2010 09:59:37 -0700
Subject: [Freeswitch-users] Polycom configuration for FreeSWITCH
In-Reply-To: <d361ca7b1001190226y70f6dfa2o3cbf4bcffff6a887@mail.gmail.com>
References: <003701ca964b$3241b100$96c51300$@net>
	<d361ca7b1001190226y70f6dfa2o3cbf4bcffff6a887@mail.gmail.com>
Message-ID: <015901ca9928$c98e2cc0$5caa8640$@net>

Thanks Yehavi!

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yehavi
Bourvine
Sent: Tuesday, January 19, 2010 3:27 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom configuration for FreeSWITCH

 

I've documented my Polycom's setup in the wiki under "Polycom
configuration".

 

                  Regards, __Yehavi:


 

2010/1/16 Adam Ford <lists at redbonez.net>

Does anyone who has successfully implemented Polycom IP301/501s, with
FreeSWITCH, have a configuration they can share with me as an example? I
have read the Polycom Admin Guide several times and understand what the
settings are/do, I am just not sure which FreeSWITCH supports, which it
doesn't, and which need special configuration to work with FreeSWITCH.

 

Thanks in advance,

-Adam

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Tue Jan 19 09:01:08 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Jan 2010 11:01:08 -0600
Subject: [Freeswitch-users] Svar: Re: How do I invite group to join
	existing call?
In-Reply-To: <4B55F231020000E1000003DB@mail.fribert.dk>
References: <4B5412C6020000E1000003D6@mail.fribert.dk>
	<87f2f3b91001181209y7a0aa68fs8a580712484c7a11@mail.gmail.com>
	<4B55F231020000E1000003DB@mail.fribert.dk>
Message-ID: <3D153366-FE85-4A7A-9351-63D5FBE3F287@freeswitch.org>

What kind of phones?

/b

On Jan 19, 2010, at 10:56 AM, mailinglist wrote:

> GRIN, ok, I'll see if I can punder a bit more on the subject.
>  
> I have a home setup, I have a couple of phones attached, as well as a couple of computers.
> The private line sends a call to the private group.
> The phones attached to the private group rings
> My wife picks it up
> She wants me to join the conversation, so she presses *11 or something :-)
> Now the phones in the private groups rings again, and I pick it up, and we have a conference with a calling party, and two local phones.
>  
> Any ideas from you guru's?
>  
> BR
> Fribse

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From mailinglist at fribert.dk  Tue Jan 19 09:02:08 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Tue, 19 Jan 2010 18:02:08 +0100
Subject: [Freeswitch-users] Home setup with home company
Message-ID: <4B55F3A0020000E1000003E0@mail.fribert.dk>

I have a very small one man constultancy company that has the occasional call, unfortunately we are getting more spam calls after hours than real calls during work hours, so I would like to set up a TOD system.
 
First step for me is just playing with the TOD example, I've gotten this far:
 
<extension name="CompanyTOD" continue="true">
   <condition field="destination_number" expression="^(8203)$"/>
      <!-- man strftime - M-F, 9AM to 5PM -->
      <condition field="${strftime(%w)}" expression="^([1-5])$"/>
      <condition field="${strftime(%H%M)}" expression="^((09|1[0-6])[0-5][0-9]|1700)$">
         <action application="set" data="call_timeout=45"/>
         <action application="set" data="domain_name=10.11.12.25"/>
         <action application="set" data="continue_on_fail=true"/>
         <action application="set" data="hangup_after_bridge=true"/>
         <action application="bridge" data="group/company@${domain_name}"/ ( 'mailto:group/company@$domain_name"/' )>
         <action application="answer" data=""/>
         <action application="sleep" data="1000"/>
         <action application="voicemail" data="default 10.11.12.25 10199"/>
         <anti-action application="voicemail" data="default 10.11.12.25 10199"/>
      </condition>
</extension>
My idea with this, was that in the time 9-17 mon-fri, it rings, the rest of the time it's sent directly to the voicemail.
I would of course like to have it not take messages outside work hours, but that's just refining :-)
 
But it picks up the call, and then nothing...
 
BR
Fribse
 
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From steveu at coppice.org  Tue Jan 19 09:10:44 2010
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 20 Jan 2010 01:10:44 +0800
Subject: [Freeswitch-users] G729 coded issues
In-Reply-To: <6D67C7A9-42CD-4F5F-AD18-E9568098FB71@freeswitch.org>
References: <cd93b3701001190831u4147214cn9832c6f469c6d17e@mail.gmail.com>
	<6D67C7A9-42CD-4F5F-AD18-E9568098FB71@freeswitch.org>
Message-ID: <4B55E794.6020909@coppice.org>

On 01/20/2010 12:36 AM, Brian West wrote:
> g729a is 100% INVALID in the sdp fix the param in your cisco SPA or your Linksys SPA phone and it will stop doing that.  Hopefully they'll fix this "bug" soon in the cisco phones to not include the a in the sdp.  The fmtp is the proper way to specify annex a or any other options for g729.
>    
Annex A only affects the inner workings of the codec. There is 
absolutely no difference whatsoever between G.729 and G.729A on the 
wire. The SDP has no reason to mention it, and the standards say it 
shouldn't.
> /b
>
> On Jan 19, 2010, at 10:31 AM, Ahmed Naji wrote:
>
>    
>> Hi everyone,
>>
>> I have the following scenario and a major customer-affecting issue thereof.
>>
>> Here is the scenario: customer traffic encoded as G.729 from a cisco gateway
>> ->  our FS (G729 passthrough) ->  remote end gw (G729)
>>
>> Calls were failing at an alarming rate, so I looked at the debug logs. It
>> transpired that the Cisco is offering G729 annex b, while the remote end can
>> only do G729a.
>>
>> Besides changing source or destination preferences, is there a way to ensure
>> that G729a is used end-end ?
>>
>> Thanks in advance.
>>      
>
Steve



From earlpinkerton at gmail.com  Tue Jan 19 00:33:04 2010
From: earlpinkerton at gmail.com (Earl Pinkerton)
Date: Tue, 19 Jan 2010 00:33:04 -0800
Subject: [Freeswitch-users] IVR returning info via event socket
Message-ID: <432c6c2b1001190033x2d4e07e7i985ef24bd6f52c7c@mail.gmail.com>

Hi All,

I am trying to call an IVR remotely over the event socket port.  I am
testing using telnet.  I call originate (similar to the following):

api originate
{origination_caller_id_number=8885551111,ignore_early_media=true}sofia/gateway/teliax/18885552222
&ivr(demo_ivr)

This works fine (places the call, runs the IVR, accepts DTMF tones to move
between menus).

My problem is that I would like the remote app to send a PIN as part of the
originate call, then have the callie enter the PIN via DTMF and indicate
whether there was a match or not back to the calling app (which is a remote
web program).  We are trying to keep it simple and have the remote app just
call the IVR and get the result, without having to control the IVR through
asynchronous events.

Is there any way for the IVR to check for a match, then somehow send info
back over the port to the calling app?

By the way, I tried using menu-exec-api inside ivr.conf.xml (to see if I
might be able to use the echo api call or something similar), but I got the
following error:

  2010-01-19 01:22:55.796658 [WARNING] switch_ivr_menu.c:704 Invalid Action
[menu-exec-api]

Thanks.
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From ahmed.ajmal at gmail.com  Tue Jan 19 07:52:22 2010
From: ahmed.ajmal at gmail.com (Ahmed Bhaila)
Date: Tue, 19 Jan 2010 20:52:22 +0500
Subject: [Freeswitch-users] Help with CDR fields:Other-Leg-Network-Addr and
	Other-Leg-Destination-Number
Message-ID: <9d22cc171001190752j797fdf39radc404510128718c@mail.gmail.com>

Hello

I am using a template for Master.csv and having problems with getting values
for the following fields: Other-Leg-Network-Addr and
Other-Leg-Destination-Number. All I am doing is just enclosing these fields
in ${} in my template definition and they always turn out to be empty but
they arent supposed to be as I am seeing values in the freeswitch console.so
I am not sure what is wrong here I have noticed that some of the channel
variables(start_stamp, end_stamp) have "variable_" prepended to them when I
see them in the console so when defining the template I omit the variable_
part and can get the values. Is this something similar? It seems like the
problem is with all field names that start with Other. Any help be greatly
appreciated.


Thanks
AB
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From brian at freeswitch.org  Tue Jan 19 09:36:36 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Jan 2010 11:36:36 -0600
Subject: [Freeswitch-users] Changes for deb and rpm files.
Message-ID: <5CA0D17B-6BD4-4BC6-B7F9-4A117A896C86@freeswitch.org>

Heads up.

http://jira.freeswitch.org/browse/FSCONFIG-17

Someone will need to update the configs for the debs and rpm's  

/b



From mailinglist at fribert.dk  Tue Jan 19 09:46:00 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Tue, 19 Jan 2010 18:46:00 +0100
Subject: [Freeswitch-users] Svar: Re: How do I invite group to join	
	existing call?
In-Reply-To: <3D153366-FE85-4A7A-9351-63D5FBE3F287@freeswitch.org>
References: <4B5412C6020000E1000003D6@mail.fribert.dk>
	<87f2f3b91001181209y7a0aa68fs8a580712484c7a11@mail.gmail.com>
	<4B55F231020000E1000003DB@mail.fribert.dk>
	<3D153366-FE85-4A7A-9351-63D5FBE3F287@freeswitch.org>
Message-ID: <4B55FDE8020000E1000003E5@mail.fribert.dk>

Hi B
 
Well, three are on siemens gigaset, then there's a linksys spa 901, then some computers with x-lite.
The gigaset can have two SIP calls established at the same time, so if she picks up a gigaset or the spa 901, shouldn't make any difference.
 
best regards
Fribse

>>> 19-01-2010 kl. 18:01 skrev Brian West <brian at freeswitch.org> i meddelelsen <3D153366-FE85-4A7A-9351-63D5FBE3F287 at freeswitch.org>:

What kind of phones?

/b

On Jan 19, 2010, at 10:56 AM, mailinglist wrote:




GRIN, ok, I'll see if I can punder a bit more on the subject.

 
I have a home setup, I have a couple of phones attached, as well as a couple of computers.
The private line sends a call to the private group.
The phones attached to the private group rings
My wife picks it up
She wants me to join the conversation, so she presses *11 or something :-)
Now the phones in the private groups rings again, and I pick it up, and we have a conference with a calling party, and two local phones.
 
Any ideas from you guru's?
 
BR
Fribse


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From lists at infosecurity.ch  Tue Jan 19 09:58:20 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Tue, 19 Jan 2010 18:58:20 +0100
Subject: [Freeswitch-users] Is possible from dialplan to detect if called
	party is online?
Message-ID: <A707BD2C-EC69-4379-9A71-4F3666221D09@infosecurity.ch>

There some condition that let to check if the called party is  
currently registered and online, before giving the called a return code?

We need to manage a condition where the called number is an iPhone and  
is usually "always offline" and we need to wakeup the voip client with  
a "push notifcation".

But before doing the push notification we should check if the user is  
already registered/online or not.

Don't know which path to follow

Fabio


From stevendt at primrosebank.net  Tue Jan 19 10:07:16 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Tue, 19 Jan 2010 18:07:16 -0000
Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors
References: <2CCD94BEC92D4DE493B6FA65A554F286@bp1.ad.bp.com><COL117-W6424736CC849832D6D9DBFB9650@phx.gbl>
	<E5A67EA005F34E68B5501C860A9B96BC@bp1.ad.bp.com>
Message-ID: <549C9A85D40D466B900D0415E7998B78@bp1.ad.bp.com>

Jeff,

just looked at the Tortoise SVN log for the 2008.Express.sln file, the last change was on the 17th with the comment . . . .

"move mod_spidermoney build to automake, fix spidermoneky dependencies (I think this really fixes -j builds), move mod_spidermonkey sub modules all under the same source directory and bundle their build together as one" (mikej). Perhaps a "gremlin" has slipped in there for the Spidemonkey build ?

(The only thing that I changed from the default configuration was to download and build the 16k and 32k music and sounds )

regards
Dave

  ----- Original Message ----- 
  From: Dave Stevenson 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, January 19, 2010 4:13 PM
  Subject: Re: [Freeswitch-users] SVN 16400 - Windows Build Errors


  Hi Jeff,

  the DLLs are not in the Debug folders pointed to by the error messages ?

  regards
  Dave

    ----- Original Message ----- 
    From: Jeff Lenk 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Tuesday, January 19, 2010 3:44 PM
    Subject: Re: [Freeswitch-users] SVN 16400 - Windows Build Errors


    Hi Dave,
     
    mod_cluechoo is not built for Windows and is not needed.
    The other files should build fine. Do they exist in your Debug folder?
     
    - Jeff
     

----------------------------------------------------------------------------
    From: stevendt at primrosebank.net
    To: freeswitch-users at lists.freeswitch.org
    Date: Tue, 19 Jan 2010 13:31:42 +0000
    Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors


    Hi,

    I have just upgraded to the latest SVN (16400) and see a couple of errors when FS starts, although the basic functionality seems to be working.
    Some DLLs have not been built, should I be worried about these errors and/or how do I correct them ?

    2010-01-19 12:51:33.280718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_cluechoo.dll
    **dll open error [126l]

    2010-01-19 12:51:33.905718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_file_string.dll
    **dll open error [126l]

    2010-01-19 12:51:34.108843 [ERR] mod_spidermonkey.c:930 Error Loading module C:\FreeSWITCH\Debug\mod\mod_spidermonkey_core_db.dll
    **dll open error [126l]


    There is nothing in the Build log on the first two, but there is an error against the last one

    88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.pdb'.
    88>Make sure that the file is not open by another process and is not write-protected.
    88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.idb'.
    88>Make sure that the file is not open by another process and is not write-protected

    regards
    Dave

    WindowsXP
    VC++ Express 2008


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From anthony.minessale at gmail.com  Tue Jan 19 11:37:58 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Jan 2010 13:37:58 -0600
Subject: [Freeswitch-users] Help with CDR fields:Other-Leg-Network-Addr
	and Other-Leg-Destination-Number
In-Reply-To: <9d22cc171001190752j797fdf39radc404510128718c@mail.gmail.com>
References: <9d22cc171001190752j797fdf39radc404510128718c@mail.gmail.com>
Message-ID: <191c3a031001191137k72530079t402ff54d03b07756@mail.gmail.com>

those are not variables only variables expand.

try bleg_network_addr
and bleg_destination_number


On Tue, Jan 19, 2010 at 9:52 AM, Ahmed Bhaila <ahmed.ajmal at gmail.com> wrote:

> Hello
>
> I am using a template for Master.csv and having problems with getting
> values for the following fields: Other-Leg-Network-Addr and
> Other-Leg-Destination-Number. All I am doing is just enclosing these fields
> in ${} in my template definition and they always turn out to be empty but
> they arent supposed to be as I am seeing values in the freeswitch console.so
> I am not sure what is wrong here I have noticed that some of the channel
> variables(start_stamp, end_stamp) have "variable_" prepended to them when I
> see them in the console so when defining the template I omit the variable_
> part and can get the values. Is this something similar? It seems like the
> problem is with all field names that start with Other. Any help be greatly
> appreciated.
>
>
> Thanks
> AB
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From john at acsol.net  Tue Jan 19 11:38:46 2010
From: john at acsol.net (John)
Date: Tue, 19 Jan 2010 12:38:46 -0700
Subject: [Freeswitch-users] Voicemail to email problems
In-Reply-To: <ab161f4749d2cf203ec197497f2ce22e@thom.fr.eu.org>
References: <ab161f4749d2cf203ec197497f2ce22e@thom.fr.eu.org>
Message-ID: <4B560A46.90000@acsol.net>

I am getting a segfault as well; however I use exim4 on Debian. Email is 
never being written to /tmp. I too can run the exim command by hand and 
it works. Any body have an idea or direction to look at?
Here is the error.
/bin/cat: write error: Broken pipe
sh: line 1:  7102 Done(1)                 /bin/cat /tmp/mail.12639191242a9f
       7103 Segmentation fault

Thanks



On 1/19/2010 9:31 AM, Fran?ois Legal wrote:
> This is not the error I get.
>
> I modified my config so that the command executed to send the email is cat
> /tmp/mail.xxxxxxx | /usr/local/freeswitch/email.sh
> with email.sh teeing the stdin to a file and to sendmail.
>
> syslog reports sendmail segfault.
>
> Fran?ois
>
> On Tue, 19 Jan 2010 09:02:30 -0700, John<john at acsol.net>  wrote:
>    
>> Voicemail to email gets a broken pipe error. I believe this is happening
>>      
>    
>> while writing the file to /tmp? The mail.xxxxxxxxxxxx file is never
>> written. Any ideas? Thanks
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>      
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    



From anthony.minessale at gmail.com  Tue Jan 19 11:43:12 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Jan 2010 13:43:12 -0600
Subject: [Freeswitch-users] Is possible from dialplan to detect if
	called party is online?
In-Reply-To: <A707BD2C-EC69-4379-9A71-4F3666221D09@infosecurity.ch>
References: <A707BD2C-EC69-4379-9A71-4F3666221D09@infosecurity.ch>
Message-ID: <191c3a031001191143w6220e6a4va3f71537133cd798@mail.gmail.com>

sofia_contact api function returns the contact addr of a registered user so
if it's blank they are not registered.




On Tue, Jan 19, 2010 at 11:58 AM, Fabio Pietrosanti (naif) <
lists at infosecurity.ch> wrote:

> There some condition that let to check if the called party is
> currently registered and online, before giving the called a return code?
>
> We need to manage a condition where the called number is an iPhone and
> is usually "always offline" and we need to wakeup the voip client with
> a "push notifcation".
>
> But before doing the push notification we should check if the user is
> already registered/online or not.
>
> Don't know which path to follow
>
> Fabio
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Tue Jan 19 11:46:04 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Jan 2010 13:46:04 -0600
Subject: [Freeswitch-users] Voicemail to email problems
In-Reply-To: <4B560A46.90000@acsol.net>
References: <ab161f4749d2cf203ec197497f2ce22e@thom.fr.eu.org>
	<4B560A46.90000@acsol.net>
Message-ID: <191c3a031001191146r257d34afq98f921c0ff07f2fc@mail.gmail.com>

sounds like exim pretending to be sendmail and not doing it very well.
I think there is a wiki page somewhere that tells you how to config it
properly.


On Tue, Jan 19, 2010 at 1:38 PM, John <john at acsol.net> wrote:

> I am getting a segfault as well; however I use exim4 on Debian. Email is
> never being written to /tmp. I too can run the exim command by hand and
> it works. Any body have an idea or direction to look at?
> Here is the error.
> /bin/cat: write error: Broken pipe
> sh: line 1:  7102 Done(1)                 /bin/cat /tmp/mail.12639191242a9f
>       7103 Segmentation fault
>
> Thanks
>
>
>
> On 1/19/2010 9:31 AM, Fran?ois Legal wrote:
> > This is not the error I get.
> >
> > I modified my config so that the command executed to send the email is
> cat
> > /tmp/mail.xxxxxxx | /usr/local/freeswitch/email.sh
> > with email.sh teeing the stdin to a file and to sendmail.
> >
> > syslog reports sendmail segfault.
> >
> > Fran?ois
> >
> > On Tue, 19 Jan 2010 09:02:30 -0700, John<john at acsol.net>  wrote:
> >
> >> Voicemail to email gets a broken pipe error. I believe this is happening
> >>
> >
> >> while writing the file to /tmp? The mail.xxxxxxxxxxxx file is never
> >> written. Any ideas? Thanks
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From help at pdscc.com  Tue Jan 19 11:50:40 2010
From: help at pdscc.com (Harondel J. Sibble)
Date: Tue, 19 Jan 2010 11:50:40 -0800
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4 (fixed)
In-Reply-To: <20100119154135.E51471DB501@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<8C20E0B1-0E56-4892-9D36-4D6DF3331244@freeswitch.org>,
	<20100119154135.E51471DB501@sinclaire.sibble.net>
Message-ID: <20100119195043.18DF01DB501@sinclaire.sibble.net>

Brian

Okay, so far so good, I'm getting consistent SAS's between devices, I'll test 
some more and report back.

I didn't rebuild libzrtp in this case.

On 19 Jan 2010 at 7:41, Harondel J. Sibble wrote:

> Okay, all I've done so far is make current from SVN and now have latest 
> installed
> 
> 2010-01-19 07:36:12.591283 [CONSOLE] switch_core.c:1565
> FreeSWITCH Version 1.0.trunk (16400) Started.
> 
> Do I need to rebuild libzrtp for Freeswitch? Or just try it as is?

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From brian at freeswitch.org  Tue Jan 19 12:14:18 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Jan 2010 14:14:18 -0600
Subject: [Freeswitch-users] Voicemail to email problems
In-Reply-To: <191c3a031001191146r257d34afq98f921c0ff07f2fc@mail.gmail.com>
References: <ab161f4749d2cf203ec197497f2ce22e@thom.fr.eu.org>
	<4B560A46.90000@acsol.net>
	<191c3a031001191146r257d34afq98f921c0ff07f2fc@mail.gmail.com>
Message-ID: <4E823F6B-75DC-44D9-BD18-C7773D77EAC9@freeswitch.org>

http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings

/b

On Jan 19, 2010, at 1:46 PM, Anthony Minessale wrote:

> sounds like exim pretending to be sendmail and not doing it very well.
> I think there is a wiki page somewhere that tells you how to config it properly.
> 
> 
> On Tue, Jan 19, 2010 at 1:38 PM, John <john at acsol.net> wrote:
> I am getting a segfault as well; however I use exim4 on Debian. Email is
> never being written to /tmp. I too can run the exim command by hand and
> it works. Any body have an idea or direction to look at?
> Here is the error.
> /bin/cat: write error: Broken pipe
> sh: line 1:  7102 Done(1)                 /bin/cat /tmp/mail.12639191242a9f
>       7103 Segmentation fault
> 
> Thanks
> 
> 
> 
> On 1/19/2010 9:31 AM, Fran?ois Legal wrote:
> > This is not the error I get.
> >
> > I modified my config so that the command executed to send the email is cat
> > /tmp/mail.xxxxxxx | /usr/local/freeswitch/email.sh
> > with email.sh teeing the stdin to a file and to sendmail.
> >
> > syslog reports sendmail segfault.
> >
> > Fran?ois
> >
> > On Tue, 19 Jan 2010 09:02:30 -0700, John<john at acsol.net>  wrote:
> >
> >> Voicemail to email gets a broken pipe error. I believe this is happening
> >>
> >
> >> while writing the file to /tmp? The mail.xxxxxxxxxxxx file is never
> >> written. Any ideas? Thanks
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jeff at jefflenk.com  Tue Jan 19 12:26:33 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 19 Jan 2010 14:26:33 -0600
Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors
In-Reply-To: <E5A67EA005F34E68B5501C860A9B96BC@bp1.ad.bp.com>
References: <2CCD94BEC92D4DE493B6FA65A554F286@bp1.ad.bp.com>,
	<COL117-W6424736CC849832D6D9DBFB9650@phx.gbl>,
	<E5A67EA005F34E68B5501C860A9B96BC@bp1.ad.bp.com>
Message-ID: <COL117-W15B08477F1FFA512F0F0B4B9650@phx.gbl>


Do a rebuild all from the solution and you should not see any errors.

 


From: stevendt at primrosebank.net
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 19 Jan 2010 16:13:44 +0000
Subject: Re: [Freeswitch-users] SVN 16400 - Windows Build Errors




Hi Jeff,
 
the DLLs are not in the Debug folders pointed to by the error messages ?
 
regards
Dave
 

----- Original Message ----- 
From: Jeff Lenk 
To: freeswitch-users at lists.freeswitch.org 
Sent: Tuesday, January 19, 2010 3:44 PM
Subject: Re: [Freeswitch-users] SVN 16400 - Windows Build Errors

Hi Dave,
 
mod_cluechoo is not built for Windows and is not needed.
The other files should build fine. Do they exist in your Debug folder?
 
- Jeff
 


From: stevendt at primrosebank.net
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 19 Jan 2010 13:31:42 +0000
Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors




Hi,
 
I have just upgraded to the latest SVN (16400) and see a couple of errors when FS starts, although the basic functionality seems to be working.
Some DLLs have not been built, should I be worried about these errors and/or how do I correct them ?
 
2010-01-19 12:51:33.280718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_cluechoo.dll
**dll open error [126l]

2010-01-19 12:51:33.905718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_file_string.dll
**dll open error [126l]

2010-01-19 12:51:34.108843 [ERR] mod_spidermonkey.c:930 Error Loading module C:\FreeSWITCH\Debug\mod\mod_spidermonkey_core_db.dll
**dll open error [126l]

 
There is nothing in the Build log on the first two, but there is an error against the last one
 
88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.pdb'.
88>Make sure that the file is not open by another process and is not write-protected.
88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.idb'.
88>Make sure that the file is not open by another process and is not write-protected
 
regards
Dave
 
WindowsXP
VC++ Express 2008


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From jerry.richards at teotech.com  Tue Jan 19 07:50:12 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 19 Jan 2010 07:50:12 -0800
Subject: [Freeswitch-users] Bypass Media True Disables MOH
In-Reply-To: <9BF355CF-C633-4BF5-BB8B-642DD81936D1@freeswitch.org>
References: <AAA8629BCAF448659E9B91C20E75CCC6@greyhawk.tonecommander.com>
	<9BF355CF-C633-4BF5-BB8B-642DD81936D1@freeswitch.org>
Message-ID: <E55CF9FDA2AC406B8799A0081BCBA4DD@greyhawk.tonecommander.com>


We are willing to pay a bounty for this.  What amount would you suggest?  We
would like the media to normally go directly between the endpoints, but when
a call is put on-hold, we would like the other end should hear MOH.

Thanks,
Jerry
  

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Tuesday, December 29, 2009 1:12 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bypass Media True Disables MOH

<param name="media-option" value="resume-media-on-hold"/>

But it doesn't go back to bypass after....   Maybe you can post a bounty for
that functionality.

/b

On Dec 29, 2009, at 2:42 PM, Jerry Richards wrote:

> 
> When I uncomment the following tag, internally held calls no longer 
> hear MOH.
> 
> <param name="inbound-bypass-media" value="true"/>
> 
> Is there a way to have the above uncommented and still provide MOH to 
> held calls?
> 
> Best Regards,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org






From freeswitch-users at digitaldan.com  Tue Jan 19 13:03:05 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Tue, 19 Jan 2010 14:03:05 -0700 (MST)
Subject: [Freeswitch-users] Lua: io.popen/read blocking in other incoming
	calls
In-Reply-To: <13515873.38.1263934963058.JavaMail.root@zimbra>
Message-ID: <19409906.41.1263934985398.JavaMail.root@zimbra>

My lua script is calling wget through lua's io.popen to send and receive data from a web service. While the f:read to wget is running, other incoming calls will block on the same io.popen call until the first call closes the pipe (with f:close()). 


I had assumed every incoming call was on its own thread and that each had its own lua instance. Is there a global lock happening here? Below is the runCommand call and the two start and stop methods that are getting called in my script when the call begins and ends (notice they even talk to different hosts, so its not the web server hanging). I have put in debugging statements and its definitely hanging trying to call io.popen and not on the f:read. I noticed on the mod_python page (http://wiki.freeswitch.org/wiki/Mod_python#Known_Bugs) that it had read issues that were fixed in 12958 , could this be related? I'm on svn trunk version 16272 on 32bit debian etch. 


Thanks, 
Dan- 
function runCommand(command)
    local f = io.popen(command) -- runs command
    local l = f:read("*a") -- read output of command
    f:close()
    return l
end function notifyStart(id)
    local url = "http://host1/start?id=" .. id
    
    local wget = "/usr/bin/wget " .. url local out = runCommand(wget) return out; end 
function notifyStop(id)
    local url = "http://host2/start?id=" .. id
    
    local wget = "/usr/bin/wget " .. url local out = runCommand(wget) return out; end 
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From stevendt at primrosebank.net  Tue Jan 19 13:03:41 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Tue, 19 Jan 2010 21:03:41 -0000
Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors
References: <2CCD94BEC92D4DE493B6FA65A554F286@bp1.ad.bp.com>,
	<COL117-W6424736CC849832D6D9DBFB9650@phx.gbl>,
	<E5A67EA005F34E68B5501C860A9B96BC@bp1.ad.bp.com>
	<COL117-W15B08477F1FFA512F0F0B4B9650@phx.gbl>
Message-ID: <FB07BA81F28C43B68925CB0F52B22ECD@bp1.ad.bp.com>

Jeff,

I had already tried to rebuild but have just done it again with the same errors on starting up FreeSwitch, i.e., the DLLs are still not there ?

regards
Dave
  ----- Original Message ----- 
  From: Jeff Lenk 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, January 19, 2010 8:26 PM
  Subject: Re: [Freeswitch-users] SVN 16400 - Windows Build Errors


  Do a rebuild all from the solution and you should not see any errors.
   

------------------------------------------------------------------------------
  From: stevendt at primrosebank.net
  To: freeswitch-users at lists.freeswitch.org
  Date: Tue, 19 Jan 2010 16:13:44 +0000
  Subject: Re: [Freeswitch-users] SVN 16400 - Windows Build Errors


  Hi Jeff,

  the DLLs are not in the Debug folders pointed to by the error messages ?

  regards
  Dave

    ----- Original Message ----- 
    From: Jeff Lenk 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Tuesday, January 19, 2010 3:44 PM
    Subject: Re: [Freeswitch-users] SVN 16400 - Windows Build Errors


    Hi Dave,
     
    mod_cluechoo is not built for Windows and is not needed.
    The other files should build fine. Do they exist in your Debug folder?
     
    - Jeff
     

----------------------------------------------------------------------------
    From: stevendt at primrosebank.net
    To: freeswitch-users at lists.freeswitch.org
    Date: Tue, 19 Jan 2010 13:31:42 +0000
    Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors


    Hi,

    I have just upgraded to the latest SVN (16400) and see a couple of errors when FS starts, although the basic functionality seems to be working.
    Some DLLs have not been built, should I be worried about these errors and/or how do I correct them ?

    2010-01-19 12:51:33.280718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_cluechoo.dll
    **dll open error [126l]

    2010-01-19 12:51:33.905718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_file_string.dll
    **dll open error [126l]

    2010-01-19 12:51:34.108843 [ERR] mod_spidermonkey.c:930 Error Loading module C:\FreeSWITCH\Debug\mod\mod_spidermonkey_core_db.dll
    **dll open error [126l]


    There is nothing in the Build log on the first two, but there is an error against the last one

    88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.pdb'.
    88>Make sure that the file is not open by another process and is not write-protected.
    88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.idb'.
    88>Make sure that the file is not open by another process and is not write-protected

    regards
    Dave

    WindowsXP
    VC++ Express 2008


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From john at acsol.net  Tue Jan 19 14:12:51 2010
From: john at acsol.net (John)
Date: Tue, 19 Jan 2010 15:12:51 -0700
Subject: [Freeswitch-users] Voicemail to email problems
In-Reply-To: <4E823F6B-75DC-44D9-BD18-C7773D77EAC9@freeswitch.org>
References: <ab161f4749d2cf203ec197497f2ce22e@thom.fr.eu.org>	<4B560A46.90000@acsol.net>	<191c3a031001191146r257d34afq98f921c0ff07f2fc@mail.gmail.com>
	<4E823F6B-75DC-44D9-BD18-C7773D77EAC9@freeswitch.org>
Message-ID: <4B562E63.7000407@acsol.net>

I have modified the Exim configuration as per the wiki. I am still 
getting the same message, in addition, the /tmp/mail.xxxxxxxx file is 
not getting created at any point. Is it possible the problem is that 
there is no file to send, so it errors out? Thank you both for your help!

/bin/cat: write error: Broken pipe
sh: line 1: 21176 Done(1)                 /bin/cat /tmp/mail.1263940067ade7
      21177 Segmentation fault      (core dumped) | exim4 -f 
1004 at voip.server.net -t jhart at server.net


On 1/19/2010 1:14 PM, Brian West wrote:
> http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings
>
> /b
>
> On Jan 19, 2010, at 1:46 PM, Anthony Minessale wrote:
>
>> sounds like exim pretending to be sendmail and not doing it very well.
>> I think there is a wiki page somewhere that tells you how to config 
>> it properly.
>>
>>
>> On Tue, Jan 19, 2010 at 1:38 PM, John <john at acsol.net 
>> <mailto:john at acsol.net>> wrote:
>>
>>     I am getting a segfault as well; however I use exim4 on Debian.
>>     Email is
>>     never being written to /tmp. I too can run the exim command by
>>     hand and
>>     it works. Any body have an idea or direction to look at?
>>     Here is the error.
>>     /bin/cat: write error: Broken pipe
>>     sh: line 1:  7102 Done(1)                 /bin/cat
>>     /tmp/mail.12639191242a9f
>>           7103 Segmentation fault
>>
>>     Thanks
>>
>>
>>
>>     On 1/19/2010 9:31 AM, Fran?ois Legal wrote:
>>     > This is not the error I get.
>>     >
>>     > I modified my config so that the command executed to send the
>>     email is cat
>>     > /tmp/mail.xxxxxxx | /usr/local/freeswitch/email.sh
>>     > with email.sh teeing the stdin to a file and to sendmail.
>>     >
>>     > syslog reports sendmail segfault.
>>     >
>>     > Fran?ois
>>     >
>>     > On Tue, 19 Jan 2010 09:02:30 -0700, John<john at acsol.net
>>     <mailto:john at acsol.net>>  wrote:
>>     >
>>     >> Voicemail to email gets a broken pipe error. I believe this is
>>     happening
>>     >>
>>     >
>>     >> while writing the file to /tmp? The mail.xxxxxxxxxxxx file is
>>     never
>>     >> written. Any ideas? Thanks
>>     >>
>>     >>
>>     >>
>>     >> _______________________________________________
>>     >> FreeSWITCH-users mailing list
>>     >> FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     >>
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     >> http://www.freeswitch.org <http://www.freeswitch.org/>
>>     >>
>>     > _______________________________________________
>>     > FreeSWITCH-users mailing list
>>     > FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     >
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     > http://www.freeswitch.org <http://www.freeswitch.org/>
>>     >
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com 
>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net <http://irc.freenode.net/> #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org 
>> <mailto:sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888 
>> <http://iax:guest at conference.freeswitch.org/888>
>> googletalk:conf+888 at conference.freeswitch.org 
>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From anthony.minessale at gmail.com  Tue Jan 19 14:43:59 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Jan 2010 16:43:59 -0600
Subject: [Freeswitch-users] Lua: io.popen/read blocking in other
	incoming calls
In-Reply-To: <19409906.41.1263934985398.JavaMail.root@zimbra>
References: <13515873.38.1263934963058.JavaMail.root@zimbra>
	<19409906.41.1263934985398.JavaMail.root@zimbra>
Message-ID: <191c3a031001191443o6184a6f4nd3f5a849ad609d5f@mail.gmail.com>

if there is any global lock it would be in the lua lib.  I know that would
not happen in C so you may want to step it in the debugger and look for any
evidence of a global mutex in the lua lib.


On Tue, Jan 19, 2010 at 3:03 PM, Dan <freeswitch-users at digitaldan.com>wrote:

> My lua script is calling wget through lua's io.popen to send
> and receive data from a web service. While the f:read to wget  is running,
> other incoming calls will block on the same io.popen call until the first
> call closes the pipe (with f:close()).
>
>  I had assumed every incoming call was on its own thread and that each had
> its own lua instance.  Is there a global lock happening here? Below is the
> runCommand call and the two start and stop methods that are getting called
> in my script when the call begins and ends (notice they even talk to
> different hosts, so its not the web server hanging).  I have put in
> debugging statements and its definitely  hanging  trying to call io.popen
> and not on the f:read.  I noticed on the mod_python page  (
> http://wiki.freeswitch.org/wiki/Mod_python#Known_Bugs) that it had read
> issues that were fixed in 12958, could this be related?  I'm on svn trunk
> version 16272 on 32bit debian etch.
>
> Thanks,
> Dan-
>
> function runCommand(command)
>     local f = io.popen(command) -- runs command
>     local l = f:read("*a") -- read output of command
>     f:close()
>     return l
> end
>
> function notifyStart(id)
>     local url = "http://host1/start?id=" .. id
>
>     local wget = "/usr/bin/wget " .. url
>
>     local out = runCommand(wget)
>
>     return out;
>
> end
>
>
> function notifyStop(id)
>     local url = "http://host2/start?id=" .. id
>
>     local wget = "/usr/bin/wget " .. url
>
>     local out = runCommand(wget)
>
>     return out;
>
> end
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From kees at mroffice.org  Tue Jan 19 14:58:15 2010
From: kees at mroffice.org (Kees Varekamp)
Date: Wed, 20 Jan 2010 11:58:15 +1300
Subject: [Freeswitch-users] SIP for Skype
Message-ID: <98d38dcf1001191458g4766d028r9540cce9d2c6f6ef@mail.gmail.com>

Hello,

I am trying to hook up my freeswitch server to SIP for skype but skype keeps
sending me 403 forbidden. I'm pretty sure that the account is OK, so I'm
wondering if they are blocking me for other reasons. Has anybody had any
success with SIP for Skype?

Thanks, Kees

Siptrace:

   ------------------------------------------------------------------------
send 878 bytes to udp/[204.9.161.164]:5060 at 22:38:31.115019:
   ------------------------------------------------------------------------
   REGISTER sip:sip.skype.com SIP/2.0
   Via: SIP/2.0/UDP 203.109.207.110;rport;branch=z9hG4bK7Da0S0XSeypHm
   Max-Forwards: 70
   From: <sip:xxxxxx at sip.skype.com <sip%3Axxxxxx at sip.skype.com>
;transport=udp>;tag=6vNF7NSDX1t7H
   To: <sip:xxxxxx at sip.skype.com <sip%3Axxxxxx at sip.skype.com>;transport=udp>
   Call-ID: 38262ed8-054b-11df-899c-d96406a83851
   CSeq: 125852020 REGISTER
   Contact: <sip:gw+sip.skype.com at 203.109.207.110:5060;transport=udp;gw=
sip.skype.com>
   Expires: 3600
   User-Agent: FreeSWITCH-mod_sofia/1.0.5-20100119-0400-16400M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Authorization: Digest username="xxxxxx", realm="sip.skype.com",
nonce="xxxxx", algorithm=MD5, uri="sip:sip.skype.com", response="xxxxx"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 378 bytes from udp/[204.9.161.164]:5060 at 22:38:31.470978:
   ------------------------------------------------------------------------
   SIP/2.0 403 Forbidden
   From: <sip:xxxxx at sip.skype.com <sip%3Axxxxx at sip.skype.com>
;transport=udp>;tag=6vNF7NSDX1t7H
   To: <sip:xxxxxx at sip.skype.com <sip%3Axxxxxx at sip.skype.com>
;transport=udp>;tag=05aed4eb43523e287156e2da6464d890.13a5
   Call-ID: 38262ed8-054b-11df-899c-d96406a83851
   CSeq: 125852020 REGISTER
   Via: SIP/2.0/UDP 203.109.207.110;rport=5060;branch=z9hG4bK7Da0S0XSeypHm
   Server: OpenSIPS
   Content-Length: 0

   ------------------------------------------------------------------------
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From brian at freeswitch.org  Tue Jan 19 15:01:15 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Jan 2010 17:01:15 -0600
Subject: [Freeswitch-users] SIP for Skype
In-Reply-To: <98d38dcf1001191458g4766d028r9540cce9d2c6f6ef@mail.gmail.com>
References: <98d38dcf1001191458g4766d028r9540cce9d2c6f6ef@mail.gmail.com>
Message-ID: <A85FC509-D282-4A8F-A0B8-1AA3E14C402B@freeswitch.org>

Yes, Please contact your provider.

/b

On Jan 19, 2010, at 4:58 PM, Kees Varekamp wrote:

> Hello,
> 
> I am trying to hook up my freeswitch server to SIP for skype but skype keeps sending me 403 forbidden. I'm pretty sure that the account is OK, so I'm wondering if they are blocking me for other reasons. Has anybody had any success with SIP for Skype?
> 
> Thanks, Kees
> 
> Siptrace:
> 



From rupa at rupa.com  Tue Jan 19 15:06:36 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Jan 2010 17:06:36 -0600
Subject: [Freeswitch-users] Lua: io.popen/read blocking in other
	incoming calls
In-Reply-To: <19409906.41.1263934985398.JavaMail.root@zimbra>
References: <13515873.38.1263934963058.JavaMail.root@zimbra>
	<19409906.41.1263934985398.JavaMail.root@zimbra>
Message-ID: <b39273531001191506v17e34e0er99d6c4098d960950@mail.gmail.com>

On Tue, Jan 19, 2010 at 3:03 PM, Dan <freeswitch-users at digitaldan.com>wrote:

> My lua script is calling wget through lua's io.popen to send
> and receive data from a web service. While the f:read to wget  is running,
> other incoming calls will block on the same io.popen call until the first
> call closes the pipe (with f:close()).
>

You might want to look at the api that mod_curl exposes to do what you want.
 No need to do an expensive system call just to call a webservice.

-- 
-Rupa
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From kees at mroffice.org  Tue Jan 19 15:14:03 2010
From: kees at mroffice.org (Kees Varekamp)
Date: Wed, 20 Jan 2010 12:14:03 +1300
Subject: [Freeswitch-users] SIP for Skype
In-Reply-To: <A85FC509-D282-4A8F-A0B8-1AA3E14C402B@freeswitch.org>
References: <98d38dcf1001191458g4766d028r9540cce9d2c6f6ef@mail.gmail.com>
	<A85FC509-D282-4A8F-A0B8-1AA3E14C402B@freeswitch.org>
Message-ID: <98d38dcf1001191514w681862baod098c21f4ae4ec9d@mail.gmail.com>

? You mean Internet provider or Skype?

Would you mind sending me an example <gateway> config?

Thanks, Kees

On Wed, Jan 20, 2010 at 12:01, Brian West <brian at freeswitch.org> wrote:

> Yes, Please contact your provider.
>
> /b
>
> On Jan 19, 2010, at 4:58 PM, Kees Varekamp wrote:
>
> > Hello,
> >
> > I am trying to hook up my freeswitch server to SIP for skype but skype
> keeps sending me 403 forbidden. I'm pretty sure that the account is OK, so
> I'm wondering if they are blocking me for other reasons. Has anybody had any
> success with SIP for Skype?
> >
> > Thanks, Kees
> >
> > Siptrace:
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch-users at digitaldan.com  Tue Jan 19 15:14:04 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Tue, 19 Jan 2010 16:14:04 -0700 (MST)
Subject: [Freeswitch-users] Lua: io.popen/read blocking in
	other	incoming calls
In-Reply-To: <191c3a031001191443o6184a6f4nd3f5a849ad609d5f@mail.gmail.com>
Message-ID: <31453864.51.1263942843967.JavaMail.root@zimbra>

Thanks for your response, I'm looking through the lua source right now. Am I correct in assuming that each incoming call has its own thread and therefore its own lua vm instance? So the only blocking that would be possible among threads is if they were calling something blocking on the core switch module. That's definitely not happening here, so I'll keep looking around. 




Thanks 
Dan- 
----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Tuesday, January 19, 2010 3:43:59 PM 
Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls 

if there is any global lock it would be in the lua lib. I know that would not happen in C so you may want to step it in the debugger and look for any evidence of a global mutex in the lua lib. 



On Tue, Jan 19, 2010 at 3:03 PM, Dan < freeswitch-users at digitaldan.com > wrote: 




My lua script is calling wget through lua's io.popen to send and receive data from a web service. While the f:read to wget is running, other incoming calls will block on the same io.popen call until the first call closes the pipe (with f:close()). 


I had assumed every incoming call was on its own thread and that each had its own lua instance. Is there a global lock happening here? Below is the runCommand call and the two start and stop methods that are getting called in my script when the call begins and ends (notice they even talk to different hosts, so its not the web server hanging). I have put in debugging statements and its definitely hanging trying to call io.popen and not on the f:read. I noticed on the mod_python page ( http://wiki.freeswitch.org/wiki/Mod_python#Known_Bugs ) that it had read issues that were fixed in 12958, could this be related? I'm on svn trunk version 16272 on 32bit debian etch. 


Thanks, 
Dan- 
function runCommand(command)
    local f = io.popen(command) -- runs command
    local l = f:read("*a") -- read output of command
    f:close()
    return l
end function notifyStart(id)
    local url = " http://host1/start?id= " .. id
    
    local wget = "/usr/bin/wget " .. url local out = runCommand(wget) return out; end 
function notifyStop(id)
    local url = " http://host2/start?id= " .. id
    
    local wget = "/usr/bin/wget " .. url local out = runCommand(wget) return out; end 
_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 




-- 
Anthony Minessale II 

FreeSWITCH http://www.freeswitch.org/ 
ClueCon http://www.cluecon.com/ 
Twitter: http://twitter.com/FreeSWITCH_wire 

AIM: anthm 
MSN:anthony_minessale at hotmail.com 
GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com 
IRC: irc.freenode.net #freeswitch 

FreeSWITCH Developer Conference 
sip:888 at conference.freeswitch.org 
iax:guest at conference.freeswitch.org/888 
googletalk:conf+888 at conference.freeswitch.org 
pstn:+19193869900 

_______________________________________________ 
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FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
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From freeswitch-users at digitaldan.com  Tue Jan 19 15:17:37 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Tue, 19 Jan 2010 16:17:37 -0700 (MST)
Subject: [Freeswitch-users] Lua: io.popen/read blocking in
	other	incoming calls
In-Reply-To: <7967781.55.1263942960863.JavaMail.root@zimbra>
Message-ID: <11387916.58.1263943057280.JavaMail.root@zimbra>

I would, but I need to post a a wav file that gets recorded, I didn't see a way to supply the location of a file to use as the post data. It looks like you have to url encode the data in the script and pass it all in the call. 


Thanks 
Dan- 
----- Original Message ----- 
From: "Rupa Schomaker" <rupa at rupa.com> 
To: "freeswitch-users" <freeswitch-users at lists.freeswitch.org> 
Sent: Tuesday, January 19, 2010 4:06:36 PM 
Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls 




On Tue, Jan 19, 2010 at 3:03 PM, Dan < freeswitch-users at digitaldan.com > wrote: 




My lua script is calling wget through lua's io.popen to send and receive data from a web service. While the f:read to wget is running, other incoming calls will block on the same io.popen call until the first call closes the pipe (with f:close()). 


You might want to look at the api that mod_curl exposes to do what you want. No need to do an expensive system call just to call a webservice. 
-- 
-Rupa 

_______________________________________________ 
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
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From brian at freeswitch.org  Tue Jan 19 15:18:27 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Jan 2010 17:18:27 -0600
Subject: [Freeswitch-users] SIP for Skype
In-Reply-To: <98d38dcf1001191514w681862baod098c21f4ae4ec9d@mail.gmail.com>
References: <98d38dcf1001191458g4766d028r9540cce9d2c6f6ef@mail.gmail.com>
	<A85FC509-D282-4A8F-A0B8-1AA3E14C402B@freeswitch.org>
	<98d38dcf1001191514w681862baod098c21f4ae4ec9d@mail.gmail.com>
Message-ID: <5B84A27E-E79D-4670-A525-074A4B1FEE43@freeswitch.org>

Skype.

/b

On Jan 19, 2010, at 5:14 PM, Kees Varekamp wrote:

> ? You mean Internet provider or Skype?
> 
> Would you mind sending me an example <gateway> config?
> 
> Thanks, Kees
>  



From msc at freeswitch.org  Tue Jan 19 15:21:55 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Jan 2010 15:21:55 -0800
Subject: [Freeswitch-users] NOTICE: mod_iax slated for unsupported on Feb. 5
Message-ID: <87f2f3b91001191521y3aec5aadp6ebee73945bb7e0c@mail.gmail.com>

To everyone in the FreeSWITCH community:

We would like to put a call out and let everyone know that unless we find
someone who wants to be the maintainer for mod_iax it will be moved to
unsupported status as of Friday February 5th. Anyone who wishes to maintain
the module will need to be in charge of all bug reports, updates, patches,
and technical support questions. If are in a position to maintain mod_iax
then please contact Brian West offlist: brian at freeswitch.org.

-Michael
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From freeswitch-users at digitaldan.com  Tue Jan 19 15:27:52 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Tue, 19 Jan 2010 16:27:52 -0700 (MST)
Subject: [Freeswitch-users] Lua: io.popen/read blocking
	in	other	incoming calls
In-Reply-To: <29464896.62.1263943497225.JavaMail.root@zimbra>
Message-ID: <22214819.65.1263943672393.JavaMail.root@zimbra>

One more question, popen on linux forks a new child process to execute a shell in, could the freeswitch environment have any influence on this? 


Thanks 
Dan- 


----- Original Message ----- 
From: "Dan" <freeswitch-users at digitaldan.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Tuesday, January 19, 2010 4:14:04 PM 
Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls 


Thanks for your response, I'm looking through the lua source right now. Am I correct in assuming that each incoming call has its own thread and therefore its own lua vm instance? So the only blocking that would be possible among threads is if they were calling something blocking on the core switch module. That's definitely not happening here, so I'll keep looking around. 




Thanks 
Dan- 
----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Tuesday, January 19, 2010 3:43:59 PM 
Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls 

if there is any global lock it would be in the lua lib. I know that would not happen in C so you may want to step it in the debugger and look for any evidence of a global mutex in the lua lib. 



On Tue, Jan 19, 2010 at 3:03 PM, Dan < freeswitch-users at digitaldan.com > wrote: 




My lua script is calling wget through lua's io.popen to send and receive data from a web service. While the f:read to wget is running, other incoming calls will block on the same io.popen call until the first call closes the pipe (with f:close()). 


I had assumed every incoming call was on its own thread and that each had its own lua instance. Is there a global lock happening here? Below is the runCommand call and the two start and stop methods that are getting called in my script when the call begins and ends (notice they even talk to different hosts, so its not the web server hanging). I have put in debugging statements and its definitely hanging trying to call io.popen and not on the f:read. I noticed on the mod_python page ( http://wiki.freeswitch.org/wiki/Mod_python#Known_Bugs ) that it had read issues that were fixed in 12958, could this be related? I'm on svn trunk version 16272 on 32bit debian etch. 


Thanks, 
Dan- 
function runCommand(command)
    local f = io.popen(command) -- runs command
    local l = f:read("*a") -- read output of command
    f:close()
    return l
end function notifyStart(id)
    local url = " http://host1/start?id= " .. id
    
    local wget = "/usr/bin/wget " .. url local out = runCommand(wget) return out; end 
function notifyStop(id)
    local url = " http://host2/start?id= " .. id
    
    local wget = "/usr/bin/wget " .. url local out = runCommand(wget) return out; end 
_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 




-- 
Anthony Minessale II 

FreeSWITCH http://www.freeswitch.org/ 
ClueCon http://www.cluecon.com/ 
Twitter: http://twitter.com/FreeSWITCH_wire 

AIM: anthm 
MSN:anthony_minessale at hotmail.com 
GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com 
IRC: irc.freenode.net #freeswitch 

FreeSWITCH Developer Conference 
sip:888 at conference.freeswitch.org 
iax:guest at conference.freeswitch.org/888 
googletalk:conf+888 at conference.freeswitch.org 
pstn:+19193869900 

_______________________________________________ 
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
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http://www.freeswitch.org 

_______________________________________________ 
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FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
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From rupa at rupa.com  Tue Jan 19 15:32:57 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Jan 2010 17:32:57 -0600
Subject: [Freeswitch-users] Lua: io.popen/read blocking in other
	incoming calls
In-Reply-To: <11387916.58.1263943057280.JavaMail.root@zimbra>
References: <7967781.55.1263942960863.JavaMail.root@zimbra>
	<11387916.58.1263943057280.JavaMail.root@zimbra>
Message-ID: <b39273531001191532s7d2ad182oc6751eedc1089d30@mail.gmail.com>

Ah... yes you do.  Patches / Bounty to implement that accepted.

On Tue, Jan 19, 2010 at 5:17 PM, Dan <freeswitch-users at digitaldan.com>wrote:

> I would, but I need to post a a wav file that gets recorded, I didn't see a
> way to supply the location of a file to use as the post data.  It looks like
> you have to url encode the data in the script and pass it all in the call.
>
> Thanks
> Dan-
>
> ----- Original Message -----
> From: "Rupa Schomaker" <rupa at rupa.com>
> To: "freeswitch-users" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, January 19, 2010 4:06:36 PM
> Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in
> other        incoming calls
>
>
>
> On Tue, Jan 19, 2010 at 3:03 PM, Dan <freeswitch-users at digitaldan.com>wrote:
>
>> My lua script is calling wget through lua's io.popen to send
>> and receive data from a web service. While the f:read to wget  is running,
>> other incoming calls will block on the same io.popen call until the first
>> call closes the pipe (with f:close()).
>>
>
> You might want to look at the api that mod_curl exposes to do what you
> want.  No need to do an expensive system call just to call a webservice.
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From msc at freeswitch.org  Tue Jan 19 16:41:40 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Jan 2010 16:41:40 -0800
Subject: [Freeswitch-users] Home setup with home company
In-Reply-To: <4B55F3A0020000E1000003E0@mail.fribert.dk>
References: <4B55F3A0020000E1000003E0@mail.fribert.dk>
Message-ID: <87f2f3b91001191641p1fea3318l3d166fe3555e760d@mail.gmail.com>

On Tue, Jan 19, 2010 at 9:02 AM, mailinglist <mailinglist at fribert.dk> wrote:

>  I have a very small one man constultancy company that has the occasional
> call, unfortunately we are getting more spam calls after hours than real
> calls during work hours, so I would like to set up a TOD system.
>
> First step for me is just playing with the TOD example, I've gotten this
> far:
>
> <extension name="CompanyTOD" continue="true">
>    <condition field="destination_number" expression="^(8203)$"/>
>       <!-- man strftime - M-F, 9AM to 5PM -->
>       <condition field="${strftime(%w)}" expression="^([1-5])$"/>
>       <condition field="${strftime(%H%M)}"
> expression="^((09|1[0-6])[0-5][0-9]|1700)$">
>          <action application="set" data="call_timeout=45"/>
>          <action application="set" data="domain_name=10.11.12.25"/>
>          <action application="set" data="continue_on_fail=true"/>
>          <action application="set" data="hangup_after_bridge=true"/>
>          <action application="bridge" data="group/company@${domain_name}"/<group/company@$%7Bdomain_name%7D%22/>
> >
>          <action application="answer" data=""/>
>          <action application="sleep" data="1000"/>
>          <action application="voicemail" data="default 10.11.12.25 10199"/>
>          <anti-action application="voicemail" data="default 10.11.12.25
> 10199"/>
>       </condition>
> </extension>
> My idea with this, was that in the time 9-17 mon-fri, it rings, the rest of
> the time it's sent directly to the voicemail.
> I would of course like to have it not take messages outside work hours, but
> that's just refining :-)
>
> But it picks up the call, and then nothing...
>
>
We have a much cleaner way of doing TOD and DOW handling. You'll need to get
to the latest FreeSWITCH version. Look in conf/dialplan/default.xml for this
example:

    <!-- Example of doing things based on time of day.

     year = 4 digit year. Example year="2009"
     yday = 1-365
     mon = 1-12
     mday = 1-31
     week = 1-52
     mweek= 1-6
     wday = 1-7
     hour = 0-23
     minute = 0-59
     minute-of-day = 1-1440

     Example:
     <condition minute-of-day="540-1080"> (9am to 6pm EVERY day)
     do something ...
     </condition>
    -->
    <extension name="tod_example" continue="true">
      <condition wday="2-6" hour="9-18">
        <action application="set" data="open=true"/>
      </condition>
    </extension>


Use that condition instead of the two conditions you're now using and see if
you have better success.

-MC
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From lakindia89 at gmail.com  Tue Jan 19 20:49:43 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 20 Jan 2010 10:19:43 +0530
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <191c3a031001190804m692f63acsaf852b8809db09d1@mail.gmail.com>
References: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
	<191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>
	<7d79b3931001132357i36bbb482jdf2bbdd3aea2a583@mail.gmail.com>
	<7d79b3931001180322p74a1227qe0c2199a77cbfe2@mail.gmail.com>
	<191c3a031001180824q6e364c72g3f789892597e9469@mail.gmail.com>
	<7d79b3931001190107o200ec04dredd76689dd235588@mail.gmail.com>
	<191c3a031001190804m692f63acsaf852b8809db09d1@mail.gmail.com>
Message-ID: <7d79b3931001192049h7e4a1a9ex1676ec415e6fd49f@mail.gmail.com>

Thanks for all your reply's.
Will give a try to on non-blocking.


On Tue, Jan 19, 2010 at 9:34 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Its nothing we can fix, that is what you must do on a failed read syscall.
> you can do a non blocking read instead and take your chances.
>
>
>
> On Tue, Jan 19, 2010 at 3:07 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> I tried with SIGUSR1, but no progress. I got SERVER_DISCONNECTED.
>> Output:
>>
>> CHILD 3814: Received USR1
>> EVENT [SERVER_DISCONNECTED]
>>
>> In esl.c, in esl_recv_event() function, line no: 824
>> if (rrval < 0) {
>>                         strerror_r(handle->errnum, handle->err,
>> sizeof(handle->err));
>>                         goto fail;
>> }
>> When the program is blocked under receive, I passed the signal. So recv
>> returns -1, and in fail: it call esl_disconnect(handle).
>>
>> Is it because of this??? If so, whether it should be fixed or not???
>>
>>
>>
>>
>> On Mon, Jan 18, 2010 at 9:54 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> try a less famous signal like SIGUSR1 it's possible something in perl
>>> still reacts to SIGINT
>>>
>>>
>>>
>>> On Mon, Jan 18, 2010 at 5:22 AM, lakshmanan ganapathy <
>>> lakindia89 at gmail.com> wrote:
>>>
>>>> Here is the result
>>>>
>>>> Program:
>>>>
>>>> require ESL;
>>>> use IO::Socket::INET;
>>>> use Data::Dumper;
>>>>
>>>> my $ip = "192.168.1.222";
>>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>>> die "Could not create socket: $!\n" unless $sock;
>>>>
>>>>
>>>> for(;;) {
>>>>     my $new_sock = $sock->accept();
>>>>     next if (not defined ($new_sock));
>>>>     my $pid = fork();
>>>>     if ($pid) {
>>>>         close($new_sock);
>>>>         next;
>>>>     }
>>>>     print "CHILD PID: $$\n";
>>>>     &register_Signals_Child();
>>>>     sub register_Signals_Child() {
>>>>         foreach ( keys %SIG ) {
>>>>             $SIG{$_} = 'Handler';
>>>>         }
>>>>     }
>>>>
>>>>     sub Handler() {
>>>>
>>>>         my $handle=$_[0];
>>>>         if($handle eq "INT") {
>>>>             print "CHILD $$: SIGNAL SIG$handle is generated\n";
>>>>         }
>>>>         else
>>>>         {
>>>>             print "CHILD $$: Received $handle\n";
>>>>
>>>>         }
>>>>     }
>>>>     my $host = $new_sock->sockhost();
>>>>     my $fd = fileno($new_sock);
>>>>
>>>>     my $con = new ESL::ESLconnection($fd);
>>>>     my $info = $con->getInfo();
>>>>
>>>>     my $uuid = $info->getHeader("unique-id");
>>>>
>>>>     printf "Connected call %s, from %s\n", $uuid,
>>>> $info->getHeader("caller-caller-id-number");
>>>>     my $r=$con->execute("answer");
>>>>     print Dumper $r;
>>>>     $con->events("plain","all");
>>>>     my
>>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>>     while($con->connected()) {
>>>>         my $e = $con->recvEvent();
>>>>
>>>>         if ($e) {
>>>>             my $name = $e->getHeader("event-name");
>>>>             print "EVENT [$name]\n";
>>>>             if ($name eq "DTMF") {
>>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>>                 print "$digit\n";
>>>>             }
>>>>         }
>>>>     }
>>>>     close($new_sock);
>>>> }
>>>>
>>>> I executed the program and the following things were printed
>>>>
>>>> CHILD PID: 6778
>>>> Connected call e0d1001a-03f4-11df-b002-db488337e0ea, from 1001
>>>> $VAR1 = 0;
>>>> EVENT [CHANNEL_EXECUTE]
>>>> EVENT [CHANNEL_ANSWER]
>>>> EVENT [CHANNEL_EXECUTE_COMPLETE]
>>>> EVENT [COMMAND]
>>>> EVENT [CHANNEL_EXECUTE]
>>>> EVENT [HEARTBEAT]
>>>> EVENT [RE_SCHEDULE]
>>>> EVENT [CHANNEL_EXECUTE_COMPLETE]
>>>>
>>>> Then from another shell I executed kill -2 6778, the result is follows
>>>> CHILD 6778: SIGNAL SIGINT is generated
>>>> EVENT [SERVER_DISCONNECTED]
>>>>
>>>> But the child process is still running as expected.
>>>> But  I don't know why I received SERVER_DISCONNECTED from freeswitch???
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Thu, Jan 14, 2010 at 1:27 PM, lakshmanan ganapathy <
>>>> lakindia89 at gmail.com> wrote:
>>>>
>>>>> I taught the signal handler will be inherited by the child process. It
>>>>> also does like that.
>>>>> After making a call, If I press ctrl + c, the above program printed
>>>>> PARENT PID: Signal SIGINT is generated
>>>>> CHILD PID: Signal SIGINT is generated.
>>>>>
>>>>> So I think the sigal handlers will be inherited to the child.
>>>>> Anyway I'll also try registering signal handlers in child also, and
>>>>> then I'll come back with that result.
>>>>>
>>>>> Thanks....
>>>>> On Wed, Jan 13, 2010 at 9:48 PM, Anthony Minessale <
>>>>> anthony.minessale at gmail.com> wrote:
>>>>>
>>>>>> you would have to register signals in your child process too
>>>>>>
>>>>>>   On Wed, Jan 13, 2010 at 3:13 AM, lakshmanan ganapathy <
>>>>>> lakindia89 at gmail.com> wrote:
>>>>>>
>>>>>>>  Hi all,
>>>>>>>
>>>>>>> I've done a sample program (In perl ESL) , which play a file to the
>>>>>>> caller and then it will call recvEvent() and print the event name. I've
>>>>>>> handled signals also.
>>>>>>>
>>>>>>> When I send SIGINT to my program (Perl), the signal handler is called
>>>>>>> and I can see the print output. But in the same time,  I received
>>>>>>> SERVER_DISCONNECTED from freeswitch as event.
>>>>>>>
>>>>>>> I don't know why I received SERVER_DISCONNECTED from freeswitch. Is
>>>>>>> it because, the recvEvent() from perl internally  calls the recvevent
>>>>>>> function in the Esl.c and when it waits to receive the information from
>>>>>>> socket, the signal occurred???
>>>>>>>
>>>>>>> Please clarify me!!
>>>>>>>
>>>>>>> Here is my program
>>>>>>> require ESL;
>>>>>>> use IO::Socket::INET;
>>>>>>> use Data::Dumper;
>>>>>>>
>>>>>>> my $ip = "192.168.1.222";
>>>>>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>>>>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>>>>>> die "Could not create socket: $!\n" unless $sock;
>>>>>>> &register_Signals();
>>>>>>>
>>>>>>> for(;;) {
>>>>>>>     my $new_sock = $sock->accept();
>>>>>>>     next if (not defined ($new_sock));
>>>>>>>     my $pid = fork();
>>>>>>>     if ($pid) {
>>>>>>>         close($new_sock);
>>>>>>>         next;
>>>>>>>     }
>>>>>>>     print "CHILD PID: $$\n";
>>>>>>>     my $host = $new_sock->sockhost();
>>>>>>>     my $fd = fileno($new_sock);
>>>>>>>
>>>>>>>     my $con = new ESL::ESLconnection($fd);
>>>>>>>     my $info = $con->getInfo();
>>>>>>>
>>>>>>>     my $uuid = $info->getHeader("unique-id");
>>>>>>>
>>>>>>>     printf "Connected call %s, from %s\n", $uuid,
>>>>>>> $info->getHeader("caller-caller-id-number");
>>>>>>>     my $r=$con->execute("answer");
>>>>>>>     print Dumper $r;
>>>>>>>     $con->events("plain","all");
>>>>>>>     my
>>>>>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>>>>>
>>>>>>>     while($con->connected()) {
>>>>>>>         my $e = $con->recvEvent();
>>>>>>>
>>>>>>>         if ($e) {
>>>>>>>             my $name = $e->getHeader("event-name");
>>>>>>>             print "EVENT [$name]\n";
>>>>>>>             if ($name eq "DTMF") {
>>>>>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>>>>>                 print "$digit\n";
>>>>>>>             }
>>>>>>>         }
>>>>>>>     }
>>>>>>>     close($new_sock);
>>>>>>> }
>>>>>>> sub register_Signals() {
>>>>>>>     foreach ( keys %SIG ) {
>>>>>>>         $SIG{$_} = 'sig_Handler';
>>>>>>>     }
>>>>>>> }
>>>>>>>
>>>>>>> sub sig_Handler() {
>>>>>>>     my $handle=$_[0];
>>>>>>>     if($handle eq "INT") {
>>>>>>>         print "$$: SIGNAL SIG$handle is generated\n";
>>>>>>>     }
>>>>>>> }
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dome at tel.co.th  Tue Jan 19 21:32:26 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 20 Jan 2010 12:32:26 +0700
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <df97b9c51001190107x39d94eaai7400635f0ac03f86@mail.gmail.com>
References: <df97b9c51001190107x39d94eaai7400635f0ac03f86@mail.gmail.com>
Message-ID: <8ccbff061001192132h344a61a4m58d2973391c8182a@mail.gmail.com>

Please try http://wiki.freeswitch.org/wiki/Webapi
you can create class and map to webapi.

Dome C.

2010/1/19 Scott Fernandez <scottferri09 at gmail.com>:
> Hi,
>
> Is there any API modules available for me to initiate a call from .Net based
> application?.
>
> The idea is to include the API modules if any with the .NET base classes so
> that the API commands will be made available on it. I know it is doable when
> I use socket programming in .NET in which Telnet session is created.
> However, this would potentially hamper the performance of the application
> because of multiple sessions that will be created for each call.
>
> Other than that, Is there any Freeswitch API modules (like plug-ins)
> available in order to include it into the .Net classes and start building
> the customized application?
>
> Any help from any one is highly appreciated.
>
> Thanks,
> Scott
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From thangappan143 at gmail.com  Tue Jan 19 21:55:33 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Wed, 20 Jan 2010 11:25:33 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma card
Message-ID: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>

Dear all,

         I have successfully configured wanpipe with freeswitch. When I was
the running wancfg_fs script the following files openzap.conf ,
autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
are created.

         I started the wanrouter command then executed the freeswitch.
         When I was executing freeswitch mod_openzap.c said the error as
"Error  for finding the span id. name:PRI_1".
         But in the openzap.conf and openzap.conf.xml files the span name is
smg_prid.

         Why the freeswitch is referring the span name as PRI_1 ?
         Whether this has to configured in anywhere?

         In the freeswitch CLI using oz command I tried to dump the PRI_1
span id but it said te error  as "PRI_1 is not found".  When I was trying
the command 'oz dump smg_prid'  all the channel states and details shown.

        It seems that smg_prid span configured in openzap perfectly (Its my
assumption). Then Why freeswitch is referring the span name as PRI_1.

DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?

Could anyone please help me?

REFERENCE:

openzap.conf
[span wanpipe smg_prid]
name => smg_prid
trunk_type =>e1
b-channel => 1:1-15
b-channel => 1:17-31


openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
  <boost_spans>
        <span id="smg_prid">
        <!--<param name="hold-music" value="$${moh_uri}"/>-->
        <param name="dialplan" value="XML"/>
        <param name="context" value="default"/>
         <!-- regex to stop dialing when it matches -->
        <!--<param name="dial-regex" value="5555"/>-->
        <!-- regex to stop dialing when it does not match -->
        <!--<param name="fail-dial-regex" value="^5"/>-->
        </span>
 </boost_spans>
 <analog_spans>
 </analog_spans>
</configuration>






-- 
Regards,
Thangappan.M
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From ahmed.ajmal at gmail.com  Tue Jan 19 22:12:51 2010
From: ahmed.ajmal at gmail.com (Ahmed Bhaila)
Date: Wed, 20 Jan 2010 11:12:51 +0500
Subject: [Freeswitch-users] Help with CDR fields:Other-Leg-Network-Addr
	and Other-Leg-Destination-Number
In-Reply-To: <191c3a031001191137k72530079t402ff54d03b07756@mail.gmail.com>
References: <9d22cc171001190752j797fdf39radc404510128718c@mail.gmail.com>
	<191c3a031001191137k72530079t402ff54d03b07756@mail.gmail.com>
Message-ID: <9d22cc171001192212rb7e617aq7f593b64ac59cd9f@mail.gmail.com>

Thanks Anthony. That worked!!!

- AB

On Wed, Jan 20, 2010 at 12:37 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> those are not variables only variables expand.
>
> try bleg_network_addr
> and bleg_destination_number
>
>
> On Tue, Jan 19, 2010 at 9:52 AM, Ahmed Bhaila <ahmed.ajmal at gmail.com>wrote:
>
>> Hello
>>
>> I am using a template for Master.csv and having problems with getting
>> values for the following fields: Other-Leg-Network-Addr and
>> Other-Leg-Destination-Number. All I am doing is just enclosing these fields
>> in ${} in my template definition and they always turn out to be empty but
>> they arent supposed to be as I am seeing values in the freeswitch console.so
>> I am not sure what is wrong here I have noticed that some of the channel
>> variables(start_stamp, end_stamp) have "variable_" prepended to them when I
>> see them in the console so when defining the template I omit the variable_
>> part and can get the values. Is this something similar? It seems like the
>> problem is with all field names that start with Other. Any help be greatly
>> appreciated.
>>
>>
>> Thanks
>> AB
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From troy at tlainvestments.com  Tue Jan 19 22:15:36 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Tue, 19 Jan 2010 23:15:36 -0700
Subject: [Freeswitch-users] Call Screening Example Broken?
In-Reply-To: <8ccbff061001192132h344a61a4m58d2973391c8182a@mail.gmail.com>
References: <df97b9c51001190107x39d94eaai7400635f0ac03f86@mail.gmail.com>
	<8ccbff061001192132h344a61a4m58d2973391c8182a@mail.gmail.com>
Message-ID: <8BE9BCF3-C766-42CE-98B9-AC5F836327FC@tlainvestments.com>

I've implemented the Example 13: Call Screening from http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_13:_Call_Screening (below) and, while the file plays fine (over and over), fs is reporting an error from switch_ivr_originate.  

[ERR] switch_ivr_originate.c:202 sofia/internal/sip:1000 at 10.0.1.100 Error Playing File!

If I remove the group_confirm_file line, it works as expected.  Could it be that the DTMF (pressing 1 to accept the call) is interrupting the playback of the file?

 <extension name="screen">
   <condition field="destination_number" expression="^(\d{4})$">
     <action application="set" data="call_screen_filename=/tmp/${caller_id_number}-name.wav"/>
     <action application="answer"/>
     <action application="sleep" data="1000"/>
     <action application="phrase" data="voicemail_record_name"/>
     <action application="playback" data="tone_stream://%(500, 0, 640)"/>
     <action application="set" data="playback_terminators=#*0123456789"/>
     <action application="record" data="${call_screen_filename} 7 200 2"/>
     <action application="set" data="group_confirm_key=1"/>
     <action application="set" data="fail_on_single_reject=true"/>
     <action application="set" data="group_confirm_file=${call_screen_filename}"/>
     <action application="set" data="continue_on_fail=true"/>
     <action application="bridge" data="user/$1"/>
     <action application="voicemail" data="default $${domain} $1"/>
     <action application="hangup"/>
   </condition>
 </extension>
-Troy
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From mike at jerris.com  Tue Jan 19 22:16:45 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 Jan 2010 01:16:45 -0500
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
Message-ID: <A6200F19-3547-408C-8EB4-BAD6BE6A9DD0@jerris.com>

grep will tell you the answer.

On Jan 20, 2010, at 12:55 AM, Thangappan.M wrote:

> Dear all,
> 
>          I have successfully configured wanpipe with freeswitch. When I was the running wancfg_fs script the following files openzap.conf , autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf are created.
> 
>          I started the wanrouter command then executed the freeswitch.
>          When I was executing freeswitch mod_openzap.c said the error as  "Error  for finding the span id. name:PRI_1".
>          But in the openzap.conf and openzap.conf.xml files the span name is smg_prid.
>  
>          Why the freeswitch is referring the span name as PRI_1 ?  
>          Whether this has to configured in anywhere? 
> 
>          In the freeswitch CLI using oz command I tried to dump the PRI_1 span id but it said te error  as "PRI_1 is not found".  When I was trying the command 'oz dump smg_prid'  all the channel states and details shown.
>  
>         It seems that smg_prid span configured in openzap perfectly (Its my assumption). Then Why freeswitch is referring the span name as PRI_1. 
> 
> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
> 



From mike at jerris.com  Tue Jan 19 22:22:11 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 Jan 2010 01:22:11 -0500
Subject: [Freeswitch-users] Voicemail to email problems
In-Reply-To: <4B562E63.7000407@acsol.net>
References: <ab161f4749d2cf203ec197497f2ce22e@thom.fr.eu.org>	<4B560A46.90000@acsol.net>	<191c3a031001191146r257d34afq98f921c0ff07f2fc@mail.gmail.com>
	<4E823F6B-75DC-44D9-BD18-C7773D77EAC9@freeswitch.org>
	<4B562E63.7000407@acsol.net>
Message-ID: <47DF4005-415B-467E-A618-E477BD519AEC@jerris.com>

if you didn't have the file it would look more like:

<9>:cat fish | more
cat: fish: No such file or directory

Broken pipe is because the thing it was piped too suddenly disappeared, like it segfaulted or something like that.

Mike


On Jan 19, 2010, at 5:12 PM, John wrote:

> I have modified the Exim configuration as per the wiki. I am still getting the same message, in addition, the /tmp/mail.xxxxxxxx file is not getting created at any point. Is it possible the problem is that there is no file to send, so it errors out? Thank you both for your help! 
> 
> /bin/cat: write error: Broken pipe
> sh: line 1: 21176 Done(1)                 /bin/cat /tmp/mail.1263940067ade7
>      21177 Segmentation fault      (core dumped) | exim4 -f 1004 at voip.server.net -t jhart at server.net
> 
> 
> On 1/19/2010 1:14 PM, Brian West wrote:
>> 
>> http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings
>> 
>> /b
>> 
>> On Jan 19, 2010, at 1:46 PM, Anthony Minessale wrote:
>> 
>>> sounds like exim pretending to be sendmail and not doing it very well.
>>> I think there is a wiki page somewhere that tells you how to config it properly.
>>> 

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From mike at jerris.com  Tue Jan 19 22:28:40 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 Jan 2010 01:28:40 -0500
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
In-Reply-To: <20100119103139.GR4767@tamay-dogan.net>
References: <20100119103139.GR4767@tamay-dogan.net>
Message-ID: <2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>

There is a debian dir in tree for our packages.  I suspect that our packages are quite far from meeting the requirements of pretty much any distro so for now, we will at least have packages for the next release available soon after release.  We don't yet have a box for debian instances for the build farm so we do not build the svn snapshot pacakges for any deb distros.

Mike

On Jan 19, 2010, at 5:31 AM, Michelle Konzack wrote:

> Hi *,
> 
> I loss my last nerv, compiling all the time FreeSWITCH from source...
> 
> Is there someone providing a Debian Package from a repository?
> 
> Also it would be nice if FreeSWITCH go into the Debian distribution.



From thangappan143 at gmail.com  Tue Jan 19 23:13:37 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Wed, 20 Jan 2010 12:43:37 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
Message-ID: <7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>

I found the error in it. The file name is used as openzap.conf.xml (
smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
openzap.conf.wiki.xml  file.

Now the another problem is raised here.
When I was using oz list command , the details of the smg_prid shown. When I
was using 'oz dump smg_prid' command it shows all the channels' details. But
all the channels' states are DOWN. why? How can I make it the states to UP?

When I was making the call , the number is busy message was get. The call
was not  at all landed to the freeswitch.

Dial plan Example:
-------------------------------
 <extension name="PRI_Testing">
        <condition field="destination_number" expression="^39114600$">
                <action application="playback"
data="ivr-welcome_to_freeswitch"/>
        </condition>
  </extension>

Please help me...........

*Output Reference:*
http://pastebin.org/79074

On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Dear all,
>
>          I have successfully configured wanpipe with freeswitch. When I was
> the running wancfg_fs script the following files openzap.conf ,
> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
> are created.
>
>          I started the wanrouter command then executed the freeswitch.
>          When I was executing freeswitch mod_openzap.c said the error as
> "Error  for finding the span id. name:PRI_1".
>          But in the openzap.conf and openzap.conf.xml files the span name
> is smg_prid.
>
>          Why the freeswitch is referring the span name as PRI_1 ?
>          Whether this has to configured in anywhere?
>
>          In the freeswitch CLI using oz command I tried to dump the PRI_1
> span id but it said te error  as "PRI_1 is not found".  When I was trying
> the command 'oz dump smg_prid'  all the channel states and details shown.
>
>         It seems that smg_prid span configured in openzap perfectly (Its my
> assumption). Then Why freeswitch is referring the span name as PRI_1.
>
> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>
> Could anyone please help me?
>
> REFERENCE:
>
> openzap.conf
> [span wanpipe smg_prid]
> name => smg_prid
> trunk_type =>e1
> b-channel => 1:1-15
> b-channel => 1:17-31
>
>
> openzap.conf.xml
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>   <settings>
>     <param name="debug" value="0"/>
>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>     <!--<param name="enable-analog-option" value="3-way"/>-->
>   </settings>
>   <boost_spans>
>         <span id="smg_prid">
>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>         <param name="dialplan" value="XML"/>
>         <param name="context" value="default"/>
>          <!-- regex to stop dialing when it matches -->
>         <!--<param name="dial-regex" value="5555"/>-->
>         <!-- regex to stop dialing when it does not match -->
>         <!--<param name="fail-dial-regex" value="^5"/>-->
>         </span>
>  </boost_spans>
>  <analog_spans>
>  </analog_spans>
> </configuration>
>
>
>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From scottferri09 at gmail.com  Tue Jan 19 23:17:50 2010
From: scottferri09 at gmail.com (Scott Fernandez)
Date: Wed, 20 Jan 2010 12:47:50 +0530
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <8ccbff061001192132h344a61a4m58d2973391c8182a@mail.gmail.com>
References: <df97b9c51001190107x39d94eaai7400635f0ac03f86@mail.gmail.com>
	<8ccbff061001192132h344a61a4m58d2973391c8182a@mail.gmail.com>
Message-ID: <df97b9c51001192317p52824e3aq368215cca75f909@mail.gmail.com>

Thanks Dome. Will try it out and get back to you if I come across any
issues.

Regards,
Scott.

On Wed, Jan 20, 2010 at 11:02 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> Please try http://wiki.freeswitch.org/wiki/Webapi
> you can create class and map to webapi.
>
> Dome C.
>
> 2010/1/19 Scott Fernandez <scottferri09 at gmail.com>:
> > Hi,
> >
> > Is there any API modules available for me to initiate a call from .Net
> based
> > application?.
> >
> > The idea is to include the API modules if any with the .NET base classes
> so
> > that the API commands will be made available on it. I know it is doable
> when
> > I use socket programming in .NET in which Telnet session is created.
> > However, this would potentially hamper the performance of the application
> > because of multiple sessions that will be created for each call.
> >
> > Other than that, Is there any Freeswitch API modules (like plug-ins)
> > available in order to include it into the .Net classes and start building
> > the customized application?
> >
> > Any help from any one is highly appreciated.
> >
> > Thanks,
> > Scott
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Wed Jan 20 00:23:43 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 Jan 2010 03:23:43 -0500
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
Message-ID: <073DBFA8-E2A0-43A6-B898-524C8AEAB296@jerris.com>

Down isn't bad, it just means no one is on that channel

On Jan 20, 2010, at 2:13 AM, "Thangappan.M" <thangappan143 at gmail.com>  
wrote:

> I found the error in it. The file name is used as openzap.conf.xml  
> ( smg_prid is specified here) and another file name as  
> openzap.conf.wiki.xml ( PRI_1 span is specified here ). FreeSWITCH  
> referred the PRI_1 span from openzap.conf.wiki.xml  file.
>
> Now the another problem is raised here.
> When I was using oz list command , the details of the smg_prid  
> shown. When I was using 'oz dump smg_prid' command it shows all the  
> channels' details. But all the channels' states are DOWN. why? How  
> can I make it the states to UP?
>
> When I was making the call , the number is busy message was get. The  
> call was not  at all landed to the freeswitch.
>
> Dial plan Example:
> -------------------------------
>  <extension name="PRI_Testing">
>         <condition field="destination_number" expression="^39114600$">
>                 <action application="playback" data="ivr- 
> welcome_to_freeswitch"/>
>         </condition>
>   </extension>
>
> Please help me...........
>
> Output Reference:
> http://pastebin.org/79074
>
> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <thangappan143 at gmail.com 
> > wrote:
> Dear all,
>
>          I have successfully configured wanpipe with freeswitch.  
> When I was the running wancfg_fs script the following files  
> openzap.conf , autoload_confg/openzap.conf.xml , /etc/wanpipe/ 
> wanpipe1.xml, smg_pri.conf are created.
>
>          I started the wanrouter command then executed the freeswitch.
>          When I was executing freeswitch mod_openzap.c said the  
> error as  "Error  for finding the span id. name:PRI_1".
>          But in the openzap.conf and openzap.conf.xml files the span  
> name is smg_prid.
>
>          Why the freeswitch is referring the span name as PRI_1 ?
>          Whether this has to configured in anywhere?
>
>          In the freeswitch CLI using oz command I tried to dump the  
> PRI_1 span id but it said te error  as "PRI_1 is not found".  When I  
> was trying the command 'oz dump smg_prid'  all the channel states  
> and details shown.
>
>         It seems that smg_prid span configured in openzap perfectly  
> (Its my assumption). Then Why freeswitch is referring the span name  
> as PRI_1.
>
> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>
> Could anyone please help me?
>
> REFERENCE:
>
> openzap.conf
> [span wanpipe smg_prid]
> name => smg_prid
> trunk_type =>e1
> b-channel => 1:1-15
> b-channel => 1:17-31
>
>
> openzap.conf.xml
> <configuration name="openzap.conf" description="OpenZAP  
> Configuration">
>   <settings>
>     <param name="debug" value="0"/>
>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>     <!--<param name="enable-analog-option" value="3-way"/>-->
>   </settings>
>   <boost_spans>
>         <span id="smg_prid">
>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>         <param name="dialplan" value="XML"/>
>         <param name="context" value="default"/>
>          <!-- regex to stop dialing when it matches -->
>         <!--<param name="dial-regex" value="5555"/>-->
>         <!-- regex to stop dialing when it does not match -->
>         <!--<param name="fail-dial-regex" value="^5"/>-->
>         </span>
>  </boost_spans>
>  <analog_spans>
>  </analog_spans>
> </configuration>
>
>
>
>
>
>
> -- 
> Regards,
> Thangappan.M
>
>
>
> -- 
> Regards,
> Thangappan.M
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From devel at thom.fr.eu.org  Wed Jan 20 00:42:01 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Wed, 20 Jan 2010 09:42:01 +0100
Subject: [Freeswitch-users] Home setup with home company
In-Reply-To: <4B55F3A0020000E1000003E0@mail.fribert.dk>
References: <4B55F3A0020000E1000003E0@mail.fribert.dk>
Message-ID: <a934cd637ee34ecf507beb261b4442cd@thom.fr.eu.org>



I think you should remove the "continue=true" in the extension
definition, as FS will continue to process the other extensions even after
this one matches, so I you have another "less restrictive" extension that
could match the call and do answer and/or bridge, then it may be processed
instead of this extension. 

Fran?ois 

On Tue, 19 Jan 2010 18:02:08 +0100,
"mailinglist"  wrote:  I have a very small one man constultancy company
that has the occasional call, unfortunately we are getting more spam calls
after hours than real calls during work hours, so I would like to set up a
TOD system.   First step for me is just playing with the TOD example, I've
gotten this far:   

 group/company@${domain_name}"/>

 My idea with this,
was that in the time 9-17 mon-fri, it rings, the rest of the time it's sent
directly to the voicemail. I would of course like to have it not take
messages outside work hours, but that's just refining :-)   But it picks up
the call, and then nothing...   BR Fribse    

 
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From mailinglist at fribert.dk  Wed Jan 20 02:45:27 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Wed, 20 Jan 2010 11:45:27 +0100
Subject: [Freeswitch-users] Svar: Re:  Home setup with home company
Message-ID: <4B56ECD7020000E1000003F1@mail.fribert.dk>

Hi Michael

It's running on pfsense, so it's kinda locked to the version it currently is.
Looks very nice though.
Looking beyond that, is the action / anti-action list corrent?

Best regards
Fribse


>>> Michael Collins  20-01-10 1:53 >>>


On Tue, Jan 19, 2010 at 9:02 AM, mailinglist <mailinglist at fribert.dk> wrote:
I have a very small one man constultancy company that has the occasional call, unfortunately we are getting more spam calls after hours than real calls during work hours, so I would like to set up a TOD system.
 
First step for me is just playing with the TOD example, I've gotten this far:
 

   
      
      
      
         
         
         
         
         group/company@${domain_name}"/>
         
         
         
         
      


My idea with this, was that in the time 9-17 mon-fri, it rings, the rest of the time it's sent directly to the voicemail.
I would of course like to have it not take messages outside work hours, but that's just refining :-)
 
But it picks up the call, and then nothing...
 

We have a much cleaner way of doing TOD and DOW handling. You'll need to get to the latest FreeSWITCH version. Look in conf/dialplan/default.xml for this example:



    
    
      
        
      
    


Use that condition instead of the two conditions you're now using and see if you have better success.

-MC

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From max.bridgewater at gmail.com  Wed Jan 20 03:23:35 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Wed, 20 Jan 2010 06:23:35 -0500
Subject: [Freeswitch-users] Port question again
Message-ID: <c55d46951001200323t18cd8c7bmc1d3df9163bbcb26@mail.gmail.com>

Hey Guys,

Thought the port question was asked a number of times, I couldn't find an
answer to this. So please bear with me. I have a Freeswitch box that is on
the Internet without any sort of NAT. I want to block as much ports as
possible on this box while still allowing Freeswitch to 1) receive calls
from Voip providers and  2) send calls to other VoIP providers.

What port can I block and what ports do I need to let open?

I know 5080 needs to be open. But can I restrict the RTP ports to, say, only
20000?

Thanks so much.
Max.
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From elihayun at gmail.com  Wed Jan 20 04:03:05 2010
From: elihayun at gmail.com (Eli Hayun)
Date: Wed, 20 Jan 2010 14:03:05 +0200
Subject: [Freeswitch-users] Module multicast fail
Message-ID: <4B56F0F9.9090808@savion.huji.ac.il>

Hi

I am trying to load module event_multicast.

I enabled it in modules and compile it. When I run FS I got this error
(ver 1.0.5pre9)


36m2010-01-20 13:26:47.194141 [NOTICE] switch_loadable_module.c:248
Adding Application 'enum'^M
^[[m^[[36m2010-01-20 13:26:47.195106 [NOTICE]
switch_loadable_module.c:270 Adding API Function 'enum'^M
^[[m^[[36m2010-01-20 13:26:47.195356 [NOTICE]
switch_loadable_module.c:270 Adding API Function 'enum_auto'^M
^[[m^[[m2010-01-20 13:26:47.204768 [CONSOLE]
switch_loadable_module.c:889 Successfully Loaded [mod_cdr_csv]^M
^[[m^[[31m2010-01-20 13:26:47.217277 [ERR] mod_event_multicast.c:410
Multicast Error^M
^[[m^[[31m2010-01-20 13:26:47.217345 [CRIT] switch_loadable_module.c:871
Error Loading module /freeswitch-1.0.5/mod/mod_event_multicast.so^M
**Module load routine returned an error**^M
^[[m^[[m2010-01-20 13:26:47.235551 [CONSOLE]
switch_loadable_module.c:889 Successfully Loaded [mod_event_socket]^M
^[[m^[[36m2010-01-20 13:26:47.235613 [NOTICE]
switch_loadable_module.c:248 Adding Application 'socket'^M
^[[m^[[36m2010-01-20 13:26:47.236229 [NOTICE]
switch_loadable_module.c:270 Adding API Function 'event_sink'^M
^[[m^[[36m2010-01-20 13:26:47.335268 [NOTICE] sofia.c:3274 Started
Profile external [sofia_reg_external]^M
^[[m^[[36m2010-01-20 13:26:47.336713 [NOTICE] sofia.c:3274 Started
Profile internal-ipv6 [sofia_reg_internal-ipv6]^M
^[[m^[[36m2010-01-20 13:26:47.339561 [NOTICE] sofia.c:1804 Adding Alias
[132.64.3.86] for profile [internal]^M



Any idea?


I tried to compile that latest trunk too and  got the same error


Thanks


Eli



From a.alalousi at gmail.com  Wed Jan 20 05:14:16 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Wed, 20 Jan 2010 13:14:16 +0000
Subject: [Freeswitch-users] G729 coded issues
In-Reply-To: <4B55E794.6020909@coppice.org>
References: <cd93b3701001190831u4147214cn9832c6f469c6d17e@mail.gmail.com>
	<6D67C7A9-42CD-4F5F-AD18-E9568098FB71@freeswitch.org>
	<4B55E794.6020909@coppice.org>
Message-ID: <cd93b3701001200514p4acc5a5bk26a463164f841ad2@mail.gmail.com>

Gents,

Thanks for the responses.

Now gotten Howler to send me a custom build that will not offer G729b in any
shape or form, as well as the customer to switch off G729b.

Cisco gateways are negotiating correctly by specifying a=fmtp:18 annexb=yes.
If this line is missing, or has an annexb=no, then they will negotiate
G729a.

Now the million dollar question: ${switch_r_sdp} will get you the SDP for
the remote leg. It's returned as a single string in that variable from what
I could tell.

What sort of regex is allowed in the match condition ? can I, for example,
simply use the Perl s/<expression>/<replacement> syntax to search for the
annexb string and get it rewritten ?

My experiments have so far failed in this regard, i.e. to rewrite the
string. Can anyone provide an example ?

I would like to handle codecs in the following way:


   1. receive inbound call
   2. return ringing tone through 3pcc and ring_back
   3. set continue_on_fail=true
   4. set hangup_after_bridge=true
   5. bridge
   6. on failure, transfer to an extension that will look at outcome of
   codec negotiation
   7. rewrite sdp of the A-leg so that remote end point will successfully
   accept it
   8. bridge the call again with the SDP appropriately written
   9. If we fail, then hangup with NORMAL_CIRCUIT_CONGESTION, otherwise just
   let the call be

Regards, and thanks once more.

Ahmed.

2010/1/19 Steve Underwood <steveu at coppice.org>

> On 01/20/2010 12:36 AM, Brian West wrote:
> > g729a is 100% INVALID in the sdp fix the param in your cisco SPA or your
> Linksys SPA phone and it will stop doing that.  Hopefully they'll fix this
> "bug" soon in the cisco phones to not include the a in the sdp.  The fmtp is
> the proper way to specify annex a or any other options for g729.
> >
> Annex A only affects the inner workings of the codec. There is
> absolutely no difference whatsoever between G.729 and G.729A on the
> wire. The SDP has no reason to mention it, and the standards say it
> shouldn't.
> > /b
> >
> > On Jan 19, 2010, at 10:31 AM, Ahmed Naji wrote:
> >
> >
> >> Hi everyone,
> >>
> >> I have the following scenario and a major customer-affecting issue
> thereof.
> >>
> >> Here is the scenario: customer traffic encoded as G.729 from a cisco
> gateway
> >> ->  our FS (G729 passthrough) ->  remote end gw (G729)
> >>
> >> Calls were failing at an alarming rate, so I looked at the debug logs.
> It
> >> transpired that the Cisco is offering G729 annex b, while the remote end
> can
> >> only do G729a.
> >>
> >> Besides changing source or destination preferences, is there a way to
> ensure
> >> that G729a is used end-end ?
> >>
> >> Thanks in advance.
> >>
> >
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From thangappan143 at gmail.com  Wed Jan 20 06:10:12 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Wed, 20 Jan 2010 19:40:12 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
Message-ID: <7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>

      I noticed the 'oz list' output  in that span type is 'ss7 (boost)'.
How can I change this to isdn?


On Wed, Jan 20, 2010 at 12:43 PM, Thangappan.M <thangappan143 at gmail.com>wrote:

> I found the error in it. The file name is used as openzap.conf.xml (
> smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
> PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
> openzap.conf.wiki.xml  file.
>
> Now the another problem is raised here.
> When I was using oz list command , the details of the smg_prid shown. When
> I was using 'oz dump smg_prid' command it shows all the channels' details.
> But all the channels' states are DOWN. why? How can I make it the states to
> UP?
>
> When I was making the call , the number is busy message was get. The call
> was not  at all landed to the freeswitch.
>
> Dial plan Example:
> -------------------------------
>  <extension name="PRI_Testing">
>         <condition field="destination_number" expression="^39114600$">
>                 <action application="playback"
> data="ivr-welcome_to_freeswitch"/>
>         </condition>
>   </extension>
>
> Please help me...........
>
> *Output Reference:*
> http://pastebin.org/79074
>
>
> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>> Dear all,
>>
>>          I have successfully configured wanpipe with freeswitch. When I
>> was the running wancfg_fs script the following files openzap.conf ,
>> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
>> are created.
>>
>>          I started the wanrouter command then executed the freeswitch.
>>          When I was executing freeswitch mod_openzap.c said the error as
>> "Error  for finding the span id. name:PRI_1".
>>          But in the openzap.conf and openzap.conf.xml files the span name
>> is smg_prid.
>>
>>          Why the freeswitch is referring the span name as PRI_1 ?
>>          Whether this has to configured in anywhere?
>>
>>          In the freeswitch CLI using oz command I tried to dump the PRI_1
>> span id but it said te error  as "PRI_1 is not found".  When I was trying
>> the command 'oz dump smg_prid'  all the channel states and details shown.
>>
>>         It seems that smg_prid span configured in openzap perfectly (Its
>> my assumption). Then Why freeswitch is referring the span name as PRI_1.
>>
>> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>>
>> Could anyone please help me?
>>
>> REFERENCE:
>>
>> openzap.conf
>> [span wanpipe smg_prid]
>> name => smg_prid
>> trunk_type =>e1
>> b-channel => 1:1-15
>> b-channel => 1:17-31
>>
>>
>> openzap.conf.xml
>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>   <settings>
>>     <param name="debug" value="0"/>
>>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>>     <!--<param name="enable-analog-option" value="3-way"/>-->
>>   </settings>
>>   <boost_spans>
>>         <span id="smg_prid">
>>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>         <param name="dialplan" value="XML"/>
>>         <param name="context" value="default"/>
>>          <!-- regex to stop dialing when it matches -->
>>         <!--<param name="dial-regex" value="5555"/>-->
>>         <!-- regex to stop dialing when it does not match -->
>>         <!--<param name="fail-dial-regex" value="^5"/>-->
>>         </span>
>>  </boost_spans>
>>  <analog_spans>
>>  </analog_spans>
>> </configuration>
>>
>>
>>
>>
>>
>>
>> --
>> Regards,
>> Thangappan.M
>>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From santiagosoares at gmail.com  Wed Jan 20 06:18:25 2010
From: santiagosoares at gmail.com (Santiago Soares)
Date: Wed, 20 Jan 2010 12:18:25 -0200
Subject: [Freeswitch-users] Port question again
In-Reply-To: <c55d46951001200323t18cd8c7bmc1d3df9163bbcb26@mail.gmail.com>
References: <c55d46951001200323t18cd8c7bmc1d3df9163bbcb26@mail.gmail.com>
Message-ID: <8ea223c01001200618v49a3ade3q621376c084e82f99@mail.gmail.com>

You can use this rule to allow media from any IP:

-A INPUT -m multiport -p udp --dport 16384:32768 -j ACCEPT

And this one to allow signaling:

-A INPUT -s aaa.bbb.ccc.ddd -p udp --dport 5080 -j ACCEPT

Where aaa.bbb.ccc.ddd is the IP address of your VoIP provider.

Santiago Soares


On Wed, Jan 20, 2010 at 9:23 AM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Hey Guys,
>
> Thought the port question was asked a number of times, I couldn't find an
> answer to this. So please bear with me. I have a Freeswitch box that is on
> the Internet without any sort of NAT. I want to block as much ports as
> possible on this box while still allowing Freeswitch to 1) receive calls
> from Voip providers and? 2) send calls to other VoIP providers.
>
> What port can I block and what ports do I need to let open?
>
> I know 5080 needs to be open. But can I restrict the RTP ports to, say, only
> 20000?
>
> Thanks so much.
> Max.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From moises.silva at gmail.com  Wed Jan 20 07:08:50 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 20 Jan 2010 10:08:50 -0500
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
Message-ID: <c4d05cbe1001200708s7f9e37eco750e66eabcd20c18@mail.gmail.com>

Hi

On Wed, Jan 20, 2010 at 9:10 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

>
>       I noticed the 'oz list' output  in that span type is 'ss7 (boost)'.
> How can I change this to isdn?


Ignore the bad name, as long as you run sangoma_prid siganling binary you
get pri signaling, the openzap side does not really know the details of the
signaling and the ss7 boost name is just a misleading name (for historic and
lame reasons).

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From brian at freeswitch.org  Wed Jan 20 07:17:07 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Jan 2010 09:17:07 -0600
Subject: [Freeswitch-users] G729 coded issues
In-Reply-To: <cd93b3701001200514p4acc5a5bk26a463164f841ad2@mail.gmail.com>
References: <cd93b3701001190831u4147214cn9832c6f469c6d17e@mail.gmail.com>
	<6D67C7A9-42CD-4F5F-AD18-E9568098FB71@freeswitch.org>
	<4B55E794.6020909@coppice.org>
	<cd93b3701001200514p4acc5a5bk26a463164f841ad2@mail.gmail.com>
Message-ID: <537338B1-B582-463E-8EFE-7BBED8165D2B@freeswitch.org>

Just an FYI.  Howler doesn't support the FreeSWITCH project in any way, shape or form.  They do not donate any proceeds or help the project at all.  That said.  We have our officially supported G729 coming out soon that will support the project.  I have it in beta if anyone is really interested in testing it please feel free to email me offlist.  We are currently working out how we want to package the lib, binary and module to make installation easy.

Thanks,
/b

On Jan 20, 2010, at 7:14 AM, Ahmed Naji wrote:

> Howler 



From dftoro at yahoo.com  Wed Jan 20 07:17:51 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Wed, 20 Jan 2010 07:17:51 -0800 (PST)
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
Message-ID: <874941.17255.qm@web33502.mail.mud.yahoo.com>

Hi, the answer is yes, you can to use mod_managed wich offer C# managed class to call control http://wiki.freeswitch.org/wiki/Mod_managed. Or using managed ESL (libs/esl/managed) which offer C# managed class to receive and send events and commands to FreeSwitch.
 
Diego Toro
http://lacarretade.blogspot.com/


--- On Wed, 1/20/10, Scott Fernandez <scottferri09 at gmail.com> wrote:

> From: Scott Fernandez <scottferri09 at gmail.com>
> Subject: Re: [Freeswitch-users] Establishing a Call from .Net based application
> To: freeswitch-users at lists.freeswitch.org
> Date: Wednesday, January 20, 2010, 2:17 AM
> Thanks Dome. Will try it out and get back to
> you if I come across any issues.
> 
> Regards,
> Scott.
> 
> On Wed, Jan 20, 2010 at 11:02 AM,
> Dome Charoenyost <dome at tel.co.th>
> wrote:
> 
> Please try http://wiki.freeswitch.org/wiki/Webapi
> 
> 
> you can create class and map to webapi.
> 
> 
> 
> Dome C.
> 
> 
> 
> 2010/1/19 Scott Fernandez <scottferri09 at gmail.com>:
> 
> > Hi,
> 
> >
> 
> > Is there any API modules available for me to initiate
> a call from .Net based
> 
> > application?.
> 
> >
> 
> > The idea is to include the API modules if any with the
> .NET base classes so
> 
> > that the API commands will be made available on it. I
> know it is doable when
> 
> > I use socket programming in .NET in which Telnet
> session is created.
> 
> > However, this would potentially hamper the performance
> of the application
> 
> > because of multiple sessions that will be created for
> each call.
> 
> >
> 
> > Other than that, Is there any Freeswitch API modules
> (like plug-ins)
> 
> > available in order to include it into the .Net classes
> and start building
> 
> > the customized application?
> 
> >
> 
> > Any help from any one is highly appreciated.
> 
> >
> 
> > Thanks,
> 
> > Scott
> 
> >
> 
> >
> _______________________________________________
> 
> > FreeSWITCH-users mailing list
> 
> > FreeSWITCH-users at lists.freeswitch.org
> 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> > http://www.freeswitch.org
> 
> >
> 
> >
> 
> 
> 
> _______________________________________________
> 
> FreeSWITCH-users mailing list
> 
> FreeSWITCH-users at lists.freeswitch.org
> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> http://www.freeswitch.org
> 
> 
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



      


From brian at freeswitch.org  Wed Jan 20 07:18:36 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Jan 2010 09:18:36 -0600
Subject: [Freeswitch-users] Module multicast fail
In-Reply-To: <4B56F0F9.9090808@savion.huji.ac.il>
References: <4B56F0F9.9090808@savion.huji.ac.il>
Message-ID: <2A8BE7A9-EC50-47CD-9FE8-172BF0F97DB6@freeswitch.org>

Please visit http://latest.freeswitch.org  and update to the latest ;)  Its the best you can get to date!  All the preX releases are gone from the download site.

/b

On Jan 20, 2010, at 6:03 AM, Eli Hayun wrote:

> (ver 1.0.5pre9)



From ecasarero at gmail.com  Wed Jan 20 07:10:43 2010
From: ecasarero at gmail.com (Eduardo Casarero)
Date: Wed, 20 Jan 2010 12:10:43 -0300
Subject: [Freeswitch-users] Port question again
In-Reply-To: <8ea223c01001200618v49a3ade3q621376c084e82f99@mail.gmail.com>
References: <c55d46951001200323t18cd8c7bmc1d3df9163bbcb26@mail.gmail.com> 
	<8ea223c01001200618v49a3ade3q621376c084e82f99@mail.gmail.com>
Message-ID: <7d9b3cf21001200710v47bf7b59pf7fc47705396abbb@mail.gmail.com>

Hi list, i'm a brand new freeswitch user (without previous asterisk/voip
experience), after reading all wiki pages, google searchs, etc i need some
help to solve a problem.

configuration:

Freeswitch -> Firewall (nat) -> internet -> Sip Provider

In my current configuration the gateway is REGED and inbound calls (from
provider to freeswitch) works ok with good audio quality. However outbound
calls don't. When i call through the gateway the destination phone rings,
and when is answered there is no audio.

I've check with "show channels" in fs_cli and i cant see any codec in the
read_codec write_codec part, they are blank. I've reviewed all sip profiles
configuration, but obviously i'm missing something.

I will really appreciate any comment,guidance,help,etc. (if someone is in
Buenos Aires/Argentina i can also offer a free beer!)

Thanks in advance.

Eduardo.
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From ecasarero at gmail.com  Wed Jan 20 07:11:55 2010
From: ecasarero at gmail.com (Eduardo Casarero)
Date: Wed, 20 Jan 2010 12:11:55 -0300
Subject: [Freeswitch-users] Problem with outbound calls
Message-ID: <7d9b3cf21001200711h6ce5eda3v1609a1487ff7dc2@mail.gmail.com>

Hi list, i'm a brand new freeswitch user (without previous asterisk/voip
experience), after reading all wiki pages, google searchs, etc i need some
help to solve a problem.

configuration:

Freeswitch -> Firewall (nat) -> internet -> Sip Provider

In my current configuration the gateway is REGED and inbound calls (from
provider to freeswitch) works ok with good audio quality. However outbound
calls don't. When i call through the gateway the destination phone rings,
and when is answered there is no audio.

I've check with "show channels" in fs_cli and i cant see any codec in the
read_codec write_codec part, they are blank. I've reviewed all sip profiles
configuration, but obviously i'm missing something.

I will really appreciate any comment,guidance,help,etc. (if someone is in
Buenos Aires/Argentina i can also offer a free beer!)

Thanks in advance.

Eduardo.
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From stevendt at primrosebank.net  Wed Jan 20 08:17:53 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 20 Jan 2010 16:17:53 -0000
Subject: [Freeswitch-users] Configuration Preservation through Trunk Updates
Message-ID: <21A169E64A904A188DC34E67B1A17651@bp1.ad.bp.com>

Hi,

What is the philosophy/technique for preserving user configuration when doing updates to the latest SVN ?

For example, under Windows, using Visual C++ 2008 Express, the program is built under the "\FreeSwitch\.\Debug" directory with all other FreeSwitch directories below that. When FreeSwitch is installed, the configuration directories, including conf\autoload_configs, conf\dialplan, conf\directory etc, are copied from the distro.

What is the best way of preserving user configuration through future rebuilds, i.e., dialplans, extensions etc. which may have been modified from the defaults ?

Can previously configs just be copied back into the appropriate directory ?

Is compatibility of configs "guaranteed" to be preserved between releases, e.g., 1.0.4 to 1.0.5 or even between SVNs of the same release ?

How do others manage this ?

Regards
Dave
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From brian at freeswitch.org  Wed Jan 20 08:28:09 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Jan 2010 10:28:09 -0600
Subject: [Freeswitch-users] Configuration Preservation through Trunk
	Updates
In-Reply-To: <21A169E64A904A188DC34E67B1A17651@bp1.ad.bp.com>
References: <21A169E64A904A188DC34E67B1A17651@bp1.ad.bp.com>
Message-ID: <C3424C68-4E69-4CA1-A494-2B143C0E8E8F@freeswitch.org>

The best bet is to never touch the installed configs.  And thats what we don on linux.

/b

On Jan 20, 2010, at 10:17 AM, Dave Stevenson wrote:

> What is the philosophy/technique for preserving user configuration when doing updates to the latest SVN ?

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From fdelawarde at wirelessmundi.com  Wed Jan 20 08:29:23 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Wed, 20 Jan 2010 17:29:23 +0100
Subject: [Freeswitch-users] Port question again
In-Reply-To: <8ea223c01001200618v49a3ade3q621376c084e82f99@mail.gmail.com>
References: <c55d46951001200323t18cd8c7bmc1d3df9163bbcb26@mail.gmail.com>
	<8ea223c01001200618v49a3ade3q621376c084e82f99@mail.gmail.com>
Message-ID: <1264004963.14614.30.camel@luna.tc.commsmundi.com>

On Wed, 2010-01-20 at 12:18 -0200, Santiago Soares wrote:
> You can use this rule to allow media from any IP:
> 
> -A INPUT -m multiport -p udp --dport 16384:32768 -j ACCEPT
> 
> And this one to allow signaling:
> 
> -A INPUT -s aaa.bbb.ccc.ddd -p udp --dport 5080 -j ACCEPT

No need to load multiport in that case:

-A INPUT -p udp --dport 5080 -j ACCEPT
-A INPUT -p udp --dport 16384:32768 -j ACCEPT

Equivalent with multiport:

-A INPUT -p udp -m multiport --dports 5080,16384:32768 -j ACCEPT

Fran?ois.



From brian at freeswitch.org  Wed Jan 20 08:32:28 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Jan 2010 10:32:28 -0600
Subject: [Freeswitch-users] Port question again
In-Reply-To: <1264004963.14614.30.camel@luna.tc.commsmundi.com>
References: <c55d46951001200323t18cd8c7bmc1d3df9163bbcb26@mail.gmail.com>
	<8ea223c01001200618v49a3ade3q621376c084e82f99@mail.gmail.com>
	<1264004963.14614.30.camel@luna.tc.commsmundi.com>
Message-ID: <256CB704-0659-46AF-B14A-E48311B17EB2@freeswitch.org>

I'm going to point out that you should open up tcp on 5080 also.  As we actually DO support TCP!

/b

On Jan 20, 2010, at 10:29 AM, Fran?ois Delawarde wrote:

> No need to load multiport in that case:
> 
> -A INPUT -p udp --dport 5080 -j ACCEPT
> -A INPUT -p udp --dport 16384:32768 -j ACCEPT
> 
> Equivalent with multiport:
> 
> -A INPUT -p udp -m multiport --dports 5080,16384:32768 -j ACCEPT
> 
> Fran?ois.



From jcasale at activenetwerx.com  Wed Jan 20 08:36:35 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 20 Jan 2010 16:36:35 +0000
Subject: [Freeswitch-users] Sip video intercom
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9083ACB@prato.activenetwerx.local>

I need to get an intercom integrated into the voip system of a highend home.
That being said, I am looking for a nice looking discrete panel to mount
outside by the front door. Anyone have any experience with these and know of
a model they recommend?

Thanks!
jlc


From wchao at yahoo.com  Wed Jan 20 08:48:08 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Wed, 20 Jan 2010 11:48:08 -0500 (EST)
Subject: [Freeswitch-users] Eavesdrop when using simring
Message-ID: <alpine.LRH.2.00.1001201139480.13391@osgood.microoffice.us>

I have eavesdrop working fine on outbound calls and also inbound calls 
where there is a single DID per IP phone. When I have a DID that rings 
multiple extensions simultaneously, what is the best way to obtain 
information about which extension has picked up the call and store that 
using hash? I can set a variable before I issue the bridge action, like 
so:

<action application="hash" data="insert/${domain_name}-spymap/646xxxyyyy-1000/${uuid}"/>
<action application="bridge" data="user/1001@${domain_name},user/1002@${domain_name},user/1003@${domain_name},user/1004@${domain_name}"/>

However, that doesn't tell me who actually picked up, so at best I can 
allow users to eavesdrop on the last incoming call to the main DID, not 
the last incoming call to a particular extension. Is there something I can 
do in the bridge that will cause it to set a variable once it knows which 
extension has picked up the call?


From jerry.richards at teotech.com  Wed Jan 20 08:59:40 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 20 Jan 2010 08:59:40 -0800
Subject: [Freeswitch-users] Freeswitch Test Tool
Message-ID: <94D82278DEC141DD9B4B951C67848708@greyhawk.tonecommander.com>


Can anyone recommend a good SIP Test Tool to test the Freeswitch PBX?  I
would like one capable of both load tests and also various call scenarios.

Thanks,
Jerry



From gmaruzz at celliax.org  Wed Jan 20 09:16:30 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 20 Jan 2010 18:16:30 +0100
Subject: [Freeswitch-users] Freeswitch Test Tool
In-Reply-To: <94D82278DEC141DD9B4B951C67848708@greyhawk.tonecommander.com>
References: <94D82278DEC141DD9B4B951C67848708@greyhawk.tonecommander.com>
Message-ID: <7b197bef1001200916yaf64a47j74b8fa7577fac53d@mail.gmail.com>

Sipp (http://sipp.sourceforge.net/)?



On Wed, Jan 20, 2010 at 5:59 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
>
> Can anyone recommend a good SIP Test Tool to test the Freeswitch PBX? ?I
> would like one capable of both load tests and also various call scenarios.
>
> Thanks,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From frank at carmickle.com  Wed Jan 20 09:27:26 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Wed, 20 Jan 2010 12:27:26 -0500
Subject: [Freeswitch-users] Port question again
In-Reply-To: <7d9b3cf21001200710v47bf7b59pf7fc47705396abbb@mail.gmail.com>
References: <c55d46951001200323t18cd8c7bmc1d3df9163bbcb26@mail.gmail.com>
	<8ea223c01001200618v49a3ade3q621376c084e82f99@mail.gmail.com>
	<7d9b3cf21001200710v47bf7b59pf7fc47705396abbb@mail.gmail.com>
Message-ID: <20100120172725.GD7006@base.carmickle.com>

Hello

On Wed, Jan 20, Eduardo Casarero wrote:
> Hi list, i'm a brand new freeswitch user (without previous asterisk/voip
> experience), after reading all wiki pages, google searchs, etc i need some
> help to solve a problem.
> 
> configuration:
> 
> Freeswitch -> Firewall (nat) -> internet -> Sip Provider
> 
> In my current configuration the gateway is REGED and inbound calls (from
> provider to freeswitch) works ok with good audio quality. However outbound
> calls don't. When i call through the gateway the destination phone rings,
> and when is answered there is no audio.
> 
> I've check with "show channels" in fs_cli and i cant see any codec in the
> read_codec write_codec part, they are blank. I've reviewed all sip profiles
> configuration, but obviously i'm missing something.

Sounds to me like your firewall is blocking outbound ports.  If it's a linux machine you'll want something like

-A OUTPUT -p udp --dport 16384:32768 -j ACCEPT

> 
> I will really appreciate any comment,guidance,help,etc. (if someone is in
> Buenos Aires/Argentina i can also offer a free beer!)

I am not but I'd sure love to try your local beer.

HTH
--FC


From stevendt at primrosebank.net  Wed Jan 20 09:29:03 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 20 Jan 2010 17:29:03 -0000
Subject: [Freeswitch-users] Configuration Preservation through
	TrunkUpdates
References: <21A169E64A904A188DC34E67B1A17651@bp1.ad.bp.com>
	<C3424C68-4E69-4CA1-A494-2B143C0E8E8F@freeswitch.org>
Message-ID: <5A041CBBECB441D094B86C935CC6C792@bp1.ad.bp.com>

Hi Thanks Brian,

OK, but I'm sure everyone has their own requirements for dialplan actions, extensions etc. If I understand you right, you're saying that I should leave all the defaults (dialplans, extensions etc.) in place and do anything specific to my installation in separate files ?

I think that I can see how that would work :-

Provided that I use extensions not already defined in dialplan\default.xml. then any extensions that I add won't be touched by FS.
What if I redefined a pre-defined extension, would it take precedence over the data in dialpan\default.xml ?

Any user dialplan actions would go into nn_xxxx.xml files in the dialplan\defaults dir and be processed after dialpan\default.xml ?

If I understand correctly, I'll head off and put things right !

regards
Dave


  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, January 20, 2010 4:28 PM
  Subject: Re: [Freeswitch-users] Configuration Preservation through TrunkUpdates


  The best bet is to never touch the installed configs.  And thats what we don on linux.


  /b


  On Jan 20, 2010, at 10:17 AM, Dave Stevenson wrote:


    What is the philosophy/technique for preserving user configuration when doing updates to the latest SVN ?




------------------------------------------------------------------------------


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From brian at freeswitch.org  Wed Jan 20 09:32:42 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Jan 2010 11:32:42 -0600
Subject: [Freeswitch-users] Port question again
In-Reply-To: <20100120172725.GD7006@base.carmickle.com>
References: <c55d46951001200323t18cd8c7bmc1d3df9163bbcb26@mail.gmail.com>
	<8ea223c01001200618v49a3ade3q621376c084e82f99@mail.gmail.com>
	<7d9b3cf21001200710v47bf7b59pf7fc47705396abbb@mail.gmail.com>
	<20100120172725.GD7006@base.carmickle.com>
Message-ID: <95B92BEF-14BA-4274-9F0C-F6DD1F18E665@freeswitch.org>

I have an even better solution:

install MiniUPnP daemon.

/b

On Jan 20, 2010, at 11:27 AM, Frank Carmickle wrote:

> Hello



From brian at freeswitch.org  Wed Jan 20 09:34:39 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Jan 2010 11:34:39 -0600
Subject: [Freeswitch-users] Configuration Preservation through
	TrunkUpdates
In-Reply-To: <5A041CBBECB441D094B86C935CC6C792@bp1.ad.bp.com>
References: <21A169E64A904A188DC34E67B1A17651@bp1.ad.bp.com>
	<C3424C68-4E69-4CA1-A494-2B143C0E8E8F@freeswitch.org>
	<5A041CBBECB441D094B86C935CC6C792@bp1.ad.bp.com>
Message-ID: <1F2B2E59-6C17-4DB8-A197-E57D2A804132@freeswitch.org>

You should NEVER install anything into a config folder if one already exists.  But you're free to do what you want locally but we will NEVER allow the install process to install extra files or overwrite existing configs its bad behavior to do so.

/b

On Jan 20, 2010, at 11:29 AM, Dave Stevenson wrote:

> Hi Thanks Brian,
>  
> OK, but I'm sure everyone has their own requirements for dialplan actions, extensions etc. If I understand you right, you're saying that I should leave all the defaults (dialplans, extensions etc.) in place and do anything specific to my installation in separate files ?
>  
> I think that I can see how that would work :-
>  
> Provided that I use extensions not already defined in dialplan\default.xml. then any extensions that I add won't be touched by FS.
> What if I redefined a pre-defined extension, would it take precedence over the data in dialpan\default.xml ?
>  
> Any user dialplan actions would go into nn_xxxx.xml files in the dialplan\defaults dir and be processed after dialpan\default.xml ?
>  
> If I understand correctly, I'll head off and put things right !
>  
> regards
> Dave
>  

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From anthony.minessale at gmail.com  Wed Jan 20 09:50:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 20 Jan 2010 11:50:24 -0600
Subject: [Freeswitch-users] Bypass Media True Disables MOH
In-Reply-To: <E55CF9FDA2AC406B8799A0081BCBA4DD@greyhawk.tonecommander.com>
References: <AAA8629BCAF448659E9B91C20E75CCC6@greyhawk.tonecommander.com>
	<9BF355CF-C633-4BF5-BB8B-642DD81936D1@freeswitch.org>
	<E55CF9FDA2AC406B8799A0081BCBA4DD@greyhawk.tonecommander.com>
Message-ID: <191c3a031001200950i228b9198o78cfaa7185fd7eb0@mail.gmail.com>

I would say $500 bounty to make it go back to bypassing after the hold is
over.
contact me directly if you wish to proceed.


On Tue, Jan 19, 2010 at 9:50 AM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> We are willing to pay a bounty for this.  What amount would you suggest?
>  We
> would like the media to normally go directly between the endpoints, but
> when
> a call is put on-hold, we would like the other end should hear MOH.
>
> Thanks,
> Jerry
>
>
> -----Original Message-----
> From: Brian West [mailto:brian at freeswitch.org]
> Sent: Tuesday, December 29, 2009 1:12 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Bypass Media True Disables MOH
>
> <param name="media-option" value="resume-media-on-hold"/>
>
> But it doesn't go back to bypass after....   Maybe you can post a bounty
> for
> that functionality.
>
> /b
>
> On Dec 29, 2009, at 2:42 PM, Jerry Richards wrote:
>
> >
> > When I uncomment the following tag, internally held calls no longer
> > hear MOH.
> >
> > <param name="inbound-bypass-media" value="true"/>
> >
> > Is there a way to have the above uncommented and still provide MOH to
> > held calls?
> >
> > Best Regards,
> > Jerry
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From linux4michelle at tamay-dogan.net  Wed Jan 20 10:07:20 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Wed, 20 Jan 2010 19:07:20 +0100
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
In-Reply-To: <2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>
References: <20100119103139.GR4767@tamay-dogan.net>
	<2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>
Message-ID: <20100120180720.GH4767@tamay-dogan.net>

Hello,

Am 2010-01-20 01:28:40, schrieb Michael Jerris:
> There is a debian dir in tree for our packages.  I suspect that our
> packages are quite far from meeting the requirements of pretty much
> any distro so for now, we will at least have packages for the next
> release available soon after release.  We don't yet have a box for
> debian instances for the build farm so we do not build the svn
> snapshot pacakges for any deb distros.

Hmmm, currently I have only one fixed IP and some VServers and  PBuilder
runing on i386 and ARM (only a small Ti Sitara AM3517 with 256 MB memory
plus SATA drive)...

Also I am trying to relocate back to Germany...
Maybe I could do the Job for Debian (i386 ARMEL) and Ubuntu (i386)

My Website is currently a backup fro 2008/12 and 2009/03 because  I  was
offine since 2009-07-23 du to my fuckingbusiness partner...

Maybe it work:
    http://www.debian.tamay-dogan.net/

I will try to reinstall the PBuilder interface which  allow  uploads  of
sources/configs and autobuilding.  Also  I  like  to  include  an  auto-
checkout from "svn" and "git" so, the author of the software should give
a signal to my interface and the build is done automaticaly.

Also I am Package Maintainer of some Debian packages...

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From stevendt at primrosebank.net  Wed Jan 20 11:44:09 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 20 Jan 2010 19:44:09 -0000
Subject: [Freeswitch-users] Configuration Preservation
	throughTrunkUpdates
References: <21A169E64A904A188DC34E67B1A17651@bp1.ad.bp.com><C3424C68-4E69-4CA1-A494-2B143C0E8E8F@freeswitch.org><5A041CBBECB441D094B86C935CC6C792@bp1.ad.bp.com>
	<1F2B2E59-6C17-4DB8-A197-E57D2A804132@freeswitch.org>
Message-ID: <DF8D6745E8E1492A9D2E345A6B42D768@bp1.ad.bp.com>

Brian,

Following on from before, there's one item that I can't see how to do outside of modifying the directory\default.xml file, and that is setting up call groups.

I thought that I'd perhaps be able to do something similar to creating user dial plans and create a new file in directory\default\  which would be loaded before the other extensions, i.e., called something like 00_groups.xml and have the call group created there. (My test file is shown below). That did not seem to work, am I on the right lines or should custom groups get created somewhere else ?

regards
Dave


<!-- Group Configuration File -->

<include>
    <groups>
      <group name="primrose">
         <users>
           <user id="101" type="pointer"/>
           <user id="111" type="pointer"/>
         </users>
      </group>

    </groups>

</include>



  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, January 20, 2010 5:34 PM
  Subject: Re: [Freeswitch-users] Configuration Preservation throughTrunkUpdates


  You should NEVER install anything into a config folder if one already exists.  But you're free to do what you want locally but we will NEVER allow the install process to install extra files or overwrite existing configs its bad behavior to do so.


  /b


  On Jan 20, 2010, at 11:29 AM, Dave Stevenson wrote:


    Hi Thanks Brian,

    OK, but I'm sure everyone has their own requirements for dialplan actions, extensions etc. If I understand you right, you're saying that I should leave all the defaults (dialplans, extensions etc.) in place and do anything specific to my installation in separate files ?

    I think that I can see how that would work :-

    Provided that I use extensions not already defined in dialplan\default.xml. then any extensions that I add won't be touched by FS.
    What if I redefined a pre-defined extension, would it take precedence over the data in dialpan\default.xml ?

    Any user dialplan actions would go into nn_xxxx.xml files in the dialplan\defaults dir and be processed after dialpan\default.xml ?

    If I understand correctly, I'll head off and put things right !

    regards
    Dave





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Wed Jan 20 12:00:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 20 Jan 2010 14:00:07 -0600
Subject: [Freeswitch-users] Freeswitch Test Tool
In-Reply-To: <94D82278DEC141DD9B4B951C67848708@greyhawk.tonecommander.com>
References: <94D82278DEC141DD9B4B951C67848708@greyhawk.tonecommander.com>
Message-ID: <191c3a031001201200t9bf3505o45bb22de368ddfe2@mail.gmail.com>

another FS box

On Wed, Jan 20, 2010 at 10:59 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

>
> Can anyone recommend a good SIP Test Tool to test the Freeswitch PBX?  I
> would like one capable of both load tests and also various call scenarios.
>
> Thanks,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Jan 20 12:09:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 20 Jan 2010 14:09:48 -0600
Subject: [Freeswitch-users] Eavesdrop when using simring
In-Reply-To: <alpine.LRH.2.00.1001201139480.13391@osgood.microoffice.us>
References: <alpine.LRH.2.00.1001201139480.13391@osgood.microoffice.us>
Message-ID: <191c3a031001201209y53d68c83t8ba066b63ff4b723@mail.gmail.com>

maybe api_on_answer var?

On Wed, Jan 20, 2010 at 10:48 AM, Wellie Chao <wchao at yahoo.com> wrote:

> I have eavesdrop working fine on outbound calls and also inbound calls
> where there is a single DID per IP phone. When I have a DID that rings
> multiple extensions simultaneously, what is the best way to obtain
> information about which extension has picked up the call and store that
> using hash? I can set a variable before I issue the bridge action, like
> so:
>
> <action application="hash"
> data="insert/${domain_name}-spymap/646xxxyyyy-1000/${uuid}"/>
> <action application="bridge" data="user/1001@${domain_name},user/1002@
> ${domain_name},user/1003@${domain_name},user/1004@${domain_name}"/>
>
> However, that doesn't tell me who actually picked up, so at best I can
> allow users to eavesdrop on the last incoming call to the main DID, not
> the last incoming call to a particular extension. Is there something I can
> do in the bridge that will cause it to set a variable once it knows which
> extension has picked up the call?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From larclap at yahoo.com  Wed Jan 20 13:02:45 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 20 Jan 2010 13:02:45 -0800
Subject: [Freeswitch-users] Debug message on console?
Message-ID: <00a801ca9a13$ea603470$bf209d50$@com>

For about the last few weeks I've noticed the following message on the
console:

 

[DEBUG] sofia_reg.c:1815 adding X-Real-IP => 192.168.10.104 to xml_curl
request

 

I am sorry but cannot understand the code. The endpoint is a SNOM 320 at
7.3.14. It is registered for two extensions. The message is output twice
every 5 minutes.

 

Does this message indicate a problem? If so, how can I correct it?

 

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

FreeSWITCH v16385

 

Thanks, Lars

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From mike at van.lammeren.net  Wed Jan 20 13:18:44 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Wed, 20 Jan 2010 16:18:44 -0500
Subject: [Freeswitch-users] Question about Lua script: How do I detect when
	someone picks up the phone?
Message-ID: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>

Hello!

Some day, I'll probably laugh at asking this question, but today I can't
figure it out.

I've written a Lua script that listens for a call, then dials a phone number
to a second person. It plays a message, then prompts the second person to
hit pound to connect. If the second person hits pound, then it bridges the
two calls together.

All that works great, but I can't figure out how to get the session for the
second person to wait until that person answers.

I'm using FreeSWITCH 1.0.4, and although there is a *getState* function
documented in the wiki, it doesn't seem to exist for me.

Any help would be appreciated!


Mike van Lammeren
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From rob4manhere at gmail.com  Wed Jan 20 13:32:18 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Wed, 20 Jan 2010 15:32:18 -0600
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
Message-ID: <e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>

Hi Mike,

I don't think v1.0.4 is supported any longer.   You'll have better luck
getting assistance by upgrading to trunk or the latest tar and reporting
back.

Good luck!
Rob

On Wed, Jan 20, 2010 at 3:18 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Hello!
>
> Some day, I'll probably laugh at asking this question, but today I can't
> figure it out.
>
> I've written a Lua script that listens for a call, then dials a phone
> number to a second person. It plays a message, then prompts the second
> person to hit pound to connect. If the second person hits pound, then it
> bridges the two calls together.
>
> All that works great, but I can't figure out how to get the session for the
> second person to wait until that person answers.
>
> I'm using FreeSWITCH 1.0.4, and although there is a *getState* function
> documented in the wiki, it doesn't seem to exist for me.
>
> Any help would be appreciated!
>
>
> Mike van Lammeren
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at van.lammeren.net  Wed Jan 20 13:45:23 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Wed, 20 Jan 2010 16:45:23 -0500
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
	<e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
Message-ID: <5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>

Hi Rob!

Unfortunately, I have the next few weeks to complete this part of the
project. Without a stable release of 1.0.5, I have to stick with 1.0.4. I'm
sure that 1.0.4 can detect and report when a phone is picked up. It's just
that I can't figure out how to get that information! Either that, or I have
something mis-configured.

Mike van Lammeren


On Wed, Jan 20, 2010 at 4:32 PM, Rob Forman <rob4manhere at gmail.com> wrote:

> Hi Mike,
>
> I don't think v1.0.4 is supported any longer.   You'll have better luck
> getting assistance by upgrading to trunk or the latest tar and reporting
> back.
>
> Good luck!
> Rob
>
> On Wed, Jan 20, 2010 at 3:18 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:
>
>> Hello!
>>
>> Some day, I'll probably laugh at asking this question, but today I can't
>> figure it out.
>>
>> I've written a Lua script that listens for a call, then dials a phone
>> number to a second person. It plays a message, then prompts the second
>> person to hit pound to connect. If the second person hits pound, then it
>> bridges the two calls together.
>>
>> All that works great, but I can't figure out how to get the session for
>> the second person to wait until that person answers.
>>
>> I'm using FreeSWITCH 1.0.4, and although there is a *getState* function
>> documented in the wiki, it doesn't seem to exist for me.
>>
>> Any help would be appreciated!
>>
>>
>> Mike van Lammeren
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at van.lammeren.net  Wed Jan 20 14:06:38 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Wed, 20 Jan 2010 17:06:38 -0500
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
	<e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
	<5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>
Message-ID: <5d2828f1001201406p4154b98ald1af1c5c25f59337@mail.gmail.com>

So, I've been reading about early media in the wiki, and have made a little
progress, which leads to more questions.

I understand now why a call is considered connected before one person has
picked up the phone. I am also able to get my script to wait for the phone
to be picked up, by setting the ignore_early_media variable when starting a
new session, like this:

customerSession =
freeswitch.Session("{ignore_early_media=true}sofia/gateway/example.com/" ..
customerPhoneNumber)


After that line, the script waits for the other phone to be picked up.

However, now I wonder what to do with calls that don't complete, get busy
signals, etc.

What do people do in this case? The only related example I can find on the
web is for a javascript dialer, which doesn't address any of these cases.

Early Media: http://wiki.freeswitch.org/wiki/Early_media
ignore_early_media variable:
http://wiki.freeswitch.org/wiki/Variable_ignore_early_media
<http://wiki.freeswitch.org/wiki/Variable_ignore_early_media>javascript
dialer: http://alexn.org/docs/dialer.html


<http://alexn.org/docs/dialer.html>Mike van Lammeren

On Wed, Jan 20, 2010 at 4:45 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Hi Rob!
>
> Unfortunately, I have the next few weeks to complete this part of the
> project. Without a stable release of 1.0.5, I have to stick with 1.0.4. I'm
> sure that 1.0.4 can detect and report when a phone is picked up. It's just
> that I can't figure out how to get that information! Either that, or I have
> something mis-configured.
>
> Mike van Lammeren
>
>
> On Wed, Jan 20, 2010 at 4:32 PM, Rob Forman <rob4manhere at gmail.com> wrote:
>
>> Hi Mike,
>>
>> I don't think v1.0.4 is supported any longer.   You'll have better luck
>> getting assistance by upgrading to trunk or the latest tar and reporting
>> back.
>>
>> Good luck!
>> Rob
>>
>> On Wed, Jan 20, 2010 at 3:18 PM, Mike van Lammeren <mike at van.lammeren.net
>> > wrote:
>>
>>>  Hello!
>>>
>>> Some day, I'll probably laugh at asking this question, but today I can't
>>> figure it out.
>>>
>>> I've written a Lua script that listens for a call, then dials a phone
>>> number to a second person. It plays a message, then prompts the second
>>> person to hit pound to connect. If the second person hits pound, then it
>>> bridges the two calls together.
>>>
>>> All that works great, but I can't figure out how to get the session for
>>> the second person to wait until that person answers.
>>>
>>> I'm using FreeSWITCH 1.0.4, and although there is a *getState* function
>>> documented in the wiki, it doesn't seem to exist for me.
>>>
>>> Any help would be appreciated!
>>>
>>>
>>> Mike van Lammeren
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From larclap at yahoo.com  Wed Jan 20 14:15:38 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 20 Jan 2010 14:15:38 -0800
Subject: [Freeswitch-users] Additional endpoints
Message-ID: <00d701ca9a1e$18afa0e0$4a0ee2a0$@com>

I know this has been answered before, but I cannot find it.

 

How do I setup more than the default 20 endpoints (1000-1019)?  Do I extend
the definition in dialplan/public.xml (public_extensions) and add the extra
in directory/default?

 

Thanks, Lars

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From anthony.minessale at gmail.com  Wed Jan 20 14:28:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 20 Jan 2010 16:28:37 -0600
Subject: [Freeswitch-users] Debug message on console?
In-Reply-To: <00a801ca9a13$ea603470$bf209d50$@com>
References: <00a801ca9a13$ea603470$bf209d50$@com>
Message-ID: <191c3a031001201428m72bcad24u6a29b0014d00f6a0@mail.gmail.com>

no it's overly chatty, i will move it up to debug level 10 so you won't see
it.


On Wed, Jan 20, 2010 at 3:02 PM, Lars Zeb <larclap at yahoo.com> wrote:

>  For about the last few weeks I?ve noticed the following message on the
> console:
>
>
>
> [DEBUG] sofia_reg.c:1815 adding X-Real-IP => 192.168.10.104 to xml_curl
> request
>
>
>
> I am sorry but cannot understand the code. The endpoint is a SNOM 320 at
> 7.3.14. It is registered for two extensions. The message is output twice
> every 5 minutes.
>
>
>
> Does this message indicate a problem? If so, how can I correct it?
>
>
>
> Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
> i386 GNU/Linux
>
> FreeSWITCH v16385
>
>
>
> Thanks, Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From rob4manhere at gmail.com  Wed Jan 20 14:29:00 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Wed, 20 Jan 2010 16:29:00 -0600
Subject: [Freeswitch-users] Additional endpoints
In-Reply-To: <00d701ca9a1e$18afa0e0$4a0ee2a0$@com>
References: <00d701ca9a1e$18afa0e0$4a0ee2a0$@com>
Message-ID: <e15d328c1001201429m6130a132vccc8e373d65cd1f6@mail.gmail.com>

Hi Lars,

For endpoint authentication, yes, copy and add more entries to
./conf/directory/default/.

For internal dialing, you'd need to change the regex expression under
"Local_Extension" in conf/dialplan/default.xml, or add additional extensions
under conf/dialplan/default/.

Rob

On Wed, Jan 20, 2010 at 4:15 PM, Lars Zeb <larclap at yahoo.com> wrote:

>  I know this has been answered before, but I cannot find it.
>
>
>
> How do I setup more than the default 20 endpoints (1000-1019)?  Do I extend
> the definition in dialplan/public.xml (public_extensions) and add the extra
> in directory/default?
>
>
>
> Thanks, Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jerry.richards at teotech.com  Wed Jan 20 14:44:24 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 20 Jan 2010 14:44:24 -0800
Subject: [Freeswitch-users] Presence Not Working After Version 1.0.5pre9?
In-Reply-To: <mailman.5334.1264017641.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.5334.1264017641.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <C726F91949324BDFA34596256B6E102A@greyhawk.tonecommander.com>

Does anyone know why I do not see NOTIFY messages with presence status being
sent out from FS for two Bria softphones?  It used to work in my old version
1.0.5pre9.  I upgraded Freeswitch on Jan 12 and Jan 20 and edited in my XML
configurations, but I do not see the NOTIFY messages since version
1.0.5pre9.

I have mostly default configuration and I added the manage-presence=true
setting.  I posted a pastebin at http://pastebin.freeswitch.org/11867.

Can anyone tell why this isn't working?

Best Regards,
Jerry



From brian at freeswitch.org  Wed Jan 20 14:50:52 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Jan 2010 16:50:52 -0600
Subject: [Freeswitch-users] Presence Not Working After Version 1.0.5pre9?
In-Reply-To: <C726F91949324BDFA34596256B6E102A@greyhawk.tonecommander.com>
References: <mailman.5334.1264017641.2495.freeswitch-users@lists.freeswitch.org>
	<C726F91949324BDFA34596256B6E102A@greyhawk.tonecommander.com>
Message-ID: <4CFA4AF5-6E5A-40AB-816F-25EE1426F3A0@freeswitch.org>

PRE9 is no longer supported please use LATEST.  http://latest.freeswitch.org

/b

On Jan 20, 2010, at 4:44 PM, Jerry Richards wrote:

> Does anyone know why I do not see NOTIFY messages with presence status being
> sent out from FS for two Bria softphones?  It used to work in my old version
> 1.0.5pre9.  I upgraded Freeswitch on Jan 12 and Jan 20 and edited in my XML
> configurations, but I do not see the NOTIFY messages since version
> 1.0.5pre9.
> 
> I have mostly default configuration and I added the manage-presence=true
> setting.  I posted a pastebin at http://pastebin.freeswitch.org/11867.
> 
> Can anyone tell why this isn't working?
> 
> Best Regards,
> Jerry

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From dftoro at yahoo.com  Wed Jan 20 15:31:35 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Wed, 20 Jan 2010 15:31:35 -0800 (PST)
Subject: [Freeswitch-users] mutiple playback files (unescape_char) Windows
Message-ID: <24068.48012.qm@web33507.mail.mud.yahoo.com>

Greetings

I have a next section dial plan:

<action application="set" data="sound_prefix=$${base_dir}\sounds\es\co\callie\" />
<action application="set" data="playback_delimiter=!"/>

<action application="playback" data="misc\8000\serror.wav!misc\8000\provide_reference_number.wav!digits\8000\5.wav" />  
 
....

I have a problem using multiple playback files on Windows, the path  misc\8000\serror.wav is changed by misc\8000 serror.wav.  I check C code on switch_utils.c, cleanup_separated_string function call to unescape_char function which change \s by ' '. This is correct, but on Windows '\' is the path separator, so is not possible to use  '\s', '\n'... into path file. I think this is possible to fix it.

Thanks

Diego Toro
http://lacarretade.blogspot.com/


      


From anthony.minessale at gmail.com  Wed Jan 20 15:31:53 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 20 Jan 2010 17:31:53 -0600
Subject: [Freeswitch-users] Presence Not Working After Version 1.0.5pre9?
In-Reply-To: <C726F91949324BDFA34596256B6E102A@greyhawk.tonecommander.com>
References: <mailman.5334.1264017641.2495.freeswitch-users@lists.freeswitch.org>
	<C726F91949324BDFA34596256B6E102A@greyhawk.tonecommander.com>
Message-ID: <191c3a031001201531g7c78cb9fw1ed1a2ba07f5773c@mail.gmail.com>

try turning on sip debug and console loglevel debug

sofia loglevel all 9
console loglevel debug

Did you try manually running the same sql stmts from the sqlite3 app?
maybe you have something misconfigured.


On Wed, Jan 20, 2010 at 4:44 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> Does anyone know why I do not see NOTIFY messages with presence status
> being
> sent out from FS for two Bria softphones?  It used to work in my old
> version
> 1.0.5pre9.  I upgraded Freeswitch on Jan 12 and Jan 20 and edited in my XML
> configurations, but I do not see the NOTIFY messages since version
> 1.0.5pre9.
>
> I have mostly default configuration and I added the manage-presence=true
> setting.  I posted a pastebin at http://pastebin.freeswitch.org/11867.
>
> Can anyone tell why this isn't working?
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From tom at tomcarlson.com  Wed Jan 20 17:38:04 2010
From: tom at tomcarlson.com (Tom Carlson)
Date: Wed, 20 Jan 2010 17:38:04 -0800
Subject: [Freeswitch-users] luacurl vs io.popen("curl ...") vs
	api:executeString( "curl ..")
Message-ID: <21e9d36c1001201738i2b2f0106w175edf4d31494f69@mail.gmail.com>

I was hoping that those knowledgeable in these things could tell me,
strategically, which curl method would be most cpu friendly for me to use to
access my restful web service.


   1. Use freeSWITCH's mod_curl and issue an api:executeString( "curl ..")
   command
   2. Use the luacurl library from http://luaforge.net/projects/luacurl/
   3. Use linux's curl through an io.popen() command


I was intending to use the luacurl library, but it occured to me that
mod_curl might have been specially engineered to provide better efficiency.

Thanks.
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From rupa at rupa.com  Wed Jan 20 18:03:39 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 20 Jan 2010 20:03:39 -0600
Subject: [Freeswitch-users] luacurl vs io.popen("curl ...") vs
	api:executeString( "curl ..")
In-Reply-To: <21e9d36c1001201738i2b2f0106w175edf4d31494f69@mail.gmail.com>
References: <21e9d36c1001201738i2b2f0106w175edf4d31494f69@mail.gmail.com>
Message-ID: <b39273531001201803g7aae7227x19245abfde31f802@mail.gmail.com>

1 and 2  both use libcurl.  2 might be better for you IF it doesn't
introduce memory leaks or other instabilities.  mod_curl is just a simple
wrapper around libcurl so nothing special.

On Wed, Jan 20, 2010 at 7:38 PM, Tom Carlson <tom at tomcarlson.com> wrote:

> I was hoping that those knowledgeable in these things could tell me,
> strategically, which curl method would be most cpu friendly for me to use to
> access my restful web service.
>
>
>    1. Use freeSWITCH's mod_curl and issue an api:executeString( "curl ..")
>    command
>    2. Use the luacurl library from http://luaforge.net/projects/luacurl/
>    3. Use linux's curl through an io.popen() command
>
>
> I was intending to use the luacurl library, but it occured to me that
> mod_curl might have been specially engineered to provide better efficiency.
>
> Thanks.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From larclap at yahoo.com  Wed Jan 20 20:48:15 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 20 Jan 2010 20:48:15 -0800
Subject: [Freeswitch-users] Debug message on console?
In-Reply-To: <191c3a031001201428m72bcad24u6a29b0014d00f6a0@mail.gmail.com>
References: <00a801ca9a13$ea603470$bf209d50$@com>
	<191c3a031001201428m72bcad24u6a29b0014d00f6a0@mail.gmail.com>
Message-ID: <019501ca9a54$f23af180$d6b0d480$@com>

Thanks for changing the message, Anthony.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, January 20, 2010 2:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Debug message on console?

 

no it's overly chatty, i will move it up to debug level 10 so you won't see
it.



On Wed, Jan 20, 2010 at 3:02 PM, Lars Zeb <larclap at yahoo.com> wrote:

For about the last few weeks I've noticed the following message on the
console:

 

[DEBUG] sofia_reg.c:1815 adding X-Real-IP => 192.168.10.104 to xml_curl
request

 

I am sorry but cannot understand the code. The endpoint is a SNOM 320 at
7.3.14. It is registered for two extensions. The message is output twice
every 5 minutes.

 

Does this message indicate a problem? If so, how can I correct it?

 

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

FreeSWITCH v16385

 

Thanks, Lars


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From thangappan143 at gmail.com  Wed Jan 20 21:04:21 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Thu, 21 Jan 2010 10:34:21 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
Message-ID: <7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>

OpenZap is loading the ss7 signalling type. As per your concern openzap does
not know the details of the signalling then how it is loading the ss7_boost
libraries?

FreeSWITCH log:
-----------------------------
2010-01-21 10:10:46.707844 [INFO] zap_io.c:2374 Configured 30 channel(s)
2010-01-21 10:10:46.708600 [INFO] zap_io.c:2468 Loading SIG from
/usr/local/freeswitch/mod/ozmod_ss7_boost.so
2010-01-21 10:10:46.709031 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost'
2010-01-21 10:10:46.709466 [DEBUG] ss7_boost_client.c:124 Creating L=
127.0.0.65:53000 R=127.0.0.66:53000
2010-01-21 10:10:46.709834 [DEBUG] ss7_boost_client.c:124 Creating L=
127.0.0.65:53001 R=127.0.0.66:53001
2010-01-21 10:10:46.710424 [WARNING] ss7_boost_client.c:244 TX EVENT (P):
SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0

The signalling type might be anything but when I used the oz list command it
showed the span details. But I am unable to make a inbound and outbound
call.

Outbound call result:
============
>  originate openzap/smg_prid/a/9940464753 openzap/smg_prid/a/9843171457
-ERR NORMAL_CIRCUIT_CONGESTION

2010-01-21 10:26:14.304816 [CRIT] ozmod_ss7_boost.c:244 SPAN is not online.
freeswitch at internal> 2010-01-21 10:26:14.304816 [ERR] mod_openzap.c:1043 No
channels available
2010-01-21 10:26:14.304816 [ERR] switch_ivr_originate.c:1510 Cannot create
outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
2010-01-21 10:26:14.304816 [DEBUG] switch_ivr_originate.c:2138 Originate
Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]

Inbound call result:
-----------------------------

     I made incoming call for the dial plan which is specified in the
earlier post at that time it said the number is busy. We did the packet
capture using the following command.

wanpipemon -i w1g1 -pcap -pcap file isdn.pcap -port ISDN -full -systime -c
trd

Here I attached the pcap file for that.


Where I did mistake or Did I miss any thing to do?
Please help me.......


On Wed, Jan 20, 2010 at 7:40 PM, Thangappan.M <thangappan143 at gmail.com>wrote:

>
>       I noticed the 'oz list' output  in that span type is 'ss7 (boost)'.
> How can I change this to isdn?
>
>
>
> On Wed, Jan 20, 2010 at 12:43 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>> I found the error in it. The file name is used as openzap.conf.xml (
>> smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
>> PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
>> openzap.conf.wiki.xml  file.
>>
>> Now the another problem is raised here.
>> When I was using oz list command , the details of the smg_prid shown. When
>> I was using 'oz dump smg_prid' command it shows all the channels' details.
>> But all the channels' states are DOWN. why? How can I make it the states to
>> UP?
>>
>> When I was making the call , the number is busy message was get. The call
>> was not  at all landed to the freeswitch.
>>
>> Dial plan Example:
>> -------------------------------
>>  <extension name="PRI_Testing">
>>         <condition field="destination_number" expression="^39114600$">
>>                 <action application="playback"
>> data="ivr-welcome_to_freeswitch"/>
>>         </condition>
>>   </extension>
>>
>> Please help me...........
>>
>> *Output Reference:*
>> http://pastebin.org/79074
>>
>>
>> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>
>>> Dear all,
>>>
>>>          I have successfully configured wanpipe with freeswitch. When I
>>> was the running wancfg_fs script the following files openzap.conf ,
>>> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
>>> are created.
>>>
>>>          I started the wanrouter command then executed the freeswitch.
>>>          When I was executing freeswitch mod_openzap.c said the error as
>>> "Error  for finding the span id. name:PRI_1".
>>>          But in the openzap.conf and openzap.conf.xml files the span name
>>> is smg_prid.
>>>
>>>          Why the freeswitch is referring the span name as PRI_1 ?
>>>          Whether this has to configured in anywhere?
>>>
>>>          In the freeswitch CLI using oz command I tried to dump the PRI_1
>>> span id but it said te error  as "PRI_1 is not found".  When I was trying
>>> the command 'oz dump smg_prid'  all the channel states and details shown.
>>>
>>>         It seems that smg_prid span configured in openzap perfectly (Its
>>> my assumption). Then Why freeswitch is referring the span name as PRI_1.
>>>
>>> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>>>
>>> Could anyone please help me?
>>>
>>> REFERENCE:
>>>
>>> openzap.conf
>>> [span wanpipe smg_prid]
>>> name => smg_prid
>>> trunk_type =>e1
>>> b-channel => 1:1-15
>>> b-channel => 1:17-31
>>>
>>>
>>> openzap.conf.xml
>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>   <settings>
>>>     <param name="debug" value="0"/>
>>>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>     <!--<param name="enable-analog-option" value="3-way"/>-->
>>>   </settings>
>>>   <boost_spans>
>>>         <span id="smg_prid">
>>>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>         <param name="dialplan" value="XML"/>
>>>         <param name="context" value="default"/>
>>>          <!-- regex to stop dialing when it matches -->
>>>         <!--<param name="dial-regex" value="5555"/>-->
>>>         <!-- regex to stop dialing when it does not match -->
>>>         <!--<param name="fail-dial-regex" value="^5"/>-->
>>>         </span>
>>>  </boost_spans>
>>>  <analog_spans>
>>>  </analog_spans>
>>> </configuration>
>>>
>>>
>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Thangappan.M
>>>
>>
>>
>>
>> --
>> Regards,
>> Thangappan.M
>>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From larclap at yahoo.com  Wed Jan 20 21:17:57 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 20 Jan 2010 21:17:57 -0800
Subject: [Freeswitch-users] Can't register Polycom
Message-ID: <01a301ca9a59$17e45fd0$47ad1f70$@com>

I am having trouble registering a Polycom 550. From the siptrace it looks
like there is no username coming from the Polycom. I configured the Polycom
via the web interface. I have inserted "1008" in the "Display Name",
"Address" and "Auth User ID" fields.

 

In conf/dialplan/default/1008.xml  the first line is <user id="1008"
cidr="192.168.10.105/24">.

 

What am I missing?

 

Thanks, Lars

 

http://pastebin.freeswitch.org/11875

 

 

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From irmatov at gmail.com  Wed Jan 20 21:59:38 2010
From: irmatov at gmail.com (Timur Irmatov)
Date: Thu, 21 Jan 2010 10:59:38 +0500
Subject: [Freeswitch-users] mod_erlang_event: disconnects
Message-ID: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>

Hi!


We have build a small and simple call center using FreeSWITCH and
mod_erlang_event. My erlang process keeps track of available agents
and routes incoming calls to them. Calls are sent to my application
via:

<action application="erlang" data="switch_event:! erlswitch at localhost" />

switch_event is a registered process, which spawns a new process for
each incoming call and returns new pid when it receives {get_pid,
UUID, Ref, From} message from FreeSWITCH.

The problem is, that pretty frequently processes which handle incoming
calls receive messages like {'EXIT', <5406.48.0>, noconnection} from
FreeSWITCH. As I understand from googling, this happens when remote C
node disconnects (and I see TCP connections from FreeSWITCH to epmd
daemon being torn down and reestablished). FreeSWITCH drops calls at
that moment.

Have anyone seen this? Is there any fix/ advice?

My system is Debian Lenny (5.0.3), 64-bit system, erlang installed
from Debian packages, no backports.


-- 
Timur Irmatov, xmpp:irmatov at jabber.ru


From mike at jerris.com  Wed Jan 20 22:48:42 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 21 Jan 2010 01:48:42 -0500
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
In-Reply-To: <20100120180720.GH4767@tamay-dogan.net>
References: <20100119103139.GR4767@tamay-dogan.net>
	<2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>
	<20100120180720.GH4767@tamay-dogan.net>
Message-ID: <BB497725-7989-44D6-8321-27B424861945@jerris.com>

We have a hudson instance doing builds, we just need boxes that can be the build drones.  And someone with a little time to set it up and bandwidth to handle it.  we build every 30 min that there is a change to svn, which is depending how long the builds take, usually 20+ times a day and upload the build results to the hudson server.

Mike

On Jan 20, 2010, at 1:07 PM, Michelle Konzack wrote:

> Hello,
> 
> Am 2010-01-20 01:28:40, schrieb Michael Jerris:
>> There is a debian dir in tree for our packages.  I suspect that our
>> packages are quite far from meeting the requirements of pretty much
>> any distro so for now, we will at least have packages for the next
>> release available soon after release.  We don't yet have a box for
>> debian instances for the build farm so we do not build the svn
>> snapshot pacakges for any deb distros.
> 
> Hmmm, currently I have only one fixed IP and some VServers and  PBuilder
> runing on i386 and ARM (only a small Ti Sitara AM3517 with 256 MB memory
> plus SATA drive)...
> 
> Also I am trying to relocate back to Germany...
> Maybe I could do the Job for Debian (i386 ARMEL) and Ubuntu (i386)
> 
> My Website is currently a backup fro 2008/12 and 2009/03 because  I  was
> offine since 2009-07-23 du to my fuckingbusiness partner...
> 
> Maybe it work:
>    http://www.debian.tamay-dogan.net/
> 
> I will try to reinstall the PBuilder interface which  allow  uploads  of
> sources/configs and autobuilding.  Also  I  like  to  include  an  auto-
> checkout from "svn" and "git" so, the author of the software should give
> a signal to my interface and the build is done automaticaly.
> 
> Also I am Package Maintainer of some Debian packages...
> 
> Thanks, Greetings and nice Day/Evening
>    Michelle Konzack
>    Systemadministrator
>    Electronic Engineer
>    Tamay Dogan Network
>    Debian GNU/Linux Consultant
> 
> -- 
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
> ##################### Debian GNU/Linux Consultant #####################
> <http://www.tamay-dogan.net/>                 Michelle Konzack
> <http://www.can4linux.org/>                   Apt. 917
> <http://www.flexray4linux.org/>               50, rue de Soultz
> Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
> IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
> ICQ    #328449886                             Tel. FR: +33  6  61925193
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Wed Jan 20 23:12:46 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 21 Jan 2010 02:12:46 -0500
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <24068.48012.qm@web33507.mail.mud.yahoo.com>
References: <24068.48012.qm@web33507.mail.mud.yahoo.com>
Message-ID: <DA570C12-70CA-43C0-B014-4B3CCBA7A7EF@jerris.com>


On Jan 20, 2010, at 6:31 PM, Diego Toro wrote:

> Greetings
> 
> I have a next section dial plan:
> 
> <action application="set" data="sound_prefix=$${base_dir}\sounds\es\co\callie\" />
> <action application="set" data="playback_delimiter=!"/>
> 
> <action application="playback" data="misc\8000\serror.wav!misc\8000\provide_reference_number.wav!digits\8000\5.wav" />  
> 
> 

<action application="playback" data="misc\\8000\\serror.wav!misc\\8000\\provide_reference_number.wav!digits\\8000\\5.wav" />  

??

> ....
> 
> I have a problem using multiple playback files on Windows, the path  misc\8000\serror.wav is changed by misc\8000 serror.wav.  I check C code on switch_utils.c, cleanup_separated_string function call to unescape_char function which change \s by ' '. This is correct, but on Windows '\' is the path separator, so is not possible to use  '\s', '\n'... into path file. I think this is possible to fix it.

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From msc at freeswitch.org  Thu Jan 21 00:25:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Jan 2010 00:25:32 -0800
Subject: [Freeswitch-users] Svar: Re: Home setup with home company
In-Reply-To: <4B56ECD7020000E1000003F1@mail.fribert.dk>
References: <4B56ECD7020000E1000003F1@mail.fribert.dk>
Message-ID: <87f2f3b91001210025t38bd679cu647b4935bef509c@mail.gmail.com>

On Wed, Jan 20, 2010 at 2:45 AM, mailinglist <mailinglist at fribert.dk> wrote:

> Hi Michael
>
> It's running on pfsense, so it's kinda locked to the version it currently
> is.
> Looks very nice though.
> Looking beyond that, is the action / anti-action list corrent?
>

I would say that you need to add an anti-action under the day of week check
and go to vm if it does not match. Right now if the DOW is 0 or 6 then the
entire extension will "fail" and the dialplan will just move on. Remember
that if any conditions fail then the entire thing extension "fails" unless
you are doing interesting things with the break= parameter. See the dialplan
page on the wiki for examples of how to use break in your conditions.
-MC


> Best regards
> Fribse
>
>
> >>> Michael Collins 20-01-10 1:53 >>>
>
>
> On Tue, Jan 19, 2010 at 9:02 AM, mailinglist <mailinglist at fribert.dk>wrote:
>
>>  I have a very small one man constultancy company that has the occasional
>> call, unfortunately we are getting more spam calls after hours than real
>> calls during work hours, so I would like to set up a TOD system.
>>
>> First step for me is just playing with the TOD example, I've gotten this
>> far:
>>
>> <extension name="CompanyTOD" continue="true">
>>    <condition field="destination_number" expression="^(8203)$"/>
>>       <!-- man strftime - M-F, 9AM to 5PM -->
>>       <condition field="${strftime(%w)}" expression="^([1-5])$"/>
>>       <condition field="${strftime(%H%M)}"
>> expression="^((09|1[0-6])[0-5][0-9]|1700)$">
>>          <action application="set" data="call_timeout=45"/>
>>          <action application="set" data="domain_name=10.11.12.25"/>
>>          <action application="set" data="continue_on_fail=true"/>
>>          <action application="set" data="hangup_after_bridge=true"/>
>>          <action application="bridge" data="
>> group/company@${domain_name}"/ <group/company@$%7Bdomain_name%7D%22/>>
>>          <action application="answer" data=""/>
>>          <action application="sleep" data="1000"/>
>>          <action application="voicemail" data="default 10.11.12.25
>> 10199"/>
>>          <anti-action application="voicemail" data="default 10.11.12.25
>> 10199"/>
>>       </condition>
>> </extension>
>> My idea with this, was that in the time 9-17 mon-fri, it rings, the rest
>> of the time it's sent directly to the voicemail.
>> I would of course like to have it not take messages outside work hours,
>> but that's just refining :-)
>>
>> But it picks up the call, and then nothing...
>>
>>
> We have a much cleaner way of doing TOD and DOW handling. You'll need to
> get to the latest FreeSWITCH version. Look in conf/dialplan/default.xml for
> this example:
>
>     <!-- Example of doing things based on time of day.
>
>      year = 4 digit year. Example year="2009"
>      yday = 1-365
>      mon = 1-12
>      mday = 1-31
>      week = 1-52
>      mweek= 1-6
>      wday = 1-7
>      hour = 0-23
>      minute = 0-59
>      minute-of-day = 1-1440
>
>      Example:
>      <condition minute-of-day="540-1080"> (9am to 6pm EVERY day)
>      do something ...
>      </condition>
>     -->
>     <extension name="tod_example" continue="true">
>       <condition wday="2-6" hour="9-18">
>         <action application="set" data="open=true"/>
>       </condition>
>     </extension>
>
>
> Use that condition instead of the two conditions you're now using and see
> if you have better success.
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From linux4michelle at tamay-dogan.net  Thu Jan 21 01:13:46 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Thu, 21 Jan 2010 10:13:46 +0100
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
In-Reply-To: <BB497725-7989-44D6-8321-27B424861945@jerris.com>
References: <20100119103139.GR4767@tamay-dogan.net>
	<2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>
	<20100120180720.GH4767@tamay-dogan.net>
	<BB497725-7989-44D6-8321-27B424861945@jerris.com>
Message-ID: <20100121091346.GF4767@tamay-dogan.net>

Hello,

Am 2010-01-21 01:48:42, schrieb Michael Jerris:
> We have a hudson instance doing builds, we just need boxes that can be
> the build drones.  And someone with a little time to set it up and
> bandwidth to handle it.  we build every 30 min that there is a change
> to svn, which is depending how long the builds take, usually 20+ times
> a day and upload the build results to the hudson server.

If I a installed in Frankfurt/Germany I will  try  to  get  as  fast  as
possibel an E3 (34Mbit) or FTTB (100Mbit) and  bandwidth  should  be  no
problem.

How many MByte is one SVN checkout?

1 Mbit bandwitdh (~320 GiB/month) cost me arround 50 Euro additional  to
the base-price.  I do bot think it wil kill my finacial resources.

My current bandwidth usage is nearly 400 kBit  because  I  have  only  a
1 Mbit access.  :-(  and I can not get a 100/50 Mbit FTTH where  I  live
otherwise you could have one of my spare Sun Fire X4100M2 (3 x 76  GByte
SAS in Raid-1 with Hotfix) as  Build-Daemon

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From jingwei.yang at gmail.com  Thu Jan 21 01:22:12 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 21 Jan 2010 17:22:12 +0800
Subject: [Freeswitch-users]  Is this queue flow correct?
Message-ID: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>

Hi All,

Please advise whether the following flow makes sense.

1. Client A calls in and parked in Queue 1
2. Originate calls to several consumers simultaneously and park them in
Queue 2
3. Intercept A's call to the first consumer of Queue 2

Basically I want A's call picked up by the first among many consumers with
no errors. Please let me know whether I'm on the right track.

Thanks and best regards,
-Jingwei
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From a.afzali2003 at gmail.com  Thu Jan 21 01:44:27 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Thu, 21 Jan 2010 13:14:27 +0330
Subject: [Freeswitch-users] Managing Presence on Gateways
Message-ID: <debff5a81001210144n4b32c19cic36415155ee52231@mail.gmail.com>

Hi Guys,

In the external profile (as in the internal) there is an option to enable
presence functionality (with setting it to passive). My question is how does
it mean presence functionality for a gateway which interfaces home domain to
another one? Does it mean that the gateway could subscribe itself for some
presence information in that domain in behaves of local users and relays
them?

Appreciate all comments,
-- afshin
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From oscav at hotmail.fr  Thu Jan 21 01:47:53 2010
From: oscav at hotmail.fr (Oscav)
Date: Thu, 21 Jan 2010 01:47:53 -0800 (PST)
Subject: [Freeswitch-users]  All channels are frozen while receiving DTMF
Message-ID: <27255181.post@talk.nabble.com>


Hi,

I'm running a script that gets some DTMF from caller. I found that when a
caller is entering DTMF , all the others channels are frozen until all the
DTMF  are received. In the logs I see that each DTMF takes 1 second. It
means that if the caller enters 10 digits then all the other running scripts
are paused for 10 seconds. The problem is exponential with traffic load.

Anyone have an idea ??

Thanks
-- 
View this message in context: http://old.nabble.com/All-channels-are-frozen-while-receiving-DTMF-tp27255181p27255181.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From oscav at hotmail.fr  Thu Jan 21 01:49:48 2010
From: oscav at hotmail.fr (Oscav)
Date: Thu, 21 Jan 2010 01:49:48 -0800 (PST)
Subject: [Freeswitch-users] Failed to connect to a SKYPE API
In-Reply-To: <27078464.post@talk.nabble.com>
References: <27062783.post@talk.nabble.com> <27078464.post@talk.nabble.com>
Message-ID: <27255195.post@talk.nabble.com>


I solved the problem. It was due to the logon session.


Oscav wrote:
> 
> Im' running FS on windows server 2003 64bits
> 
> 
> Oscav wrote:
>> 
>> Hi,
>> 
>> I'm trying to use to SkypeIAX. When I load the mod_skypiax, I got the
>> following error :
>> 
>>  Failed to connect to a SKYPE API for interface_id=1, no SKYPE client
>> running, please (re)start Skype client. Skypiax exiting
>> 
>> Skype is running with the correct account and skypiax.conf use the same
>> account. I was expecting a permission request from the Skype user but
>> nothing happens.
>> 
>> Somebody knows how I can solve this ??
>> 
>> Many thanks.
>> 
> 
> 

-- 
View this message in context: http://old.nabble.com/Failed-to-connect-to-a-SKYPE-API-tp27062783p27255195.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From a.alalousi at gmail.com  Thu Jan 21 02:10:59 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Thu, 21 Jan 2010 10:10:59 +0000
Subject: [Freeswitch-users] G729 coded issues
In-Reply-To: <537338B1-B582-463E-8EFE-7BBED8165D2B@freeswitch.org>
References: <cd93b3701001190831u4147214cn9832c6f469c6d17e@mail.gmail.com>
	<6D67C7A9-42CD-4F5F-AD18-E9568098FB71@freeswitch.org>
	<4B55E794.6020909@coppice.org>
	<cd93b3701001200514p4acc5a5bk26a463164f841ad2@mail.gmail.com>
	<537338B1-B582-463E-8EFE-7BBED8165D2B@freeswitch.org>
Message-ID: <cd93b3701001210210v2ddba8afwc6a310540d2e5180@mail.gmail.com>

Hi Brian,

All for it, so ye, let me have the beta.

The only reason I went Howler is because of a pressing need.

Moreover, I'm willing to put mine and my teams resources into the project. I
and others cut code in C++/C/Perl/....etc. As an organisation, we are
actively involved in telecoms consultancy and software development, and I
really would like to put our backs into pushing the case for FS wherever
possible.

Ping me offline on the subscription e-mail for this account, and let's
exchange details if there is interest your side.

Support ? we are all sold on FS down here believe me, so you can count on
it.

Regards,

Ahmed.


2010/1/20 Brian West <brian at freeswitch.org>

> Just an FYI.  Howler doesn't support the FreeSWITCH project in any way,
> shape or form.  They do not donate any proceeds or help the project at all.
>  That said.  We have our officially supported G729 coming out soon that will
> support the project.  I have it in beta if anyone is really interested in
> testing it please feel free to email me offlist.  We are currently working
> out how we want to package the lib, binary and module to make installation
> easy.
>
> Thanks,
> /b
>
> On Jan 20, 2010, at 7:14 AM, Ahmed Naji wrote:
>
> > Howler
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From nicolas at medularis.com  Thu Jan 21 03:35:07 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Thu, 21 Jan 2010 08:35:07 -0300
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <5d2828f1001201406p4154b98ald1af1c5c25f59337@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
	<e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
	<5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>
	<5d2828f1001201406p4154b98ald1af1c5c25f59337@mail.gmail.com>
Message-ID: <1b46b4e81001210335l42baef16r8a2952aa5b92f6e6@mail.gmail.com>

On Wed, Jan 20, 2010 at 7:06 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> So, I've been reading about early media in the wiki, and have made a little
> progress, which leads to more questions.
>
> I understand now why a call is considered connected before one person has
> picked up the phone. I am also able to get my script to wait for the phone
> to be picked up, by setting the ignore_early_media variable when starting a
> new session, like this:
>
> customerSession =
> freeswitch.Session("{ignore_early_media=true}sofia/gateway/example.com/"
> .. customerPhoneNumber)
>
>
> After that line, the script waits for the other phone to be picked up.
>
> However, now I wonder what to do with calls that don't complete, get busy
> signals, etc.
>
> What do people do in this case? The only related example I can find on the
> web is for a javascript dialer, which doesn't address any of these cases.
>


I guess it depends on what you want to do. For example I have a lua script
very similar to what you describe, although there is no confirmation
involved. Depending on the hangup cause the session gets, it might try
redialing with a different gateway, try again or just hangup.

Take a look here http://wiki.freeswitch.org/wiki/Hangup_causes to see what
each hangup cause means. You don't need to have a special case for all of
them, only the ones you are interested in.

Here's an example in code which retries a call depending on the hangup
cause. It retries max_retries1 times and alternates between 2 different
gateways:

session1 = null;
max_retries1 = 3;
retries = 0;
ostr = "";
repeat
        retries = retries + 1;
        if (retries % 2) then ostr = originate_str1;
        else ostr = originate_str12; end
        freeswitch.consoleLog("notice", "*********** Dialing Leg1: " .. ostr
.. " - Try: "..retries.." ***********\n");
        session1 = freeswitch.Session(ostr);
        local hcause = session1:hangupCause();
        freeswitch.consoleLog("notice", "*********** Leg1: " .. hcause .. "
- Try: "..retries.." ***********\n");
until not ((hcause == 'NO_ROUTE_DESTINATION' or hcause ==
'RECOVERY_ON_TIMER_EXPIRE' or hcause == 'INCOMPATIBLE_DESTINATION' or hcause
== 'CALL_REJECTED' or hcause == 'NORMAL_TEMPORARY_FAILURE') and (retries <
max_retriesl1))


Note: originate_str1 and originate_str2 are two different dial strings for 2
different gateways.



>
> Early Media: http://wiki.freeswitch.org/wiki/Early_media
> ignore_early_media variable:
> http://wiki.freeswitch.org/wiki/Variable_ignore_early_media
> <http://wiki.freeswitch.org/wiki/Variable_ignore_early_media>javascript
> dialer: http://alexn.org/docs/dialer.html
>
>
> <http://alexn.org/docs/dialer.html>Mike van Lammeren
>
>
> On Wed, Jan 20, 2010 at 4:45 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:
>
>> Hi Rob!
>>
>> Unfortunately, I have the next few weeks to complete this part of the
>> project. Without a stable release of 1.0.5, I have to stick with 1.0.4. I'm
>> sure that 1.0.4 can detect and report when a phone is picked up. It's just
>> that I can't figure out how to get that information! Either that, or I have
>> something mis-configured.
>>
>> Mike van Lammeren
>>
>>
>> On Wed, Jan 20, 2010 at 4:32 PM, Rob Forman <rob4manhere at gmail.com>wrote:
>>
>>> Hi Mike,
>>>
>>> I don't think v1.0.4 is supported any longer.   You'll have better luck
>>> getting assistance by upgrading to trunk or the latest tar and reporting
>>> back.
>>>
>>> Good luck!
>>> Rob
>>>
>>> On Wed, Jan 20, 2010 at 3:18 PM, Mike van Lammeren <
>>> mike at van.lammeren.net> wrote:
>>>
>>>>  Hello!
>>>>
>>>> Some day, I'll probably laugh at asking this question, but today I can't
>>>> figure it out.
>>>>
>>>> I've written a Lua script that listens for a call, then dials a phone
>>>> number to a second person. It plays a message, then prompts the second
>>>> person to hit pound to connect. If the second person hits pound, then it
>>>> bridges the two calls together.
>>>>
>>>> All that works great, but I can't figure out how to get the session for
>>>> the second person to wait until that person answers.
>>>>
>>>> I'm using FreeSWITCH 1.0.4, and although there is a *getState* function
>>>> documented in the wiki, it doesn't seem to exist for me.
>>>>
>>>> Any help would be appreciated!
>>>>
>>>>
>>>> Mike van Lammeren
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Thu Jan 21 03:41:40 2010
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 21 Jan 2010 19:41:40 +0800
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
Message-ID: <23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>

what no errors mean? how do you originate calls to consumers? I don't
understand your scenario.

2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
> Hi All,
>
> Please advise whether the following flow makes sense.
>
> 1. Client A calls in and parked in Queue 1
> 2. Originate calls to several consumers simultaneously and park them in
> Queue 2
> 3. Intercept A's call to the first consumer of Queue 2
>
> Basically I want A's call picked up by the first among many consumers with
> no errors. Please let me know whether I'm on the right track.
>
> Thanks and best regards,
> -Jingwei
>


From mikael at bjerkeland.com  Thu Jan 21 04:20:28 2010
From: mikael at bjerkeland.com (Mikael Bjerkeland)
Date: Thu, 21 Jan 2010 13:20:28 +0100
Subject: [Freeswitch-users] Sip video intercom
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9083ACB@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E9083ACB@prato.activenetwerx.local>
Message-ID: <cb79a141001210420o196ae36ck615e7a5c69791bb@mail.gmail.com>

http://www.2n.cz/products/door-lift-phones/door-entry-systems/ip-communication.html

2010/1/20 Joseph L. Casale <jcasale at activenetwerx.com>

> I need to get an intercom integrated into the voip system of a highend
> home.
> That being said, I am looking for a nice looking discrete panel to mount
> outside by the front door. Anyone have any experience with these and know
> of
> a model they recommend?
>
> Thanks!
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dftoro at yahoo.com  Thu Jan 21 05:02:40 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Thu, 21 Jan 2010 05:02:40 -0800 (PST)
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <DA570C12-70CA-43C0-B014-4B3CCBA7A7EF@jerris.com>
Message-ID: <37012.20543.qm@web33507.mail.mud.yahoo.com>

Hi MikeJ, using '\\' the behavior is the same, '\\s' is replaced by ' '.

<action application="playback" data="misc\\8000\\serror.wav!misc\\8000\\provide_reference_number.wav!digits\\8000\\5.wav"/>  

Console output error is:

[ERR] mod_sndfile.c:194 Error Opening File [d:\fs\fs_trunk_20100118\Debug\sounds\es\co\callie\misc\8000 error.wav]


S.O.: Windows 7
FreeSwitch: Trunk (svn latest version)

Thank you


Diego Toro
http://lacarretade.blogspot.com/


--- On Thu, 1/21/10, Michael Jerris <mike at jerris.com> wrote:

> From: Michael Jerris <mike at jerris.com>
> Subject: Re: [Freeswitch-users] mutiple playback files (unescape_char) Windows
> To: freeswitch-users at lists.freeswitch.org
> Date: Thursday, January 21, 2010, 2:12 AM
> 
> On Jan 20, 2010, at 6:31 PM, Diego Toro
> wrote:
> Greetings
> 
> I have a next section dial plan:
> 
> <action application="set"
> data="sound_prefix=$${base_dir}\sounds\es\co\callie\"
> />
> <action application="set"
> data="playback_delimiter=!"/>
> 
> <action application="playback"
> data="misc\8000\serror.wav!misc\8000\provide_reference_number.wav!digits\8000\5.wav"
> /> ?
> 
> 
> 
> <action
> application="playback"
> data="misc\\8000\\serror.wav!misc\\8000\\provide_reference_number.wav!digits\\8000\\5.wav"
> /> ?
> ??
> ....
> 
> I have a problem using multiple playback files on Windows,
> the path ?misc\8000\serror.wav is changed by
> misc\8000 serror.wav. ?I check C code on
> switch_utils.c, cleanup_separated_string function call to
> unescape_char function which change \s by ' '.
> This is correct, but on Windows '\' is the path
> separator, so is not possible to use ?'\s',
> '\n'... into path file. I think this is possible
> to fix it.
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From andrew at hijacked.us  Thu Jan 21 05:42:41 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 21 Jan 2010 08:42:41 -0500
Subject: [Freeswitch-users] mod_erlang_event: disconnects
In-Reply-To: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>
References: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>
Message-ID: <20100121134241.GD1036@hijacked.us>

On Thu, Jan 21, 2010 at 10:59:38AM +0500, Timur Irmatov wrote:
> We have build a small and simple call center using FreeSWITCH and
> mod_erlang_event. My erlang process keeps track of available agents
> and routes incoming calls to them. Calls are sent to my application
> via:
> 
> <action application="erlang" data="switch_event:! erlswitch at localhost" />
> 
> switch_event is a registered process, which spawns a new process for
> each incoming call and returns new pid when it receives {get_pid,
> UUID, Ref, From} message from FreeSWITCH.

That all looks fine.
> 
> The problem is, that pretty frequently processes which handle incoming
> calls receive messages like {'EXIT', <5406.48.0>, noconnection} from
> FreeSWITCH. As I understand from googling, this happens when remote C
> node disconnects (and I see TCP connections from FreeSWITCH to epmd
> daemon being torn down and reestablished). FreeSWITCH drops calls at
> that moment.

Does it drop ALL calls being handled in erlang, or just that one?
> 
> Have anyone seen this? Is there any fix/ advice?

I haven't seen this before, how many calls are involved? I'm willing to
help you troubleshoot though. Is there anything relevant in the logs
(even at DEBUG)?

> 
> My system is Debian Lenny (5.0.3), 64-bit system, erlang installed
> from Debian packages, no backports.
> 
What OTP release does that equate to, R12 or R13?

Andrew (mod_erlang_event author)


From andrew at hijacked.us  Thu Jan 21 05:44:51 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 21 Jan 2010 08:44:51 -0500
Subject: [Freeswitch-users] mod_erlang_event: disconnects
In-Reply-To: <20100121134241.GD1036@hijacked.us>
References: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>
	<20100121134241.GD1036@hijacked.us>
Message-ID: <20100121134451.GE1036@hijacked.us>

Also, what FS version are you running?

Andrew


From Russell.Mosemann at cune.org  Thu Jan 21 06:28:17 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 21 Jan 2010 14:28:17 -0000
Subject: [Freeswitch-users] Can't register Polycom
In-Reply-To: <01a301ca9a59$17e45fd0$47ad1f70$@com>
Message-ID: <20100121142817.CDAB42A865A@cuneorg-email.cune.pri>

Lars Zeb <larclap at yahoo.com> said:

> cidr="192.168.10.105/24"

The IP address and mask don't make sense. Either it needs to be
192.168.10.0/24 or 192.168.10.105.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From Claudio.Cavalera at italtel.it  Thu Jan 21 07:17:49 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Thu, 21 Jan 2010 16:17:49 +0100
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <191c3a031001140837m7cffcdd5w71886d6c8ba1dafe@mail.gmail.com>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F36308@BESONE.corp.dom>

Anthony,
I've noticed that you did not mention javascript, is it the exception?
Is the C++ file with the api
src/include/switch_swigable_cpp.h ?
Thx,
Claudio
 


________________________________

	From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
	Sent: Thursday, January 14, 2010 5:38 PM
	To: freeswitch-users at lists.freeswitch.org
	Subject: Re: [Freeswitch-users] playing with sessions in lua
	
	
	Don't forget that lua,perl,python,managed,java all share the
same exact C++ source file with swig so the same exact api applies to
all of the above.
	
	


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From brian at freeswitch.org  Thu Jan 21 07:26:09 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 21 Jan 2010 09:26:09 -0600
Subject: [Freeswitch-users] Can't register Polycom
In-Reply-To: <20100121142817.CDAB42A865A@cuneorg-email.cune.pri>
References: <20100121142817.CDAB42A865A@cuneorg-email.cune.pri>
Message-ID: <77267B4B-6A0C-422F-ACA5-34240FCCDD37@freeswitch.org>

It will still take that mask and make it work. 

/b

On Jan 21, 2010, at 8:28 AM, <Russell.Mosemann at cune.org> <Russell.Mosemann at cune.org> wrote:

> Lars Zeb <larclap at yahoo.com> said:
> 
>> cidr="192.168.10.105/24"
> 
> The IP address and mask don't make sense. Either it needs to be
> 192.168.10.0/24 or 192.168.10.105.
> 
> -- 
> Russell Mosemann

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From jingwei.yang at gmail.com  Thu Jan 21 07:39:07 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 21 Jan 2010 23:39:07 +0800
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
Message-ID: <13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>

Sorry about the confusion, I'm just trying to think over all the abnormal
situations before the implementation and hope the flow has no design flaws.

Client A is parked in Queue 1, multiple consumers will be ringed to answer
him. And once the first one is connected, all the rest will hang up. This is
the targeted function.

To achieve this, I'm considering to originate a call to each consumer and
put the calls in Queue 2. Then intercept client A to the first element of
Queue 2.

I'm not sure if it's the usual or the best way. If you feel not, please
don't hesitate to correct me. Any thoughts are warmly appreciated.

On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com> wrote:

> what no errors mean? how do you originate calls to consumers? I don't
> understand your scenario.
>
> 2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
> > Hi All,
> >
> > Please advise whether the following flow makes sense.
> >
> > 1. Client A calls in and parked in Queue 1
> > 2. Originate calls to several consumers simultaneously and park them in
> > Queue 2
> > 3. Intercept A's call to the first consumer of Queue 2
> >
> > Basically I want A's call picked up by the first among many consumers
> with
> > no errors. Please let me know whether I'm on the right track.
> >
> > Thanks and best regards,
> > -Jingwei
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fdelawarde at wirelessmundi.com  Thu Jan 21 08:50:20 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Thu, 21 Jan 2010 17:50:20 +0100
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
Message-ID: <1264092620.14614.73.camel@luna.tc.commsmundi.com>

Why do you need 2 fifos? You could have callback agents connected to the
fifo and send incoming calls there, mod_fifo will do the rest.

To configure agents for callback:
http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback

To place a call into a fifo:
http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue

Fran?ois.

On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
> Sorry about the confusion, I'm just trying to think over all the
> abnormal situations before the implementation and hope the flow has no
> design flaws.
> 
> Client A is parked in Queue 1, multiple consumers will be ringed to
> answer him. And once the first one is connected, all the rest will
> hang up. This is the targeted function. 
> 
> To achieve this, I'm considering to originate a call to each consumer
> and put the calls in Queue 2. Then intercept client A to the first
> element of Queue 2. 
> 
> I'm not sure if it's the usual or the best way. If you feel not,
> please don't hesitate to correct me. Any thoughts are warmly
> appreciated.
> 
> On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com> wrote:
>         what no errors mean? how do you originate calls to consumers?
>         I don't
>         understand your scenario.
>         
>         2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
>         
>         > Hi All,
>         >
>         > Please advise whether the following flow makes sense.
>         >
>         > 1. Client A calls in and parked in Queue 1
>         > 2. Originate calls to several consumers simultaneously and
>         park them in
>         > Queue 2
>         > 3. Intercept A's call to the first consumer of Queue 2
>         >
>         > Basically I want A's call picked up by the first among many
>         consumers with
>         > no errors. Please let me know whether I'm on the right
>         track.
>         >
>         > Thanks and best regards,
>         > -Jingwei
>         >
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From larclap at yahoo.com  Thu Jan 21 09:01:25 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 21 Jan 2010 09:01:25 -0800
Subject: [Freeswitch-users] Can't register Polycom
In-Reply-To: <77267B4B-6A0C-422F-ACA5-34240FCCDD37@freeswitch.org>
References: <20100121142817.CDAB42A865A@cuneorg-email.cune.pri>
	<77267B4B-6A0C-422F-ACA5-34240FCCDD37@freeswitch.org>
Message-ID: <014501ca9abb$5ddae970$1990bc50$@com>

I removed the /24 from the cidr and still 403 Forbidden.

 

I configured this extension on the Polycom on Line 4. I removed this
definition and put it on Line 3. It then registered. 

 

There must be something wrong with the phone.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, January 21, 2010 7:26 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't register Polycom

 

It will still take that mask and make it work. 

 

/b

 

On Jan 21, 2010, at 8:28 AM, <Russell.Mosemann at cune.org>
<Russell.Mosemann at cune.org> wrote:





Lars Zeb <larclap at yahoo.com> said:




cidr="192.168.10.105/24"


The IP address and mask don't make sense. Either it needs to be
192.168.10.0/24 or 192.168.10.105.

-- 
Russell Mosemann



 

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From mike at van.lammeren.net  Thu Jan 21 09:03:08 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 21 Jan 2010 12:03:08 -0500
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <1b46b4e81001210335l42baef16r8a2952aa5b92f6e6@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
	<e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
	<5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>
	<5d2828f1001201406p4154b98ald1af1c5c25f59337@mail.gmail.com>
	<1b46b4e81001210335l42baef16r8a2952aa5b92f6e6@mail.gmail.com>
Message-ID: <5d2828f1001210903r2e5ec264q44945e17b48dda50@mail.gmail.com>

Awesome example code, Nicolas! Thanks!

On Thu, Jan 21, 2010 at 6:35 AM, Nicolas Brenner <nicolas at medularis.com>wrote:

>
> On Wed, Jan 20, 2010 at 7:06 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:
>
>> So, I've been reading about early media in the wiki, and have made a
>> little progress, which leads to more questions.
>>
>> I understand now why a call is considered connected before one person has
>> picked up the phone. I am also able to get my script to wait for the phone
>> to be picked up, by setting the ignore_early_media variable when starting a
>> new session, like this:
>>
>> customerSession =
>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/example.com/"
>> .. customerPhoneNumber)
>>
>>
>> After that line, the script waits for the other phone to be picked up.
>>
>> However, now I wonder what to do with calls that don't complete, get busy
>> signals, etc.
>>
>> What do people do in this case? The only related example I can find on the
>> web is for a javascript dialer, which doesn't address any of these cases.
>>
>
>
> I guess it depends on what you want to do. For example I have a lua script
> very similar to what you describe, although there is no confirmation
> involved. Depending on the hangup cause the session gets, it might try
> redialing with a different gateway, try again or just hangup.
>
> Take a look here http://wiki.freeswitch.org/wiki/Hangup_causes to see what
> each hangup cause means. You don't need to have a special case for all of
> them, only the ones you are interested in.
>
> Here's an example in code which retries a call depending on the hangup
> cause. It retries max_retries1 times and alternates between 2 different
> gateways:
>
> session1 = null;
> max_retries1 = 3;
> retries = 0;
> ostr = "";
> repeat
>         retries = retries + 1;
>         if (retries % 2) then ostr = originate_str1;
>         else ostr = originate_str12; end
>         freeswitch.consoleLog("notice", "*********** Dialing Leg1: " ..
> ostr .. " - Try: "..retries.." ***********\n");
>         session1 = freeswitch.Session(ostr);
>         local hcause = session1:hangupCause();
>         freeswitch.consoleLog("notice", "*********** Leg1: " .. hcause .. "
> - Try: "..retries.." ***********\n");
> until not ((hcause == 'NO_ROUTE_DESTINATION' or hcause ==
> 'RECOVERY_ON_TIMER_EXPIRE' or hcause == 'INCOMPATIBLE_DESTINATION' or hcause
> == 'CALL_REJECTED' or hcause == 'NORMAL_TEMPORARY_FAILURE') and (retries <
> max_retriesl1))
>
>
> Note: originate_str1 and originate_str2 are two different dial strings for
> 2 different gateways.
>
>
>
>>
>> Early Media: http://wiki.freeswitch.org/wiki/Early_media
>> ignore_early_media variable:
>> http://wiki.freeswitch.org/wiki/Variable_ignore_early_media
>> <http://wiki.freeswitch.org/wiki/Variable_ignore_early_media>javascript
>> dialer: http://alexn.org/docs/dialer.html
>>
>>
>> <http://alexn.org/docs/dialer.html>Mike van Lammeren
>>
>>
>> On Wed, Jan 20, 2010 at 4:45 PM, Mike van Lammeren <mike at van.lammeren.net
>> > wrote:
>>
>>> Hi Rob!
>>>
>>> Unfortunately, I have the next few weeks to complete this part of the
>>> project. Without a stable release of 1.0.5, I have to stick with 1.0.4. I'm
>>> sure that 1.0.4 can detect and report when a phone is picked up. It's just
>>> that I can't figure out how to get that information! Either that, or I have
>>> something mis-configured.
>>>
>>> Mike van Lammeren
>>>
>>>
>>> On Wed, Jan 20, 2010 at 4:32 PM, Rob Forman <rob4manhere at gmail.com>wrote:
>>>
>>>> Hi Mike,
>>>>
>>>> I don't think v1.0.4 is supported any longer.   You'll have better luck
>>>> getting assistance by upgrading to trunk or the latest tar and reporting
>>>> back.
>>>>
>>>> Good luck!
>>>> Rob
>>>>
>>>> On Wed, Jan 20, 2010 at 3:18 PM, Mike van Lammeren <
>>>> mike at van.lammeren.net> wrote:
>>>>
>>>>>  Hello!
>>>>>
>>>>> Some day, I'll probably laugh at asking this question, but today I
>>>>> can't figure it out.
>>>>>
>>>>> I've written a Lua script that listens for a call, then dials a phone
>>>>> number to a second person. It plays a message, then prompts the second
>>>>> person to hit pound to connect. If the second person hits pound, then it
>>>>> bridges the two calls together.
>>>>>
>>>>> All that works great, but I can't figure out how to get the session for
>>>>> the second person to wait until that person answers.
>>>>>
>>>>> I'm using FreeSWITCH 1.0.4, and although there is a *getState*function documented in the wiki, it doesn't seem to exist for me.
>>>>>
>>>>> Any help would be appreciated!
>>>>>
>>>>>
>>>>> Mike van Lammeren
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tayeb.meftah at gmail.com  Thu Jan 21 09:09:35 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Thu, 21 Jan 2010 18:09:35 +0100
Subject: [Freeswitch-users] Can't register Polycom
In-Reply-To: <01a301ca9a59$17e45fd0$47ad1f70$@com>
References: <01a301ca9a59$17e45fd0$47ad1f70$@com>
Message-ID: <4B588A4F.5060303@gmail.com>

hi,
make sure that you are using the user: 1008 and pass: 1008
don't forgot to add autorisation user: 1008
the dialplan don't have any relation with the user/pass acording to your 
post, you are montioning the dialplan path but you need to edit the 
directory (conf/directory/default/1008.xml)
chedck your freeswitch ip and configure your policom ip acording to your 
fs ip
good luck
Le 21/01/2010 06:17, Lars Zeb a ?crit :
>
> I am having trouble registering a Polycom 550. From the siptrace it 
> looks like there is no username coming from the Polycom. I configured 
> the Polycom via the web interface. I have inserted "1008" in the 
> "Display Name", "Address" and "Auth User ID" fields.
>
> In conf/dialplan/default/1008.xml  the first line is <user id="1008" 
> cidr="192.168.10.105/24">.
>
> What am I missing?
>
> Thanks, Lars
>
> http://pastebin.freeswitch.org/11875
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From regs at kinetix.gr  Thu Jan 21 10:18:22 2010
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Thu, 21 Jan 2010 20:18:22 +0200
Subject: [Freeswitch-users] reloadxml/rescan profile  and  killgw
Message-ID: <4B589A6E.8010205@kinetix.gr>

Wouldn't it be nice a "reloadxml"/"rescan profile" to delete all 
nonexistent gateways (when they are absent in the xml config) without 
having to issue a "sofia profile xxxxx killgw yyyyyy" command?

I always seem to find forgotten gateway's in the profile because of 
this... Any thoughts?


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
-------------------------------------------


From sos at sokhapkin.dyndns.org  Thu Jan 21 10:41:09 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 21 Jan 2010 13:41:09 -0500
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION?
Message-ID: <201001211341.09739.sos@sokhapkin.dyndns.org>

I often see in FS log the following problem (bypass_media=true), SVN r16340:

SDP sent out to gateway (INVITE):

   v=0
   o=bandx-msw3 0 0 IN IP4 213.166.9.4
   s=sip call
   c=IN IP4 213.166.9.6
   t=0 0
   m=audio 56032 RTP/AVP 0 8 18
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=yes
   a=silenceSupp:on - - - -


SDP response from gateway (183 Session Progress):

   v=0
   o=- 3473087019 3473087037 IN IP4 67.203.64.182
   s=-
   c=IN IP4 67.203.64.182
   t=0 0
   m=audio 14116 RTP/AVP 0
   a=sendrecv
   a=ptime:20
   a=rtpmap:0 PCMU/8000

Freeswitch hangs up the call with hangup cause INCOMPATIBLE_DESTINATION. Where 
is incompatibility? There is common codec 0.


From anthony.minessale at gmail.com  Thu Jan 21 10:53:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Jan 2010 12:53:23 -0600
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION?
In-Reply-To: <201001211341.09739.sos@sokhapkin.dyndns.org>
References: <201001211341.09739.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031001211053i6670cec3l8b8f317dac4c6072@mail.gmail.com>

if you use bypass_media=true from the dialplan without late-negotiation set
in the profile, it still tries to match the codecs locally on the inbound
leg and the variable does not work if the call has established media before
making the outbound leg.

It's hard to tell you the exact answer without a console trace on debug
level.


On Thu, Jan 21, 2010 at 12:41 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> I often see in FS log the following problem (bypass_media=true), SVN
> r16340:
>
> SDP sent out to gateway (INVITE):
>
>   v=0
>   o=bandx-msw3 0 0 IN IP4 213.166.9.4
>   s=sip call
>   c=IN IP4 213.166.9.6
>   t=0 0
>   m=audio 56032 RTP/AVP 0 8 18
>   a=rtpmap:18 G729/8000
>   a=fmtp:18 annexb=yes
>   a=silenceSupp:on - - - -
>
>
> SDP response from gateway (183 Session Progress):
>
>   v=0
>   o=- 3473087019 3473087037 IN IP4 67.203.64.182
>   s=-
>   c=IN IP4 67.203.64.182
>   t=0 0
>   m=audio 14116 RTP/AVP 0
>   a=sendrecv
>   a=ptime:20
>   a=rtpmap:0 PCMU/8000
>
> Freeswitch hangs up the call with hangup cause INCOMPATIBLE_DESTINATION.
> Where
> is incompatibility? There is common codec 0.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From sos at sokhapkin.dyndns.org  Thu Jan 21 11:05:19 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 21 Jan 2010 14:05:19 -0500
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION?
In-Reply-To: <191c3a031001211053i6670cec3l8b8f317dac4c6072@mail.gmail.com>
References: <201001211341.09739.sos@sokhapkin.dyndns.org>
	<191c3a031001211053i6670cec3l8b8f317dac4c6072@mail.gmail.com>
Message-ID: <201001211405.19271.sos@sokhapkin.dyndns.org>

Late negotiation is set. I will try to enable debug when the traffic will be 
low and open a problem on jira.

On Thursday 21 January 2010, Anthony Minessale wrote:
> if you use bypass_media=true from the dialplan without late-negotiation set
> in the profile, it still tries to match the codecs locally on the inbound
> leg and the variable does not work if the call has established media before
> making the outbound leg.
>
> It's hard to tell you the exact answer without a console trace on debug
> level.
>
>
> On Thu, Jan 21, 2010 at 12:41 PM, Sergey Okhapkin
>
> <sos at sokhapkin.dyndns.org>wrote:
> > I often see in FS log the following problem (bypass_media=true), SVN
> > r16340:
> >
> > SDP sent out to gateway (INVITE):
> >
> >   v=0
> >   o=bandx-msw3 0 0 IN IP4 213.166.9.4
> >   s=sip call
> >   c=IN IP4 213.166.9.6
> >   t=0 0
> >   m=audio 56032 RTP/AVP 0 8 18
> >   a=rtpmap:18 G729/8000
> >   a=fmtp:18 annexb=yes
> >   a=silenceSupp:on - - - -
> >
> >
> > SDP response from gateway (183 Session Progress):
> >
> >   v=0
> >   o=- 3473087019 3473087037 IN IP4 67.203.64.182
> >   s=-
> >   c=IN IP4 67.203.64.182
> >   t=0 0
> >   m=audio 14116 RTP/AVP 0
> >   a=sendrecv
> >   a=ptime:20
> >   a=rtpmap:0 PCMU/8000
> >
> > Freeswitch hangs up the call with hangup cause INCOMPATIBLE_DESTINATION.
> > Where
> > is incompatibility? There is common codec 0.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From msc at freeswitch.org  Thu Jan 21 11:20:06 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Jan 2010 11:20:06 -0800
Subject: [Freeswitch-users] Configuration Preservation
	throughTrunkUpdates
In-Reply-To: <DF8D6745E8E1492A9D2E345A6B42D768@bp1.ad.bp.com>
References: <21A169E64A904A188DC34E67B1A17651@bp1.ad.bp.com>
	<C3424C68-4E69-4CA1-A494-2B143C0E8E8F@freeswitch.org>
	<5A041CBBECB441D094B86C935CC6C792@bp1.ad.bp.com>
	<1F2B2E59-6C17-4DB8-A197-E57D2A804132@freeswitch.org>
	<DF8D6745E8E1492A9D2E345A6B42D768@bp1.ad.bp.com>
Message-ID: <87f2f3b91001211120q478a6d71nc8b2afb2799e91ec@mail.gmail.com>

On Wed, Jan 20, 2010 at 11:44 AM, Dave Stevenson
<stevendt at primrosebank.net>wrote:

>  Brian,
>
> Following on from before, there's one item that I can't see how to do
> outside of modifying the directory\default.xml file, and that is setting up
> call groups.
>
> I thought that I'd perhaps be able to do something similar to creating user
> dial plans and create a new file in directory\default\  which would be
> loaded before the other extensions, i.e., called something like
> 00_groups.xml and have the call group created there. (My test file is shown
> below). That did not seem to work, am I on the right lines or should custom
> groups get created somewhere else ?
>
>

This leads to the bigger question of where people should put their
customizations without breaking the defaults. A good example of the issue is
with ivr.conf.xml and the corresponding phrase macros in
conf/lang/en/en.xml. In some cases it is impossible to add customizations
without modifying the default config files.

I will bring this up on tomorrow's conference call. If you want to be heard
on this topic then please join us on the call. I will get the agenda posted
ASAP.

-MC
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From stevendt at primrosebank.net  Thu Jan 21 11:41:48 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Thu, 21 Jan 2010 19:41:48 -0000
Subject: [Freeswitch-users] Configuration PreservationthroughTrunkUpdates
References: <21A169E64A904A188DC34E67B1A17651@bp1.ad.bp.com><C3424C68-4E69-4CA1-A494-2B143C0E8E8F@freeswitch.org><5A041CBBECB441D094B86C935CC6C792@bp1.ad.bp.com><1F2B2E59-6C17-4DB8-A197-E57D2A804132@freeswitch.org><DF8D6745E8E1492A9D2E345A6B42D768@bp1.ad.bp.com>
	<87f2f3b91001211120q478a6d71nc8b2afb2799e91ec@mail.gmail.com>
Message-ID: <EE53C4F234D848EBA490E0A82847257F@bp1.ad.bp.com>

Hi Michael,

thanks for picking this up - I'm glad that it's something that is relevant to a wider audience than just me !

I'll try and join the call tomorrow - if only to listen in,

regards
Dave
  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, January 21, 2010 7:20 PM
  Subject: Re: [Freeswitch-users] Configuration PreservationthroughTrunkUpdates





  On Wed, Jan 20, 2010 at 11:44 AM, Dave Stevenson <stevendt at primrosebank.net> wrote:

    Brian,

    Following on from before, there's one item that I can't see how to do outside of modifying the directory\default.xml file, and that is setting up call groups.

    I thought that I'd perhaps be able to do something similar to creating user dial plans and create a new file in directory\default\  which would be loaded before the other extensions, i.e., called something like 00_groups.xml and have the call group created there. (My test file is shown below). That did not seem to work, am I on the right lines or should custom groups get created somewhere else ?


  This leads to the bigger question of where people should put their customizations without breaking the defaults. A good example of the issue is with ivr.conf.xml and the corresponding phrase macros in conf/lang/en/en.xml. In some cases it is impossible to add customizations without modifying the default config files. 

  I will bring this up on tomorrow's conference call. If you want to be heard on this topic then please join us on the call. I will get the agenda posted ASAP.

  -MC




------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From msc at freeswitch.org  Thu Jan 21 11:59:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Jan 2010 11:59:49 -0800
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
Message-ID: <87f2f3b91001211159x55f9be96qb9278c4516b4c5ec@mail.gmail.com>

Can you do two things:

Get a debug log of openzap loading, put it on pastebin. (you can unload
mod_openzap and then press F8 and type load mod_openzap, capturing the
output. Also do "oz list" and "oz dump 1" and pastebin the output.

Pastebin your openzap.conf.xml, openzap.conf, and smg_prid.conf files.

-MC

On Wed, Jan 20, 2010 at 9:04 PM, Thangappan.M <thangappan143 at gmail.com>wrote:

> OpenZap is loading the ss7 signalling type. As per your concern openzap
> does not know the details of the signalling then how it is loading the
> ss7_boost libraries?
>
> FreeSWITCH log:
> -----------------------------
> 2010-01-21 10:10:46.707844 [INFO] zap_io.c:2374 Configured 30 channel(s)
> 2010-01-21 10:10:46.708600 [INFO] zap_io.c:2468 Loading SIG from
> /usr/local/freeswitch/mod/ozmod_ss7_boost.so
> 2010-01-21 10:10:46.709031 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost'
> 2010-01-21 10:10:46.709466 [DEBUG] ss7_boost_client.c:124 Creating L=
> 127.0.0.65:53000 R=127.0.0.66:53000
> 2010-01-21 10:10:46.709834 [DEBUG] ss7_boost_client.c:124 Creating L=
> 127.0.0.65:53001 R=127.0.0.66:53001
> 2010-01-21 10:10:46.710424 [WARNING] ss7_boost_client.c:244 TX EVENT (P):
> SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>
> The signalling type might be anything but when I used the oz list command
> it showed the span details. But I am unable to make a inbound and outbound
> call.
>
> Outbound call result:
> ============
> >  originate openzap/smg_prid/a/9940464753 openzap/smg_prid/a/9843171457
> -ERR NORMAL_CIRCUIT_CONGESTION
>
> 2010-01-21 10:26:14.304816 [CRIT] ozmod_ss7_boost.c:244 SPAN is not online.
> freeswitch at internal> 2010-01-21 10:26:14.304816 [ERR] mod_openzap.c:1043
> No channels available
> 2010-01-21 10:26:14.304816 [ERR] switch_ivr_originate.c:1510 Cannot create
> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
> 2010-01-21 10:26:14.304816 [DEBUG] switch_ivr_originate.c:2138 Originate
> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>
> Inbound call result:
> -----------------------------
>
>      I made incoming call for the dial plan which is specified in the
> earlier post at that time it said the number is busy. We did the packet
> capture using the following command.
>
> wanpipemon -i w1g1 -pcap -pcap file isdn.pcap -port ISDN -full -systime -c
> trd
>
> Here I attached the pcap file for that.
>
>
> Where I did mistake or Did I miss any thing to do?
> Please help me.......
>
>
>
> On Wed, Jan 20, 2010 at 7:40 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>>
>>       I noticed the 'oz list' output  in that span type is 'ss7 (boost)'.
>> How can I change this to isdn?
>>
>>
>>
>> On Wed, Jan 20, 2010 at 12:43 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>
>>> I found the error in it. The file name is used as openzap.conf.xml (
>>> smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
>>> PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
>>> openzap.conf.wiki.xml  file.
>>>
>>> Now the another problem is raised here.
>>> When I was using oz list command , the details of the smg_prid shown.
>>> When I was using 'oz dump smg_prid' command it shows all the channels'
>>> details. But all the channels' states are DOWN. why? How can I make it the
>>> states to UP?
>>>
>>> When I was making the call , the number is busy message was get. The call
>>> was not  at all landed to the freeswitch.
>>>
>>> Dial plan Example:
>>> -------------------------------
>>>  <extension name="PRI_Testing">
>>>         <condition field="destination_number" expression="^39114600$">
>>>                 <action application="playback"
>>> data="ivr-welcome_to_freeswitch"/>
>>>         </condition>
>>>   </extension>
>>>
>>> Please help me...........
>>>
>>> *Output Reference:*
>>> http://pastebin.org/79074
>>>
>>>
>>> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>>
>>>> Dear all,
>>>>
>>>>          I have successfully configured wanpipe with freeswitch. When I
>>>> was the running wancfg_fs script the following files openzap.conf ,
>>>> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
>>>> are created.
>>>>
>>>>          I started the wanrouter command then executed the freeswitch.
>>>>          When I was executing freeswitch mod_openzap.c said the error
>>>> as  "Error  for finding the span id. name:PRI_1".
>>>>          But in the openzap.conf and openzap.conf.xml files the span
>>>> name is smg_prid.
>>>>
>>>>          Why the freeswitch is referring the span name as PRI_1 ?
>>>>          Whether this has to configured in anywhere?
>>>>
>>>>          In the freeswitch CLI using oz command I tried to dump the
>>>> PRI_1 span id but it said te error  as "PRI_1 is not found".  When I was
>>>> trying the command 'oz dump smg_prid'  all the channel states and details
>>>> shown.
>>>>
>>>>         It seems that smg_prid span configured in openzap perfectly (Its
>>>> my assumption). Then Why freeswitch is referring the span name as PRI_1.
>>>>
>>>> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>>>>
>>>> Could anyone please help me?
>>>>
>>>> REFERENCE:
>>>>
>>>> openzap.conf
>>>> [span wanpipe smg_prid]
>>>> name => smg_prid
>>>> trunk_type =>e1
>>>> b-channel => 1:1-15
>>>> b-channel => 1:17-31
>>>>
>>>>
>>>> openzap.conf.xml
>>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>>   <settings>
>>>>     <param name="debug" value="0"/>
>>>>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>>     <!--<param name="enable-analog-option" value="3-way"/>-->
>>>>   </settings>
>>>>   <boost_spans>
>>>>         <span id="smg_prid">
>>>>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>         <param name="dialplan" value="XML"/>
>>>>         <param name="context" value="default"/>
>>>>          <!-- regex to stop dialing when it matches -->
>>>>         <!--<param name="dial-regex" value="5555"/>-->
>>>>         <!-- regex to stop dialing when it does not match -->
>>>>         <!--<param name="fail-dial-regex" value="^5"/>-->
>>>>         </span>
>>>>  </boost_spans>
>>>>  <analog_spans>
>>>>  </analog_spans>
>>>> </configuration>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Thangappan.M
>>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Thangappan.M
>>>
>>
>>
>>
>> --
>> Regards,
>> Thangappan.M
>>
>
>
>
> --
> Regards,
> Thangappan.M
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Thu Jan 21 12:08:37 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 21 Jan 2010 14:08:37 -0600
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <87f2f3b91001211159x55f9be96qb9278c4516b4c5ec@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
	<87f2f3b91001211159x55f9be96qb9278c4516b4c5ec@mail.gmail.com>
Message-ID: <1873B030-24AA-4B2E-8444-E4678591D2B4@freeswitch.org>

You're running old FreeSWITCH you'll need the very latest.

/b

On Jan 21, 2010, at 1:59 PM, Michael Collins wrote:

> 2010-01-21 10:10:46.710424 [WARNING] ss7_boost_client.c:244 TX EVENT (P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0



From lists at redbonez.net  Thu Jan 21 12:12:30 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 21 Jan 2010 13:12:30 -0700
Subject: [Freeswitch-users] Can't register Polycom
In-Reply-To: <4B588A4F.5060303@gmail.com>
References: <01a301ca9a59$17e45fd0$47ad1f70$@com> <4B588A4F.5060303@gmail.com>
Message-ID: <003601ca9ad6$108bb2b0$31a31810$@net>

I dunno about the 550s, but on my Polycom IP501 I had to configure the auth
username both from the phone interface and the web interface to get it to
stick.

 

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Meftah
Tayeb
Sent: Thursday, January 21, 2010 10:10 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't register Polycom

 

hi,
make sure that you are using the user: 1008 and pass: 1008
don't forgot to add autorisation user: 1008
the dialplan don't have any relation with the user/pass acording to your
post, you are montioning the dialplan path but you need to edit the
directory (conf/directory/default/1008.xml)
chedck your freeswitch ip and configure your policom ip acording to your fs
ip
good luck
Le 21/01/2010 06:17, Lars Zeb a ?crit : 

I am having trouble registering a Polycom 550. >From the siptrace it looks
like there is no username coming from the Polycom. I configured the Polycom
via the web interface. I have inserted ?1008? in the ?Display Name?,
?Address? and ?Auth User ID? fields.

 

In conf/dialplan/default/1008.xml  the first line is <user id="1008"
cidr="192.168.10.105/24">.

 

What am I missing?

 

Thanks, Lars

 

http://pastebin.freeswitch.org/11875

 

 

 
 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  

 

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From msc at freeswitch.org  Thu Jan 21 12:45:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Jan 2010 12:45:18 -0800
Subject: [Freeswitch-users] Additional endpoints
In-Reply-To: <00d701ca9a1e$18afa0e0$4a0ee2a0$@com>
References: <00d701ca9a1e$18afa0e0$4a0ee2a0$@com>
Message-ID: <87f2f3b91001211245l1b724266ibe99f7001d757b33@mail.gmail.com>

On Wed, Jan 20, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:

>  I know this has been answered before, but I cannot find it.
>
>
>
> How do I setup more than the default 20 endpoints (1000-1019)?  Do I extend
> the definition in dialplan/public.xml (public_extensions) and add the extra
> in directory/default?
>
> This topic is covered in the FreeSWITCH article in Linux Mag:
http://bit.ly/EpVrv

-MC
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From msc at freeswitch.org  Thu Jan 21 12:49:36 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Jan 2010 12:49:36 -0800
Subject: [Freeswitch-users] All channels are frozen while receiving DTMF
In-Reply-To: <27255181.post@talk.nabble.com>
References: <27255181.post@talk.nabble.com>
Message-ID: <87f2f3b91001211249o531eda47m68658b016cacdfef@mail.gmail.com>

On Thu, Jan 21, 2010 at 1:47 AM, Oscav <oscav at hotmail.fr> wrote:

>
> Hi,
>
> I'm running a script that gets some DTMF from caller. I found that when a
> caller is entering DTMF , all the others channels are frozen until all the
> DTMF  are received. In the logs I see that each DTMF takes 1 second. It
> means that if the caller enters 10 digits then all the other running
> scripts
> are paused for 10 seconds. The problem is exponential with traffic load.
>
> Anyone have an idea ??
>
> What version of FS? What OS are you running? Which scripting language?
Please pastebin your script and a debug log of this symptom happening. Of
course, you should be on the latest version of FreeSWITCH first because the
devs frequently fix bugs and add features. :)

-MC
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From msc at freeswitch.org  Thu Jan 21 13:00:34 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Jan 2010 13:00:34 -0800
Subject: [Freeswitch-users] G729 coded issues
In-Reply-To: <cd93b3701001210210v2ddba8afwc6a310540d2e5180@mail.gmail.com>
References: <cd93b3701001190831u4147214cn9832c6f469c6d17e@mail.gmail.com>
	<6D67C7A9-42CD-4F5F-AD18-E9568098FB71@freeswitch.org>
	<4B55E794.6020909@coppice.org>
	<cd93b3701001200514p4acc5a5bk26a463164f841ad2@mail.gmail.com>
	<537338B1-B582-463E-8EFE-7BBED8165D2B@freeswitch.org>
	<cd93b3701001210210v2ddba8afwc6a310540d2e5180@mail.gmail.com>
Message-ID: <87f2f3b91001211300q18449180hfe7e51d2e86b74f9@mail.gmail.com>

On Thu, Jan 21, 2010 at 2:10 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:

> Hi Brian,
>
> All for it, so ye, let me have the beta.
>
> The only reason I went Howler is because of a pressing need.
>
> Moreover, I'm willing to put mine and my teams resources into the project.
> I and others cut code in C++/C/Perl/....etc. As an organisation, we are
> actively involved in telecoms consultancy and software development, and I
> really would like to put our backs into pushing the case for FS wherever
> possible.
>
> Ping me offline on the subscription e-mail for this account, and let's
> exchange details if there is interest your side.
>
> Support ? we are all sold on FS down here believe me, so you can count on
> it.
>
> Regards,
>
> Ahmed.
>
Ahmed, that's great to hear. You may want to join the community conference
calls that we have every Friday. It's 11:00AM CST (or 16:00 GMT) and we'd
love to have new people join up. The info can be found here:
http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_22

-MC
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From msc at freeswitch.org  Thu Jan 21 14:12:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Jan 2010 14:12:23 -0800
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <1b46b4e81001210335l42baef16r8a2952aa5b92f6e6@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
	<e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
	<5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>
	<5d2828f1001201406p4154b98ald1af1c5c25f59337@mail.gmail.com>
	<1b46b4e81001210335l42baef16r8a2952aa5b92f6e6@mail.gmail.com>
Message-ID: <87f2f3b91001211412h7aa0a84ageadb9557b869ca01@mail.gmail.com>

On Thu, Jan 21, 2010 at 3:35 AM, Nicolas Brenner <nicolas at medularis.com>wrote:

>
> On Wed, Jan 20, 2010 at 7:06 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:
>
>> So, I've been reading about early media in the wiki, and have made a
>> little progress, which leads to more questions.
>>
>> I understand now why a call is considered connected before one person has
>> picked up the phone. I am also able to get my script to wait for the phone
>> to be picked up, by setting the ignore_early_media variable when starting a
>> new session, like this:
>>
>> customerSession =
>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/example.com/"
>> .. customerPhoneNumber)
>>
>>
>> After that line, the script waits for the other phone to be picked up.
>>
>> However, now I wonder what to do with calls that don't complete, get busy
>> signals, etc.
>>
>> What do people do in this case? The only related example I can find on the
>> web is for a javascript dialer, which doesn't address any of these cases.
>>
>
>
> I guess it depends on what you want to do. For example I have a lua script
> very similar to what you describe, although there is no confirmation
> involved. Depending on the hangup cause the session gets, it might try
> redialing with a different gateway, try again or just hangup.
>
> Take a look here http://wiki.freeswitch.org/wiki/Hangup_causes to see what
> each hangup cause means. You don't need to have a special case for all of
> them, only the ones you are interested in.
>
> Here's an example in code which retries a call depending on the hangup
> cause. It retries max_retries1 times and alternates between 2 different
> gateways:
>
> session1 = null;
> max_retries1 = 3;
> retries = 0;
> ostr = "";
> repeat
>         retries = retries + 1;
>         if (retries % 2) then ostr = originate_str1;
>         else ostr = originate_str12; end
>         freeswitch.consoleLog("notice", "*********** Dialing Leg1: " ..
> ostr .. " - Try: "..retries.." ***********\n");
>         session1 = freeswitch.Session(ostr);
>         local hcause = session1:hangupCause();
>         freeswitch.consoleLog("notice", "*********** Leg1: " .. hcause .. "
> - Try: "..retries.." ***********\n");
> until not ((hcause == 'NO_ROUTE_DESTINATION' or hcause ==
> 'RECOVERY_ON_TIMER_EXPIRE' or hcause == 'INCOMPATIBLE_DESTINATION' or hcause
> == 'CALL_REJECTED' or hcause == 'NORMAL_TEMPORARY_FAILURE') and (retries <
> max_retriesl1))
>
>
> Note: originate_str1 and originate_str2 are two different dial strings for
> 2 different gateways.
>
>
Nicolas,

This is really nice. Would you be willing to add this script and a brief
explanation to the wiki? You could create a whole new page and just link to
it from here: http://wiki.freeswitch.org/wiki/Mod_lua#More_Samples

If you have any questions please let me know!
-MC
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From jerry.richards at teotech.com  Thu Jan 21 14:39:28 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 21 Jan 2010 14:39:28 -0800
Subject: [Freeswitch-users] Presence Not Working After Version 1.0.5pre9?
In-Reply-To: <191c3a031001201531g7c78cb9fw1ed1a2ba07f5773c@mail.gmail.com>
References: <mailman.5334.1264017641.2495.freeswitch-users@lists.freeswitch.org><C726F91949324BDFA34596256B6E102A@greyhawk.tonecommander.com>
	<191c3a031001201531g7c78cb9fw1ed1a2ba07f5773c@mail.gmail.com>
Message-ID: <B4EB2C4365CD4CA79B54134ECF232EED@greyhawk.tonecommander.com>

Yes you are correct.  The Bria Softphone has a setting under
ContactProfile/Advanced.../Account menu which is required to be the
softphone's extension (not blank and not the extension that is being
subscribed to).  After I set this field to the softphone's extension, FS
starting reporting the the contact's presence status.
 
Thanks and Best Regards,
Jerry
 


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Wednesday, January 20, 2010 3:32 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Not Working After Version
1.0.5pre9?


try turning on sip debug and console loglevel debug

sofia loglevel all 9
console loglevel debug

Did you try manually running the same sql stmts from the sqlite3 app?
maybe you have something misconfigured.



On Wed, Jan 20, 2010 at 4:44 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


Does anyone know why I do not see NOTIFY messages with presence status being
sent out from FS for two Bria softphones?  It used to work in my old version
1.0.5pre9.  I upgraded Freeswitch on Jan 12 and Jan 20 and edited in my XML
configurations, but I do not see the NOTIFY messages since version
1.0.5pre9.

I have mostly default configuration and I added the manage-presence=true
setting.  I posted a pastebin at http://pastebin.freeswitch.org/11867.

Can anyone tell why this isn't working?

Best Regards,
Jerry


_______________________________________________
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http://www.freeswitch.org





-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


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From mike at jerris.com  Thu Jan 21 14:56:21 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 21 Jan 2010 17:56:21 -0500
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <37012.20543.qm@web33507.mail.mud.yahoo.com>
References: <37012.20543.qm@web33507.mail.mud.yahoo.com>
Message-ID: <5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>

How about with svn r16440

On Jan 21, 2010, at 8:02 AM, Diego Toro wrote:

> Hi MikeJ, using '\\' the behavior is the same, '\\s' is replaced by ' '.
> 
> <action application="playback" data="misc\\8000\\serror.wav!misc\\8000\\provide_reference_number.wav!digits\\8000\\5.wav"/>  
> 
> Console output error is:
> 
> [ERR] mod_sndfile.c:194 Error Opening File [d:\fs\fs_trunk_20100118\Debug\sounds\es\co\callie\misc\8000 error.wav]
> 
> 
> S.O.: Windows 7
> FreeSwitch: Trunk (svn latest version)

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From troy at tlainvestments.com  Thu Jan 21 15:21:27 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 21 Jan 2010 16:21:27 -0700
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
In-Reply-To: <5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>
References: <37012.20543.qm@web33507.mail.mud.yahoo.com>
	<5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>
Message-ID: <517DA285-7E9B-4226-8AEF-C2DAB52976CF@tlainvestments.com>

Hello,

I'm on the latest trunk version (16440) and having an issue with Polycom and transferring.  The dial plan is set up so that unanswered calls go to voicemail.  When I answer a call with a polycom phone and then transfer that call to another phone, if the other phone doesn't pick up and the voicemail app starts, then I hit transfer again with the intent of having the caller leave a voicemail, the call is dropped.  If the phone does pick up during the transfer, it works fine.

I also have an Aastra phone, and when I do the same thing, but from the Aastra phone, it works as expected.  Is this known to be a problem with Polycom?

Thanks!
Troy

From larclap at yahoo.com  Thu Jan 21 15:36:19 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 21 Jan 2010 15:36:19 -0800
Subject: [Freeswitch-users] Can't register Polycom
In-Reply-To: <003601ca9ad6$108bb2b0$31a31810$@net>
References: <01a301ca9a59$17e45fd0$47ad1f70$@com> <4B588A4F.5060303@gmail.com>
	<003601ca9ad6$108bb2b0$31a31810$@net>
Message-ID: <02e101ca9af2$887829e0$99687da0$@com>

Thanks for that hint. In the meantime I just used Line 3. I?ll try your
suggestion soon.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam
Ford
Sent: Thursday, January 21, 2010 12:13 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't register Polycom

 

I dunno about the 550s, but on my Polycom IP501 I had to configure the auth
username both from the phone interface and the web interface to get it to
stick.

 

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Meftah
Tayeb
Sent: Thursday, January 21, 2010 10:10 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't register Polycom

 

hi,
make sure that you are using the user: 1008 and pass: 1008
don't forgot to add autorisation user: 1008
the dialplan don't have any relation with the user/pass acording to your
post, you are montioning the dialplan path but you need to edit the
directory (conf/directory/default/1008.xml)
chedck your freeswitch ip and configure your policom ip acording to your fs
ip
good luck
Le 21/01/2010 06:17, Lars Zeb a ?crit : 

I am having trouble registering a Polycom 550. >From the siptrace it looks
like there is no username coming from the Polycom. I configured the Polycom
via the web interface. I have inserted ?1008? in the ?Display Name?,
?Address? and ?Auth User ID? fields.

 

In conf/dialplan/default/1008.xml  the first line is <user id="1008"
cidr="192.168.10.105/24">.

 

What am I missing?

 

Thanks, Lars

 

http://pastebin.freeswitch.org/11875

 

 

 
 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  

 

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From lists at redbonez.net  Thu Jan 21 15:42:39 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 21 Jan 2010 16:42:39 -0700
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
In-Reply-To: <517DA285-7E9B-4226-8AEF-C2DAB52976CF@tlainvestments.com>
References: <37012.20543.qm@web33507.mail.mud.yahoo.com>	<5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>
	<517DA285-7E9B-4226-8AEF-C2DAB52976CF@tlainvestments.com>
Message-ID: <004e01ca9af3$6c5a2d20$450e8760$@net>

Yes it is a known issue with Polycom phones. Polycom supports a non-standard
transfer method which does not work with FreeSWITCH. 

See this article for further details -
http://www.junctionnetworks.com/knowledgebase/onsip/phones-routers-and-devic
es/phone-configuration/polycom/polycom-disabling-non-standard-transfer-meth

I ran into the same problem, disabling
voIpProt.SIP.allowTransferOnProceeding as suggested in that article resolved
the issue for me.

-Adam

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Troy
Anderson
Sent: Thursday, January 21, 2010 4:21 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail

Hello,

I'm on the latest trunk version (16440) and having an issue with Polycom and
transferring.  The dial plan is set up so that unanswered calls go to
voicemail.  When I answer a call with a polycom phone and then transfer that
call to another phone, if the other phone doesn't pick up and the voicemail
app starts, then I hit transfer again with the intent of having the caller
leave a voicemail, the call is dropped.  If the phone does pick up during
the transfer, it works fine.

I also have an Aastra phone, and when I do the same thing, but from the
Aastra phone, it works as expected.  Is this known to be a problem with
Polycom?

Thanks!
Troy
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From brian at freeswitch.org  Thu Jan 21 15:45:32 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 21 Jan 2010 17:45:32 -0600
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
In-Reply-To: <004e01ca9af3$6c5a2d20$450e8760$@net>
References: <37012.20543.qm@web33507.mail.mud.yahoo.com>	<5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>
	<517DA285-7E9B-4226-8AEF-C2DAB52976CF@tlainvestments.com>
	<004e01ca9af3$6c5a2d20$450e8760$@net>
Message-ID: <84550BFB-81A3-4CF4-8CD0-9CC4F29A66F0@freeswitch.org>

No this works.  But not to an IVR I suspect.

This is when people don't know what blind vs attended means and they could just press one more button and get the same effect.

/b

On Jan 21, 2010, at 5:42 PM, Adam Ford wrote:

> voIpProt.SIP.allowTransferOnProceeding 



From lists at redbonez.net  Thu Jan 21 16:00:06 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 21 Jan 2010 17:00:06 -0700
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
In-Reply-To: <004e01ca9af3$6c5a2d20$450e8760$@net>
References: <37012.20543.qm@web33507.mail.mud.yahoo.com>	<5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>	<517DA285-7E9B-4226-8AEF-C2DAB52976CF@tlainvestments.com>
	<004e01ca9af3$6c5a2d20$450e8760$@net>
Message-ID: <005b01ca9af5$dc759980$9560cc80$@net>

That link didn't come through very well, here is a shortened one -
http://bit.ly/6wDAXD

-Adam

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam
Ford
Sent: Thursday, January 21, 2010 4:43 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom Consultative Transfer and Voicemail

Yes it is a known issue with Polycom phones. Polycom supports a non-standard
transfer method which does not work with FreeSWITCH. 

See this article for further details -
http://www.junctionnetworks.com/knowledgebase/onsip/phones-routers-and-devic
es/phone-configuration/polycom/polycom-disabling-non-standard-transfer-meth

I ran into the same problem, disabling
voIpProt.SIP.allowTransferOnProceeding as suggested in that article resolved
the issue for me.

-Adam

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Troy
Anderson
Sent: Thursday, January 21, 2010 4:21 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail

Hello,

I'm on the latest trunk version (16440) and having an issue with Polycom and
transferring.  The dial plan is set up so that unanswered calls go to
voicemail.  When I answer a call with a polycom phone and then transfer that
call to another phone, if the other phone doesn't pick up and the voicemail
app starts, then I hit transfer again with the intent of having the caller
leave a voicemail, the call is dropped.  If the phone does pick up during
the transfer, it works fine.

I also have an Aastra phone, and when I do the same thing, but from the
Aastra phone, it works as expected.  Is this known to be a problem with
Polycom?

Thanks!
Troy
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From mouncifbb at gmail.com  Thu Jan 21 15:58:37 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Thu, 21 Jan 2010 18:58:37 -0500
Subject: [Freeswitch-users] Javascript self.session.getVariable
Message-ID: <ade741d61001211558h7b84cb96q16754650812a80e2@mail.gmail.com>

I have the following in javascript:

caller_id = self.session.getVariable("caller_id_number")


for some reasons it returns:
ReferenceError: self is not defined, I am following this page:
http://wiki.freeswitch.org/wiki/Session_getVariable


any ideas?
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From anthony.minessale at gmail.com  Thu Jan 21 16:48:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Jan 2010 18:48:17 -0600
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
In-Reply-To: <005b01ca9af5$dc759980$9560cc80$@net>
References: <37012.20543.qm@web33507.mail.mud.yahoo.com>
	<5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>
	<517DA285-7E9B-4226-8AEF-C2DAB52976CF@tlainvestments.com>
	<004e01ca9af3$6c5a2d20$450e8760$@net>
	<005b01ca9af5$dc759980$9560cc80$@net>
Message-ID: <191c3a031001211648i5ed86253k8640c49f29121e0c@mail.gmail.com>

if you used the loopback endpoint to loop around to voicemail or made a
looped sip call back to your own box you could xfer it as desired.


bridge to "loopback/app=voicemail:default ${domain_name}
${dialed_extension}"

That will make the vm app run as a channel instead of an inline app.

This is an undocumented feature because it's not well tested so if it
doesn't work *shrug* =D



On Thu, Jan 21, 2010 at 6:00 PM, Adam Ford <lists at redbonez.net> wrote:

> That link didn't come through very well, here is a shortened one -
> http://bit.ly/6wDAXD
>
> -Adam
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam
> Ford
> Sent: Thursday, January 21, 2010 4:43 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
>
> Yes it is a known issue with Polycom phones. Polycom supports a
> non-standard
> transfer method which does not work with FreeSWITCH.
>
> See this article for further details -
>
> http://www.junctionnetworks.com/knowledgebase/onsip/phones-routers-and-devic
> es/phone-configuration/polycom/polycom-disabling-non-standard-transfer-meth<http://www.junctionnetworks.com/knowledgebase/onsip/phones-routers-and-devic%0Aes/phone-configuration/polycom/polycom-disabling-non-standard-transfer-meth>
>
> I ran into the same problem, disabling
> voIpProt.SIP.allowTransferOnProceeding as suggested in that article
> resolved
> the issue for me.
>
> -Adam
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Troy
> Anderson
> Sent: Thursday, January 21, 2010 4:21 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
>
> Hello,
>
> I'm on the latest trunk version (16440) and having an issue with Polycom
> and
> transferring.  The dial plan is set up so that unanswered calls go to
> voicemail.  When I answer a call with a polycom phone and then transfer
> that
> call to another phone, if the other phone doesn't pick up and the voicemail
> app starts, then I hit transfer again with the intent of having the caller
> leave a voicemail, the call is dropped.  If the phone does pick up during
> the transfer, it works fine.
>
> I also have an Aastra phone, and when I do the same thing, but from the
> Aastra phone, it works as expected.  Is this known to be a problem with
> Polycom?
>
> Thanks!
> Troy
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From styli1 at hotmail.com  Thu Jan 21 16:55:21 2010
From: styli1 at hotmail.com (styli1 at hotmail.com)
Date: Thu, 21 Jan 2010 16:55:21 -0800
Subject: [Freeswitch-users] Vacation reply
In-Reply-To: <191c3a031001211648i5ed86253k8640c49f29121e0c@mail.gmail.com>
Message-ID: <SNT0-MC1-F211A2E5BC7F772D7C5E81A85620@phx.gbl>

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From jingwei.yang at gmail.com  Thu Jan 21 17:05:13 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Fri, 22 Jan 2010 09:05:13 +0800
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <1264092620.14614.73.camel@luna.tc.commsmundi.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
	<1264092620.14614.73.camel@luna.tc.commsmundi.com>
Message-ID: <13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>

Thanks for the reply. All the agents are dynamic and I can't predefine them
in the config file.

Regards,
-Jingwei

On Fri, Jan 22, 2010 at 12:50 AM, Fran?ois Delawarde <
fdelawarde at wirelessmundi.com> wrote:

> Why do you need 2 fifos? You could have callback agents connected to the
> fifo and send incoming calls there, mod_fifo will do the rest.
>
> To configure agents for callback:
> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>
> To place a call into a fifo:
> http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue
>
> Fran?ois.
>
> On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
> > Sorry about the confusion, I'm just trying to think over all the
> > abnormal situations before the implementation and hope the flow has no
> > design flaws.
> >
> > Client A is parked in Queue 1, multiple consumers will be ringed to
> > answer him. And once the first one is connected, all the rest will
> > hang up. This is the targeted function.
> >
> > To achieve this, I'm considering to originate a call to each consumer
> > and put the calls in Queue 2. Then intercept client A to the first
> > element of Queue 2.
> >
> > I'm not sure if it's the usual or the best way. If you feel not,
> > please don't hesitate to correct me. Any thoughts are warmly
> > appreciated.
> >
> > On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com> wrote:
> >         what no errors mean? how do you originate calls to consumers?
> >         I don't
> >         understand your scenario.
> >
> >         2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
> >
> >         > Hi All,
> >         >
> >         > Please advise whether the following flow makes sense.
> >         >
> >         > 1. Client A calls in and parked in Queue 1
> >         > 2. Originate calls to several consumers simultaneously and
> >         park them in
> >         > Queue 2
> >         > 3. Intercept A's call to the first consumer of Queue 2
> >         >
> >         > Basically I want A's call picked up by the first among many
> >         consumers with
> >         > no errors. Please let me know whether I'm on the right
> >         track.
> >         >
> >         > Thanks and best regards,
> >         > -Jingwei
> >         >
> >
> >
> >         _______________________________________________
> >         FreeSWITCH-users mailing list
> >         FreeSWITCH-users at lists.freeswitch.org
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From styli1 at hotmail.com  Thu Jan 21 17:10:58 2010
From: styli1 at hotmail.com (styli1 at hotmail.com)
Date: Thu, 21 Jan 2010 17:10:58 -0800
Subject: [Freeswitch-users] Vacation reply
In-Reply-To: <13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
Message-ID: <COL0-MC4-F33AC87B2CCB44670A656DB85620@phx.gbl>

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From jmesquita at freeswitch.org  Thu Jan 21 17:21:14 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 21 Jan 2010 22:21:14 -0300
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
	<1264092620.14614.73.camel@luna.tc.commsmundi.com>
	<13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
Message-ID: <ebbc2d551001211721n51cb5c54m996e0d494842c383@mail.gmail.com>

Jingwei, check my contrib dir. I think it may help you with one FIFO since
we are able there to sign in and sign off dynamic agents as well as
customize what we do when the FIFO makes a call to them.

Regards,
Jo?o Mesquita
FSComm Developer

On Thu, Jan 21, 2010 at 10:05 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:

> Thanks for the reply. All the agents are dynamic and I can't predefine them
> in the config file.
>
> Regards,
> -Jingwei
>
>
> On Fri, Jan 22, 2010 at 12:50 AM, Fran?ois Delawarde <
> fdelawarde at wirelessmundi.com> wrote:
>
>> Why do you need 2 fifos? You could have callback agents connected to the
>> fifo and send incoming calls there, mod_fifo will do the rest.
>>
>> To configure agents for callback:
>> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>>
>> To place a call into a fifo:
>> http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue
>>
>> Fran?ois.
>>
>> On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
>> > Sorry about the confusion, I'm just trying to think over all the
>> > abnormal situations before the implementation and hope the flow has no
>> > design flaws.
>> >
>> > Client A is parked in Queue 1, multiple consumers will be ringed to
>> > answer him. And once the first one is connected, all the rest will
>> > hang up. This is the targeted function.
>> >
>> > To achieve this, I'm considering to originate a call to each consumer
>> > and put the calls in Queue 2. Then intercept client A to the first
>> > element of Queue 2.
>> >
>> > I'm not sure if it's the usual or the best way. If you feel not,
>> > please don't hesitate to correct me. Any thoughts are warmly
>> > appreciated.
>> >
>> > On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com> wrote:
>> >         what no errors mean? how do you originate calls to consumers?
>> >         I don't
>> >         understand your scenario.
>> >
>> >         2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
>> >
>> >         > Hi All,
>> >         >
>> >         > Please advise whether the following flow makes sense.
>> >         >
>> >         > 1. Client A calls in and parked in Queue 1
>> >         > 2. Originate calls to several consumers simultaneously and
>> >         park them in
>> >         > Queue 2
>> >         > 3. Intercept A's call to the first consumer of Queue 2
>> >         >
>> >         > Basically I want A's call picked up by the first among many
>> >         consumers with
>> >         > no errors. Please let me know whether I'm on the right
>> >         track.
>> >         >
>> >         > Thanks and best regards,
>> >         > -Jingwei
>> >         >
>> >
>> >
>> >         _______________________________________________
>> >         FreeSWITCH-users mailing list
>> >         FreeSWITCH-users at lists.freeswitch.org
>> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >         UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >         http://www.freeswitch.org
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jingwei.yang at gmail.com  Thu Jan 21 18:06:22 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Fri, 22 Jan 2010 10:06:22 +0800
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <ebbc2d551001211721n51cb5c54m996e0d494842c383@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
	<1264092620.14614.73.camel@luna.tc.commsmundi.com>
	<13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
	<ebbc2d551001211721n51cb5c54m996e0d494842c383@mail.gmail.com>
Message-ID: <13529f9d1001211806o35fc3435j93b71bbb2ede3028@mail.gmail.com>

Hi Jo?o, thanks for the reply. I'll try it out.

Regards,
-Jingwei

2010/1/22 Jo?o Mesquita <jmesquita at freeswitch.org>

> Jingwei, check my contrib dir. I think it may help you with one FIFO since
> we are able there to sign in and sign off dynamic agents as well as
> customize what we do when the FIFO makes a call to them.
>
> Regards,
> Jo?o Mesquita
> FSComm Developer
>
>
> On Thu, Jan 21, 2010 at 10:05 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>
>> Thanks for the reply. All the agents are dynamic and I can't predefine
>> them in the config file.
>>
>> Regards,
>> -Jingwei
>>
>>
>> On Fri, Jan 22, 2010 at 12:50 AM, Fran?ois Delawarde <
>> fdelawarde at wirelessmundi.com> wrote:
>>
>>> Why do you need 2 fifos? You could have callback agents connected to the
>>> fifo and send incoming calls there, mod_fifo will do the rest.
>>>
>>> To configure agents for callback:
>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>>>
>>> To place a call into a fifo:
>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue
>>>
>>> Fran?ois.
>>>
>>> On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
>>> > Sorry about the confusion, I'm just trying to think over all the
>>> > abnormal situations before the implementation and hope the flow has no
>>> > design flaws.
>>> >
>>> > Client A is parked in Queue 1, multiple consumers will be ringed to
>>> > answer him. And once the first one is connected, all the rest will
>>> > hang up. This is the targeted function.
>>> >
>>> > To achieve this, I'm considering to originate a call to each consumer
>>> > and put the calls in Queue 2. Then intercept client A to the first
>>> > element of Queue 2.
>>> >
>>> > I'm not sure if it's the usual or the best way. If you feel not,
>>> > please don't hesitate to correct me. Any thoughts are warmly
>>> > appreciated.
>>> >
>>> > On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com> wrote:
>>> >         what no errors mean? how do you originate calls to consumers?
>>> >         I don't
>>> >         understand your scenario.
>>> >
>>> >         2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
>>> >
>>> >         > Hi All,
>>> >         >
>>> >         > Please advise whether the following flow makes sense.
>>> >         >
>>> >         > 1. Client A calls in and parked in Queue 1
>>> >         > 2. Originate calls to several consumers simultaneously and
>>> >         park them in
>>> >         > Queue 2
>>> >         > 3. Intercept A's call to the first consumer of Queue 2
>>> >         >
>>> >         > Basically I want A's call picked up by the first among many
>>> >         consumers with
>>> >         > no errors. Please let me know whether I'm on the right
>>> >         track.
>>> >         >
>>> >         > Thanks and best regards,
>>> >         > -Jingwei
>>> >         >
>>> >
>>> >
>>> >         _______________________________________________
>>> >         FreeSWITCH-users mailing list
>>> >         FreeSWITCH-users at lists.freeswitch.org
>>> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >         UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >         http://www.freeswitch.org
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fvillarroel at yahoo.com  Thu Jan 21 20:26:10 2010
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Thu, 21 Jan 2010 20:26:10 -0800 (PST)
Subject: [Freeswitch-users] CDR Gateways
Message-ID: <956716.34674.qm@web34301.mail.mud.yahoo.com>

Dear All.

I have defined various gateways in ~/sip-profiles/external

My questions is if a gateway named foo that send calls from diferents IP address like x.x.x.x and x.x.x.y. How i can doing a group for both ip address where i can doing later a sql like accountcode=foo?

In this moment if i need know the traffic of customer foo, i should doing two differents cdr, one for every ip address.

It's fine or i can doing of different way?

I hope anyone could me explain how i can doing. 

my gateway foo.xml

<include>
    <gateway name="foo">
      <param name="username" value="foo"/>
      <param name="password" value="foo"/>
      <param name="realm" value="x.x.x.x"/>
      <param name="register" value="false"/>
      <param name="context" value="public"/>
    </gateway>
</include>

foo1.xml

<include>
    <gateway name="foo1">
      <param name="username" value="foo1"/>
      <param name="password" value="foo1"/>
      <param name="realm" value="x.x.x.y"/>
      <param name="register" value="false"/>
      <param name="context" value="public"/>
    </gateway>
</include>

Both gateways foo and foo1 are the same customer

my cdr_csv.conf.xml

<configuration name="cdr_csv.conf" description="CDR CSV Format">
  <settings>
    <!-- 'cdr-csv' will always be appended to log-base -->
    <!--<param name="log-base" value="/var/log"/>-->
    <param name="default-template" value="sql"/>
    <!-- This is like the info app but after the call is hung up -->
    <!--<param name="debug" value="true"/>-->
    <param name="rotate-on-hup" value="false"/>
    <!-- may be a b or ab -->
    <param name="legs" value="ab"/>
  </settings>
  <templates>
        <template name="sql">INSERT INTO v_cdr (v_id, caller_id_name, caller_id_number, destination_number, context, start_stamp, answer_stamp, end_stamp,
duration, billsec, hangup_cause, uuid, bleg_uuid, accountcode, read_codec, write_codec, remote_media_ip, network_addr) VALUES ("{v_id}", "${caller_id_name}"
,"${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}"
,"${uuid}","${bleg_uuid}", "${accountcode}", "${read_codec}", "${write_codec}", "${remote_media_ip}", "${network_addr}");</template>

The argument accountcode on my database is Blank or None for all records of gateways.
Regards.


      


From thangappan143 at gmail.com  Thu Jan 21 20:45:58 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Fri, 22 Jan 2010 10:15:58 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
Message-ID: <7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>

The following link have the openzap.conf,openzap.conf.xml ,smg_prid.conf ,
debug log of mod_openzap , oz list and oz dump 1 output.

http://pastebin.org/80095


On Thu, Jan 21, 2010 at 10:34 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> OpenZap is loading the ss7 signalling type. As per your concern openzap
> does not know the details of the signalling then how it is loading the
> ss7_boost libraries?
>
> FreeSWITCH log:
> -----------------------------
> 2010-01-21 10:10:46.707844 [INFO] zap_io.c:2374 Configured 30 channel(s)
> 2010-01-21 10:10:46.708600 [INFO] zap_io.c:2468 Loading SIG from
> /usr/local/freeswitch/mod/ozmod_ss7_boost.so
> 2010-01-21 10:10:46.709031 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost'
> 2010-01-21 10:10:46.709466 [DEBUG] ss7_boost_client.c:124 Creating L=
> 127.0.0.65:53000 R=127.0.0.66:53000
> 2010-01-21 10:10:46.709834 [DEBUG] ss7_boost_client.c:124 Creating L=
> 127.0.0.65:53001 R=127.0.0.66:53001
> 2010-01-21 10:10:46.710424 [WARNING] ss7_boost_client.c:244 TX EVENT (P):
> SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>
> The signalling type might be anything but when I used the oz list command
> it showed the span details. But I am unable to make a inbound and outbound
> call.
>
> Outbound call result:
> ============
> >  originate openzap/smg_prid/a/9940464753 openzap/smg_prid/a/9843171457
> -ERR NORMAL_CIRCUIT_CONGESTION
>
> 2010-01-21 10:26:14.304816 [CRIT] ozmod_ss7_boost.c:244 SPAN is not online.
> freeswitch at internal> 2010-01-21 10:26:14.304816 [ERR] mod_openzap.c:1043
> No channels available
> 2010-01-21 10:26:14.304816 [ERR] switch_ivr_originate.c:1510 Cannot create
> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
> 2010-01-21 10:26:14.304816 [DEBUG] switch_ivr_originate.c:2138 Originate
> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>
> Inbound call result:
> -----------------------------
>
>      I made incoming call for the dial plan which is specified in the
> earlier post at that time it said the number is busy. We did the packet
> capture using the following command.
>
> wanpipemon -i w1g1 -pcap -pcap file isdn.pcap -port ISDN -full -systime -c
> trd
>
> Here I attached the pcap file for that.
>
>
> Where I did mistake or Did I miss any thing to do?
> Please help me.......
>
>
>
> On Wed, Jan 20, 2010 at 7:40 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>>
>>       I noticed the 'oz list' output  in that span type is 'ss7 (boost)'.
>> How can I change this to isdn?
>>
>>
>>
>> On Wed, Jan 20, 2010 at 12:43 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>
>>> I found the error in it. The file name is used as openzap.conf.xml (
>>> smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
>>> PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
>>> openzap.conf.wiki.xml  file.
>>>
>>> Now the another problem is raised here.
>>> When I was using oz list command , the details of the smg_prid shown.
>>> When I was using 'oz dump smg_prid' command it shows all the channels'
>>> details. But all the channels' states are DOWN. why? How can I make it the
>>> states to UP?
>>>
>>> When I was making the call , the number is busy message was get. The call
>>> was not  at all landed to the freeswitch.
>>>
>>> Dial plan Example:
>>> -------------------------------
>>>  <extension name="PRI_Testing">
>>>         <condition field="destination_number" expression="^39114600$">
>>>                 <action application="playback"
>>> data="ivr-welcome_to_freeswitch"/>
>>>         </condition>
>>>   </extension>
>>>
>>> Please help me...........
>>>
>>> *Output Reference:*
>>> http://pastebin.org/79074
>>>
>>>
>>> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>>
>>>> Dear all,
>>>>
>>>>          I have successfully configured wanpipe with freeswitch. When I
>>>> was the running wancfg_fs script the following files openzap.conf ,
>>>> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
>>>> are created.
>>>>
>>>>          I started the wanrouter command then executed the freeswitch.
>>>>          When I was executing freeswitch mod_openzap.c said the error
>>>> as  "Error  for finding the span id. name:PRI_1".
>>>>          But in the openzap.conf and openzap.conf.xml files the span
>>>> name is smg_prid.
>>>>
>>>>          Why the freeswitch is referring the span name as PRI_1 ?
>>>>          Whether this has to configured in anywhere?
>>>>
>>>>          In the freeswitch CLI using oz command I tried to dump the
>>>> PRI_1 span id but it said te error  as "PRI_1 is not found".  When I was
>>>> trying the command 'oz dump smg_prid'  all the channel states and details
>>>> shown.
>>>>
>>>>         It seems that smg_prid span configured in openzap perfectly (Its
>>>> my assumption). Then Why freeswitch is referring the span name as PRI_1.
>>>>
>>>> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>>>>
>>>> Could anyone please help me?
>>>>
>>>> REFERENCE:
>>>>
>>>> openzap.conf
>>>> [span wanpipe smg_prid]
>>>> name => smg_prid
>>>> trunk_type =>e1
>>>> b-channel => 1:1-15
>>>> b-channel => 1:17-31
>>>>
>>>>
>>>> openzap.conf.xml
>>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>>   <settings>
>>>>     <param name="debug" value="0"/>
>>>>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>>     <!--<param name="enable-analog-option" value="3-way"/>-->
>>>>   </settings>
>>>>   <boost_spans>
>>>>         <span id="smg_prid">
>>>>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>         <param name="dialplan" value="XML"/>
>>>>         <param name="context" value="default"/>
>>>>          <!-- regex to stop dialing when it matches -->
>>>>         <!--<param name="dial-regex" value="5555"/>-->
>>>>         <!-- regex to stop dialing when it does not match -->
>>>>         <!--<param name="fail-dial-regex" value="^5"/>-->
>>>>         </span>
>>>>  </boost_spans>
>>>>  <analog_spans>
>>>>  </analog_spans>
>>>> </configuration>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Thangappan.M
>>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Thangappan.M
>>>
>>
>>
>>
>> --
>> Regards,
>> Thangappan.M
>>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From troy at tlainvestments.com  Thu Jan 21 20:57:49 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 21 Jan 2010 21:57:49 -0700
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
In-Reply-To: <191c3a031001211648i5ed86253k8640c49f29121e0c@mail.gmail.com>
References: <37012.20543.qm@web33507.mail.mud.yahoo.com>
	<5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>
	<517DA285-7E9B-4226-8AEF-C2DAB52976CF@tlainvestments.com>
	<004e01ca9af3$6c5a2d20$450e8760$@net>
	<005b01ca9af5$dc759980$9560cc80$@net>
	<191c3a031001211648i5ed86253k8640c49f29121e0c@mail.gmail.com>
Message-ID: <8589C894-EE61-4FA7-92E1-5CB9C52EBD60@tlainvestments.com>

I get the idea, but can't seem to get it to work.

I tried doing a bridge to "loopback/app=bridge ${sofia_contact(${dialed_user}@${dialed_domain}", but fs complained => Cannot create outgoing channel of type [loopback=app:sofia]
Also, I tried "loopback/app=voicemail:default ${domain_name} ${dialed_extension}" and it did get to voicemail, but it didn't prompt me for any info - just immediately complained that the recording was too short.

I tried the suggestions about modifying the sip.cfg for the phones and that does work (thanks!), but that forces a Consultative Transfer.  It would be nice to get this method working, which would result in the transfer button doing a Consultive Transfer unless you hang up, then it would be like a blind transfer.

Thanks,
Troy

On Jan 21, 2010, at 5:48 PM, Anthony Minessale wrote:

> if you used the loopback endpoint to loop around to voicemail or made a looped sip call back to your own box you could xfer it as desired.
> 
> 
> bridge to "loopback/app=voicemail:default ${domain_name} ${dialed_extension}"
> 
> That will make the vm app run as a channel instead of an inline app.
> 
> This is an undocumented feature because it's not well tested so if it doesn't work *shrug* =D
> 
> 
> 
> On Thu, Jan 21, 2010 at 6:00 PM, Adam Ford <lists at redbonez.net> wrote:
> That link didn't come through very well, here is a shortened one -
> http://bit.ly/6wDAXD
> 
> -Adam
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam
> Ford
> Sent: Thursday, January 21, 2010 4:43 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
> 
> Yes it is a known issue with Polycom phones. Polycom supports a non-standard
> transfer method which does not work with FreeSWITCH.
> 
> See this article for further details -
> http://www.junctionnetworks.com/knowledgebase/onsip/phones-routers-and-devic
> es/phone-configuration/polycom/polycom-disabling-non-standard-transfer-meth
> 
> I ran into the same problem, disabling
> voIpProt.SIP.allowTransferOnProceeding as suggested in that article resolved
> the issue for me.
> 
> -Adam
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Troy
> Anderson
> Sent: Thursday, January 21, 2010 4:21 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
> 
> Hello,
> 
> I'm on the latest trunk version (16440) and having an issue with Polycom and
> transferring.  The dial plan is set up so that unanswered calls go to
> voicemail.  When I answer a call with a polycom phone and then transfer that
> call to another phone, if the other phone doesn't pick up and the voicemail
> app starts, then I hit transfer again with the intent of having the caller
> leave a voicemail, the call is dropped.  If the phone does pick up during
> the transfer, it works fine.
> 
> I also have an Aastra phone, and when I do the same thing, but from the
> Aastra phone, it works as expected.  Is this known to be a problem with
> Polycom?
> 
> Thanks!
> Troy
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lakindia89 at gmail.com  Thu Jan 21 21:50:02 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 22 Jan 2010 11:20:02 +0530
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <191c3a031001190804m692f63acsaf852b8809db09d1@mail.gmail.com>
References: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
	<191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>
	<7d79b3931001132357i36bbb482jdf2bbdd3aea2a583@mail.gmail.com>
	<7d79b3931001180322p74a1227qe0c2199a77cbfe2@mail.gmail.com>
	<191c3a031001180824q6e364c72g3f789892597e9469@mail.gmail.com>
	<7d79b3931001190107o200ec04dredd76689dd235588@mail.gmail.com>
	<191c3a031001190804m692f63acsaf852b8809db09d1@mail.gmail.com>
Message-ID: <7d79b3931001212150q2533f49l1725ee1d9cd5848f@mail.gmail.com>

Hi all,
I've solved that problem by adding.

use POSIX;
POSIX::sigaction( SIGINT, POSIX::SigAction->new( \&Handler, 0,
POSIX::SA_RESTART),);

This will restart the system calls if that is failed because of the SIGINT
signal.

Provided here as an information...
Thanks all...


On Tue, Jan 19, 2010 at 9:34 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Its nothing we can fix, that is what you must do on a failed read syscall.
> you can do a non blocking read instead and take your chances.
>
>
>
> On Tue, Jan 19, 2010 at 3:07 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> I tried with SIGUSR1, but no progress. I got SERVER_DISCONNECTED.
>> Output:
>>
>> CHILD 3814: Received USR1
>> EVENT [SERVER_DISCONNECTED]
>>
>> In esl.c, in esl_recv_event() function, line no: 824
>> if (rrval < 0) {
>>                         strerror_r(handle->errnum, handle->err,
>> sizeof(handle->err));
>>                         goto fail;
>> }
>> When the program is blocked under receive, I passed the signal. So recv
>> returns -1, and in fail: it call esl_disconnect(handle).
>>
>> Is it because of this??? If so, whether it should be fixed or not???
>>
>>
>>
>>
>> On Mon, Jan 18, 2010 at 9:54 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> try a less famous signal like SIGUSR1 it's possible something in perl
>>> still reacts to SIGINT
>>>
>>>
>>>
>>> On Mon, Jan 18, 2010 at 5:22 AM, lakshmanan ganapathy <
>>> lakindia89 at gmail.com> wrote:
>>>
>>>> Here is the result
>>>>
>>>> Program:
>>>>
>>>> require ESL;
>>>> use IO::Socket::INET;
>>>> use Data::Dumper;
>>>>
>>>> my $ip = "192.168.1.222";
>>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>>> die "Could not create socket: $!\n" unless $sock;
>>>>
>>>>
>>>> for(;;) {
>>>>     my $new_sock = $sock->accept();
>>>>     next if (not defined ($new_sock));
>>>>     my $pid = fork();
>>>>     if ($pid) {
>>>>         close($new_sock);
>>>>         next;
>>>>     }
>>>>     print "CHILD PID: $$\n";
>>>>     &register_Signals_Child();
>>>>     sub register_Signals_Child() {
>>>>         foreach ( keys %SIG ) {
>>>>             $SIG{$_} = 'Handler';
>>>>         }
>>>>     }
>>>>
>>>>     sub Handler() {
>>>>
>>>>         my $handle=$_[0];
>>>>         if($handle eq "INT") {
>>>>             print "CHILD $$: SIGNAL SIG$handle is generated\n";
>>>>         }
>>>>         else
>>>>         {
>>>>             print "CHILD $$: Received $handle\n";
>>>>
>>>>         }
>>>>     }
>>>>     my $host = $new_sock->sockhost();
>>>>     my $fd = fileno($new_sock);
>>>>
>>>>     my $con = new ESL::ESLconnection($fd);
>>>>     my $info = $con->getInfo();
>>>>
>>>>     my $uuid = $info->getHeader("unique-id");
>>>>
>>>>     printf "Connected call %s, from %s\n", $uuid,
>>>> $info->getHeader("caller-caller-id-number");
>>>>     my $r=$con->execute("answer");
>>>>     print Dumper $r;
>>>>     $con->events("plain","all");
>>>>     my
>>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>>     while($con->connected()) {
>>>>         my $e = $con->recvEvent();
>>>>
>>>>         if ($e) {
>>>>             my $name = $e->getHeader("event-name");
>>>>             print "EVENT [$name]\n";
>>>>             if ($name eq "DTMF") {
>>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>>                 print "$digit\n";
>>>>             }
>>>>         }
>>>>     }
>>>>     close($new_sock);
>>>> }
>>>>
>>>> I executed the program and the following things were printed
>>>>
>>>> CHILD PID: 6778
>>>> Connected call e0d1001a-03f4-11df-b002-db488337e0ea, from 1001
>>>> $VAR1 = 0;
>>>> EVENT [CHANNEL_EXECUTE]
>>>> EVENT [CHANNEL_ANSWER]
>>>> EVENT [CHANNEL_EXECUTE_COMPLETE]
>>>> EVENT [COMMAND]
>>>> EVENT [CHANNEL_EXECUTE]
>>>> EVENT [HEARTBEAT]
>>>> EVENT [RE_SCHEDULE]
>>>> EVENT [CHANNEL_EXECUTE_COMPLETE]
>>>>
>>>> Then from another shell I executed kill -2 6778, the result is follows
>>>> CHILD 6778: SIGNAL SIGINT is generated
>>>> EVENT [SERVER_DISCONNECTED]
>>>>
>>>> But the child process is still running as expected.
>>>> But  I don't know why I received SERVER_DISCONNECTED from freeswitch???
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Thu, Jan 14, 2010 at 1:27 PM, lakshmanan ganapathy <
>>>> lakindia89 at gmail.com> wrote:
>>>>
>>>>> I taught the signal handler will be inherited by the child process. It
>>>>> also does like that.
>>>>> After making a call, If I press ctrl + c, the above program printed
>>>>> PARENT PID: Signal SIGINT is generated
>>>>> CHILD PID: Signal SIGINT is generated.
>>>>>
>>>>> So I think the sigal handlers will be inherited to the child.
>>>>> Anyway I'll also try registering signal handlers in child also, and
>>>>> then I'll come back with that result.
>>>>>
>>>>> Thanks....
>>>>> On Wed, Jan 13, 2010 at 9:48 PM, Anthony Minessale <
>>>>> anthony.minessale at gmail.com> wrote:
>>>>>
>>>>>> you would have to register signals in your child process too
>>>>>>
>>>>>>   On Wed, Jan 13, 2010 at 3:13 AM, lakshmanan ganapathy <
>>>>>> lakindia89 at gmail.com> wrote:
>>>>>>
>>>>>>>  Hi all,
>>>>>>>
>>>>>>> I've done a sample program (In perl ESL) , which play a file to the
>>>>>>> caller and then it will call recvEvent() and print the event name. I've
>>>>>>> handled signals also.
>>>>>>>
>>>>>>> When I send SIGINT to my program (Perl), the signal handler is called
>>>>>>> and I can see the print output. But in the same time,  I received
>>>>>>> SERVER_DISCONNECTED from freeswitch as event.
>>>>>>>
>>>>>>> I don't know why I received SERVER_DISCONNECTED from freeswitch. Is
>>>>>>> it because, the recvEvent() from perl internally  calls the recvevent
>>>>>>> function in the Esl.c and when it waits to receive the information from
>>>>>>> socket, the signal occurred???
>>>>>>>
>>>>>>> Please clarify me!!
>>>>>>>
>>>>>>> Here is my program
>>>>>>> require ESL;
>>>>>>> use IO::Socket::INET;
>>>>>>> use Data::Dumper;
>>>>>>>
>>>>>>> my $ip = "192.168.1.222";
>>>>>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>>>>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>>>>>> die "Could not create socket: $!\n" unless $sock;
>>>>>>> &register_Signals();
>>>>>>>
>>>>>>> for(;;) {
>>>>>>>     my $new_sock = $sock->accept();
>>>>>>>     next if (not defined ($new_sock));
>>>>>>>     my $pid = fork();
>>>>>>>     if ($pid) {
>>>>>>>         close($new_sock);
>>>>>>>         next;
>>>>>>>     }
>>>>>>>     print "CHILD PID: $$\n";
>>>>>>>     my $host = $new_sock->sockhost();
>>>>>>>     my $fd = fileno($new_sock);
>>>>>>>
>>>>>>>     my $con = new ESL::ESLconnection($fd);
>>>>>>>     my $info = $con->getInfo();
>>>>>>>
>>>>>>>     my $uuid = $info->getHeader("unique-id");
>>>>>>>
>>>>>>>     printf "Connected call %s, from %s\n", $uuid,
>>>>>>> $info->getHeader("caller-caller-id-number");
>>>>>>>     my $r=$con->execute("answer");
>>>>>>>     print Dumper $r;
>>>>>>>     $con->events("plain","all");
>>>>>>>     my
>>>>>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>>>>>
>>>>>>>     while($con->connected()) {
>>>>>>>         my $e = $con->recvEvent();
>>>>>>>
>>>>>>>         if ($e) {
>>>>>>>             my $name = $e->getHeader("event-name");
>>>>>>>             print "EVENT [$name]\n";
>>>>>>>             if ($name eq "DTMF") {
>>>>>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>>>>>                 print "$digit\n";
>>>>>>>             }
>>>>>>>         }
>>>>>>>     }
>>>>>>>     close($new_sock);
>>>>>>> }
>>>>>>> sub register_Signals() {
>>>>>>>     foreach ( keys %SIG ) {
>>>>>>>         $SIG{$_} = 'sig_Handler';
>>>>>>>     }
>>>>>>> }
>>>>>>>
>>>>>>> sub sig_Handler() {
>>>>>>>     my $handle=$_[0];
>>>>>>>     if($handle eq "INT") {
>>>>>>>         print "$$: SIGNAL SIG$handle is generated\n";
>>>>>>>     }
>>>>>>> }
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Jan 21 22:00:11 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 22 Jan 2010 00:00:11 -0600
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
In-Reply-To: <191c3a031001212158s4c574324m4c8545f238967907@mail.gmail.com>
References: <37012.20543.qm@web33507.mail.mud.yahoo.com>
	<5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>
	<517DA285-7E9B-4226-8AEF-C2DAB52976CF@tlainvestments.com>
	<004e01ca9af3$6c5a2d20$450e8760$@net>
	<005b01ca9af5$dc759980$9560cc80$@net>
	<191c3a031001211648i5ed86253k8640c49f29121e0c@mail.gmail.com>
	<8589C894-EE61-4FA7-92E1-5CB9C52EBD60@tlainvestments.com>
	<191c3a031001212158s4c574324m4c8545f238967907@mail.gmail.com>
Message-ID: <191c3a031001212200k3744d2bctb0e54eb41897ee74@mail.gmail.com>

You don't need loopback if you were calling over sofia back to your own box,
you just need to use a dest that could be regexed to right to vm when it
comes back around. The loopback to an app should work you probably just have
a syntax err

On Jan 21, 2010 11:03 PM, "Troy Anderson" <troy at tlainvestments.com> wrote:

I get the idea, but can't seem to get it to work.

I tried doing a bridge to
"loopback/app=bridge ${sofia_contact(${dialed_user}@${dialed_domain}", but
fs complained => Cannot create outgoing channel of type [loopback=app:sofia]
Also, I tried "loopback/app=voicemail:default ${domain_name}
${dialed_extension}" and it did get to voicemail, but it didn't prompt me
for any info - just immediately complained that the recording was too short.

I tried the suggestions about modifying the sip.cfg for the phones and that
does work (thanks!), but that forces a Consultative Transfer.  It would be
nice to get this method working, which would result in the transfer button
doing a Consultive Transfer unless you hang up, then it would be like a
blind transfer.

Thanks,
Troy

On Jan 21, 2010, at 5:48 PM, Anthony Minessale wrote: > if you used the
loopback endpoint to loop...

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From mailinglist at fribert.dk  Fri Jan 22 03:14:41 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Fri, 22 Jan 2010 12:14:41 +0100
Subject: [Freeswitch-users] Svar: Re: Svar: Re: Home setup with home	 company
Message-ID: <4B5996B1020000E100000404@mail.fribert.dk>

Ok, I set it up like this:

   
     
      
         
      
         
         
         
         
         
         
         
         
         
      


But now it gives me:

2010-01-22 11:52:08.667564 [NOTICE] switch_channel.c:602 New Channel sofia/external/2680xxxx at 87.54.25.116 [16baec2f-4407-df11-8fb3-000c29b7b4cb]
2010-01-22 11:52:08.800123 [INFO] mod_dialplan_xml.c:252 Processing 2680xxxx->4692xxxx in context public
2010-01-22 11:52:08.830071 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/2680xxxx at 87.54.25.116 to XML[8203 at default]
2010-01-22 11:52:08.830071 [INFO] mod_dialplan_xml.c:252 Processing 2680xxxx->8203 in context default
2010-01-22 11:52:08.830071 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/2680xxxx at 87.54.25.116 to enum[8203 at default]
2010-01-22 11:52:09.163439 [INFO] switch_core_state_machine.c:136 No Route, Aborting

huh


>>> Michael Collins  21-01-10 9:39 >>>


On Wed, Jan 20, 2010 at 2:45 AM, mailinglist <mailinglist at fribert.dk> wrote:
Hi Michael

It's running on pfsense, so it's kinda locked to the version it currently is.
Looks very nice though.
Looking beyond that, is the action / anti-action list corrent?


I would say that you need to add an anti-action under the day of week check and go to vm if it does not match. Right now if the DOW is 0 or 6 then the entire extension will "fail" and the dialplan will just move on. Remember that if any conditions fail then the entire thing extension "fails" unless you are doing interesting things with the break= parameter. See the dialplan page on the wiki for examples of how to use break in your conditions.
-MC



Best regards
Fribse


>>> Michael Collins  20-01-10 1:53 >>>


On Tue, Jan 19, 2010 at 9:02 AM, mailinglist <mailinglist at fribert.dk> wrote:
I have a very small one man constultancy company that has the occasional call, unfortunately we are getting more spam calls after hours than real calls during work hours, so I would like to set up a TOD system.
 
First step for me is just playing with the TOD example, I've gotten this far:
 

   
      
      
      
         
         
         
         
         group/company@${domain_name}"/>
         
         
         
         
      


My idea with this, was that in the time 9-17 mon-fri, it rings, the rest of the time it's sent directly to the voicemail.
I would of course like to have it not take messages outside work hours, but that's just refining :-)
 
But it picks up the call, and then nothing...
 

We have a much cleaner way of doing TOD and DOW handling. You'll need to get to the latest FreeSWITCH version. Look in conf/dialplan/default.xml for this example:



    
    
      
        
      
    


Use that condition instead of the two conditions you're now using and see if you have better success.

-MC


_______________________________________________
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From mayamatakeshi at gmail.com  Fri Jan 22 03:22:00 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Fri, 22 Jan 2010 20:22:00 +0900
Subject: [Freeswitch-users] Distinguishing Blind REFER from application
	transfer
Message-ID: <15b9404e1001220322h5636c1aat968f515e0601d769@mail.gmail.com>

Hello,
I'm using mod_xml_curl to provide dialplan.
In my application, I need to differentiate if a call has entered the
dialplan again due to uuid_transfer or due to Blind Transfer.
I know I can recognize a Blind Transfer by checking
variable_sip_h_Referred-By and variable_sip_refer_to.
However, these variables are not cleaned up when the call reenters the
dialplan due to application transfer or uuid_transfer.

I realize I can differentiate them by adding some prefix like this:
uuid_transfer TRANSFER,DestinationNumber XML default

Or it is my responsibility to call
application set or uuid_setvar to unset the variable(s) in this case?
it seems to me the bad thing here would be to have to add lots of preventive

 <action application="set" data="sip_refer_to=" />
in my dialplan.

Before doing so I thought in ask if the REFER-related variables being
preserved upon dialplan reentry would not be a bug (well, it could be a
feature useful in some scenarios I suspect).

br,
takeshi
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From mailinglist at fribert.dk  Fri Jan 22 04:12:46 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Fri, 22 Jan 2010 13:12:46 +0100
Subject: [Freeswitch-users] Svar: Re: How do I invite group to join	
	existing call?
Message-ID: <4B59A44E020000E100000413@mail.fribert.dk>

Hi Michael et al.

I found the 'attended xfer' example in the documentation, and it looks like rxactly to what I want to do.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer

The example 2:
Do I add the first part to my dialplan/default.xml, and the second part to my dialplan/features.xml

How does it affect if I try to do a normal local dial if it just reacts on the fact that three digits have been dialed (which I should change to 2, as I use 2 digit local numbers).

Best regards
Fribse

>>> Michael Collins  18-01-10 21:19 >>>


On Sun, Jan 17, 2010 at 10:50 PM, mailinglist <mailinglist at fribert.dk> wrote:
Hi All

I would like to be able to invite a group / global to join an existing call, but how do I accomplish this, can it be done?


Malfunction! Need Input!

Could you give us a few more details on what you're trying to accomplish? What's the big picture? I'm curious what problem you're trying to solve. I'm sure the gang here will have thoughts to pass along. :)

-MC




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From irmatov at gmail.com  Fri Jan 22 05:22:21 2010
From: irmatov at gmail.com (Timur Irmatov)
Date: Fri, 22 Jan 2010 18:22:21 +0500
Subject: [Freeswitch-users] mod_erlang_event: disconnects
In-Reply-To: <20100121134241.GD1036@hijacked.us>
References: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>
	<20100121134241.GD1036@hijacked.us>
Message-ID: <241d382f1001220522g27f67ef5p5ba86ebb3afe63c@mail.gmail.com>

On Thu, Jan 21, 2010 at 6:42 PM, Andrew Thompson <andrew at hijacked.us> wrote:
>> The problem is, that pretty frequently processes which handle incoming
>> calls receive messages like {'EXIT', <5406.48.0>, noconnection} from
>> FreeSWITCH. As I understand from googling, this happens when remote C
>> node disconnects (and I see TCP connections from FreeSWITCH to epmd
>> daemon being torn down and reestablished). FreeSWITCH drops calls at
>> that moment.
>
> Does it drop ALL calls being handled in erlang, or just that one?

It seems at it drops all calls handled in erlang. At the moment we
have only 7 calls maximum, and my applications logs several such exits
happening at exactly same time, up to 7 at once.

>> Have anyone seen this? Is there any fix/ advice?
>
> I haven't seen this before, how many calls are involved? I'm willing to
> help you troubleshoot though. Is there anything relevant in the logs
> (even at DEBUG)?

It is good and bad to know that you haven't seen this before.. :)
Good, because then it seems like local anomaly which, hopefully, can
be debugged and fixed. Bad, because it means we should debug it and
there's no ready fix.

As for the load, it is small. FreeSWITCH has 7 sip registrations to
our upstream, so 7 simultaneous calls is maximum.

I will send you logs offlist. This is small excerpt:

2010-01-22 16:40:00.059202 [DEBUG] mod_sofia.c:293
sofia/external/1504291 at 10.0.2.5 SOFIA DESTROY
2010-01-22 16:40:00.059202 [DEBUG] switch_core_state_machine.c:60
sofia/external/1504291 at 10.0.2.5 Standard DESTROY
2010-01-22 16:40:00.059202 [DEBUG] switch_core_state_machine.c:434
(sofia/external/1504291 at 10.0.2.5) State DESTROY going to sleep
2010-01-22 16:40:00.091201 [WARNING] mod_erlang_event.c:489 Can't
locate session df4f60c0-074a-11df-af37-b1ee9ca9c744
2010-01-22 16:40:00.091201 [DEBUG] mod_erlang_event.c:541 Notifying
new session failed
2010-01-22 16:40:00.091201 [DEBUG] mod_erlang_event.c:842
check_attached_sessions requested exit
2010-01-22 16:40:00.091201 [DEBUG] mod_erlang_event.c:920 Session
complete, waiting for children
2010-01-22 16:40:00.091201 [DEBUG] mod_erlang_event.c:930 Connection Closed
2010-01-22 16:40:00.095844 [DEBUG] mod_erlang_event.c:1723 Launching
listener, connection from node erlswitch at localhost, ip 127.0.0.1
2010-01-22 16:40:00.095844 [DEBUG] mod_erlang_event.c:910 Connection
Open from 127.0.0.1
2010-01-22 16:40:00.095844 [DEBUG] switch_core_session.c:857 Send
signal sofia/internal/sip:106 at 192.168.1.107:5060 [BREAK]
2010-01-22 16:40:00.095844 [DEBUG] switch_core_session.c:857 Send
signal sofia/internal/sip:108 at 192.168.1.108 [BREAK]
2010-01-22 16:40:00.102708 [DEBUG] switch_ivr_bridge.c:315
sofia/internal/sip:108 at 192.168.1.108 receive message [UNBRIDGE]

>> My system is Debian Lenny (5.0.3), 64-bit system, erlang installed
>> from Debian packages, no backports.
> What OTP release does that equate to, R12 or R13?

I guess this corresponds to R12B3:

$ apt-cache show erlang-nox|grep Version
Version: 1:12.b.3-dfsg-4

May be I should try to build latest erlang from source, rebuild
FreeSWITCH and see if it helps..

>Also, what FS version are you running?

'version' output in fs_cli does not reveal it's version:
freeswitch at internal> version
FreeSWITCH Version 1.0.trunk (hacked)

It was build from svn, i guess it is revision 16041.


-- 
Timur Irmatov, xmpp:irmatov at jabber.ru


From brian at freeswitch.org  Fri Jan 22 06:17:45 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 22 Jan 2010 08:17:45 -0600
Subject: [Freeswitch-users] Distinguishing Blind REFER from application
	transfer
In-Reply-To: <15b9404e1001220322h5636c1aat968f515e0601d769@mail.gmail.com>
References: <15b9404e1001220322h5636c1aat968f515e0601d769@mail.gmail.com>
Message-ID: <9F3D655F-BF09-4503-9709-953A839DC65C@freeswitch.org>

Have you done a uuid_dump to see all the variables?

/b

On Jan 22, 2010, at 5:22 AM, mayamatakeshi wrote:

> Before doing so I thought in ask if the REFER-related variables being preserved upon dialplan reentry would not be a bug (well, it could be a feature useful in some scenarios I suspect).



From dftoro at yahoo.com  Fri Jan 22 06:18:18 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Fri, 22 Jan 2010 06:18:18 -0800 (PST)
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>
Message-ID: <984278.36075.qm@web33504.mail.mud.yahoo.com>

Hi, with svn r16440 the problem persists, I creted a jira report http://jira.freeswitch.org/browse/LBSNDF-8 this is a minor issue, but activing playback delimiter no audio file can be played. On FS the audio files are placed in the \sound\ directory, building the path on Windows would be \sound '\s' which is replaced by 'ound'.

Thank you

Diego Toro
http://lacarretade.blogspot.com/


--- On Thu, 1/21/10, Michael Jerris <mike at jerris.com> wrote:

> From: Michael Jerris <mike at jerris.com>
> Subject: Re: [Freeswitch-users] mutiple playback files (unescape_char) Windows
> To: freeswitch-users at lists.freeswitch.org
> Date: Thursday, January 21, 2010, 5:56 PM
> How about with svn
> r16440
> On Jan 21, 2010, at 8:02
> AM, Diego Toro wrote:
> Hi MikeJ, using '\\' the
> behavior is the same, '\\s' is replaced by
> ' '.
> 
> <action application="playback"
> data="misc\\8000\\serror.wav!misc\\8000\\provide_reference_number.wav!digits\\8000\\5.wav"/>
> ?
> 
> Console output error is:
> 
> [ERR] mod_sndfile.c:194 Error Opening File
> [d:\fs\fs_trunk_20100118\Debug\sounds\es\co\callie\misc\8000
> error.wav]
> 
> 
> S.O.: Windows 7
> FreeSwitch: Trunk (svn latest version)
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From andrew at hijacked.us  Fri Jan 22 07:46:58 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Fri, 22 Jan 2010 10:46:58 -0500
Subject: [Freeswitch-users] mod_erlang_event: disconnects
In-Reply-To: <241d382f1001220522g27f67ef5p5ba86ebb3afe63c@mail.gmail.com>
References: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>
	<20100121134241.GD1036@hijacked.us>
	<241d382f1001220522g27f67ef5p5ba86ebb3afe63c@mail.gmail.com>
Message-ID: <20100122154658.GC25693@hijacked.us>

Give this patch a shot:

http://eagle.bsd.st/~andrew/erlang_session_fix.diff

And see if it makes a difference.

Andrew


From mayamatakeshi at gmail.com  Fri Jan 22 07:47:34 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Sat, 23 Jan 2010 00:47:34 +0900
Subject: [Freeswitch-users] Distinguishing Blind REFER from application
	transfer
In-Reply-To: <9F3D655F-BF09-4503-9709-953A839DC65C@freeswitch.org>
References: <15b9404e1001220322h5636c1aat968f515e0601d769@mail.gmail.com>
	<9F3D655F-BF09-4503-9709-953A839DC65C@freeswitch.org>
Message-ID: <15b9404e1001220747y79571117qab826152a2df1dc8@mail.gmail.com>

On Fri, Jan 22, 2010 at 11:17 PM, Brian West <brian at freeswitch.org> wrote:

> Have you done a uuid_dump to see all the variables?
>

I just tried that with trunk. I can see the REFER variables stay set till
the end of the call.
They will show up in CHANNEL_HANGUP_COMPLETE:

variable_sip_h_Referred-By: user2 <sip:user2 at test1.com<sip%3Auser2 at test1.com>
>
variable_sip_refer_to: <sip:2000 at test1.com <sip%3A2000 at test1.com>>

I suppose the only thing that will change them is another blind refer. But
they will never be unset.


>
> On Jan 22, 2010, at 5:22 AM, mayamatakeshi wrote:
>
> > Before doing so I thought in ask if the REFER-related variables being
> preserved upon dialplan reentry would not be a bug (well, it could be a
> feature useful in some scenarios I suspect).
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From robert.hadley at teotech.com  Fri Jan 22 08:21:55 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 22 Jan 2010 08:21:55 -0800
Subject: [Freeswitch-users] No external sangoma calls running FS as daemon
Message-ID: <B654371338324F17A504DF26271798B9@greyhawk.tonecommander.com>

I have freeswitch running as daemon as user freeswitch.  Internal calls
work.  I get 503 service unavailable when I make external calls through
sangoma.  If I change the FS daemon to run as root then external sangoma
calls work.  Does anybody know what permissions I need to fix?

 

Thanks,

Robert

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From msc at freeswitch.org  Fri Jan 22 08:29:46 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 22 Jan 2010 08:29:46 -0800
Subject: [Freeswitch-users] FreeSWITCH Weekly Call Starting in 30 min
Message-ID: <87f2f3b91001220829k32f60609kbd76c07c93db435e@mail.gmail.com>

Hello all,

The weekly conference call will be starting in a bit. Here's the agenda:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_22

I'll be on just after 9am my time as I have to get my kids off to school.
Talk to you all soon.

-Michael
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From moises.silva at gmail.com  Fri Jan 22 09:45:54 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 22 Jan 2010 12:45:54 -0500
Subject: [Freeswitch-users] No external sangoma calls running FS as
	daemon
In-Reply-To: <B654371338324F17A504DF26271798B9@greyhawk.tonecommander.com>
References: <B654371338324F17A504DF26271798B9@greyhawk.tonecommander.com>
Message-ID: <c4d05cbe1001220945v265ee803x3969595ece7e218b@mail.gmail.com>

On Fri, Jan 22, 2010 at 11:21 AM, Robert Hadley
<robert.hadley at teotech.com>wrote:

>  I have freeswitch running as daemon as user freeswitch.  Internal calls
> work.  I get 503 service unavailable when I make external calls through
> sangoma.  If I change the FS daemon to run as root then external sangoma
> calls work.  Does anybody know what permissions I need to fix?
>

Check the FS logs and pastebin a call attempt. In any case, my guess is
/dev/wanpipe* devices permissions must be changed using udev.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From oscav at hotmail.fr  Fri Jan 22 09:53:25 2010
From: oscav at hotmail.fr (Oscav)
Date: Fri, 22 Jan 2010 09:53:25 -0800 (PST)
Subject: [Freeswitch-users] Script ends when originate receives
 INVALID_NUMBER_FORMAT
Message-ID: <27277429.post@talk.nabble.com>


Hi,

My script ends when I received a 484 INVALID_NUMBER_FORMAT, and doesn't
continue even the script or a hangup Hook.

Here is the script :

session.setVariable("hangup_after_bridge",false);
...
session.setVariable("continue_on_fail","true");
...
session.preAnswer();
...
new_session = new Session(route,session);
new_session.waitForAnswer(15000);
if (new_session.ready()) {	
  bridge(session, new_session);
}
...
console_log("info","call ended")
...

Is there something missing on my script ??
Thanks.
-- 
View this message in context: http://old.nabble.com/Script-ends-when-originate-receives-INVALID_NUMBER_FORMAT-tp27277429p27277429.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From brian at freeswitch.org  Fri Jan 22 09:54:22 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 22 Jan 2010 11:54:22 -0600
Subject: [Freeswitch-users] IAX2 Support Removed.
Message-ID: <0C2714A2-355A-42C7-B589-F0704D436607@freeswitch.org>

Due to lack of support for the libiax2 being updated to support the newer protocol changes and the lack of interest from anyone willing to actually work on it.  I have moved mod_iax to unsupported where it will stay until someone steps up to rewrite a new IAX2 lib.  

Thanks,
Brian

From jerry.richards at teotech.com  Fri Jan 22 10:17:18 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 22 Jan 2010 10:17:18 -0800
Subject: [Freeswitch-users] Wanpipe Driver Install Before FS Install
In-Reply-To: <mailman.5417.1264166574.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.5417.1264166574.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <2D7783AC2C3A4E2E8E0222572286DE51@greyhawk.tonecommander.com>


Is it true that the Sangoma Wanpipe Driver should be installed before
Freeswitch, because the Freeswitch build will autodetect the wanpipe
drivers?

Thanks,
Jerry



From moises.silva at gmail.com  Fri Jan 22 10:33:15 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 22 Jan 2010 13:33:15 -0500
Subject: [Freeswitch-users] Wanpipe Driver Install Before FS Install
In-Reply-To: <2D7783AC2C3A4E2E8E0222572286DE51@greyhawk.tonecommander.com>
References: <mailman.5417.1264166574.2495.freeswitch-users@lists.freeswitch.org>
	<2D7783AC2C3A4E2E8E0222572286DE51@greyhawk.tonecommander.com>
Message-ID: <c4d05cbe1001221033y73aa99del14194540d818a041@mail.gmail.com>

On Fri, Jan 22, 2010 at 1:17 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> Is it true that the Sangoma Wanpipe Driver should be installed before
> Freeswitch, because the Freeswitch build will autodetect the wanpipe
> drivers?
>

If you want to use mod_openzap ( the endpoint used by FreeSWITCH to make
calls using analog and TDM telephony) and you are using Sangoma cards, yes,
you need to install the Wanpipe drivers before even boostraping (./boostrap)
FreeSWITCH. OpenZAP will look during the ./configure state for libsangoma
(which is installed along with the drivers) and then when compiling it will
need the Wanpipe headers, if not found, ozmod_wanpipe will not be compiled
and FreeSWITCH will not have Sangoma cards support for analog and TDM.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From Prometheus001 at gmx.net  Fri Jan 22 11:06:51 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 22 Jan 2010 20:06:51 +0100
Subject: [Freeswitch-users] skypiax and xml-curl
Message-ID: <4B59F74B.1010201@gmx.net>

Hello,

is there a way to manage skypiax via XML-curl besides the dialplan?

Best regards
Peter



From mailinglist at fribert.dk  Fri Jan 22 12:50:17 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Fri, 22 Jan 2010 21:50:17 +0100
Subject: [Freeswitch-users] Svar: Re: How do I invite group to join		
	existing call?
In-Reply-To: <4B59A44E020000E100000413@mail.fribert.dk>
References: <4B59A44E020000E100000413@mail.fribert.dk>
Message-ID: <4B5A1D99020000E100000418@mail.fribert.dk>

Hmm, I don't get it, it might not do the right thing.
 
The situation is that I receive a call from the outside, answers it on a phone, and then wants to ask a third (local) party to join the conversation.
 
I thought from the example that I should just press *3, and then the extension I want to invite, but nothing happens.
 
I haven't the faintest how I accomplish this :-o


>>> 22-01-2010 kl. 13:12 skrev "mailinglist" <mailinglist at fribert.dk> i meddelelsen <4B59A44E020000E100000413 at mail.fribert.dk>:

Hi Michael et al.

I found the 'attended xfer' example in the documentation, and it looks like rxactly to what I want to do.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer

The example 2:
Do I add the first part to my dialplan/default.xml, and the second part to my dialplan/features.xml

How does it affect if I try to do a normal local dial if it just reacts on the fact that three digits have been dialed (which I should change to 2, as I use 2 digit local numbers).

Best regards
Fribse

>>> Michael Collins 18-01-10 21:19 >>>


On Sun, Jan 17, 2010 at 10:50 PM, mailinglist <mailinglist at fribert.dk> wrote:


Hi All

I would like to be able to invite a group / global to join an existing call, but how do I accomplish this, can it be done?


Malfunction! Need Input!

Could you give us a few more details on what you're trying to accomplish? What's the big picture? I'm curious what problem you're trying to solve. I'm sure the gang here will have thoughts to pass along. :)

-MC

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From jerry.richards at teotech.com  Fri Jan 22 13:05:35 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 22 Jan 2010 13:05:35 -0800
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 43, Issue 187
In-Reply-To: <mailman.5425.1264184283.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.5425.1264184283.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <FABCD242F6D74C3D906C886156FEDFE2@greyhawk.tonecommander.com>


I tried sending this Email earlier in the week but it got kicked back by our
Mail Server, so if this is a duplicate, my apologies...

Are you still working toward a release version 1.0.5?

Best Regards,
Jerry



From nicolas at medularis.com  Fri Jan 22 13:22:46 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 22 Jan 2010 18:22:46 -0300
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <87f2f3b91001211412h7aa0a84ageadb9557b869ca01@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
	<e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
	<5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>
	<5d2828f1001201406p4154b98ald1af1c5c25f59337@mail.gmail.com>
	<1b46b4e81001210335l42baef16r8a2952aa5b92f6e6@mail.gmail.com>
	<87f2f3b91001211412h7aa0a84ageadb9557b869ca01@mail.gmail.com>
Message-ID: <1b46b4e81001221322w5da04799s2ea86000c5c4a9a@mail.gmail.com>

No problem, here it is:

- http://wiki.freeswitch.org/wiki/Call_retry_based_on_hangup_cause

It is linked from your reference (
http://wiki.freeswitch.org/wiki/Mod_lua#More_Samples).

Sorry I didn't do it early, I hadn't seen your email.

I also added another, more complete, example here (also linked):

- http://wiki.freeswitch.org/wiki/Bridging_two_calls_with_retry



On Thu, Jan 21, 2010 at 7:12 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, Jan 21, 2010 at 3:35 AM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>>
>> On Wed, Jan 20, 2010 at 7:06 PM, Mike van Lammeren <mike at van.lammeren.net
>> > wrote:
>>
>>> So, I've been reading about early media in the wiki, and have made a
>>> little progress, which leads to more questions.
>>>
>>> I understand now why a call is considered connected before one person has
>>> picked up the phone. I am also able to get my script to wait for the phone
>>> to be picked up, by setting the ignore_early_media variable when starting a
>>> new session, like this:
>>>
>>> customerSession =
>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/example.com/"
>>> .. customerPhoneNumber)
>>>
>>>
>>> After that line, the script waits for the other phone to be picked up.
>>>
>>> However, now I wonder what to do with calls that don't complete, get busy
>>> signals, etc.
>>>
>>> What do people do in this case? The only related example I can find on
>>> the web is for a javascript dialer, which doesn't address any of these
>>> cases.
>>>
>>
>>
>> I guess it depends on what you want to do. For example I have a lua script
>> very similar to what you describe, although there is no confirmation
>> involved. Depending on the hangup cause the session gets, it might try
>> redialing with a different gateway, try again or just hangup.
>>
>> Take a look here http://wiki.freeswitch.org/wiki/Hangup_causes to see
>> what each hangup cause means. You don't need to have a special case for all
>> of them, only the ones you are interested in.
>>
>> Here's an example in code which retries a call depending on the hangup
>> cause. It retries max_retries1 times and alternates between 2 different
>> gateways:
>>
>> session1 = null;
>> max_retries1 = 3;
>> retries = 0;
>> ostr = "";
>> repeat
>>         retries = retries + 1;
>>         if (retries % 2) then ostr = originate_str1;
>>         else ostr = originate_str12; end
>>         freeswitch.consoleLog("notice", "*********** Dialing Leg1: " ..
>> ostr .. " - Try: "..retries.." ***********\n");
>>         session1 = freeswitch.Session(ostr);
>>         local hcause = session1:hangupCause();
>>         freeswitch.consoleLog("notice", "*********** Leg1: " .. hcause ..
>> " - Try: "..retries.." ***********\n");
>> until not ((hcause == 'NO_ROUTE_DESTINATION' or hcause ==
>> 'RECOVERY_ON_TIMER_EXPIRE' or hcause == 'INCOMPATIBLE_DESTINATION' or hcause
>> == 'CALL_REJECTED' or hcause == 'NORMAL_TEMPORARY_FAILURE') and (retries <
>> max_retriesl1))
>>
>>
>> Note: originate_str1 and originate_str2 are two different dial strings for
>> 2 different gateways.
>>
>>
> Nicolas,
>
> This is really nice. Would you be willing to add this script and a brief
> explanation to the wiki? You could create a whole new page and just link to
> it from here: http://wiki.freeswitch.org/wiki/Mod_lua#More_Samples
>
> If you have any questions please let me know!
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From michal.zubac at comgate.cz  Fri Jan 22 10:20:20 2010
From: michal.zubac at comgate.cz (=?ISO-8859-2?Q?Michal_Zub=E1=E8?=)
Date: Fri, 22 Jan 2010 19:20:20 +0100
Subject: [Freeswitch-users] E1 hangups
Message-ID: <4B59EC64.3080907@comgate.cz>

Hi.

I'm trying to correct this behaviour, but can't figure out, where is the
problem. Here's the scenario:

- we're trying to execute simple dialplan
    * answer
    * play sound
    * wait for 3 seconds
    * hangup

- for SIP caller it works as expected
- problems are when, we try to call into it from E1 line
- for E1 we're using sangoma winpipe & openzap
- dialplan in freeswitch console is done in a moment ending with hangup
- on the E1 line I hear nothing and after 2 seconds it disconnects
- similar problem when there's only bridge to another number (E1)
- it rings (on the destination phone) for a short moment (0.5-1s), but
then hangs up spontaneously

Thanks for any clues.

mZubac
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From tim at communicatefreely.net  Fri Jan 22 12:30:17 2010
From: tim at communicatefreely.net (Tim St. Pierre)
Date: Fri, 22 Jan 2010 15:30:17 -0500
Subject: [Freeswitch-users] Choppy conference audio
Message-ID: <4B5A0AD9.8020700@communicatefreely.net>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello all,

I'm having a tough time sorting out the audio in mod_conference, I'm hoping someone can point me in
the right direction.  I took a good look around the wiki, but couldn't really find what I was
looking for.

I have a test system set up, with three Aastra 9133i, a 6731i, and a 57i.  All register, call fine, etc.

Music on hold audio is perfect.  Calls between phones are perfect.

CPU load on the lab machine is about 8%, so not really busy.

Conference audio is choppy, in the way that things get choppy when there is some sort of timing
issue.  Even with one member in the conference, the prompts and moh sound choppy.

I played around with the rate and interval values, there wasn't any change.  I tried disabling the
monotonic timer, and that didn't change anything either.

I'm running FreeBSD 7, with the latest release of FS downloaded last week.

Thanks for any help.  Let me know if I need to post any other information.

- -Tim


-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.4 (FreeBSD)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iQCVAwUBS1oK2YqVcvNCnHOrAQLriAP/csjIfD/VP0CA3ePyRBKXbDPfxEiqx/cP
iWILZ65F0bxaryKTYT2ZV8W7u+wH6YxMCaoej+H+yd1XG08hZr4kLUcewUaCTMna
S8Zd24ReknEmL8d0AXCMgospf2wmwZZx2pJMFmbPN3IlXup20dKWOESBp/Dru3vL
63GE8s6AdyA=
=ndAy
-----END PGP SIGNATURE-----


From anthony.minessale at gmail.com  Fri Jan 22 14:32:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 22 Jan 2010 16:32:43 -0600
Subject: [Freeswitch-users] Choppy conference audio
In-Reply-To: <4B5A0AD9.8020700@communicatefreely.net>
References: <4B5A0AD9.8020700@communicatefreely.net>
Message-ID: <191c3a031001221432t1d8f9614r1d86e0379f82aa2d@mail.gmail.com>

try combos of -vm and -nocal flags
one or both of each


On Fri, Jan 22, 2010 at 2:30 PM, Tim St. Pierre
<tim at communicatefreely.net>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello all,
>
> I'm having a tough time sorting out the audio in mod_conference, I'm hoping
> someone can point me in
> the right direction.  I took a good look around the wiki, but couldn't
> really find what I was
> looking for.
>
> I have a test system set up, with three Aastra 9133i, a 6731i, and a 57i.
>  All register, call fine, etc.
>
> Music on hold audio is perfect.  Calls between phones are perfect.
>
> CPU load on the lab machine is about 8%, so not really busy.
>
> Conference audio is choppy, in the way that things get choppy when there is
> some sort of timing
> issue.  Even with one member in the conference, the prompts and moh sound
> choppy.
>
> I played around with the rate and interval values, there wasn't any change.
>  I tried disabling the
> monotonic timer, and that didn't change anything either.
>
> I'm running FreeBSD 7, with the latest release of FS downloaded last week.
>
> Thanks for any help.  Let me know if I need to post any other information.
>
> - -Tim
>
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v2.0.4 (FreeBSD)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iQCVAwUBS1oK2YqVcvNCnHOrAQLriAP/csjIfD/VP0CA3ePyRBKXbDPfxEiqx/cP
> iWILZ65F0bxaryKTYT2ZV8W7u+wH6YxMCaoej+H+yd1XG08hZr4kLUcewUaCTMna
> S8Zd24ReknEmL8d0AXCMgospf2wmwZZx2pJMFmbPN3IlXup20dKWOESBp/Dru3vL
> 63GE8s6AdyA=
> =ndAy
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Jan 22 14:36:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 22 Jan 2010 16:36:23 -0600
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 43, Issue 187
In-Reply-To: <FABCD242F6D74C3D906C886156FEDFE2@greyhawk.tonecommander.com>
References: <mailman.5425.1264184283.2495.freeswitch-users@lists.freeswitch.org>
	<FABCD242F6D74C3D906C886156FEDFE2@greyhawk.tonecommander.com>
Message-ID: <191c3a031001221436m2089a333ne0aafe99136845fa@mail.gmail.com>

Current Scheduled release date is the week of Feb 8th


On Fri, Jan 22, 2010 at 3:05 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> I tried sending this Email earlier in the week but it got kicked back by
> our
> Mail Server, so if this is a duplicate, my apologies...
>
> Are you still working toward a release version 1.0.5?
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From thangappan143 at gmail.com  Fri Jan 22 20:32:32 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Sat, 23 Jan 2010 10:02:32 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
	<7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>
Message-ID: <7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>

Updated the latest version of freeswitch ( 1.0.5-20100121-0400) and run the
wanrouter then freeswitch. While executing the freeswtich it said the
following error.

[ERR] zap_io.c:2562 Error
loading/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
[/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
file: No such file or directory]
[ERR] zap_io.c:2722 can't find 'sangoma_boost




Searched about this in freeswitch mailing list and found one post was there
regarding the same problem. Finally found the problem. I missed to install
the SCTP packages. Installed it and compiled the freeswitch again now the
inbound call was landed on freeswitch.

But I am unable to make a outbound call. When I was trying the following was
get.

freeswitch at internal> originate
openzap/smg_prid/a/9940464753 at g1openzap/smg_prid/a/9940464753 at g1
-ERR NORMAL_CIRCUIT_CONGESTION

2010-01-23 10:00:45.688854 [WARNING] ozmod_sangoma_boost.c:348 TX EVENT:
CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[9940464753]
Ci=[0000000000] Rdnis=[]
freeswitch at internal> 2010-01-23 10:00:46.709355 [WARNING]
ozmod_sangoma_boost.c:1373 RX EVENT (N): CALL_START_NACK:(82) [w256g256]
Rc=0 CSid=2 Seq=2
2010-01-23 10:00:46.709355 [WARNING] sangoma_boost_client.c:220 TX EVENT
(N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=2 Seq=3
2010-01-23 10:00:46.709355 [ERR] mod_openzap.c:1162 No channels available
2010-01-23 10:00:46.709355 [ERR] switch_ivr_originate.c:2411 Cannot create
outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
2010-01-23 10:00:46.709355 [DEBUG] switch_ivr_originate.c:3211 Originate
Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]

Please help me...........


On Fri, Jan 22, 2010 at 10:15 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> The following link have the openzap.conf,openzap.conf.xml ,smg_prid.conf ,
> debug log of mod_openzap , oz list and oz dump 1 output.
>
> http://pastebin.org/80095
>
>
>
> On Thu, Jan 21, 2010 at 10:34 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>> OpenZap is loading the ss7 signalling type. As per your concern openzap
>> does not know the details of the signalling then how it is loading the
>> ss7_boost libraries?
>>
>> FreeSWITCH log:
>> -----------------------------
>> 2010-01-21 10:10:46.707844 [INFO] zap_io.c:2374 Configured 30 channel(s)
>> 2010-01-21 10:10:46.708600 [INFO] zap_io.c:2468 Loading SIG from
>> /usr/local/freeswitch/mod/ozmod_ss7_boost.so
>> 2010-01-21 10:10:46.709031 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost'
>> 2010-01-21 10:10:46.709466 [DEBUG] ss7_boost_client.c:124 Creating L=
>> 127.0.0.65:53000 R=127.0.0.66:53000
>> 2010-01-21 10:10:46.709834 [DEBUG] ss7_boost_client.c:124 Creating L=
>> 127.0.0.65:53001 R=127.0.0.66:53001
>> 2010-01-21 10:10:46.710424 [WARNING] ss7_boost_client.c:244 TX EVENT (P):
>> SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>>
>> The signalling type might be anything but when I used the oz list command
>> it showed the span details. But I am unable to make a inbound and outbound
>> call.
>>
>> Outbound call result:
>> ============
>> >  originate openzap/smg_prid/a/9940464753 openzap/smg_prid/a/9843171457
>> -ERR NORMAL_CIRCUIT_CONGESTION
>>
>> 2010-01-21 10:26:14.304816 [CRIT] ozmod_ss7_boost.c:244 SPAN is not
>> online.
>> freeswitch at internal> 2010-01-21 10:26:14.304816 [ERR] mod_openzap.c:1043
>> No channels available
>> 2010-01-21 10:26:14.304816 [ERR] switch_ivr_originate.c:1510 Cannot create
>> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>> 2010-01-21 10:26:14.304816 [DEBUG] switch_ivr_originate.c:2138 Originate
>> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>>
>> Inbound call result:
>> -----------------------------
>>
>>      I made incoming call for the dial plan which is specified in the
>> earlier post at that time it said the number is busy. We did the packet
>> capture using the following command.
>>
>> wanpipemon -i w1g1 -pcap -pcap file isdn.pcap -port ISDN -full -systime -c
>> trd
>>
>> Here I attached the pcap file for that.
>>
>>
>> Where I did mistake or Did I miss any thing to do?
>> Please help me.......
>>
>>
>>
>> On Wed, Jan 20, 2010 at 7:40 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>
>>>
>>>       I noticed the 'oz list' output  in that span type is 'ss7 (boost)'.
>>> How can I change this to isdn?
>>>
>>>
>>>
>>> On Wed, Jan 20, 2010 at 12:43 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>>
>>>> I found the error in it. The file name is used as openzap.conf.xml (
>>>> smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
>>>> PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
>>>> openzap.conf.wiki.xml  file.
>>>>
>>>> Now the another problem is raised here.
>>>> When I was using oz list command , the details of the smg_prid shown.
>>>> When I was using 'oz dump smg_prid' command it shows all the channels'
>>>> details. But all the channels' states are DOWN. why? How can I make it the
>>>> states to UP?
>>>>
>>>> When I was making the call , the number is busy message was get. The
>>>> call was not  at all landed to the freeswitch.
>>>>
>>>> Dial plan Example:
>>>> -------------------------------
>>>>  <extension name="PRI_Testing">
>>>>         <condition field="destination_number" expression="^39114600$">
>>>>                 <action application="playback"
>>>> data="ivr-welcome_to_freeswitch"/>
>>>>         </condition>
>>>>   </extension>
>>>>
>>>> Please help me...........
>>>>
>>>> *Output Reference:*
>>>> http://pastebin.org/79074
>>>>
>>>>
>>>> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <thangappan143 at gmail.com
>>>> > wrote:
>>>>
>>>>> Dear all,
>>>>>
>>>>>          I have successfully configured wanpipe with freeswitch. When I
>>>>> was the running wancfg_fs script the following files openzap.conf ,
>>>>> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
>>>>> are created.
>>>>>
>>>>>          I started the wanrouter command then executed the freeswitch.
>>>>>          When I was executing freeswitch mod_openzap.c said the error
>>>>> as  "Error  for finding the span id. name:PRI_1".
>>>>>          But in the openzap.conf and openzap.conf.xml files the span
>>>>> name is smg_prid.
>>>>>
>>>>>          Why the freeswitch is referring the span name as PRI_1 ?
>>>>>          Whether this has to configured in anywhere?
>>>>>
>>>>>          In the freeswitch CLI using oz command I tried to dump the
>>>>> PRI_1 span id but it said te error  as "PRI_1 is not found".  When I was
>>>>> trying the command 'oz dump smg_prid'  all the channel states and details
>>>>> shown.
>>>>>
>>>>>         It seems that smg_prid span configured in openzap perfectly
>>>>> (Its my assumption). Then Why freeswitch is referring the span name as
>>>>> PRI_1.
>>>>>
>>>>> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>>>>>
>>>>> Could anyone please help me?
>>>>>
>>>>> REFERENCE:
>>>>>
>>>>> openzap.conf
>>>>> [span wanpipe smg_prid]
>>>>> name => smg_prid
>>>>> trunk_type =>e1
>>>>> b-channel => 1:1-15
>>>>> b-channel => 1:17-31
>>>>>
>>>>>
>>>>> openzap.conf.xml
>>>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>>>   <settings>
>>>>>     <param name="debug" value="0"/>
>>>>>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>>>     <!--<param name="enable-analog-option" value="3-way"/>-->
>>>>>   </settings>
>>>>>   <boost_spans>
>>>>>         <span id="smg_prid">
>>>>>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>>         <param name="dialplan" value="XML"/>
>>>>>         <param name="context" value="default"/>
>>>>>          <!-- regex to stop dialing when it matches -->
>>>>>         <!--<param name="dial-regex" value="5555"/>-->
>>>>>         <!-- regex to stop dialing when it does not match -->
>>>>>         <!--<param name="fail-dial-regex" value="^5"/>-->
>>>>>         </span>
>>>>>  </boost_spans>
>>>>>  <analog_spans>
>>>>>  </analog_spans>
>>>>> </configuration>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Regards,
>>>>> Thangappan.M
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Thangappan.M
>>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Thangappan.M
>>>
>>
>>
>>
>> --
>> Regards,
>> Thangappan.M
>>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From nazim.agabekov at gmail.com  Fri Jan 22 15:38:49 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Sat, 23 Jan 2010 03:38:49 +0400
Subject: [Freeswitch-users] CDR Gateways
In-Reply-To: <956716.34674.qm@web34301.mail.mud.yahoo.com>
References: <956716.34674.qm@web34301.mail.mud.yahoo.com>
Message-ID: <4B5A3709.4090304@gmail.com>

Hello Fernando,

I have coded a small xml_cdr FCGI logger for this purpose, it receives
mod_xml_cdr's data and inserts it into mysql table.
Basically it's a FastCgi and Libxml's XPATH hacked together. Example
config file is pretty easy to understand.
Software is of pre-alfa quality, but works. You could get the svn
snapshot from blog.buta-tech.com.

Best Regards,
Nazim Aghabayov


On 01/22/2010 08:26 AM, FERNANDO VILLARROEL wrote:
> Dear All.
>
> I have defined various gateways in ~/sip-profiles/external
>
> My questions is if a gateway named foo that send calls from diferents IP address like x.x.x.x and x.x.x.y. How i can doing a group for both ip address where i can doing later a sql like accountcode=foo?
>
> In this moment if i need know the traffic of customer foo, i should doing two differents cdr, one for every ip address.
>
> It's fine or i can doing of different way?
>
> I hope anyone could me explain how i can doing. 
>
> my gateway foo.xml
>
> <include>
>     <gateway name="foo">
>       <param name="username" value="foo"/>
>       <param name="password" value="foo"/>
>       <param name="realm" value="x.x.x.x"/>
>       <param name="register" value="false"/>
>       <param name="context" value="public"/>
>     </gateway>
> </include>
>
> foo1.xml
>
> <include>
>     <gateway name="foo1">
>       <param name="username" value="foo1"/>
>       <param name="password" value="foo1"/>
>       <param name="realm" value="x.x.x.y"/>
>       <param name="register" value="false"/>
>       <param name="context" value="public"/>
>     </gateway>
> </include>
>
> Both gateways foo and foo1 are the same customer
>
> my cdr_csv.conf.xml
>
> <configuration name="cdr_csv.conf" description="CDR CSV Format">
>   <settings>
>     <!-- 'cdr-csv' will always be appended to log-base -->
>     <!--<param name="log-base" value="/var/log"/>-->
>     <param name="default-template" value="sql"/>
>     <!-- This is like the info app but after the call is hung up -->
>     <!--<param name="debug" value="true"/>-->
>     <param name="rotate-on-hup" value="false"/>
>     <!-- may be a b or ab -->
>     <param name="legs" value="ab"/>
>   </settings>
>   <templates>
>         <template name="sql">INSERT INTO v_cdr (v_id, caller_id_name, caller_id_number, destination_number, context, start_stamp, answer_stamp, end_stamp,
> duration, billsec, hangup_cause, uuid, bleg_uuid, accountcode, read_codec, write_codec, remote_media_ip, network_addr) VALUES ("{v_id}", "${caller_id_name}"
> ,"${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}"
> ,"${uuid}","${bleg_uuid}", "${accountcode}", "${read_codec}", "${write_codec}", "${remote_media_ip}", "${network_addr}");</template>
>
> The argument accountcode on my database is Blank or None for all records of gateways.
> Regards.
>
>
>       
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From jingwei.yang at gmail.com  Fri Jan 22 22:00:36 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Sat, 23 Jan 2010 14:00:36 +0800
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <13529f9d1001211806o35fc3435j93b71bbb2ede3028@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
	<1264092620.14614.73.camel@luna.tc.commsmundi.com>
	<13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
	<ebbc2d551001211721n51cb5c54m996e0d494842c383@mail.gmail.com>
	<13529f9d1001211806o35fc3435j93b71bbb2ede3028@mail.gmail.com>
Message-ID: <13529f9d1001222200p44afd96ema21729d0038e89ba@mail.gmail.com>

Hi Jo?o, do you know how to sign the agent off automatically when either
party hangs up the call?

Here's how I originate the call to the agent and sign him up in ACD1:

originate skypiax/ANY/jingwei.yang 6*1

However, I found the user_name property is empty. May I know how it is set?

Thanks and best regards,
-Jingwei

On Fri, Jan 22, 2010 at 10:06 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:

> Hi Jo?o, thanks for the reply. I'll try it out.
>
> Regards,
> -Jingwei
>
> 2010/1/22 Jo?o Mesquita <jmesquita at freeswitch.org>
>
> Jingwei, check my contrib dir. I think it may help you with one FIFO since
>> we are able there to sign in and sign off dynamic agents as well as
>> customize what we do when the FIFO makes a call to them.
>>
>> Regards,
>> Jo?o Mesquita
>> FSComm Developer
>>
>>
>> On Thu, Jan 21, 2010 at 10:05 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>>
>>> Thanks for the reply. All the agents are dynamic and I can't predefine
>>> them in the config file.
>>>
>>> Regards,
>>> -Jingwei
>>>
>>>
>>> On Fri, Jan 22, 2010 at 12:50 AM, Fran?ois Delawarde <
>>> fdelawarde at wirelessmundi.com> wrote:
>>>
>>>> Why do you need 2 fifos? You could have callback agents connected to the
>>>> fifo and send incoming calls there, mod_fifo will do the rest.
>>>>
>>>> To configure agents for callback:
>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>>>>
>>>> To place a call into a fifo:
>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue
>>>>
>>>> Fran?ois.
>>>>
>>>> On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
>>>> > Sorry about the confusion, I'm just trying to think over all the
>>>> > abnormal situations before the implementation and hope the flow has no
>>>> > design flaws.
>>>> >
>>>> > Client A is parked in Queue 1, multiple consumers will be ringed to
>>>> > answer him. And once the first one is connected, all the rest will
>>>> > hang up. This is the targeted function.
>>>> >
>>>> > To achieve this, I'm considering to originate a call to each consumer
>>>> > and put the calls in Queue 2. Then intercept client A to the first
>>>> > element of Queue 2.
>>>> >
>>>> > I'm not sure if it's the usual or the best way. If you feel not,
>>>> > please don't hesitate to correct me. Any thoughts are warmly
>>>> > appreciated.
>>>> >
>>>> > On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com>
>>>> wrote:
>>>> >         what no errors mean? how do you originate calls to consumers?
>>>> >         I don't
>>>> >         understand your scenario.
>>>> >
>>>> >         2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
>>>> >
>>>> >         > Hi All,
>>>> >         >
>>>> >         > Please advise whether the following flow makes sense.
>>>> >         >
>>>> >         > 1. Client A calls in and parked in Queue 1
>>>> >         > 2. Originate calls to several consumers simultaneously and
>>>> >         park them in
>>>> >         > Queue 2
>>>> >         > 3. Intercept A's call to the first consumer of Queue 2
>>>> >         >
>>>> >         > Basically I want A's call picked up by the first among many
>>>> >         consumers with
>>>> >         > no errors. Please let me know whether I'm on the right
>>>> >         track.
>>>> >         >
>>>> >         > Thanks and best regards,
>>>> >         > -Jingwei
>>>> >         >
>>>> >
>>>> >
>>>> >         _______________________________________________
>>>> >         FreeSWITCH-users mailing list
>>>> >         FreeSWITCH-users at lists.freeswitch.org
>>>> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >         UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >         http://www.freeswitch.org
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From a.afzali2003 at gmail.com  Sat Jan 23 06:49:30 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sat, 23 Jan 2010 18:19:30 +0330
Subject: [Freeswitch-users] Possibly Bug in mod_sofia
Message-ID: <debff5a81001230649v5dcd1127nd46ea112b03b4c1c@mail.gmail.com>

Hi,

In sofia_reg.c line 190 :

nua_handle_bind(gateway_ptr->nh, gateway_ptr->sofia_private);

It seems the right handle be gateway_ptr->sub_nh. It causes every SUBSCRIBE
message's response print "Gateway information missing" error.

-- afshin
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From satish_lx at hotmail.com  Fri Jan 22 23:08:14 2010
From: satish_lx at hotmail.com (satish patel)
Date: Sat, 23 Jan 2010 07:08:14 +0000
Subject: [Freeswitch-users] mod_radius_cdr module load error
Message-ID: <BAY141-W1AF84CAAD730644C7A09790610@phx.gbl>


Hi All,
I am following this wiki http://wiki.freeswitch.org/wiki/Mod_radius_cdr to hook up freeradius with freeswitch but i am getting following error in log
2010-01-23 01:56:25.717201 [ERR] mod_radius_cdr.c:662 Open of mod_radius_cdr.conf failed2010-01-23 01:56:25.717225 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_radius_cdr.so**Module load routine returned an error**

Any idea Team?
Appreciate your help.
Best,
S. 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
http://clk.atdmt.com/GBL/go/196390706/direct/01/
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From jmesquita at freeswitch.org  Sat Jan 23 08:24:39 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 23 Jan 2010 14:24:39 -0200
Subject: [Freeswitch-users] Call for help on FSComm
Message-ID: <ebbc2d551001230824p53f4319dj85bc4b44254c4a18@mail.gmail.com>

FreeSWITCH?ers,

I believe that everyone already heard about FSComm by now. Development is
moving fast and features are being added very fast. Nonetheless, we need
help to get this software the way we want.

If you are a GUI Designer or have any design skills and want to contribute,
please, get in touch with me by email, IM, IRC on #fscomm or even pigeon if
you like. I believe everyone wants a sexy looking softphone with all the
features FreeSWITCH? is able to provide.

Regards,
Jo?o Mesquita
FSComm Developer
GTalk: jmesquita at gmail.com
PayPal: jmesquita at gmail.com
IRC: jmesquita on #fscomm @freenode
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From brian at freeswitch.org  Sat Jan 23 08:29:31 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 23 Jan 2010 10:29:31 -0600
Subject: [Freeswitch-users] Possibly Bug in mod_sofia
In-Reply-To: <debff5a81001230649v5dcd1127nd46ea112b03b4c1c@mail.gmail.com>
References: <debff5a81001230649v5dcd1127nd46ea112b03b4c1c@mail.gmail.com>
Message-ID: <73D97305-56C0-4E68-B452-92FC0D1A438C@freeswitch.org>

No I'm pretty sure its correct. Their are options on gateways to make or take inbound subscriptions and route them its not documented.

Also that line is NOT an error level its a Debug level log.

If it were dangerous it would be CRIT or ERROR level logging.

Look at parse_gateway_subscriptions

/b


On Jan 23, 2010, at 8:49 AM, afshin afzali wrote:

> Hi,
> 
> In sofia_reg.c line 190 : 
> 
> nua_handle_bind(gateway_ptr->nh, gateway_ptr->sofia_private);
> 
> It seems the right handle be gateway_ptr->sub_nh. It causes every SUBSCRIBE message's response print "Gateway information missing" error.
> 
> -- afshin

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From yehavi.bourvine at gmail.com  Sat Jan 23 09:06:12 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 23 Jan 2010 19:06:12 +0200
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
Message-ID: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>

Hello,

  We are running 1.0.5pre10 for a while, and today I tried to move to the
latest tarball (from January 22nd). The software crashes with a core dump
after a few seconds. The core dump. The two relevant lines (to my opinion)
are:

#4  0xb7547d2d in nua_bye_server_report (sr=0xb6c910b0, tags=0x0)
    at nua_session.c:3938
#5  0xb7541bb3 in nua_server_report (sr=0x6) at nua_server.c:643
line 3938 is an assert() statement. Any idea?

                          Thanks! __Yehavi:
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From brian at freeswitch.org  Sat Jan 23 09:28:20 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 23 Jan 2010 11:28:20 -0600
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
Message-ID: <45626378-10E7-4723-A2D3-B05A831CD8E9@freeswitch.org>

Please collect the sip trace, console logs and everything you can up till the crash. Do not make decisions on what you think is relevant collect everything you can and open a jira.  

Also did you do a fresh checkout? or a make current?

Thanks,
Brian

On Jan 23, 2010, at 11:06 AM, Yehavi Bourvine wrote:

> Hello,
>  
>   We are running 1.0.5pre10 for a while, and today I tried to move to the latest tarball (from January 22nd). The software crashes with a core dump after a few seconds. The core dump. The two relevant lines (to my opinion) are:
>  
> #4  0xb7547d2d in nua_bye_server_report (sr=0xb6c910b0, tags=0x0)
>     at nua_session.c:3938
> #5  0xb7541bb3 in nua_server_report (sr=0x6) at nua_server.c:643
> line 3938 is an assert() statement. Any idea?
>  
>                           Thanks! __Yehavi:
>  



From sos at sokhapkin.dyndns.org  Sat Jan 23 09:35:05 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 23 Jan 2010 12:35:05 -0500
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
Message-ID: <201001231235.05300.sos@sokhapkin.dyndns.org>

Could you confirm that you have an issue described in 
http://jira.freeswitch.org/browse/SFSIP-197 ?

Seems like you're not the only unlucky...

On Saturday 23 January 2010, Yehavi Bourvine wrote:
> Hello,
>
>   We are running 1.0.5pre10 for a while, and today I tried to move to the
> latest tarball (from January 22nd). The software crashes with a core dump
> after a few seconds. The core dump. The two relevant lines (to my opinion)
> are:
>
> #4  0xb7547d2d in nua_bye_server_report (sr=0xb6c910b0, tags=0x0)
>     at nua_session.c:3938
> #5  0xb7541bb3 in nua_server_report (sr=0x6) at nua_server.c:643
> line 3938 is an assert() statement. Any idea?
>
>                           Thanks! __Yehavi:




From moises.silva at gmail.com  Sat Jan 23 09:41:53 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 23 Jan 2010 12:41:53 -0500
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
	<7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>
	<7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>
Message-ID: <c4d05cbe1001230941ud426917m77aa774430110a2b@mail.gmail.com>

On Fri, Jan 22, 2010 at 11:32 PM, Thangappan.M <thangappan143 at gmail.com>wrote:

> But I am unable to make a outbound call. When I was trying the following
> was get.
>
> freeswitch at internal> originate openzap/smg_prid/a/9940464753 at g1openzap/smg_prid/a/9940464753 at g1
> -ERR NORMAL_CIRCUIT_CONGESTION
>
> 2010-01-23 10:00:45.688854 [WARNING] ozmod_sangoma_boost.c:348 TX EVENT:
> CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[9940464753]
> Ci=[0000000000] Rdnis=[]
> freeswitch at internal> 2010-01-23 10:00:46.709355 [WARNING]
> ozmod_sangoma_boost.c:1373 RX EVENT (N): CALL_START_NACK:(82) [w256g256]
> Rc=0 CSid=2 Seq=2
> 2010-01-23 10:00:46.709355 [WARNING] sangoma_boost_client.c:220 TX EVENT
> (N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=2 Seq=3
> 2010-01-23 10:00:46.709355 [ERR] mod_openzap.c:1162 No channels available
> 2010-01-23 10:00:46.709355 [ERR] switch_ivr_originate.c:2411 Cannot create
> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
> 2010-01-23 10:00:46.709355 [DEBUG] switch_ivr_originate.c:3211 Originate
> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>
> Please help me...........
>

Did you define group 1 in /etc/wanpipe/smg_prid.conf, pastebin the file plz.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From yehavi.bourvine at gmail.com  Sat Jan 23 10:02:09 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 23 Jan 2010 20:02:09 +0200
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <201001231235.05300.sos@sokhapkin.dyndns.org>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<201001231235.05300.sos@sokhapkin.dyndns.org>
Message-ID: <d361ca7b1001231002q216a23d5j96bbc87e01c01749@mail.gmail.com>

Yes, it looks exactly the same, with the same code in retval. It happens
just when an incoming INVITE arrives.
Since it has already a jira issue opened, do I still have to provide the
traces?

About how I upgrade: I've downloded the tarball of the latest version into a
fresh directory, built it, and in order to install it:
deleted everyhting in bin, mod and lib, and then made "make install".

                     Thanks, __Yehavi:




2010/1/23 Sergey Okhapkin <sos at sokhapkin.dyndns.org>

> Could you confirm that you have an issue described in
> http://jira.freeswitch.org/browse/SFSIP-197 ?
>
> Seems like you're not the only unlucky...
>
> On Saturday 23 January 2010, Yehavi Bourvine wrote:
> > Hello,
> >
> >   We are running 1.0.5pre10 for a while, and today I tried to move to the
> > latest tarball (from January 22nd). The software crashes with a core dump
> > after a few seconds. The core dump. The two relevant lines (to my
> opinion)
> > are:
> >
> > #4  0xb7547d2d in nua_bye_server_report (sr=0xb6c910b0, tags=0x0)
> >     at nua_session.c:3938
> > #5  0xb7541bb3 in nua_server_report (sr=0x6) at nua_server.c:643
> > line 3938 is an assert() statement. Any idea?
> >
> >                           Thanks! __Yehavi:
>
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From a.afzali2003 at gmail.com  Sat Jan 23 10:23:48 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sat, 23 Jan 2010 21:53:48 +0330
Subject: [Freeswitch-users] Possibly Bug in mod_sofia
In-Reply-To: <73D97305-56C0-4E68-B452-92FC0D1A438C@freeswitch.org>
References: <debff5a81001230649v5dcd1127nd46ea112b03b4c1c@mail.gmail.com>
	<73D97305-56C0-4E68-B452-92FC0D1A438C@freeswitch.org>
Message-ID: <debff5a81001231023n7c491282ub878b645ed38b09a@mail.gmail.com>

Thanks to your reply,

I've paid attention to that error because the successful subscription,
immediately does unsubscribe by sending another SUBSCRIBE message. By
inspection in the function which the error does log (sofia_presence : 2198)
it appears that it is a precondition to accept any SUBSCRIBE message
response. and finally I did just this modification : nh -> sub_nh and the
result is the stable subscription operation.

of course you are right  :)
-- afshin

On Sat, Jan 23, 2010 at 7:59 PM, Brian West <brian at freeswitch.org> wrote:

> No I'm pretty sure its correct. Their are options on gateways to make or
> take inbound subscriptions and route them its not documented.
>
> Also that line is NOT an error level its a Debug level log.
>
> If it were dangerous it would be CRIT or ERROR level logging.
>
> Look at parse_gateway_subscriptions
>
> /b
>
>
> On Jan 23, 2010, at 8:49 AM, afshin afzali wrote:
>
> Hi,
>
> In sofia_reg.c line 190 :
>
> nua_handle_bind(gateway_ptr->nh, gateway_ptr->sofia_private);
>
> It seems the right handle be gateway_ptr->sub_nh. It causes every SUBSCRIBE
> message's response print "Gateway information missing" error.
>
> -- afshin
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sos at sokhapkin.dyndns.org  Sat Jan 23 10:25:53 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 23 Jan 2010 13:25:53 -0500
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001231002q216a23d5j96bbc87e01c01749@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<201001231235.05300.sos@sokhapkin.dyndns.org>
	<d361ca7b1001231002q216a23d5j96bbc87e01c01749@mail.gmail.com>
Message-ID: <201001231325.54564.sos@sokhapkin.dyndns.org>

Add your traces and all information you can provide to 
http://jira.freeswitch.org/browse/SFSIP-197 , the more information developers 
will get, the sooner they will be able to find and fix the issue.

I wish my FS crash "in a few seconds" like yours, then I will be able to debug 
myself, but it crashes very rarely...

On Saturday 23 January 2010, Yehavi Bourvine wrote:
> Yes, it looks exactly the same, with the same code in retval. It happens
> just when an incoming INVITE arrives.
> Since it has already a jira issue opened, do I still have to provide the
> traces?
>
> About how I upgrade: I've downloded the tarball of the latest version into
> a fresh directory, built it, and in order to install it:
> deleted everyhting in bin, mod and lib, and then made "make install".
>
>                      Thanks, __Yehavi:
>
>
>
>
> 2010/1/23 Sergey Okhapkin <sos at sokhapkin.dyndns.org>
>
> > Could you confirm that you have an issue described in
> > http://jira.freeswitch.org/browse/SFSIP-197 ?
> >
> > Seems like you're not the only unlucky...
> >
> > On Saturday 23 January 2010, Yehavi Bourvine wrote:
> > > Hello,
> > >
> > >   We are running 1.0.5pre10 for a while, and today I tried to move to
> > > the latest tarball (from January 22nd). The software crashes with a
> > > core dump after a few seconds. The core dump. The two relevant lines
> > > (to my
> >
> > opinion)
> >
> > > are:
> > >
> > > #4  0xb7547d2d in nua_bye_server_report (sr=0xb6c910b0, tags=0x0)
> > >     at nua_session.c:3938
> > > #5  0xb7541bb3 in nua_server_report (sr=0x6) at nua_server.c:643
> > > line 3938 is an assert() statement. Any idea?
> > >
> > >                           Thanks! __Yehavi:
> >
> >  _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From brian at freeswitch.org  Sat Jan 23 10:29:57 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 23 Jan 2010 12:29:57 -0600
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001231002q216a23d5j96bbc87e01c01749@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<201001231235.05300.sos@sokhapkin.dyndns.org>
	<d361ca7b1001231002q216a23d5j96bbc87e01c01749@mail.gmail.com>
Message-ID: <AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>

Yes please provide traces the more information we have the clearer the picture and possibility of fixing it.  

/b

On Jan 23, 2010, at 12:02 PM, Yehavi Bourvine wrote:

> Yes, it looks exactly the same, with the same code in retval. It happens just when an incoming INVITE arrives.
> Since it has already a jira issue opened, do I still have to provide the traces?
>  
> About how I upgrade: I've downloded the tarball of the latest version into a fresh directory, built it, and in order to install it:
> deleted everyhting in bin, mod and lib, and then made "make install".
>  
>                      Thanks, __Yehavi:
>  
> 



From brian at freeswitch.org  Sat Jan 23 10:31:02 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 23 Jan 2010 12:31:02 -0600
Subject: [Freeswitch-users] Possibly Bug in mod_sofia
In-Reply-To: <debff5a81001231023n7c491282ub878b645ed38b09a@mail.gmail.com>
References: <debff5a81001230649v5dcd1127nd46ea112b03b4c1c@mail.gmail.com>
	<73D97305-56C0-4E68-B452-92FC0D1A438C@freeswitch.org>
	<debff5a81001231023n7c491282ub878b645ed38b09a@mail.gmail.com>
Message-ID: <A46ABB96-5355-44DE-9F0E-B098AAB5B44D@freeswitch.org>

If its still a bug or you had to change code please post a jira.. its the correct place to debate and talk about issues so we can track them properly.  ;)

Thanks,
Brian

On Jan 23, 2010, at 12:23 PM, afshin afzali wrote:

> Thanks to your reply,
> 
> I've paid attention to that error because the successful subscription, immediately does unsubscribe by sending another SUBSCRIBE message. By inspection in the function which the error does log (sofia_presence : 2198) it appears that it is a precondition to accept any SUBSCRIBE message response. and finally I did just this modification : nh -> sub_nh and the result is the stable subscription operation.
> 
> of course you are right  :)
> -- afshin     



From yehavi.bourvine at gmail.com  Sat Jan 23 10:38:27 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 23 Jan 2010 20:38:27 +0200
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<201001231235.05300.sos@sokhapkin.dyndns.org>
	<d361ca7b1001231002q216a23d5j96bbc87e01c01749@mail.gmail.com>
	<AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
Message-ID: <d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>

Just uploaded. It happens when I call our voicemail number and disconnects
the call. The voicemail application answers, and I disconnect the call the
Freeswitch crash.

                 Thanks, __Yehavi:

2010/1/23 Brian West <brian at freeswitch.org>

> Yes please provide traces the more information we have the clearer the
> picture and possibility of fixing it.
>
> /b
>
> On Jan 23, 2010, at 12:02 PM, Yehavi Bourvine wrote:
>
> > Yes, it looks exactly the same, with the same code in retval. It happens
> just when an incoming INVITE arrives.
> > Since it has already a jira issue opened, do I still have to provide the
> traces?
> >
> > About how I upgrade: I've downloded the tarball of the latest version
> into a fresh directory, built it, and in order to install it:
> > deleted everyhting in bin, mod and lib, and then made "make install".
> >
> >                      Thanks, __Yehavi:
> >
> >
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sos at sokhapkin.dyndns.org  Sat Jan 23 10:55:30 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 23 Jan 2010 13:55:30 -0500
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
	<d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
Message-ID: <201001231355.31032.sos@sokhapkin.dyndns.org>

GOOD! You have a consistent way to reproduce the problem!

On Saturday 23 January 2010, Yehavi Bourvine wrote:
> Just uploaded. It happens when I call our voicemail number and disconnects
> the call. The voicemail application answers, and I disconnect the call the
> Freeswitch crash.
>
>                  Thanks, __Yehavi:
>
> 2010/1/23 Brian West <brian at freeswitch.org>
>
> > Yes please provide traces the more information we have the clearer the
> > picture and possibility of fixing it.
> >
> > /b
> >
> > On Jan 23, 2010, at 12:02 PM, Yehavi Bourvine wrote:
> > > Yes, it looks exactly the same, with the same code in retval. It
> > > happens
> >
> > just when an incoming INVITE arrives.
> >
> > > Since it has already a jira issue opened, do I still have to provide
> > > the
> >
> > traces?
> >
> > > About how I upgrade: I've downloded the tarball of the latest version
> >
> > into a fresh directory, built it, and in order to install it:
> > > deleted everyhting in bin, mod and lib, and then made "make install".
> > >
> > >                      Thanks, __Yehavi:
> >
> >  _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From brian at freeswitch.org  Sat Jan 23 11:00:13 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 23 Jan 2010 13:00:13 -0600
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<201001231235.05300.sos@sokhapkin.dyndns.org>
	<d361ca7b1001231002q216a23d5j96bbc87e01c01749@mail.gmail.com>
	<AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
	<d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
Message-ID: <FE53682D-179E-4F6D-AF2F-714913C76B6B@freeswitch.org>

Please get a full pcap trace console log and attach it.

/b

On Jan 23, 2010, at 12:38 PM, Yehavi Bourvine wrote:

> Just uploaded. It happens when I call our voicemail number and disconnects the call. The voicemail application answers, and I disconnect the call the Freeswitch crash.
>  
>                  Thanks, __Yehavi:



From brian at freeswitch.org  Sat Jan 23 11:02:53 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 23 Jan 2010 13:02:53 -0600
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <201001231355.31032.sos@sokhapkin.dyndns.org>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
	<d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
	<201001231355.31032.sos@sokhapkin.dyndns.org>
Message-ID: <AC76DB08-C278-4F59-90D0-2C3724CDB39A@freeswitch.org>

Best to find us on IRC when anthm is around and lets get into your box and fix this.

/b

On Jan 23, 2010, at 12:55 PM, Sergey Okhapkin wrote:

> GOOD! You have a consistent way to reproduce the problem!



From camilin2212 at hotmail.com  Sat Jan 23 14:08:57 2010
From: camilin2212 at hotmail.com (juan camilo ospina quintero)
Date: Sat, 23 Jan 2010 17:08:57 -0500
Subject: [Freeswitch-users] FreeSwitch Integration With Sailfin
Message-ID: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>


hi


im having serious trouble trying to integrate freeswitch with sailfin, if someone could help me, taht would be awesome. 
I guess is with a bridge statement in the dialplan , but not sure how to do that, and if that is the right solution.

thanks
 		 	   		  
_________________________________________________________________
Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail?.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009
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From brian at freeswitch.org  Sat Jan 23 14:34:04 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 23 Jan 2010 16:34:04 -0600
Subject: [Freeswitch-users] FreeSwitch Integration With Sailfin
In-Reply-To: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>
References: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>
Message-ID: <CFA0D82A-33B3-4B50-B6AE-BAA61E0862A4@freeswitch.org>

When asking a question on the list you'll need to provide some back story on what exactly you're doing.  Outline any issues you're having and try to include as much info as possible.  Nobody can help you otherwise.  For example what is Sailfin?  What bridge line?  What do your logs say?

Thanks,
/b

On Jan 23, 2010, at 4:08 PM, juan camilo ospina quintero wrote:

> hi
> 
> 
> im having serious trouble trying to integrate freeswitch with sailfin, if someone could help me, taht would be awesome. 
> I guess is with a bridge statement in the dialplan , but not sure how to do that, and if that is the right solution.
> 
> thanks

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From camilin2212 at hotmail.com  Sat Jan 23 14:48:42 2010
From: camilin2212 at hotmail.com (juan camilo ospina quintero)
Date: Sat, 23 Jan 2010 17:48:42 -0500
Subject: [Freeswitch-users] FreeSwitch Integration With Sailfin
In-Reply-To: <CFA0D82A-33B3-4B50-B6AE-BAA61E0862A4@freeswitch.org>
References: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>,
	<CFA0D82A-33B3-4B50-B6AE-BAA61E0862A4@freeswitch.org>
Message-ID: <BAY143-W13760770FF3FCA224AACEA1610@phx.gbl>


OK

sorry, is my first time in the mailing list, well what i need to do, is to integrate freeswitch with sailfin, sailfin is an application server SIP based, i need to forwards the SIP traffic from freeswitch to sailfin. the registration can be done in freeswitch. But i need to redirect the Sip invite methods to sailfin. i have no log because i have not been able to do anything. i have freeswitch installed and runnig, also sailfin. i can make calls between users with freeswitch. thats all. i read that adding a bridge statement to the dial plan would help, but i dont know how to do that, im new in freeswitch

thanks to all

From: brian at freeswitch.org
Date: Sat, 23 Jan 2010 16:34:04 -0600
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSwitch Integration With Sailfin



When asking a question on the list you'll need to provide some back story on what exactly you're doing.  Outline any issues you're having and try to include as much info as possible.  Nobody can help you otherwise.  For example what is Sailfin?  What bridge line?  What do your logs say?
Thanks,/b
On Jan 23, 2010, at 4:08 PM, juan camilo ospina quintero wrote:hi


im having serious trouble trying to integrate freeswitch with sailfin, if someone could help me, taht would be awesome. 
I guess is with a bridge statement in the dialplan , but not sure how to do that, and if that is the right solution.

thanks
 		 	   		  
_________________________________________________________________
Windows Live: Keep your friends up to date with what you do online.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_1:092010
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From edpimentl at gmail.com  Sat Jan 23 15:01:07 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Sat, 23 Jan 2010 18:01:07 -0500
Subject: [Freeswitch-users] FreeSwitch Integration With Sailfin
In-Reply-To: <BAY143-W13760770FF3FCA224AACEA1610@phx.gbl>
References: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>
	<CFA0D82A-33B3-4B50-B6AE-BAA61E0862A4@freeswitch.org> 
	<BAY143-W13760770FF3FCA224AACEA1610@phx.gbl>
Message-ID: <9dc4a1671001231501t65b1281epeeb7e7fa1b0bf5bf@mail.gmail.com>

It would be good if you include the following and then inject your question
http://wiki.glassfish.java.net/Wiki.jsp?page=GlassFishWiki
http://wiki.glassfish.java.net/Wiki.jsp?page=SailFin
https://sailfin.dev.java.net/


-E
http://vCardCloud.com




On Sat, Jan 23, 2010 at 5:48 PM, juan camilo ospina quintero <
camilin2212 at hotmail.com> wrote:

>  OK
>
> sorry, is my first time in the mailing list, well what i need to do, is to
> integrate freeswitch with sailfin, sailfin is an application server SIP
> based, i need to forwards the SIP traffic from freeswitch to sailfin. the
> registration can be done in freeswitch. But i need to redirect the Sip
> invite methods to sailfin. i have no log because i have not been able to do
> anything. i have freeswitch installed and runnig, also sailfin. i can make
> calls between users with freeswitch. thats all. i read that adding a bridge
> statement to the dial plan would help, but i dont know how to do that, im
> new in freeswitch
>
> thanks to all
>
> ------------------------------
> From: brian at freeswitch.org
> Date: Sat, 23 Jan 2010 16:34:04 -0600
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FreeSwitch Integration With Sailfin
>
> When asking a question on the list you'll need to provide some back story
> on what exactly you're doing.  Outline any issues you're having and try to
> include as much info as possible.  Nobody can help you otherwise.  For
> example what is Sailfin?  What bridge line?  What do your logs say?
>
> Thanks,
> /b
>
> On Jan 23, 2010, at 4:08 PM, juan camilo ospina quintero wrote:
>
> hi
>
>
> im having serious trouble trying to integrate freeswitch with sailfin, if
> someone could help me, taht would be awesome.
> I guess is with a bridge statement in the dialplan , but not sure how to do
> that, and if that is the right solution.
>
> thanks
>
>
>
> ------------------------------
> Windows Live: Keep your friends up to date with what you do online.<http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_1:092010>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wchao at yahoo.com  Sat Jan 23 19:11:10 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Sat, 23 Jan 2010 22:11:10 -0500 (EST)
Subject: [Freeswitch-users] Eavesdrop when using simring
In-Reply-To: <191c3a031001201209y53d68c83t8ba066b63ff4b723@mail.gmail.com>
References: <alpine.LRH.2.00.1001201139480.13391@osgood.microoffice.us>
	<191c3a031001201209y53d68c83t8ba066b63ff4b723@mail.gmail.com>
Message-ID: <alpine.LRH.2.00.1001232153410.26382@osgood.microoffice.us>

Thanks. Your answer helped me find execute_on_answer, which worked for me. 
Using execute_on_answer, I was able to get it to work exactly as I wanted, 
but I had to hardcode the variable per bridge target. I am wondering if 
there's a better way to handle it. Here is what I have now (which works):

<action application="bridge" 
data="[execute_on_answer='hash 
insert/${domain_name}-spymap/646xxxyyyy-1000/${uuid}']user/1001@${domain_name},[execute_on_answer='hash 
insert/${domain_name}-spymap/646xxxyyyy-1001/${uuid}']user/1002@${domain_name},[execute_on_answer='hash 
insert/${domain_name}-spymap/646xxxyyyy-1002/${uuid}']user/1003@${domain_name},[execute_on_answer='hash 
insert/${domain_name}-spymap/646xxxyyyy-1003/${uuid}']user/1004@${domain_name}"/>

What I am wondering is whether there's a way to do it like this instead:

<action application="export" data="execute_on_answer='hash 
insert/${domain_name}-spymap/646xxxyyyy-${answered_username}/${uuid}'"/>
<action application="bridge" 
data="user/1001@${domain_name},user/1002@${domain_name},user/1003@${domain_name},user/1004@${domain_name}"/>

So, the two questions are:

[1] How would I get the username of the bridge target that picks up the 
call?

[2] Is there a way to defer variable substitution/evaluation until after 
the call is answered?


Date: Wed, 20 Jan 2010 14:09:48 -0600
From: Anthony Minessale <anthony.minessale at gmail.com>
Reply-To: freeswitch-users at lists.freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Eavesdrop when using simring

maybe api_on_answer var?

On Wed, Jan 20, 2010 at 10:48 AM, Wellie Chao <wchao at yahoo.com> wrote:
      I have eavesdrop working fine on outbound calls and also inbound calls
      where there is a single DID per IP phone. When I have a DID that rings
      multiple extensions simultaneously, what is the best way to obtain
      information about which extension has picked up the call and store that
      using hash? I can set a variable before I issue the bridge action, like
      so:

      <action application="hash" data="insert/${domain_name}-spymap/646xxxyyyy-1000/${uuid}"/>
      <action application="bridge"
      data="user/1001@${domain_name},user/1002@${domain_name},user/1003@${domain_name},user/1004@${domain_name}"/>

      However, that doesn't tell me who actually picked up, so at best I can
      allow users to eavesdrop on the last incoming call to the main DID, not
      the last incoming call to a particular extension. Is there something I can
      do in the bridge that will cause it to set a variable once it knows which
      extension has picked up the call?

      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900



From camilin2212 at hotmail.com  Sat Jan 23 20:48:42 2010
From: camilin2212 at hotmail.com (juan camilo ospina quintero)
Date: Sat, 23 Jan 2010 23:48:42 -0500
Subject: [Freeswitch-users] FreeSwitch Integration With Sailfin
In-Reply-To: <9dc4a1671001231501t65b1281epeeb7e7fa1b0bf5bf@mail.gmail.com>
References: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>,
	<CFA0D82A-33B3-4B50-B6AE-BAA61E0862A4@freeswitch.org>,
	<BAY143-W13760770FF3FCA224AACEA1610@phx.gbl>,
	<9dc4a1671001231501t65b1281epeeb7e7fa1b0bf5bf@mail.gmail.com>
Message-ID: <BAY143-W760D1216FA26D7EFD8E9DA1600@phx.gbl>


Hi 
thanks for fast answers
i already have sailfin installed and runnig, also freeswitch, both on the same machine.
now my quiestion is how to redirect o forward the SIP flow from freeswitch to salifin, the softphones must register in freeswitch, but to make a call the invite should go through sailfin and get back to freeswitch, this is for implementing some services of VoIP, for a project i'm on.

From: edpimentl at gmail.com
Date: Sat, 23 Jan 2010 18:01:07 -0500
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSwitch Integration With Sailfin

It would be good if you include the following and then inject your question
http://wiki.glassfish.java.net/Wiki.jsp?page=GlassFishWiki
http://wiki.glassfish.java.net/Wiki.jsp?page=SailFin


https://sailfin.dev.java.net/

-E
http://vCardCloud.com





On Sat, Jan 23, 2010 at 5:48 PM, juan camilo ospina quintero <camilin2212 at hotmail.com> wrote:







OK

sorry, is my first time in the mailing list, well what i need to do, is to integrate freeswitch with sailfin, sailfin is an application server SIP based, i need to forwards the SIP traffic from freeswitch to sailfin. the registration can be done in freeswitch. But i need to redirect the Sip invite methods to sailfin. i have no log because i have not been able to do anything. i have freeswitch installed and runnig, also sailfin. i can make calls between users with freeswitch. thats all. i read that adding a bridge statement to the dial plan would help, but i dont know how to do that, im new in freeswitch



thanks to all

From: brian at freeswitch.org
Date: Sat, 23 Jan 2010 16:34:04 -0600
To: freeswitch-users at lists.freeswitch.org


Subject: Re: [Freeswitch-users] FreeSwitch Integration With Sailfin



When asking a question on the list you'll need to provide some back story on what exactly you're doing.  Outline any issues you're having and try to include as much info as possible.  Nobody can help you otherwise.  For example what is Sailfin?  What bridge line?  What do your logs say?


Thanks,/b
On Jan 23, 2010, at 4:08 PM, juan camilo ospina quintero wrote:
hi




im having serious trouble trying to integrate freeswitch with sailfin, if someone could help me, taht would be awesome. 
I guess is with a bridge statement in the dialplan , but not sure how to do that, and if that is the right solution.



thanks
 		 	   		  
Windows Live: Keep your friends up to date with what you do online.



_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org



 		 	   		  
_________________________________________________________________
Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail?.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009
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From mike at jerris.com  Sat Jan 23 22:54:50 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 01:54:50 -0500
Subject: [Freeswitch-users] Managing Presence on Gateways
In-Reply-To: <debff5a81001210144n4b32c19cic36415155ee52231@mail.gmail.com>
References: <debff5a81001210144n4b32c19cic36415155ee52231@mail.gmail.com>
Message-ID: <326ADCAA-AC7E-4625-886B-3FFF7E30FAD2@jerris.com>

No, we don't have the functionality to gateway presence.

On Jan 21, 2010, at 4:44 AM, afshin afzali <a.afzali2003 at gmail.com>  
wrote:

> Hi Guys,
>
> In the external profile (as in the internal) there is an option to  
> enable presence functionality (with setting it to passive). My  
> question is how does it mean presence functionality for a gateway  
> which interfaces home domain to another one? Does it mean that the  
> gateway could subscribe itself for some presence information in that  
> domain in behaves of local users and relays them?
>
> Appreciate all comments,
> -- afshin
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From mike at jerris.com  Sun Jan 24 01:48:13 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 04:48:13 -0500
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION?
In-Reply-To: <201001211405.19271.sos@sokhapkin.dyndns.org>
References: <201001211341.09739.sos@sokhapkin.dyndns.org>
	<191c3a031001211053i6670cec3l8b8f317dac4c6072@mail.gmail.com>
	<201001211405.19271.sos@sokhapkin.dyndns.org>
Message-ID: <6206D9B0-EA6B-4CAC-A42A-727BAF6254FC@jerris.com>

Try this out again with current trunk before taking logs and potentially filing a bug.  I suspect this is fixed now if it is what I think it is.

Mike

On Jan 21, 2010, at 2:05 PM, Sergey Okhapkin wrote:

> Late negotiation is set. I will try to enable debug when the traffic will be 
> low and open a problem on jira.
> 
> On Thursday 21 January 2010, Anthony Minessale wrote:
>> if you use bypass_media=true from the dialplan without late-negotiation set
>> in the profile, it still tries to match the codecs locally on the inbound
>> leg and the variable does not work if the call has established media before
>> making the outbound leg.
>> 
>> It's hard to tell you the exact answer without a console trace on debug
>> level.
>> 
>> 
>> On Thu, Jan 21, 2010 at 12:41 PM, Sergey Okhapkin
>> 
>> <sos at sokhapkin.dyndns.org>wrote:
>>> I often see in FS log the following problem (bypass_media=true), SVN
>>> r16340:
>>> 
>>> SDP sent out to gateway (INVITE):
>>> 
>>>  v=0
>>>  o=bandx-msw3 0 0 IN IP4 213.166.9.4
>>>  s=sip call
>>>  c=IN IP4 213.166.9.6
>>>  t=0 0
>>>  m=audio 56032 RTP/AVP 0 8 18
>>>  a=rtpmap:18 G729/8000
>>>  a=fmtp:18 annexb=yes
>>>  a=silenceSupp:on - - - -
>>> 
>>> 
>>> SDP response from gateway (183 Session Progress):
>>> 
>>>  v=0
>>>  o=- 3473087019 3473087037 IN IP4 67.203.64.182
>>>  s=-
>>>  c=IN IP4 67.203.64.182
>>>  t=0 0
>>>  m=audio 14116 RTP/AVP 0
>>>  a=sendrecv
>>>  a=ptime:20
>>>  a=rtpmap:0 PCMU/8000
>>> 
>>> Freeswitch hangs up the call with hangup cause INCOMPATIBLE_DESTINATION.
>>> Where
>>> is incompatibility? There is common codec 0.
>>> 


From mike at jerris.com  Sun Jan 24 01:50:00 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 04:50:00 -0500
Subject: [Freeswitch-users] reloadxml/rescan profile  and  killgw
In-Reply-To: <4B589A6E.8010205@kinetix.gr>
References: <4B589A6E.8010205@kinetix.gr>
Message-ID: <3F378D25-85F8-48B9-B814-7F7979EFE840@jerris.com>

A lot of things would be nice.  Patches for example.  I find them quite nice.  This should probably be a different command or argument.

Mike

On Jan 21, 2010, at 1:18 PM, Apostolos Pantsiopoulos wrote:
  
> Wouldn't it be nice a "reloadxml"/"rescan profile" to delete all 
> nonexistent gateways (when they are absent in the xml config) without 
> having to issue a "sofia profile xxxxx killgw yyyyyy" command?
> 
> I always seem to find forgotten gateway's in the profile because of 
> this... Any thoughts?



From mike at jerris.com  Sun Jan 24 01:53:13 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 04:53:13 -0500
Subject: [Freeswitch-users] Presence Not Working After Version 1.0.5pre9?
In-Reply-To: <B4EB2C4365CD4CA79B54134ECF232EED@greyhawk.tonecommander.com>
References: <mailman.5334.1264017641.2495.freeswitch-users@lists.freeswitch.org><C726F91949324BDFA34596256B6E102A@greyhawk.tonecommander.com>
	<191c3a031001201531g7c78cb9fw1ed1a2ba07f5773c@mail.gmail.com>
	<B4EB2C4365CD4CA79B54134ECF232EED@greyhawk.tonecommander.com>
Message-ID: <75590ED4-F843-48F3-93B8-3EFEE107D411@jerris.com>

If you could document the configuration requirements on the wiki I would appreciate it.

Mike

On Jan 21, 2010, at 5:39 PM, Jerry Richards wrote:

> Yes you are correct.  The Bria Softphone has a setting under ContactProfile/Advanced.../Account menu which is required to be the softphone's extension (not blank and not the extension that is being subscribed to).  After I set this field to the softphone's extension, FS starting reporting the the contact's presence status.
>  
> Thanks and Best Regards,
> Jerry
>  

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From mike at jerris.com  Sun Jan 24 01:57:17 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 04:57:17 -0500
Subject: [Freeswitch-users] Javascript self.session.getVariable
In-Reply-To: <ade741d61001211558h7b84cb96q16754650812a80e2@mail.gmail.com>
References: <ade741d61001211558h7b84cb96q16754650812a80e2@mail.gmail.com>
Message-ID: <ECA755AB-6BE3-46DF-9272-5085271A5F4F@jerris.com>

it should be just session.getVariable

var base_dir = session.getVariable ("base_dir"); 

example taken from http://svn.freeswitch.org/svn/freeswitch/trunk/scripts/javascript/aadir/aadir.js

This of course assumes that you are running as an application and you have a session there.

Mike


On Jan 21, 2010, at 6:58 PM, Mouncif Benniane wrote:

> I have the following in javascript:
> 
> caller_id = self.session.getVariable("caller_id_number")
> 
> 
> for some reasons it returns: 
> ReferenceError: self is not defined, I am following this page:
> 
> http://wiki.freeswitch.org/wiki/Session_getVariable
> 
> 
> any ideas?
> 

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From mike at jerris.com  Sun Jan 24 01:59:06 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 04:59:06 -0500
Subject: [Freeswitch-users] Svar: Re: Svar: Re: Home setup with home	
	company
In-Reply-To: <4B5996B1020000E100000404@mail.fribert.dk>
References: <4B5996B1020000E100000404@mail.fribert.dk>
Message-ID: <DA8197DE-0287-4097-AFE4-00295589DDE3@jerris.com>

Debug logs should help you figure out what is and is not matching in your conditions.

Mike

On Jan 22, 2010, at 6:14 AM, mailinglist wrote:

> Ok, I set it up like this:
> <extension name="CompanyIn">
>    <condition field="destination_number" expression="^(8203)$" continue="true"/>
>      <!-- man strftime - M-F, 9AM to 5PM -->
>       <condition field="${strftime(%w)}" expression="^([1-2])$"/>
>          <anti-action application="voicemail" data="default 10.11.12.25 10199"/>
>       <condition field="${strftime(%H%M)}" expression="^((09|1[0-6])[0-5][0-9]|1700)$">
>          <action application="set" data="call_timeout=45"/>
>          <action application="set" data="domain_name=10.11.12.25"/>
>          <action application="set" data="continue_on_fail=true"/>
>          <action application="set" data="hangup_after_bridge=true"/>
>          <action application="bridge" data="group/company@${domain_name}"/>
>          <action application="answer" data=""/>
>          <action application="sleep" data="1000"/>
>          <action application="voicemail" data="default 10.11.12.25 10199"/>
>          <anti-action application="voicemail" data="default 10.11.12.25 10199"/>
>       </condition>
> </extension>
> 
> But now it gives me:
> 
> 2010-01-22 11:52:08.667564 [NOTICE] switch_channel.c:602 New Channel sofia/external/2680xxxx at 87.54.25.116 [16baec2f-4407-df11-8fb3-000c29b7b4cb]
> 2010-01-22 11:52:08.800123 [INFO] mod_dialplan_xml.c:252 Processing 2680xxxx->4692xxxx in context public
> 2010-01-22 11:52:08.830071 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/2680xxxx at 87.54.25.116 to XML[8203 at default]
> 2010-01-22 11:52:08.830071 [INFO] mod_dialplan_xml.c:252 Processing 2680xxxx->8203 in context default
> 2010-01-22 11:52:08.830071 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/2680xxxx at 87.54.25.116 to enum[8203 at default]
> 2010-01-22 11:52:09.163439 [INFO] switch_core_state_machine.c:136 No Route, Aborting
> 
> huh

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From mike at jerris.com  Sun Jan 24 02:03:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 05:03:18 -0500
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <984278.36075.qm@web33504.mail.mud.yahoo.com>
References: <984278.36075.qm@web33504.mail.mud.yahoo.com>
Message-ID: <9881D312-67D1-40D3-B169-A178202F4E6C@jerris.com>

As noted on that bug, you should be able to either use \\ or / for the path separator there and it should work.

Mike

On Jan 22, 2010, at 9:18 AM, Diego Toro wrote:

> Hi, with svn r16440 the problem persists, I creted a jira report http://jira.freeswitch.org/browse/LBSNDF-8 this is a minor issue, but activing playback delimiter no audio file can be played. On FS the audio files are placed in the \sound\ directory, building the path on Windows would be \sound '\s' which is replaced by 'ound'.
> 
> Thank you
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 



From mike at jerris.com  Sun Jan 24 02:08:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 05:08:20 -0500
Subject: [Freeswitch-users] Distinguishing Blind REFER from application
	transfer
In-Reply-To: <15b9404e1001220747y79571117qab826152a2df1dc8@mail.gmail.com>
References: <15b9404e1001220322h5636c1aat968f515e0601d769@mail.gmail.com>
	<9F3D655F-BF09-4503-9709-953A839DC65C@freeswitch.org>
	<15b9404e1001220747y79571117qab826152a2df1dc8@mail.gmail.com>
Message-ID: <78A051B8-57E4-44ED-BFCC-17AB7C41B366@jerris.com>

I think they are intended to be there as long as you want them to be, up until they can be used in the cdr modules.  If you are looking for this much control, you would have to set them off to other vars, use some scripting language to manipulate, or use some socket based control mechanism.  

Mike


On Jan 22, 2010, at 10:47 AM, mayamatakeshi wrote:

> 
> On Fri, Jan 22, 2010 at 11:17 PM, Brian West <brian at freeswitch.org> wrote:
> Have you done a uuid_dump to see all the variables?
> 
> I just tried that with trunk. I can see the REFER variables stay set till the end of the call.
> They will show up in CHANNEL_HANGUP_COMPLETE:
> 
> variable_sip_h_Referred-By: user2 <sip:user2 at test1.com>
> variable_sip_refer_to: <sip:2000 at test1.com>
> 
> I suppose the only thing that will change them is another blind refer. But they will never be unset.
> 
> 
> 
> On Jan 22, 2010, at 5:22 AM, mayamatakeshi wrote:
> 
> > Before doing so I thought in ask if the REFER-related variables being preserved upon dialplan reentry would not be a bug (well, it could be a feature useful in some scenarios I suspect).
> 
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From mike at jerris.com  Sun Jan 24 02:13:32 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 05:13:32 -0500
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <c4d05cbe1001230941ud426917m77aa774430110a2b@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
	<7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>
	<7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>
	<c4d05cbe1001230941ud426917m77aa774430110a2b@mail.gmail.com>
Message-ID: <AED7091C-7115-4B19-B555-02E8BDBFD1EE@jerris.com>

Also note, we just made a backwards incompatible change to boost in the latest svn..  this will require sangoma_prid version 1.48 or later.  I don't think this is packaged up with wanpipe anywhere yet.  We are doing more validation of the new code, and updated driver packages should be available very soon.

Mike

On Jan 23, 2010, at 12:41 PM, Moises Silva wrote:

> On Fri, Jan 22, 2010 at 11:32 PM, Thangappan.M <thangappan143 at gmail.com> wrote:
> But I am unable to make a outbound call. When I was trying the following was get.
> 
> freeswitch at internal> originate openzap/smg_prid/a/9940464753 at g1 openzap/smg_prid/a/9940464753 at g1
> -ERR NORMAL_CIRCUIT_CONGESTION
> 
> 2010-01-23 10:00:45.688854 [WARNING] ozmod_sangoma_boost.c:348 TX EVENT: CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[9940464753] Ci=[0000000000] Rdnis=[]
> freeswitch at internal> 2010-01-23 10:00:46.709355 [WARNING] ozmod_sangoma_boost.c:1373 RX EVENT (N): CALL_START_NACK:(82) [w256g256] Rc=0 CSid=2 Seq=2
> 2010-01-23 10:00:46.709355 [WARNING] sangoma_boost_client.c:220 TX EVENT (N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=2 Seq=3
> 2010-01-23 10:00:46.709355 [ERR] mod_openzap.c:1162 No channels available
> 2010-01-23 10:00:46.709355 [ERR] switch_ivr_originate.c:2411 Cannot create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
> 2010-01-23 10:00:46.709355 [DEBUG] switch_ivr_originate.c:3211 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
> 
> Please help me...........
> 
> Did you define group 1 in /etc/wanpipe/smg_prid.conf, pastebin the file plz.
> 
> -- 
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Sun Jan 24 02:17:33 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 05:17:33 -0500
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <AC76DB08-C278-4F59-90D0-2C3724CDB39A@freeswitch.org>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
	<d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
	<201001231355.31032.sos@sokhapkin.dyndns.org>
	<AC76DB08-C278-4F59-90D0-2C3724CDB39A@freeswitch.org>
Message-ID: <5F6D7DFA-2432-4542-813D-3E77FFF92DCD@jerris.com>

I have looked into the code and backtrace related to this issue, and the best I can see, its impossible...  We will definitely need access to a box this is reproducible on to fix this issue.

Mike

On Jan 23, 2010, at 2:02 PM, Brian West wrote:

> Best to find us on IRC when anthm is around and lets get into your box and fix this.
> 
> /b
> 
> On Jan 23, 2010, at 12:55 PM, Sergey Okhapkin wrote:
> 
>> GOOD! You have a consistent way to reproduce the problem!
> 



From b_ball_henry at hotmail.com  Sun Jan 24 02:18:04 2010
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Sun, 24 Jan 2010 02:18:04 -0800
Subject: [Freeswitch-users] Accessing Sangoma card inside a openVZ container
Message-ID: <59ad9ca11001240218w3accbb12r462e5c2fa7511024@mail.gmail.com>

I know there are a couple of OpenVZ expert here on the mailing list.

Has anyone of you tried to run freeswitch in a container with access to
physical Sangoma card?
The reason for this is that I would like to create an "all in one box" for
small to medium size company that takes care of the most essential services.
Things like an IP PBX would be a huge plus. I have been playing with OpenVZ
for a while now, and I know how to let containers access devices in the /dev
directory from the physical box. But to be able to use something like
sangoma card, I think there are some things in the /proc directory need to
be made available to the containers. But I don't know how to do this yet.
Maybe some of you know..

Thanks,

-- 
Henry Huang
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From mike at jerris.com  Sun Jan 24 02:20:23 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 05:20:23 -0500
Subject: [Freeswitch-users] FreeSwitch Integration With Sailfin
In-Reply-To: <BAY143-W760D1216FA26D7EFD8E9DA1600@phx.gbl>
References: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>,
	<CFA0D82A-33B3-4B50-B6AE-BAA61E0862A4@freeswitch.org>,
	<BAY143-W13760770FF3FCA224AACEA1610@phx.gbl>,
	<9dc4a1671001231501t65b1281epeeb7e7fa1b0bf5bf@mail.gmail.com>
	<BAY143-W760D1216FA26D7EFD8E9DA1600@phx.gbl>
Message-ID: <84AB66EA-956C-4B42-87C8-42660CBEDCE3@jerris.com>

This is a good place to start reading on how to configure dialplan:

http://wiki.freeswitch.org/wiki/Dialplan
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#dialplan

Mike

On Jan 23, 2010, at 11:48 PM, juan camilo ospina quintero wrote:

> Hi 
> thanks for fast answers
> i already have sailfin installed and runnig, also freeswitch, both on the same machine.
> now my quiestion is how to redirect o forward the SIP flow from freeswitch to salifin, the softphones must register in freeswitch, but to make a call the invite should go through sailfin and get back to freeswitch, this is for implementing some services of VoIP, for a project i'm on.

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From yehavi.bourvine at gmail.com  Sun Jan 24 02:23:51 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 24 Jan 2010 12:23:51 +0200
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <AC76DB08-C278-4F59-90D0-2C3724CDB39A@freeswitch.org>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
	<d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
	<201001231355.31032.sos@sokhapkin.dyndns.org>
	<AC76DB08-C278-4F59-90D0-2C3724CDB39A@freeswitch.org>
Message-ID: <d361ca7b1001240223s6d88d86ardc05b9c573987559@mail.gmail.com>

About getting into my box: It is a production machine with quite tight
access control, so this is close to impossible...
I am trying to reproduce it on a backup system. If it works, then I might be
able to give access to it. I am working on it now...

                          __Yehavi:

2010/1/23 Brian West <brian at freeswitch.org>

> Best to find us on IRC when anthm is around and lets get into your box and
> fix this.
>
> /b
>
> On Jan 23, 2010, at 12:55 PM, Sergey Okhapkin wrote:
>
> > GOOD! You have a consistent way to reproduce the problem!
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mcampbellsmith at gmail.com  Sun Jan 24 02:47:51 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 24 Jan 2010 21:47:51 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
Message-ID: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>

Hi!

Is there any way that a custom NOTIFY message can be built and sent in
FS without cutting code?

I have a bunch of Linksys SPA's and I want to implement the
Resync_From_SIP option, which enables a resync to be triggered via a
SIP NOTIFY message. (ie Event: resync in the NOTIFY message)

Is this at all possible?

Thanks!


From mcampbellsmith at gmail.com  Sun Jan 24 03:28:57 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 24 Jan 2010 22:28:57 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
Message-ID: <33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>

Actually I just found http://wiki.freeswitch.org/wiki/Mod_event_socket

If I telnet to FS as described
http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do I
just need to enter somthing like:

sendevent NOTIFY
profile: internal
event-string: resync
user: 1000
host: 192.168.1.121
content-type: application/simple-message-summary

where 192.168.1.121 is the ip address of one of the Linksys devices?

I don't see any messages sent when I do this.  What am I doing wrong?

Thanks

On Sun, Jan 24, 2010 at 9:47 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Hi!
>
> Is there any way that a custom NOTIFY message can be built and sent in
> FS without cutting code?
>
> I have a bunch of Linksys SPA's and I want to implement the
> Resync_From_SIP option, which enables a resync to be triggered via a
> SIP NOTIFY message. (ie Event: resync in the NOTIFY message)
>
> Is this at all possible?
>
> Thanks!
>


From mcampbellsmith at gmail.com  Sun Jan 24 03:41:26 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 24 Jan 2010 22:41:26 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
Message-ID: <33c87fa31001240341x58f1c953n8a1e958a867591cd@mail.gmail.com>

Sorry for the spam ...

Playing around with this a bit more and I noticed that host should be
the ip address of FS, not the Linksys ATA.

How do I authorize the NOTIFY message?  I see FS tries to Authorize
but uses one of my external sip profiles in the authorization details,
instead of the extension details.

For example the following is sent:

   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bKevv1r7F0SNSUm
   Max-Forwards: 70
   From: <sip:1000 at 192.168.1.120>;tag=ZHS035c6yc4rD
   To: <sip:1000 at 192.168.1.120>
   Call-ID: 488d261f-837f-122d-7ba9-00e04c0312e9
   CSeq: 126048101 NOTIFY
   Contact: <sip:mod_sofia at 192.168.1.120:5060>
   Expires: 3590
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Event: resync
   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Subscription-State: terminated;reason=timeout
   Authorization: Digest username="XXXXXXXXXXXX",
realm="192.168.1.120", nonce="efe79af6",
cnonce="SJIYXoN/Ei2pewDgTAMS6Q", algorithm=MD5,
uri="sip:1000 at 192.168.1.121:5060",
response="9b27c57170fca740df4f538634e2e407", qop=auth, nc=00000001
   Content-Type: application/simple-message-summary
   Content-Length: 0

The username XXXXXXXXXXXX is one of my SIP profiles - I would have
expected FS to use the profile for extension 1000 here.  How can I do
that?

Thanks!

On Sun, Jan 24, 2010 at 10:28 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Actually I just found http://wiki.freeswitch.org/wiki/Mod_event_socket
>
> If I telnet to FS as described
> http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do I
> just need to enter somthing like:
>
> sendevent NOTIFY
> profile: internal
> event-string: resync
> user: 1000
> host: 192.168.1.121
> content-type: application/simple-message-summary
>
> where 192.168.1.121 is the ip address of one of the Linksys devices?
>
> I don't see any messages sent when I do this. ?What am I doing wrong?
>
> Thanks
>
> On Sun, Jan 24, 2010 at 9:47 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>> Hi!
>>
>> Is there any way that a custom NOTIFY message can be built and sent in
>> FS without cutting code?
>>
>> I have a bunch of Linksys SPA's and I want to implement the
>> Resync_From_SIP option, which enables a resync to be triggered via a
>> SIP NOTIFY message. (ie Event: resync in the NOTIFY message)
>>
>> Is this at all possible?
>>
>> Thanks!
>>
>


From tzury.by at reguluslabs.com  Sun Jan 24 04:25:14 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Sun, 24 Jan 2010 14:25:14 +0200
Subject: [Freeswitch-users] strange behavior of openzap/libpri (is that a
	bug?)
Message-ID: <10128ef11001240425x5481679ahcebb94efce745933@mail.gmail.com>

Hi All,

In my environment I have the following components:
* FreeSWITCH Version 1.0.4
* latest libpri
* latest openZAP

I have noticed that when originating calls to a specific number, the
system sometimes reports this as UNALLOCATED_NUMBER and sometimes not.
That is, I am calling a number which certainly exists, however, in
most of the attempts to cal this number I get the UNALLOCATED_NUMBER
error, while every 4th or 5th attempt the line get connected.

I paste-bin the log at http://pastebin.freeswitch.org/11928.
There is one with oz debug output at http://pastebin.freeswitch.org/11929

please advise,
Tzury


From yehavi.bourvine at gmail.com  Sun Jan 24 06:42:23 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 24 Jan 2010 16:42:23 +0200
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001240223s6d88d86ardc05b9c573987559@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
	<d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
	<201001231355.31032.sos@sokhapkin.dyndns.org>
	<AC76DB08-C278-4F59-90D0-2C3724CDB39A@freeswitch.org>
	<d361ca7b1001240223s6d88d86ardc05b9c573987559@mail.gmail.com>
Message-ID: <d361ca7b1001240642s680e770ao749dd9949cc14850@mail.gmail.com>

I've tried printing the value of retval and it is 1 (and not 6...).  BTW,
after adding the printf()'s I cannot examine anymore the value of retval in
gdb as it is optimized away. Maybe this is the reason for a value of 6 which
does not exist in the code that returns it.

nua_base_server_report() returns one as "terminated" variable is false.
Hope this helps.

                           Regards, __Yehavi:
2010/1/24 Yehavi Bourvine <yehavi.bourvine at gmail.com>

> About getting into my box: It is a production machine with quite tight
> access control, so this is close to impossible...
> I am trying to reproduce it on a backup system. If it works, then I might
> be able to give access to it. I am working on it now...
>
>                           __Yehavi:
>
>  2010/1/23 Brian West <brian at freeswitch.org>
>
>> Best to find us on IRC when anthm is around and lets get into your box and
>> fix this.
>>
>>
>> /b
>>
>> On Jan 23, 2010, at 12:55 PM, Sergey Okhapkin wrote:
>>
>> > GOOD! You have a consistent way to reproduce the problem!
>>
>>
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
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From Russell.Mosemann at cune.org  Sun Jan 24 07:13:30 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sun, 24 Jan 2010 09:13:30 -0600
Subject: [Freeswitch-users] strange behavior of openzap/libpri (is that
	abug?)
In-Reply-To: <10128ef11001240425x5481679ahcebb94efce745933@mail.gmail.com>
References: <10128ef11001240425x5481679ahcebb94efce745933@mail.gmail.com>
Message-ID: <DA783286028641449A0B00D75CA40183@cune.pri>

Tzury Bar Yochay wrote:

> In my environment I have the following components:
> * FreeSWITCH Version 1.0.4

That is an old version. Upgrade to the latest version and try it, again.

--
Russell Mosemann



From a.afzali2003 at gmail.com  Sun Jan 24 07:39:16 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sun, 24 Jan 2010 19:09:16 +0330
Subject: [Freeswitch-users] How to get chat message via event
Message-ID: <debff5a81001240739q3dbe4680uf15fa1ce4ee79bf7@mail.gmail.com>

Hi,

It seems that the chat messages don't fire via events by default and just
exchange between parties.
Is it true? Is it possible to enable those via events?

appreciate all,
-- afshin
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From gmaruzz at celliax.org  Sun Jan 24 07:48:32 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 24 Jan 2010 16:48:32 +0100
Subject: [Freeswitch-users] How to get chat message via event
In-Reply-To: <debff5a81001240739q3dbe4680uf15fa1ce4ee79bf7@mail.gmail.com>
References: <debff5a81001240739q3dbe4680uf15fa1ce4ee79bf7@mail.gmail.com>
Message-ID: <7b197bef1001240748r248f53b8i5794f7cbc1fedfb1@mail.gmail.com>

you subscribe to them as MESSAGE events

eg, from a telnet session:

telnet localhost 8021
auth ClueCon
events plain message

then those events will show up in your telnet session.
-gm

On Sun, Jan 24, 2010 at 4:39 PM, afshin afzali <a.afzali2003 at gmail.com> wrote:
> Hi,
>
> It seems that the chat messages don't fire via events by default and just
> exchange between parties.
> Is it true? Is it possible to enable those via events?
>
> appreciate all,
> -- afshin
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From mailinglist at fribert.dk  Sun Jan 24 08:39:47 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Sun, 24 Jan 2010 17:39:47 +0100
Subject: [Freeswitch-users] Svar: Re: How do I invite group to join			
	existing call?
In-Reply-To: <4B5A1D99020000E100000418@mail.fribert.dk>
References: <4B59A44E020000E100000413@mail.fribert.dk>
	<4B5A1D99020000E100000418@mail.fribert.dk>
Message-ID: <4B5C85E3020000E100000425@mail.fribert.dk>

Somebody help me understand this features expression, it sounds like it sort of does what I need, I could just dial a group to have the group ring, and invite a third party to the conversation.
 
In the pfsense package there is this example:
 
<!-- In call Transfer for phones without a transfer button -->
<extension name="dx">
      <condition field="destination_number" expression="^dx$">
       <action application="answer"/>
       <action application="read" data="11 11 'tone_stream://%(10000,0,350,440)' digits 5000 #"/>
       <action application="execute_extension" data="is_transfer XML features"/>
      </condition>
</extension>
But what does it mean, how do I use it during a call?


>>> 22-01-2010 kl. 21:50 skrev "mailinglist" <mailinglist at fribert.dk> i meddelelsen <4B5A1D99020000E100000418 at mail.fribert.dk>:

Hmm, I don't get it, it might not do the right thing.
 
The situation is that I receive a call from the outside, answers it on a phone, and then wants to ask a third (local) party to join the conversation.
 
I thought from the example that I should just press *3, and then the extension I want to invite, but nothing happens.
 
I haven't the faintest how I accomplish this :-o


>>> 22-01-2010 kl. 13:12 skrev "mailinglist" <mailinglist at fribert.dk> i meddelelsen <4B59A44E020000E100000413 at mail.fribert.dk>:

Hi Michael et al.

I found the 'attended xfer' example in the documentation, and it looks like rxactly to what I want to do.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer

The example 2:
Do I add the first part to my dialplan/default.xml, and the second part to my dialplan/features.xml

How does it affect if I try to do a normal local dial if it just reacts on the fact that three digits have been dialed (which I should change to 2, as I use 2 digit local numbers).

Best regards
Fribse

>>> Michael Collins 18-01-10 21:19 >>>


On Sun, Jan 17, 2010 at 10:50 PM, mailinglist <mailinglist at fribert.dk> wrote:


Hi All

I would like to be able to invite a group / global to join an existing call, but how do I accomplish this, can it be done?


Malfunction! Need Input!

Could you give us a few more details on what you're trying to accomplish? What's the big picture? I'm curious what problem you're trying to solve. I'm sure the gang here will have thoughts to pass along. :)

-MC

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From camilin2212 at hotmail.com  Sun Jan 24 08:48:43 2010
From: camilin2212 at hotmail.com (juan camilo ospina quintero)
Date: Sun, 24 Jan 2010 11:48:43 -0500
Subject: [Freeswitch-users] FreeSwitch Integration With Sailfin
In-Reply-To: <84AB66EA-956C-4B42-87C8-42660CBEDCE3@jerris.com>
References: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>, ,
	<CFA0D82A-33B3-4B50-B6AE-BAA61E0862A4@freeswitch.org>, ,
	<BAY143-W13760770FF3FCA224AACEA1610@phx.gbl>, ,
	<9dc4a1671001231501t65b1281epeeb7e7fa1b0bf5bf@mail.gmail.com>,
	<BAY143-W760D1216FA26D7EFD8E9DA1600@phx.gbl>,
	<84AB66EA-956C-4B42-87C8-42660CBEDCE3@jerris.com>
Message-ID: <BAY143-W27CC94D97F584D192C5680A1600@phx.gbl>


Thanks mike,

i already read about dialplan, and it seems that the bridge application
is the one i need, but now i want to know, how to modify dialplan to use bridge
and where to put this in the dialplan:

<action application="bridge" data="sofia/sip/9998881111 at sip.yourprovider.com"/>

this is an example, but i think thats what i need, if someone has work with bridging,
please help with this.
thanks


From: mike at jerris.com
Date: Sun, 24 Jan 2010 05:20:23 -0500
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSwitch Integration With Sailfin



This is a good place to start reading on how to configure dialplan:
http://wiki.freeswitch.org/wiki/Dialplanhttp://wiki.freeswitch.org/wiki/Getting_Started_Guide#dialplan
Mike
On Jan 23, 2010, at 11:48 PM, juan camilo ospina quintero wrote:Hi 
thanks for fast answers
i already have sailfin installed and runnig, also freeswitch, both on the same machine.
now my quiestion is how to redirect o forward the SIP flow from freeswitch to salifin, the softphones must register in freeswitch, but to make a call the invite should go through sailfin and get back to freeswitch, this is for implementing some services of VoIP, for a project i'm on.

 		 	   		  
_________________________________________________________________
Windows Live: Make it easier for your friends to see what you?re up to on Facebook.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009
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From mike at jerris.com  Sun Jan 24 09:44:26 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 12:44:26 -0500
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001240642s680e770ao749dd9949cc14850@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
	<d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
	<201001231355.31032.sos@sokhapkin.dyndns.org>
	<AC76DB08-C278-4F59-90D0-2C3724CDB39A@freeswitch.org>
	<d361ca7b1001240223s6d88d86ardc05b9c573987559@mail.gmail.com>
	<d361ca7b1001240642s680e770ao749dd9949cc14850@mail.gmail.com>
Message-ID: <D56D65EB-B150-4A59-8B66-68BF4C58108B@jerris.com>

Yes, ssh is best.  Are you able to meet up with me on irc so we can  
discuss this issue?

On Jan 24, 2010, at 9:42 AM, Yehavi Bourvine  
<yehavi.bourvine at gmail.com> wrote:

> I've tried printing the value of retval and it is 1 (and not 6...).   
> BTW, after adding the printf()'s I cannot examine anymore the value  
> of retval in gdb as it is optimized away. Maybe this is the reason  
> for a value of 6 which does not exist in the code that returns it.
>
> nua_base_server_report() returns one as "terminated" variable is  
> false.
> Hope this helps.
>
>                            Regards, __Yehavi:
> 2010/1/24 Yehavi Bourvine <yehavi.bourvine at gmail.com>
> About getting into my box: It is a production machine with quite  
> tight access control, so this is close to impossible...
> I am trying to reproduce it on a backup system. If it works, then I  
> might be able to give access to it. I am working on it now...
>
>                           __Yehavi:
>
> 2010/1/23 Brian West <brian at freeswitch.org>
> Best to find us on IRC when anthm is around and lets get into your  
> box and fix this.
>
>
> /b
>
> On Jan 23, 2010, at 12:55 PM, Sergey Okhapkin wrote:
>
> > GOOD! You have a consistent way to reproduce the problem!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From brian at freeswitch.org  Sun Jan 24 09:46:13 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 24 Jan 2010 11:46:13 -0600
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
Message-ID: <99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>

or sofia profile xxx flush_inbound_reg callid reboot

callid you can get from sofia status profile xxx

/b

On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:

> Actually I just found http://wiki.freeswitch.org/wiki/Mod_event_socket
> 
> If I telnet to FS as described
> http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do I
> just need to enter somthing like:
> 
> sendevent NOTIFY
> profile: internal
> event-string: resync
> user: 1000
> host: 192.168.1.121
> content-type: application/simple-message-summary
> 
> where 192.168.1.121 is the ip address of one of the Linksys devices?
> 
> I don't see any messages sent when I do this.  What am I doing wrong?
> 
> Thanks

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From mike at jerris.com  Sun Jan 24 09:46:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 12:46:20 -0500
Subject: [Freeswitch-users] strange behavior of openzap/libpri (is that
	a bug?)
In-Reply-To: <10128ef11001240425x5481679ahcebb94efce745933@mail.gmail.com>
References: <10128ef11001240425x5481679ahcebb94efce745933@mail.gmail.com>
Message-ID: <6A2CAE5A-90FE-441C-B383-DD8ED9B71F25@jerris.com>

Does this same behavior happen with svn trunk?

On Jan 24, 2010, at 7:25 AM, Tzury Bar Yochay  
<tzury.by at reguluslabs.com> wrote:

> Hi All,
>
> In my environment I have the following components:
> * FreeSWITCH Version 1.0.4
> * latest libpri
> * latest openZAP
>
> I have noticed that when originating calls to a specific number, the
> system sometimes reports this as UNALLOCATED_NUMBER and sometimes not.
> That is, I am calling a number which certainly exists, however, in
> most of the attempts to cal this number I get the UNALLOCATED_NUMBER
> error, while every 4th or 5th attempt the line get connected.
>
> I paste-bin the log at http://pastebin.freeswitch.org/11928.
> There is one with oz debug output at http://pastebin.freeswitch.org/11929
>
> please advise,
> Tzury
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From scott.torr.fs at letterboxes.org  Sun Jan 24 09:47:14 2010
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Mon, 25 Jan 2010 04:47:14 +1100
Subject: [Freeswitch-users] Application="record_session" records 16000Hz in
	mono now?
Message-ID: <1264355234.24126.1356300893@webmail.messagingengine.com>

Hi,

When recording a session in the past the caller audio would be in the
right channel and the callee audio in the left of a stereo recording.

<action application="record_session" data="/root/recordings/test.wav" />


A 8000Hz recording has remained the same and is in stereo,

but a 16000Hz recording is now in mono?


Noticed change since FreeSWITCH Version 1.0.trunk (16195).


What do I need to do to get stereo recording back?

Why the change?



Regards,
Scott Torr









From adam.falcone at gmail.com  Sun Jan 24 09:50:59 2010
From: adam.falcone at gmail.com (AFalcon)
Date: Sun, 24 Jan 2010 09:50:59 -0800 (PST)
Subject: [Freeswitch-users] Freeswitch process hangs, losses connection.
Message-ID: <1264355459996-4449929.post@n2.nabble.com>


Hi I am new here but have successfully configured Freeswitch to run on Snow
Leopard. I have configured Snow Leopard so that the computer will never go
to sleep. My issue though is that Freeswitch losses it connection at some
point while running. When I call in I get a busy signal or a message saying
the phone number is not in operation. 

To get around this I wrote a launchd process that restarts Freeswitch and my
softphone every 4 hours. 

I don't like this and am wondering how to go about trouble shooting the
issue of Freeswitch losing connectivity after a period of a few hours. 

Any help would be appreciated. Thanks.
-- 
View this message in context: http://n2.nabble.com/Freeswitch-process-hangs-losses-connection-tp4449929p4449929.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Sun Jan 24 09:52:29 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 24 Jan 2010 11:52:29 -0600
Subject: [Freeswitch-users] Application="record_session" records 16000Hz
	in mono now?
In-Reply-To: <1264355234.24126.1356300893@webmail.messagingengine.com>
References: <1264355234.24126.1356300893@webmail.messagingengine.com>
Message-ID: <7A25419A-110C-4390-A2B4-9AF8D9700B22@freeswitch.org>

Set the variable record_stereo=true

/b

On Jan 24, 2010, at 11:47 AM, Scott Torr wrote:

> Hi,
> 
> When recording a session in the past the caller audio would be in the
> right channel and the callee audio in the left of a stereo recording.
> 
> <action application="record_session" data="/root/recordings/test.wav" />
> 
> 
> A 8000Hz recording has remained the same and is in stereo,
> 
> but a 16000Hz recording is now in mono?
> 
> 
> Noticed change since FreeSWITCH Version 1.0.trunk (16195).
> 
> 
> What do I need to do to get stereo recording back?
> 
> Why the change?
> 
> 
> 
> Regards,
> Scott Torr
> 
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From scott.torr.fs at letterboxes.org  Sun Jan 24 10:27:05 2010
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Mon, 25 Jan 2010 05:27:05 +1100
Subject: [Freeswitch-users] Application="record_session" records 16000Hz
 in mono now?
In-Reply-To: <7A25419A-110C-4390-A2B4-9AF8D9700B22@freeswitch.org>
References: <1264355234.24126.1356300893@webmail.messagingengine.com>
	<7A25419A-110C-4390-A2B4-9AF8D9700B22@freeswitch.org>
Message-ID: <1264357625.29510.1356306415@webmail.messagingengine.com>

Yeap,
That did the trick :)

<action application="set" data="RECORD_STEREO=true"/>
<action application="record_session" data="/root/recordings/test.wav" />

D'oh!
And right there in the wiki:(
It was an unexpected change to the previous default, thanks.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session


regards,
Scott Torr


On Sun, 24 Jan 2010 11:52 -0600, "Brian West" <brian at freeswitch.org>
wrote:
> Set the variable record_stereo=true
> 
> /b
> 
> On Jan 24, 2010, at 11:47 AM, Scott Torr wrote:
> 
> > Hi,
> > 
> > When recording a session in the past the caller audio would be in the
> > right channel and the callee audio in the left of a stereo recording.
> > 
> > <action application="record_session" data="/root/recordings/test.wav" />
> > 
> > 
> > A 8000Hz recording has remained the same and is in stereo,
> > 
> > but a 16000Hz recording is now in mono?
> > 
> > 
> > Noticed change since FreeSWITCH Version 1.0.trunk (16195).
> > 
> > 
> > What do I need to do to get stereo recording back?
> > 
> > Why the change?
> > 
> > 
> > 
> > Regards,
> > Scott Torr
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From tayeb.meftah at gmail.com  Sun Jan 24 10:31:38 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 24 Jan 2010 19:31:38 +0100
Subject: [Freeswitch-users] strange behavior of openzap/libpri (is that
 a bug?)
In-Reply-To: <10128ef11001240425x5481679ahcebb94efce745933@mail.gmail.com>
References: <10128ef11001240425x5481679ahcebb94efce745933@mail.gmail.com>
Message-ID: <4B5C920A.2040602@gmail.com>

hi,
upgrade to the latest 1.0.5 release at:
http://latest.freeswitch.org
thanks
24/01/2010 13:25, Tzury Bar Yochay a ?crit :
> Hi All,
>
> In my environment I have the following components:
> * FreeSWITCH Version 1.0.4
> * latest libpri
> * latest openZAP
>
> I have noticed that when originating calls to a specific number, the
> system sometimes reports this as UNALLOCATED_NUMBER and sometimes not.
> That is, I am calling a number which certainly exists, however, in
> most of the attempts to cal this number I get the UNALLOCATED_NUMBER
> error, while every 4th or 5th attempt the line get connected.
>
> I paste-bin the log at http://pastebin.freeswitch.org/11928.
> There is one with oz debug output at http://pastebin.freeswitch.org/11929
>
> please advise,
> Tzury
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    



From brian at freeswitch.org  Sun Jan 24 10:39:45 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 24 Jan 2010 12:39:45 -0600
Subject: [Freeswitch-users] Application="record_session" records 16000Hz
	in mono now?
In-Reply-To: <1264357625.29510.1356306415@webmail.messagingengine.com>
References: <1264355234.24126.1356300893@webmail.messagingengine.com>
	<7A25419A-110C-4390-A2B4-9AF8D9700B22@freeswitch.org>
	<1264357625.29510.1356306415@webmail.messagingengine.com>
Message-ID: <DA0DDF32-DBA1-4753-8357-AB86DA516C98@freeswitch.org>

It didn't change.  I suspect you removed the directory entry or its an inbound call to a user... in the defaults its always been in conf/directory/default.xml as a variable.

/b

On Jan 24, 2010, at 12:27 PM, Scott Torr wrote:

> It was an unexpected change to the previous default, thanks.



From mastermind202 at gmail.com  Sun Jan 24 10:55:36 2010
From: mastermind202 at gmail.com (mm_202)
Date: Sun, 24 Jan 2010 13:55:36 -0500
Subject: [Freeswitch-users] reloadxml/rescan profile and killgw
In-Reply-To: <3F378D25-85F8-48B9-B814-7F7979EFE840@jerris.com>
References: <4B589A6E.8010205@kinetix.gr>
	<3F378D25-85F8-48B9-B814-7F7979EFE840@jerris.com>
Message-ID: <63de75711001241055p129739behbefda343b2435a08@mail.gmail.com>

I agree that it should be a different command (breaking backwards
compatibility is bad).

I would suggest something like 'sofia profile [profilename] update' that
would first run rescan, then any gateways that were not present can be
considered stale and deleted.

Apostolos:  How are you C skills?  ;)

-- mm_202.

On Sun, Jan 24, 2010 at 4:50 AM, Michael Jerris <mike at jerris.com> wrote:

> A lot of things would be nice.  Patches for example.  I find them quite
> nice.  This should probably be a different command or argument.
>
> Mike
>
> On Jan 21, 2010, at 1:18 PM, Apostolos Pantsiopoulos wrote:
>
> > Wouldn't it be nice a "reloadxml"/"rescan profile" to delete all
> > nonexistent gateways (when they are absent in the xml config) without
> > having to issue a "sofia profile xxxxx killgw yyyyyy" command?
> >
> > I always seem to find forgotten gateway's in the profile because of
> > this... Any thoughts?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From testa at voicetechnology.com.br  Sun Jan 24 11:21:51 2010
From: testa at voicetechnology.com.br (Fernando Gregianin Testa)
Date: Sun, 24 Jan 2010 17:21:51 -0200
Subject: [Freeswitch-users] Lua: io.popen/read blocking in
	other	incoming calls
In-Reply-To: <11387916.58.1263943057280.JavaMail.root@zimbra>
References: <11387916.58.1263943057280.JavaMail.root@zimbra>
Message-ID: <6BA36994-7FA2-44F2-9592-78597BB9E9A4@voicetechnology.com.br>

You may consider use lua socket.http package as an alternative to popen+wget. Check:
 https://web.tecgraf.puc-rio.br/luasocket/ 
http://www.tecgraf.puc-rio.br/~diego/professional/luasocket/http.html 

Maybe you can be interested also in http://github.com/fertesta/restinlua 

Em 19/01/2010, ?s 21:17, Dan escreveu:

> I would, but I need to post a a wav file that gets recorded, I didn't see a way to supply the location of a file to use as the post data.  It looks like you have to url encode the data in the script and pass it all in the call. 
> 
> Thanks
> Dan-
> ----- Original Message -----
> From: "Rupa Schomaker" <rupa at rupa.com>
> To: "freeswitch-users" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, January 19, 2010 4:06:36 PM
> Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other        incoming calls
> 
> 
> 
> On Tue, Jan 19, 2010 at 3:03 PM, Dan <freeswitch-users at digitaldan.com> wrote:
> My lua script is calling wget through lua's io.popen to send and receive data from a web service. While the f:read to wget  is running, other incoming calls will block on the same io.popen call until the first call closes the pipe (with f:close()).  
> 
> You might want to look at the api that mod_curl exposes to do what you want.  No need to do an expensive system call just to call a webservice.
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Fernando Gregianin Testa
testa at voicetechnology.com.br
+55 11 35882166



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From mrene_lists at avgs.ca  Sun Jan 24 11:42:40 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sun, 24 Jan 2010 14:42:40 -0500
Subject: [Freeswitch-users] Accessing Sangoma card inside a openVZ
	container
In-Reply-To: <59ad9ca11001240218w3accbb12r462e5c2fa7511024@mail.gmail.com>
References: <59ad9ca11001240218w3accbb12r462e5c2fa7511024@mail.gmail.com>
Message-ID: <E93E560E-3EF8-475C-B7C7-72155E600355@avgs.ca>

As far as openzap is concerned you only have to expose the proper  
device files in /dev, and allow then in your VE's config. The driver  
always runs on the host's kernel.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 24-Jan-10, at 5:18 AM, Henry Huang wrote:

> I know there are a couple of OpenVZ expert here on the mailing list.
>
> Has anyone of you tried to run freeswitch in a container with access  
> to physical Sangoma card?
> The reason for this is that I would like to create an "all in one  
> box" for small to medium size company that takes care of the most  
> essential services. Things like an IP PBX would be a huge plus. I  
> have been playing with OpenVZ for a while now, and I know how to let  
> containers access devices in the /dev directory from the physical  
> box. But to be able to use something like sangoma card, I think  
> there are some things in the /proc directory need to be made  
> available to the containers. But I don't know how to do this yet.  
> Maybe some of you know..
>
> Thanks,
>
> -- 
> Henry Huang
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From a.afzali2003 at gmail.com  Sun Jan 24 12:59:10 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 25 Jan 2010 00:29:10 +0330
Subject: [Freeswitch-users] How to get chat message via event
In-Reply-To: <7b197bef1001240748r248f53b8i5794f7cbc1fedfb1@mail.gmail.com>
References: <debff5a81001240739q3dbe4680uf15fa1ce4ee79bf7@mail.gmail.com>
	<7b197bef1001240748r248f53b8i5794f7cbc1fedfb1@mail.gmail.com>
Message-ID: <debff5a81001241259q3bc2c82dl40c73ac81e43b40b@mail.gmail.com>

Hi,

As you say, I've already done and unfortunately did not get the message
events although other events are fired as expected :(

-- afshin

On Sun, Jan 24, 2010 at 7:18 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> you subscribe to them as MESSAGE events
>
> eg, from a telnet session:
>
> telnet localhost 8021
> auth ClueCon
> events plain message
>
> then those events will show up in your telnet session.
> -gm
>
> On Sun, Jan 24, 2010 at 4:39 PM, afshin afzali <a.afzali2003 at gmail.com>
> wrote:
> > Hi,
> >
> > It seems that the chat messages don't fire via events by default and just
> > exchange between parties.
> > Is it true? Is it possible to enable those via events?
> >
> > appreciate all,
> > -- afshin
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at celliax.org  Sun Jan 24 13:05:54 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 24 Jan 2010 22:05:54 +0100
Subject: [Freeswitch-users] How to get chat message via event
In-Reply-To: <debff5a81001241259q3bc2c82dl40c73ac81e43b40b@mail.gmail.com>
References: <debff5a81001240739q3dbe4680uf15fa1ce4ee79bf7@mail.gmail.com> 
	<7b197bef1001240748r248f53b8i5794f7cbc1fedfb1@mail.gmail.com> 
	<debff5a81001241259q3bc2c82dl40c73ac81e43b40b@mail.gmail.com>
Message-ID: <7b197bef1001241305o3ee9bc9cq7decdc412f90575c@mail.gmail.com>

Which events you don't get? From which channel in which circumstances?
(I mean what you do and what do you expect?)

-giovanni

On Sun, Jan 24, 2010 at 9:59 PM, afshin afzali <a.afzali2003 at gmail.com> wrote:
> Hi,
>
> As you say, I've already done and unfortunately did not get the message
> events although other events are fired as expected :(
>
> -- afshin
>
> On Sun, Jan 24, 2010 at 7:18 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> you subscribe to them as MESSAGE events
>>
>> eg, from a telnet session:
>>
>> telnet localhost 8021
>> auth ClueCon
>> events plain message
>>
>> then those events will show up in your telnet session.
>> -gm
>>
>> On Sun, Jan 24, 2010 at 4:39 PM, afshin afzali <a.afzali2003 at gmail.com>
>> wrote:
>> > Hi,
>> >
>> > It seems that the chat messages don't fire via events by default and
>> > just
>> > exchange between parties.
>> > Is it true? Is it possible to enable those via events?
>> >
>> > appreciate all,
>> > -- afshin
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From tjardick at vanderkraan.net  Sun Jan 24 12:51:17 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Sun, 24 Jan 2010 21:51:17 +0100
Subject: [Freeswitch-users] Compile error sofia on Mac OS X
Message-ID: <F026A1E8-211C-444E-B6FA-AAFCDBE806C5@vanderkraan.net>

Hi,

I'm trying to compile freeswitch on my MacBook pro to have a local dev  
instance, but i run in to the following compile error during the make:

Compiling mod_sofia.c ...
cc1: warnings being treated as errors
mod_sofia.c: In function 'sofia_receive_message':
mod_sofia.c:1446: warning: 'from_host' may be used uninitialized in  
this function
mod_sofia.c:1446: warning: 'from_user' may be used uninitialized in  
this function
make[5]: *** [mod_sofia_la-mod_sofia.lo] Error 1
make[4]: *** [all] Error 2
make[3]: *** [mod_sofia-all] Error 1
make[2]: *** [all-recursive] Error 1


It's an MBP running Leopard version 10.5.8

Any help would be appreciated.

Kind regards,

Tjardick
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From brian at freeswitch.org  Sun Jan 24 13:15:15 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 24 Jan 2010 15:15:15 -0600
Subject: [Freeswitch-users] Compile error sofia on Mac OS X
In-Reply-To: <F026A1E8-211C-444E-B6FA-AAFCDBE806C5@vanderkraan.net>
References: <F026A1E8-211C-444E-B6FA-AAFCDBE806C5@vanderkraan.net>
Message-ID: <9AEDA6E3-E875-4E63-A4A5-9457317C1D1A@freeswitch.org>

Update this has been fixed now.

Thanks,
Brian

On Jan 24, 2010, at 2:51 PM, Tjardick van der Kraan wrote:

> Hi,
> 
> I'm trying to compile freeswitch on my MacBook pro to have a local dev instance, but i run in to the following compile error during the make:
> 
> Compiling mod_sofia.c ...
> cc1: warnings being treated as errors
> mod_sofia.c: In function 'sofia_receive_message':
> mod_sofia.c:1446: warning: 'from_host' may be used uninitialized in this function
> mod_sofia.c:1446: warning: 'from_user' may be used uninitialized in this function
> make[5]: *** [mod_sofia_la-mod_sofia.lo] Error 1
> make[4]: *** [all] Error 2
> make[3]: *** [mod_sofia-all] Error 1
> make[2]: *** [all-recursive] Error 1
> 
> 
> It's an MBP running Leopard version 10.5.8
> 
> Any help would be appreciated.
> 
> Kind regards,
> 
> Tjardick
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jmesquita at freeswitch.org  Sun Jan 24 15:47:16 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 24 Jan 2010 21:47:16 -0200
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <13529f9d1001222200p44afd96ema21729d0038e89ba@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
	<1264092620.14614.73.camel@luna.tc.commsmundi.com>
	<13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
	<ebbc2d551001211721n51cb5c54m996e0d494842c383@mail.gmail.com>
	<13529f9d1001211806o35fc3435j93b71bbb2ede3028@mail.gmail.com>
	<13529f9d1001222200p44afd96ema21729d0038e89ba@mail.gmail.com>
Message-ID: <ebbc2d551001241547j7e11d4e4se093ce5a60e008b5@mail.gmail.com>

What user_name? I don't understand that statement.

I think that you can always use
http://wiki.freeswitch.org/wiki/Channel_Variables#api_hangup_hook

Regards,
Jo?o Mesquita
FSComm Developer


On Sat, Jan 23, 2010 at 4:00 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:

> Hi Jo?o, do you know how to sign the agent off automatically when either
> party hangs up the call?
>
> Here's how I originate the call to the agent and sign him up in ACD1:
>
> originate skypiax/ANY/jingwei.yang 6*1
>
> However, I found the user_name property is empty. May I know how it is set?
>
>
> Thanks and best regards,
> -Jingwei
>
> On Fri, Jan 22, 2010 at 10:06 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>
>> Hi Jo?o, thanks for the reply. I'll try it out.
>>
>> Regards,
>> -Jingwei
>>
>> 2010/1/22 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> Jingwei, check my contrib dir. I think it may help you with one FIFO since
>>> we are able there to sign in and sign off dynamic agents as well as
>>> customize what we do when the FIFO makes a call to them.
>>>
>>> Regards,
>>> Jo?o Mesquita
>>> FSComm Developer
>>>
>>>
>>> On Thu, Jan 21, 2010 at 10:05 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>>>
>>>> Thanks for the reply. All the agents are dynamic and I can't predefine
>>>> them in the config file.
>>>>
>>>> Regards,
>>>> -Jingwei
>>>>
>>>>
>>>> On Fri, Jan 22, 2010 at 12:50 AM, Fran?ois Delawarde <
>>>> fdelawarde at wirelessmundi.com> wrote:
>>>>
>>>>> Why do you need 2 fifos? You could have callback agents connected to
>>>>> the
>>>>> fifo and send incoming calls there, mod_fifo will do the rest.
>>>>>
>>>>> To configure agents for callback:
>>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>>>>>
>>>>> To place a call into a fifo:
>>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue
>>>>>
>>>>> Fran?ois.
>>>>>
>>>>> On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
>>>>> > Sorry about the confusion, I'm just trying to think over all the
>>>>> > abnormal situations before the implementation and hope the flow has
>>>>> no
>>>>> > design flaws.
>>>>> >
>>>>> > Client A is parked in Queue 1, multiple consumers will be ringed to
>>>>> > answer him. And once the first one is connected, all the rest will
>>>>> > hang up. This is the targeted function.
>>>>> >
>>>>> > To achieve this, I'm considering to originate a call to each consumer
>>>>> > and put the calls in Queue 2. Then intercept client A to the first
>>>>> > element of Queue 2.
>>>>> >
>>>>> > I'm not sure if it's the usual or the best way. If you feel not,
>>>>> > please don't hesitate to correct me. Any thoughts are warmly
>>>>> > appreciated.
>>>>> >
>>>>> > On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com>
>>>>> wrote:
>>>>> >         what no errors mean? how do you originate calls to consumers?
>>>>> >         I don't
>>>>> >         understand your scenario.
>>>>> >
>>>>> >         2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
>>>>> >
>>>>> >         > Hi All,
>>>>> >         >
>>>>> >         > Please advise whether the following flow makes sense.
>>>>> >         >
>>>>> >         > 1. Client A calls in and parked in Queue 1
>>>>> >         > 2. Originate calls to several consumers simultaneously and
>>>>> >         park them in
>>>>> >         > Queue 2
>>>>> >         > 3. Intercept A's call to the first consumer of Queue 2
>>>>> >         >
>>>>> >         > Basically I want A's call picked up by the first among many
>>>>> >         consumers with
>>>>> >         > no errors. Please let me know whether I'm on the right
>>>>> >         track.
>>>>> >         >
>>>>> >         > Thanks and best regards,
>>>>> >         > -Jingwei
>>>>> >         >
>>>>> >
>>>>> >
>>>>> >         _______________________________________________
>>>>> >         FreeSWITCH-users mailing list
>>>>> >         FreeSWITCH-users at lists.freeswitch.org
>>>>> >
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >         UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >         http://www.freeswitch.org
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Sun Jan 24 15:58:35 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 25 Jan 2010 10:58:35 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
Message-ID: <33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>

Thanks Brian.. this still does not work.  Maybe I need to open a Jira?
 Notice the username in the authorization field.  It should be 1000.

Cheers
Mark

freeswitch at internal> sofia status profile internal user 1000 at 192.168.1.120

Registrations:
=================================================================================================
Call-ID:        bd783b73-66877627 at 192.168.1.121
User:           1000 at 192.168.1.120
Contact:        1000 <sip:1000 at 192.168.1.121:5060>
Agent:          Linksys/PAP2T-5.1.6(LS)
Status:         Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
Host:           freeswitch
IP:             192.168.1.121
Port:           5060
Auth-User:      1000
Auth-Realm:     192.168.1.120
MWI-Account:    1000 at 192.168.1.120

=================================================================================================

freeswitch at internal> sofia profile internal flush_inbound_reg
bd783b73-66877627 at 192.168.1.121 reboot
+OK rebooting all registrations matching specified call_id

freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060 at
23:55:49.012627:
   ------------------------------------------------------------------------
   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
   Max-Forwards: 70
   From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
   To: <sip:1000 at 192.168.1.120>
   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
   CSeq: 126070338 NOTIFY
   Contact: <sip:mod_sofia at 192.168.1.120:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Event: reboot_now
   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Subscription-State: terminated;reason=timeout
   Content-Type: application/simple-message-summary
   Content-Length: 0

   ------------------------------------------------------------------------
recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
   From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
   CSeq: 126070338 NOTIFY
   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
   Server: Linksys/PAP2T-5.1.6(LS)
   WWW-Authenticate: Digest realm="192.168.1.120", nonce="8e54805b",
qop="auth", algorithm=md5
   Content-Length: 0

   ------------------------------------------------------------------------
send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
   ------------------------------------------------------------------------
   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
   Max-Forwards: 70
   From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
   To: <sip:1000 at 192.168.1.120>
   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
   CSeq: 126070339 NOTIFY
   Contact: <sip:mod_sofia at 192.168.1.120:5060>
   Expires: 3590
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Event: reboot_now
   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Subscription-State: terminated;reason=timeout
   Authorization: Digest username="1115633124", realm="192.168.1.120",
nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
uri="sip:1000 at 192.168.1.121:5060",
response="747b4d04544c84535dbbd987f2999ca7", qop=auth, nc=00000001
   Content-Type: application/simple-message-summary
   Content-Length: 0

   ------------------------------------------------------------------------
recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
   From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
   CSeq: 126070339 NOTIFY
   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
   Server: Linksys/PAP2T-5.1.6(LS)
   WWW-Authenticate: Digest realm="192.168.1.120", nonce="5339c7ba",
qop="auth", algorithm=md5
   Content-Length: 0

   ------------------------------------------------------------------------

On Mon, Jan 25, 2010 at 4:46 AM, Brian West <brian at freeswitch.org> wrote:
> or sofia profile xxx flush_inbound_reg callid reboot
> callid you can get from sofia status profile xxx
> /b
> On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
>
> Actually I just found?http://wiki.freeswitch.org/wiki/Mod_event_socket
>
> If I telnet to FS as described
> http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do I
> just need to enter somthing like:
>
> sendevent NOTIFY
> profile: internal
> event-string: resync
> user: 1000
> host: 192.168.1.121
> content-type: application/simple-message-summary
>
> where 192.168.1.121 is the ip address of one of the Linksys devices?
>
> I don't see any messages sent when I do this. ?What am I doing wrong?
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From oseslija at gmail.com  Sun Jan 24 16:15:13 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 25 Jan 2010 01:15:13 +0100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
Message-ID: <4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>

You  should not authenticate those NOTIFYs (this will work only with SPA9000
afaik). The option to change for this is in EXT tabs:

Auth Resync-Reboot: No

Also, FSs code will do a reboot of a phone, not resync (it sends reboot_now
event). For that to work a patch is required.

I've just tried to reboot my 942 (rev 16506) and it definitely works.

Regards,
Ognjen


On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Thanks Brian.. this still does not work.  Maybe I need to open a Jira?
>  Notice the username in the authorization field.  It should be 1000.
>
> Cheers
> Mark
>
> freeswitch at internal> sofia status profile internal user 1000 at 192.168.1.120
>
> Registrations:
>
> =================================================================================================
> Call-ID:        bd783b73-66877627 at 192.168.1.121
> User:           1000 at 192.168.1.120
> Contact:        1000 <sip:1000 at 192.168.1.121:5060>
> Agent:          Linksys/PAP2T-5.1.6(LS)
> Status:         Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
> Host:           freeswitch
> IP:             192.168.1.121
> Port:           5060
> Auth-User:      1000
> Auth-Realm:     192.168.1.120
> MWI-Account:    1000 at 192.168.1.120
>
>
> =================================================================================================
>
> freeswitch at internal> sofia profile internal flush_inbound_reg
> bd783b73-66877627 at 192.168.1.121 reboot
> +OK rebooting all registrations matching specified call_id
>
> freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060 at
> 23:55:49.012627:
>   ------------------------------------------------------------------------
>    NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
>   Max-Forwards: 70
>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
>    To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
>    Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>   CSeq: 126070338 NOTIFY
>    Contact: <sip:mod_sofia at 192.168.1.120:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>    Event: reboot_now
>    Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>   Subscription-State: terminated;reason=timeout
>    Content-Type: application/simple-message-summary
>   Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
>   ------------------------------------------------------------------------
>   SIP/2.0 401 Unauthorized
>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>   CSeq: 126070338 NOTIFY
>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
>   Server: Linksys/PAP2T-5.1.6(LS)
>   WWW-Authenticate: Digest realm="192.168.1.120", nonce="8e54805b",
> qop="auth", algorithm=md5
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
>   ------------------------------------------------------------------------
>    NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
>   Max-Forwards: 70
>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
>    To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
>    Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>   CSeq: 126070339 NOTIFY
>    Contact: <sip:mod_sofia at 192.168.1.120:5060>
>   Expires: 3590
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>    Event: reboot_now
>    Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>   Subscription-State: terminated;reason=timeout
>    Authorization: Digest username="1115633124", realm="192.168.1.120",
> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
> uri="sip:1000 at 192.168.1.121:5060",
> response="747b4d04544c84535dbbd987f2999ca7", qop=auth, nc=00000001
>    Content-Type: application/simple-message-summary
>   Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
>   ------------------------------------------------------------------------
>   SIP/2.0 401 Unauthorized
>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>   CSeq: 126070339 NOTIFY
>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
>   Server: Linksys/PAP2T-5.1.6(LS)
>   WWW-Authenticate: Digest realm="192.168.1.120", nonce="5339c7ba",
> qop="auth", algorithm=md5
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
>
> On Mon, Jan 25, 2010 at 4:46 AM, Brian West <brian at freeswitch.org> wrote:
> > or sofia profile xxx flush_inbound_reg callid reboot
> > callid you can get from sofia status profile xxx
> > /b
> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
> >
> > Actually I just found http://wiki.freeswitch.org/wiki/Mod_event_socket
> >
> > If I telnet to FS as described
> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do I
> > just need to enter somthing like:
> >
> > sendevent NOTIFY
> > profile: internal
> > event-string: resync
> > user: 1000
> > host: 192.168.1.121
> > content-type: application/simple-message-summary
> >
> > where 192.168.1.121 is the ip address of one of the Linksys devices?
> >
> > I don't see any messages sent when I do this.  What am I doing wrong?
> >
> > Thanks
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mcampbellsmith at gmail.com  Sun Jan 24 16:29:46 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 25 Jan 2010 11:29:46 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
Message-ID: <33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>

Hi Ognjen,

Thanks for the tip on the resync under the EXT tab.  It now works
using mod_event_socket and the following:

sendevent NOTIFY
profile: internal
event-string: resync
user: 1000
host: 192.168.1.121
content-type: application/simple-message-summary

However, if AUTH is required, why does FS send the wrong information to the SPA?

On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija <oseslija at gmail.com> wrote:
> You? should not authenticate those NOTIFYs (this will work only with SPA9000
> afaik). The option to change for this is in EXT tabs:
>
> Auth Resync-Reboot: No
>
> Also, FSs code will do a reboot of a phone, not resync (it sends reboot_now
> event). For that to work a patch is required.
>
> I've just tried to reboot my 942 (rev 16506) and it definitely works.
>
> Regards,
> Ognjen
>
>
> On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>>
>> Thanks Brian.. this still does not work. ?Maybe I need to open a Jira?
>> ?Notice the username in the authorization field. ?It should be 1000.
>>
>> Cheers
>> Mark
>>
>> freeswitch at internal> sofia status profile internal user 1000 at 192.168.1.120
>>
>> Registrations:
>>
>> =================================================================================================
>> Call-ID: ? ? ? ?bd783b73-66877627 at 192.168.1.121
>> User: ? ? ? ? ? 1000 at 192.168.1.120
>> Contact: ? ? ? ?1000 <sip:1000 at 192.168.1.121:5060>
>> Agent: ? ? ? ? ?Linksys/PAP2T-5.1.6(LS)
>> Status: ? ? ? ? Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
>> Host: ? ? ? ? ? freeswitch
>> IP: ? ? ? ? ? ? 192.168.1.121
>> Port: ? ? ? ? ? 5060
>> Auth-User: ? ? ?1000
>> Auth-Realm: ? ? 192.168.1.120
>> MWI-Account: ? ?1000 at 192.168.1.120
>>
>>
>> =================================================================================================
>>
>> freeswitch at internal> sofia profile internal flush_inbound_reg
>> bd783b73-66877627 at 192.168.1.121 reboot
>> +OK rebooting all registrations matching specified call_id
>>
>> freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060 at
>> 23:55:49.012627:
>> ? ------------------------------------------------------------------------
>> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> ? Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
>> ? Max-Forwards: 70
>> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> ? To: <sip:1000 at 192.168.1.120>
>> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> ? CSeq: 126070338 NOTIFY
>> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> ? Supported: timer, precondition, path, replaces
>> ? Event: reboot_now
>> ? Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer
>> ? Subscription-State: terminated;reason=timeout
>> ? Content-Type: application/simple-message-summary
>> ? Content-Length: 0
>>
>> ? ------------------------------------------------------------------------
>> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
>> ? ------------------------------------------------------------------------
>> ? SIP/2.0 401 Unauthorized
>> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> ? CSeq: 126070338 NOTIFY
>> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
>> ? Server: Linksys/PAP2T-5.1.6(LS)
>> ? WWW-Authenticate: Digest realm="192.168.1.120", nonce="8e54805b",
>> qop="auth", algorithm=md5
>> ? Content-Length: 0
>>
>> ? ------------------------------------------------------------------------
>> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
>> ? ------------------------------------------------------------------------
>> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> ? Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
>> ? Max-Forwards: 70
>> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> ? To: <sip:1000 at 192.168.1.120>
>> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> ? CSeq: 126070339 NOTIFY
>> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> ? Expires: 3590
>> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> ? Supported: timer, precondition, path, replaces
>> ? Event: reboot_now
>> ? Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer
>> ? Subscription-State: terminated;reason=timeout
>> ? Authorization: Digest username="1115633124", realm="192.168.1.120",
>> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
>> uri="sip:1000 at 192.168.1.121:5060",
>> response="747b4d04544c84535dbbd987f2999ca7", qop=auth, nc=00000001
>> ? Content-Type: application/simple-message-summary
>> ? Content-Length: 0
>>
>> ? ------------------------------------------------------------------------
>> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
>> ? ------------------------------------------------------------------------
>> ? SIP/2.0 401 Unauthorized
>> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> ? CSeq: 126070339 NOTIFY
>> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
>> ? Server: Linksys/PAP2T-5.1.6(LS)
>> ? WWW-Authenticate: Digest realm="192.168.1.120", nonce="5339c7ba",
>> qop="auth", algorithm=md5
>> ? Content-Length: 0
>>
>> ? ------------------------------------------------------------------------
>>
>> On Mon, Jan 25, 2010 at 4:46 AM, Brian West <brian at freeswitch.org> wrote:
>> > or sofia profile xxx flush_inbound_reg callid reboot
>> > callid you can get from sofia status profile xxx
>> > /b
>> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
>> >
>> > Actually I just found?http://wiki.freeswitch.org/wiki/Mod_event_socket
>> >
>> > If I telnet to FS as described
>> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do I
>> > just need to enter somthing like:
>> >
>> > sendevent NOTIFY
>> > profile: internal
>> > event-string: resync
>> > user: 1000
>> > host: 192.168.1.121
>> > content-type: application/simple-message-summary
>> >
>> > where 192.168.1.121 is the ip address of one of the Linksys devices?
>> >
>> > I don't see any messages sent when I do this. ?What am I doing wrong?
>> >
>> > Thanks
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Sun Jan 24 16:39:49 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 24 Jan 2010 18:39:49 -0600
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
Message-ID: <A5B60A4C-C827-4025-823B-1007E178DB2E@freeswitch.org>

Its not WRONG its just we don't know how to answer the challenge.

/b

On Jan 24, 2010, at 6:29 PM, Mark Campbell-Smith wrote:

> However, if AUTH is required, why does FS send the wrong information to the SPA?



From oseslija at gmail.com  Sun Jan 24 16:50:06 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 25 Jan 2010 01:50:06 +0100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
Message-ID: <4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>

The phone is asking FS to authenticate prior then accepting a NOTIFY from
it.
The authentication of notify's from spa endpoints work (afaik) only with
Linksys SPA9000 PBX , and FS doesn't have the code for that for obvious
reasons.
If you have SPA9000 maybe you can collect SIP traces.

Ognjen





On Mon, Jan 25, 2010 at 1:29 AM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Hi Ognjen,
>
> Thanks for the tip on the resync under the EXT tab.  It now works
> using mod_event_socket and the following:
>
> sendevent NOTIFY
> profile: internal
> event-string: resync
> user: 1000
> host: 192.168.1.121
> content-type: application/simple-message-summary
>
> However, if AUTH is required, why does FS send the wrong information to the
> SPA?
>
> On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija <oseslija at gmail.com>
> wrote:
> > You  should not authenticate those NOTIFYs (this will work only with
> SPA9000
> > afaik). The option to change for this is in EXT tabs:
> >
> > Auth Resync-Reboot: No
> >
> > Also, FSs code will do a reboot of a phone, not resync (it sends
> reboot_now
> > event). For that to work a patch is required.
> >
> > I've just tried to reboot my 942 (rev 16506) and it definitely works.
> >
> > Regards,
> > Ognjen
> >
> >
> > On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
> > <mcampbellsmith at gmail.com> wrote:
> >>
> >> Thanks Brian.. this still does not work.  Maybe I need to open a Jira?
> >>  Notice the username in the authorization field.  It should be 1000.
> >>
> >> Cheers
> >> Mark
> >>
> >> freeswitch at internal> sofia status profile internal user
> 1000 at 192.168.1.120
> >>
> >> Registrations:
> >>
> >>
> =================================================================================================
> >> Call-ID:        bd783b73-66877627 at 192.168.1.121
> >> User:           1000 at 192.168.1.120
> >> Contact:        1000 <sip:1000 at 192.168.1.121:5060>
> >> Agent:          Linksys/PAP2T-5.1.6(LS)
> >> Status:         Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
> >> Host:           freeswitch
> >> IP:             192.168.1.121
> >> Port:           5060
> >> Auth-User:      1000
> >> Auth-Realm:     192.168.1.120
> >> MWI-Account:    1000 at 192.168.1.120
> >>
> >>
> >>
> =================================================================================================
> >>
> >> freeswitch at internal> sofia profile internal flush_inbound_reg
> >> bd783b73-66877627 at 192.168.1.121 reboot
> >> +OK rebooting all registrations matching specified call_id
> >>
> >> freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060 at
> >> 23:55:49.012627:
> >>
> ------------------------------------------------------------------------
> >>   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
> >>   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
> >>   Max-Forwards: 70
> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >>   CSeq: 126070338 NOTIFY
> >>   Contact: <sip:mod_sofia at 192.168.1.120:5060>
> >>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
> >>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >>   Supported: timer, precondition, path, replaces
> >>   Event: reboot_now
> >>   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >> include-session-description, presence.winfo, message-summary, refer
> >>   Subscription-State: terminated;reason=timeout
> >>   Content-Type: application/simple-message-summary
> >>   Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
> >>
> ------------------------------------------------------------------------
> >>   SIP/2.0 401 Unauthorized
> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >>   CSeq: 126070338 NOTIFY
> >>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
> >>   Server: Linksys/PAP2T-5.1.6(LS)
> >>   WWW-Authenticate: Digest realm="192.168.1.120", nonce="8e54805b",
> >> qop="auth", algorithm=md5
> >>   Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
> >>
> ------------------------------------------------------------------------
> >>   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
> >>   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
> >>   Max-Forwards: 70
> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >>   CSeq: 126070339 NOTIFY
> >>   Contact: <sip:mod_sofia at 192.168.1.120:5060>
> >>   Expires: 3590
> >>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
> >>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >>   Supported: timer, precondition, path, replaces
> >>   Event: reboot_now
> >>   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >> include-session-description, presence.winfo, message-summary, refer
> >>   Subscription-State: terminated;reason=timeout
> >>   Authorization: Digest username="1115633124", realm="192.168.1.120",
> >> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
> >> uri="sip:1000 at 192.168.1.121:5060",
> >> response="747b4d04544c84535dbbd987f2999ca7", qop=auth, nc=00000001
> >>   Content-Type: application/simple-message-summary
> >>   Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
> >>
> ------------------------------------------------------------------------
> >>   SIP/2.0 401 Unauthorized
> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >>   CSeq: 126070339 NOTIFY
> >>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
> >>   Server: Linksys/PAP2T-5.1.6(LS)
> >>   WWW-Authenticate: Digest realm="192.168.1.120", nonce="5339c7ba",
> >> qop="auth", algorithm=md5
> >>   Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >>
> >> On Mon, Jan 25, 2010 at 4:46 AM, Brian West <brian at freeswitch.org>
> wrote:
> >> > or sofia profile xxx flush_inbound_reg callid reboot
> >> > callid you can get from sofia status profile xxx
> >> > /b
> >> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
> >> >
> >> > Actually I just found
> http://wiki.freeswitch.org/wiki/Mod_event_socket
> >> >
> >> > If I telnet to FS as described
> >> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do I
> >> > just need to enter somthing like:
> >> >
> >> > sendevent NOTIFY
> >> > profile: internal
> >> > event-string: resync
> >> > user: 1000
> >> > host: 192.168.1.121
> >> > content-type: application/simple-message-summary
> >> >
> >> > where 192.168.1.121 is the ip address of one of the Linksys devices?
> >> >
> >> > I don't see any messages sent when I do this.  What am I doing wrong?
> >> >
> >> > Thanks
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Sun Jan 24 20:04:41 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 23:04:41 -0500
Subject: [Freeswitch-users] E1 hangups
In-Reply-To: <4B59EC64.3080907@comgate.cz>
References: <4B59EC64.3080907@comgate.cz>
Message-ID: <A13D010D-25B0-4D05-BC14-5388F4AB4234@jerris.com>

We just fixed this issue friday and have been testing it more over the weekend.  The new drivers that should go with the latest openzap code have not been released yet but should be early next week.

Mike

On Jan 22, 2010, at 1:20 PM, Michal Zub?? wrote:

> Hi.
> 
> I'm trying to correct this behaviour, but can't figure out, where is the
> problem. Here's the scenario:
> 
> - we're trying to execute simple dialplan
>    * answer
>    * play sound
>    * wait for 3 seconds
>    * hangup
> 
> - for SIP caller it works as expected
> - problems are when, we try to call into it from E1 line
> - for E1 we're using sangoma winpipe & openzap
> - dialplan in freeswitch console is done in a moment ending with hangup
> - on the E1 line I hear nothing and after 2 seconds it disconnects
> - similar problem when there's only bridge to another number (E1)
> - it rings (on the destination phone) for a short moment (0.5-1s), but
> then hangs up spontaneously
> 
> Thanks for any clues.
> 
> mZubac
> <problem_background.zip>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Sun Jan 24 20:17:27 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 23:17:27 -0500
Subject: [Freeswitch-users] FreeSwitch Integration With Sailfin
In-Reply-To: <BAY143-W27CC94D97F584D192C5680A1600@phx.gbl>
References: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>, ,
	<CFA0D82A-33B3-4B50-B6AE-BAA61E0862A4@freeswitch.org>, ,
	<BAY143-W13760770FF3FCA224AACEA1610@phx.gbl>, ,
	<9dc4a1671001231501t65b1281epeeb7e7fa1b0bf5bf@mail.gmail.com>,
	<BAY143-W760D1216FA26D7EFD8E9DA1600@phx.gbl>,
	<84AB66EA-956C-4B42-87C8-42660CBEDCE3@jerris.com>
	<BAY143-W27CC94D97F584D192C5680A1600@phx.gbl>
Message-ID: <F8985501-2A06-4596-913C-081E899B396A@jerris.com>

There is more information on how to create a proper bridge string for sofia here:

http://wiki.freeswitch.org/wiki/Sofia#Syntax

You should put it in the dialplan in the context that you need to call it from.

Mie


On Jan 24, 2010, at 11:48 AM, juan camilo ospina quintero wrote:

> Thanks mike,
> 
> i already read about dialplan, and it seems that the bridge application
> is the one i need, but now i want to know, how to modify dialplan to use bridge
> and where to put this in the dialplan:
> 
> <action application="bridge" data="sofia/sip/9998881111 at sip.yourprovider.com"/>
> 
> this is an example, but i think thats what i need, if someone has work with bridging,
> please help with this.
> thanks
> 
> 
> From: mike at jerris.com
> Date: Sun, 24 Jan 2010 05:20:23 -0500
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FreeSwitch Integration With Sailfin
> 
> This is a good place to start reading on how to configure dialplan:
> 
> http://wiki.freeswitch.org/wiki/Dialplan
> http://wiki.freeswitch.org/wiki/Getting_Started_Guide#dialplan
> 

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From thangappan143 at gmail.com  Sun Jan 24 20:25:59 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Mon, 25 Jan 2010 09:55:59 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
	<7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>
	<7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>
Message-ID: <7aa29e791001242025y2adc91aes622bd22ea8d5ae1c@mail.gmail.com>

Here I mentioned the link which has the details of /etc/wanpipe/smg_pri.conf

http://www.pastebin.org/81895

On Sat, Jan 23, 2010 at 10:02 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Updated the latest version of freeswitch ( 1.0.5-20100121-0400) and run the
> wanrouter then freeswitch. While executing the freeswtich it said the
> following error.
>
> [ERR] zap_io.c:2562 Error loading/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
>
> [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
> file: No such file or directory]
> [ERR] zap_io.c:2722 can't find 'sangoma_boost
>
>
>
>
> Searched about this in freeswitch mailing list and found one post was there
> regarding the same problem. Finally found the problem. I missed to install
> the SCTP packages. Installed it and compiled the freeswitch again now the
> inbound call was landed on freeswitch.
>
> But I am unable to make a outbound call. When I was trying the following
> was get.
>
> freeswitch at internal> originate openzap/smg_prid/a/9940464753 at g1openzap/smg_prid/a/9940464753 at g1
> -ERR NORMAL_CIRCUIT_CONGESTION
>
> 2010-01-23 10:00:45.688854 [WARNING] ozmod_sangoma_boost.c:348 TX EVENT:
> CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[9940464753]
> Ci=[0000000000] Rdnis=[]
> freeswitch at internal> 2010-01-23 10:00:46.709355 [WARNING]
> ozmod_sangoma_boost.c:1373 RX EVENT (N): CALL_START_NACK:(82) [w256g256]
> Rc=0 CSid=2 Seq=2
> 2010-01-23 10:00:46.709355 [WARNING] sangoma_boost_client.c:220 TX EVENT
> (N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=2 Seq=3
> 2010-01-23 10:00:46.709355 [ERR] mod_openzap.c:1162 No channels available
> 2010-01-23 10:00:46.709355 [ERR] switch_ivr_originate.c:2411 Cannot create
> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
> 2010-01-23 10:00:46.709355 [DEBUG] switch_ivr_originate.c:3211 Originate
> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>
> Please help me...........
>
>
>
> On Fri, Jan 22, 2010 at 10:15 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>> The following link have the openzap.conf,openzap.conf.xml ,smg_prid.conf ,
>> debug log of mod_openzap , oz list and oz dump 1 output.
>>
>> http://pastebin.org/80095
>>
>>
>>
>> On Thu, Jan 21, 2010 at 10:34 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>
>>> OpenZap is loading the ss7 signalling type. As per your concern openzap
>>> does not know the details of the signalling then how it is loading the
>>> ss7_boost libraries?
>>>
>>> FreeSWITCH log:
>>> -----------------------------
>>> 2010-01-21 10:10:46.707844 [INFO] zap_io.c:2374 Configured 30 channel(s)
>>> 2010-01-21 10:10:46.708600 [INFO] zap_io.c:2468 Loading SIG from
>>> /usr/local/freeswitch/mod/ozmod_ss7_boost.so
>>> 2010-01-21 10:10:46.709031 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost'
>>> 2010-01-21 10:10:46.709466 [DEBUG] ss7_boost_client.c:124 Creating L=
>>> 127.0.0.65:53000 R=127.0.0.66:53000
>>> 2010-01-21 10:10:46.709834 [DEBUG] ss7_boost_client.c:124 Creating L=
>>> 127.0.0.65:53001 R=127.0.0.66:53001
>>> 2010-01-21 10:10:46.710424 [WARNING] ss7_boost_client.c:244 TX EVENT (P):
>>> SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>>>
>>> The signalling type might be anything but when I used the oz list command
>>> it showed the span details. But I am unable to make a inbound and outbound
>>> call.
>>>
>>> Outbound call result:
>>> ============
>>> >  originate openzap/smg_prid/a/9940464753 openzap/smg_prid/a/9843171457
>>> -ERR NORMAL_CIRCUIT_CONGESTION
>>>
>>> 2010-01-21 10:26:14.304816 [CRIT] ozmod_ss7_boost.c:244 SPAN is not
>>> online.
>>> freeswitch at internal> 2010-01-21 10:26:14.304816 [ERR] mod_openzap.c:1043
>>> No channels available
>>> 2010-01-21 10:26:14.304816 [ERR] switch_ivr_originate.c:1510 Cannot
>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>> 2010-01-21 10:26:14.304816 [DEBUG] switch_ivr_originate.c:2138 Originate
>>> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>>>
>>> Inbound call result:
>>> -----------------------------
>>>
>>>      I made incoming call for the dial plan which is specified in the
>>> earlier post at that time it said the number is busy. We did the packet
>>> capture using the following command.
>>>
>>> wanpipemon -i w1g1 -pcap -pcap file isdn.pcap -port ISDN -full -systime
>>> -c trd
>>>
>>> Here I attached the pcap file for that.
>>>
>>>
>>> Where I did mistake or Did I miss any thing to do?
>>> Please help me.......
>>>
>>>
>>>
>>> On Wed, Jan 20, 2010 at 7:40 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>>
>>>>
>>>>       I noticed the 'oz list' output  in that span type is 'ss7
>>>> (boost)'. How can I change this to isdn?
>>>>
>>>>
>>>>
>>>> On Wed, Jan 20, 2010 at 12:43 PM, Thangappan.M <thangappan143 at gmail.com
>>>> > wrote:
>>>>
>>>>> I found the error in it. The file name is used as openzap.conf.xml (
>>>>> smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
>>>>> PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
>>>>> openzap.conf.wiki.xml  file.
>>>>>
>>>>> Now the another problem is raised here.
>>>>> When I was using oz list command , the details of the smg_prid shown.
>>>>> When I was using 'oz dump smg_prid' command it shows all the channels'
>>>>> details. But all the channels' states are DOWN. why? How can I make it the
>>>>> states to UP?
>>>>>
>>>>> When I was making the call , the number is busy message was get. The
>>>>> call was not  at all landed to the freeswitch.
>>>>>
>>>>> Dial plan Example:
>>>>> -------------------------------
>>>>>  <extension name="PRI_Testing">
>>>>>         <condition field="destination_number" expression="^39114600$">
>>>>>                 <action application="playback"
>>>>> data="ivr-welcome_to_freeswitch"/>
>>>>>         </condition>
>>>>>   </extension>
>>>>>
>>>>> Please help me...........
>>>>>
>>>>> *Output Reference:*
>>>>> http://pastebin.org/79074
>>>>>
>>>>>
>>>>> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <
>>>>> thangappan143 at gmail.com> wrote:
>>>>>
>>>>>> Dear all,
>>>>>>
>>>>>>          I have successfully configured wanpipe with freeswitch. When
>>>>>> I was the running wancfg_fs script the following files openzap.conf ,
>>>>>> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
>>>>>> are created.
>>>>>>
>>>>>>          I started the wanrouter command then executed the freeswitch.
>>>>>>          When I was executing freeswitch mod_openzap.c said the error
>>>>>> as  "Error  for finding the span id. name:PRI_1".
>>>>>>          But in the openzap.conf and openzap.conf.xml files the span
>>>>>> name is smg_prid.
>>>>>>
>>>>>>          Why the freeswitch is referring the span name as PRI_1 ?
>>>>>>          Whether this has to configured in anywhere?
>>>>>>
>>>>>>          In the freeswitch CLI using oz command I tried to dump the
>>>>>> PRI_1 span id but it said te error  as "PRI_1 is not found".  When I was
>>>>>> trying the command 'oz dump smg_prid'  all the channel states and details
>>>>>> shown.
>>>>>>
>>>>>>         It seems that smg_prid span configured in openzap perfectly
>>>>>> (Its my assumption). Then Why freeswitch is referring the span name as
>>>>>> PRI_1.
>>>>>>
>>>>>> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>>>>>>
>>>>>> Could anyone please help me?
>>>>>>
>>>>>> REFERENCE:
>>>>>>
>>>>>> openzap.conf
>>>>>> [span wanpipe smg_prid]
>>>>>> name => smg_prid
>>>>>> trunk_type =>e1
>>>>>> b-channel => 1:1-15
>>>>>> b-channel => 1:17-31
>>>>>>
>>>>>>
>>>>>> openzap.conf.xml
>>>>>> <configuration name="openzap.conf" description="OpenZAP
>>>>>> Configuration">
>>>>>>   <settings>
>>>>>>     <param name="debug" value="0"/>
>>>>>>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>>>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>>>>     <!--<param name="enable-analog-option" value="3-way"/>-->
>>>>>>   </settings>
>>>>>>   <boost_spans>
>>>>>>         <span id="smg_prid">
>>>>>>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>>>         <param name="dialplan" value="XML"/>
>>>>>>         <param name="context" value="default"/>
>>>>>>          <!-- regex to stop dialing when it matches -->
>>>>>>         <!--<param name="dial-regex" value="5555"/>-->
>>>>>>         <!-- regex to stop dialing when it does not match -->
>>>>>>         <!--<param name="fail-dial-regex" value="^5"/>-->
>>>>>>         </span>
>>>>>>  </boost_spans>
>>>>>>  <analog_spans>
>>>>>>  </analog_spans>
>>>>>> </configuration>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Regards,
>>>>>> Thangappan.M
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Regards,
>>>>> Thangappan.M
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Thangappan.M
>>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Thangappan.M
>>>
>>
>>
>>
>> --
>> Regards,
>> Thangappan.M
>>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From thangappan143 at gmail.com  Sun Jan 24 21:50:02 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Mon, 25 Jan 2010 11:20:02 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001242025y2adc91aes622bd22ea8d5ae1c@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
	<7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>
	<7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>
	<7aa29e791001242025y2adc91aes622bd22ea8d5ae1c@mail.gmail.com>
Message-ID: <7aa29e791001242150o3a582e2end69f1f5e9f1aef1c@mail.gmail.com>

The following link have the openzap.conf,openzap.conf.xml
,smg_pri.conf, output of oz list
and oz dump.

http://www.pastebin.org/81929



On Mon, Jan 25, 2010 at 9:55 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Here I mentioned the link which has the details of
> /etc/wanpipe/smg_pri.conf
> http://www.pastebin.org/81895
>
>
> On Sat, Jan 23, 2010 at 10:02 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>> Updated the latest version of freeswitch ( 1.0.5-20100121-0400) and run
>> the wanrouter then freeswitch. While executing the freeswtich it said the
>> following error.
>>
>> [ERR] zap_io.c:2562 Error loading/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
>>
>> [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
>> file: No such file or directory]
>> [ERR] zap_io.c:2722 can't find 'sangoma_boost
>>
>>
>>
>>
>> Searched about this in freeswitch mailing list and found one post was
>> there regarding the same problem. Finally found the problem. I missed to
>> install the SCTP packages. Installed it and compiled the freeswitch again
>> now the inbound call was landed on freeswitch.
>>
>> But I am unable to make a outbound call. When I was trying the following
>> was get.
>>
>> freeswitch at internal> originate openzap/smg_prid/a/9940464753 at g1openzap/smg_prid/a/9940464753 at g1
>> -ERR NORMAL_CIRCUIT_CONGESTION
>>
>> 2010-01-23 10:00:45.688854 [WARNING] ozmod_sangoma_boost.c:348 TX EVENT:
>> CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[9940464753]
>> Ci=[0000000000] Rdnis=[]
>> freeswitch at internal> 2010-01-23 10:00:46.709355 [WARNING]
>> ozmod_sangoma_boost.c:1373 RX EVENT (N): CALL_START_NACK:(82) [w256g256]
>> Rc=0 CSid=2 Seq=2
>> 2010-01-23 10:00:46.709355 [WARNING] sangoma_boost_client.c:220 TX EVENT
>> (N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=2 Seq=3
>> 2010-01-23 10:00:46.709355 [ERR] mod_openzap.c:1162 No channels available
>> 2010-01-23 10:00:46.709355 [ERR] switch_ivr_originate.c:2411 Cannot create
>> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>> 2010-01-23 10:00:46.709355 [DEBUG] switch_ivr_originate.c:3211 Originate
>> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>>
>> Please help me...........
>>
>>
>>
>> On Fri, Jan 22, 2010 at 10:15 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>
>>> The following link have the openzap.conf,openzap.conf.xml ,smg_prid.conf
>>> , debug log of mod_openzap , oz list and oz dump 1 output.
>>>
>>> http://pastebin.org/80095
>>>
>>>
>>>
>>> On Thu, Jan 21, 2010 at 10:34 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>>
>>>> OpenZap is loading the ss7 signalling type. As per your concern openzap
>>>> does not know the details of the signalling then how it is loading the
>>>> ss7_boost libraries?
>>>>
>>>> FreeSWITCH log:
>>>> -----------------------------
>>>> 2010-01-21 10:10:46.707844 [INFO] zap_io.c:2374 Configured 30 channel(s)
>>>> 2010-01-21 10:10:46.708600 [INFO] zap_io.c:2468 Loading SIG from
>>>> /usr/local/freeswitch/mod/ozmod_ss7_boost.so
>>>> 2010-01-21 10:10:46.709031 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost'
>>>> 2010-01-21 10:10:46.709466 [DEBUG] ss7_boost_client.c:124 Creating L=
>>>> 127.0.0.65:53000 R=127.0.0.66:53000
>>>> 2010-01-21 10:10:46.709834 [DEBUG] ss7_boost_client.c:124 Creating L=
>>>> 127.0.0.65:53001 R=127.0.0.66:53001
>>>> 2010-01-21 10:10:46.710424 [WARNING] ss7_boost_client.c:244 TX EVENT
>>>> (P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>>>>
>>>> The signalling type might be anything but when I used the oz list
>>>> command it showed the span details. But I am unable to make a inbound and
>>>> outbound call.
>>>>
>>>> Outbound call result:
>>>> ============
>>>> >  originate openzap/smg_prid/a/9940464753 openzap/smg_prid/a/9843171457
>>>> -ERR NORMAL_CIRCUIT_CONGESTION
>>>>
>>>> 2010-01-21 10:26:14.304816 [CRIT] ozmod_ss7_boost.c:244 SPAN is not
>>>> online.
>>>> freeswitch at internal> 2010-01-21 10:26:14.304816 [ERR]
>>>> mod_openzap.c:1043 No channels available
>>>> 2010-01-21 10:26:14.304816 [ERR] switch_ivr_originate.c:1510 Cannot
>>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>>> 2010-01-21 10:26:14.304816 [DEBUG] switch_ivr_originate.c:2138 Originate
>>>> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>>>>
>>>> Inbound call result:
>>>> -----------------------------
>>>>
>>>>      I made incoming call for the dial plan which is specified in the
>>>> earlier post at that time it said the number is busy. We did the packet
>>>> capture using the following command.
>>>>
>>>> wanpipemon -i w1g1 -pcap -pcap file isdn.pcap -port ISDN -full -systime
>>>> -c trd
>>>>
>>>> Here I attached the pcap file for that.
>>>>
>>>>
>>>> Where I did mistake or Did I miss any thing to do?
>>>> Please help me.......
>>>>
>>>>
>>>>
>>>> On Wed, Jan 20, 2010 at 7:40 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>>>
>>>>>
>>>>>       I noticed the 'oz list' output  in that span type is 'ss7
>>>>> (boost)'. How can I change this to isdn?
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Jan 20, 2010 at 12:43 PM, Thangappan.M <
>>>>> thangappan143 at gmail.com> wrote:
>>>>>
>>>>>> I found the error in it. The file name is used as openzap.conf.xml (
>>>>>> smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
>>>>>> PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
>>>>>> openzap.conf.wiki.xml  file.
>>>>>>
>>>>>> Now the another problem is raised here.
>>>>>> When I was using oz list command , the details of the smg_prid shown.
>>>>>> When I was using 'oz dump smg_prid' command it shows all the channels'
>>>>>> details. But all the channels' states are DOWN. why? How can I make it the
>>>>>> states to UP?
>>>>>>
>>>>>> When I was making the call , the number is busy message was get. The
>>>>>> call was not  at all landed to the freeswitch.
>>>>>>
>>>>>> Dial plan Example:
>>>>>> -------------------------------
>>>>>>  <extension name="PRI_Testing">
>>>>>>         <condition field="destination_number" expression="^39114600$">
>>>>>>                 <action application="playback"
>>>>>> data="ivr-welcome_to_freeswitch"/>
>>>>>>         </condition>
>>>>>>   </extension>
>>>>>>
>>>>>> Please help me...........
>>>>>>
>>>>>> *Output Reference:*
>>>>>> http://pastebin.org/79074
>>>>>>
>>>>>>
>>>>>> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <
>>>>>> thangappan143 at gmail.com> wrote:
>>>>>>
>>>>>>> Dear all,
>>>>>>>
>>>>>>>          I have successfully configured wanpipe with freeswitch. When
>>>>>>> I was the running wancfg_fs script the following files openzap.conf ,
>>>>>>> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
>>>>>>> are created.
>>>>>>>
>>>>>>>          I started the wanrouter command then executed the
>>>>>>> freeswitch.
>>>>>>>          When I was executing freeswitch mod_openzap.c said the error
>>>>>>> as  "Error  for finding the span id. name:PRI_1".
>>>>>>>          But in the openzap.conf and openzap.conf.xml files the span
>>>>>>> name is smg_prid.
>>>>>>>
>>>>>>>          Why the freeswitch is referring the span name as PRI_1 ?
>>>>>>>          Whether this has to configured in anywhere?
>>>>>>>
>>>>>>>          In the freeswitch CLI using oz command I tried to dump the
>>>>>>> PRI_1 span id but it said te error  as "PRI_1 is not found".  When I was
>>>>>>> trying the command 'oz dump smg_prid'  all the channel states and details
>>>>>>> shown.
>>>>>>>
>>>>>>>         It seems that smg_prid span configured in openzap perfectly
>>>>>>> (Its my assumption). Then Why freeswitch is referring the span name as
>>>>>>> PRI_1.
>>>>>>>
>>>>>>> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>>>>>>>
>>>>>>> Could anyone please help me?
>>>>>>>
>>>>>>> REFERENCE:
>>>>>>>
>>>>>>> openzap.conf
>>>>>>> [span wanpipe smg_prid]
>>>>>>> name => smg_prid
>>>>>>> trunk_type =>e1
>>>>>>> b-channel => 1:1-15
>>>>>>> b-channel => 1:17-31
>>>>>>>
>>>>>>>
>>>>>>> openzap.conf.xml
>>>>>>> <configuration name="openzap.conf" description="OpenZAP
>>>>>>> Configuration">
>>>>>>>   <settings>
>>>>>>>     <param name="debug" value="0"/>
>>>>>>>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>>>>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>>>>>     <!--<param name="enable-analog-option" value="3-way"/>-->
>>>>>>>   </settings>
>>>>>>>   <boost_spans>
>>>>>>>         <span id="smg_prid">
>>>>>>>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>>>>         <param name="dialplan" value="XML"/>
>>>>>>>         <param name="context" value="default"/>
>>>>>>>          <!-- regex to stop dialing when it matches -->
>>>>>>>         <!--<param name="dial-regex" value="5555"/>-->
>>>>>>>         <!-- regex to stop dialing when it does not match -->
>>>>>>>         <!--<param name="fail-dial-regex" value="^5"/>-->
>>>>>>>         </span>
>>>>>>>  </boost_spans>
>>>>>>>  <analog_spans>
>>>>>>>  </analog_spans>
>>>>>>> </configuration>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Regards,
>>>>>>> Thangappan.M
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Regards,
>>>>>> Thangappan.M
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Regards,
>>>>> Thangappan.M
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Thangappan.M
>>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Thangappan.M
>>>
>>
>>
>>
>> --
>> Regards,
>> Thangappan.M
>>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From elihayun at gmail.com  Sun Jan 24 22:17:33 2010
From: elihayun at gmail.com (Eli Hayun)
Date: Mon, 25 Jan 2010 08:17:33 +0200
Subject: [Freeswitch-users] Module multicast fail
In-Reply-To: <2A8BE7A9-EC50-47CD-9FE8-172BF0F97DB6@freeswitch.org>
References: <4B56F0F9.9090808@savion.huji.ac.il>
	<2A8BE7A9-EC50-47CD-9FE8-172BF0F97DB6@freeswitch.org>
Message-ID: <4B5D377D.8050408@savion.huji.ac.il>

On 01/20/2010 05:18 PM, Brian West wrote:
> Please visit http://latest.freeswitch.org  and update to the latest ;)  Its the best you can get to date!  All the preX releases are gone from the download site.
>
> /b
>
> On Jan 20, 2010, at 6:03 AM, Eli Hayun wrote:
>
>   
>> (ver 1.0.5pre9)
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
Hi
I run the latest version and still getting an error

2010-01-25 07:42:53.449509 [ERR] mod_event_multicast.c:410 Multicast Error^M
2010-01-25 07:42:53.449761 [CRIT] switch_loadable_module.c:872 Error
Loading module /freeswitch-1.0.5_20100120-0400/mod/mod_event_multicast.so^M
**Module load routine returned an error**^M
2010-01-25 07:42:53.450424 [CONSOLE] switch_loadable_module.c:890
Successfully Loaded [mod_event_socket]^M

Eli




From brian at freeswitch.org  Sun Jan 24 22:34:55 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 25 Jan 2010 00:34:55 -0600
Subject: [Freeswitch-users] Module multicast fail
In-Reply-To: <4B5D377D.8050408@savion.huji.ac.il>
References: <4B56F0F9.9090808@savion.huji.ac.il>
	<2A8BE7A9-EC50-47CD-9FE8-172BF0F97DB6@freeswitch.org>
	<4B5D377D.8050408@savion.huji.ac.il>
Message-ID: <6F3B0496-C94E-4515-ADE3-89D12EAD6379@freeswitch.org>

+OK log level  [7]
freeswitch at internal> load mod_event_multicast
+OK

2010-01-25 00:34:37.581408 [CONSOLE] switch_loadable_module.c:890 Successfully Loaded [mod_event_multicast]
freeswitch at internal> 2010-01-25 00:34:37.581408 [NOTICE] switch_loadable_module.c:271 Adding API Function 'multicast_peers'

What distro are you on?

/b

On Jan 25, 2010, at 12:17 AM, Eli Hayun wrote:

> mod_event_multicast



From a.afzali2003 at gmail.com  Sun Jan 24 22:36:21 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 25 Jan 2010 10:06:21 +0330
Subject: [Freeswitch-users] How to get chat message via event
In-Reply-To: <7b197bef1001241305o3ee9bc9cq7decdc412f90575c@mail.gmail.com>
References: <debff5a81001240739q3dbe4680uf15fa1ce4ee79bf7@mail.gmail.com>
	<7b197bef1001240748r248f53b8i5794f7cbc1fedfb1@mail.gmail.com>
	<debff5a81001241259q3bc2c82dl40c73ac81e43b40b@mail.gmail.com>
	<7b197bef1001241305o3ee9bc9cq7decdc412f90575c@mail.gmail.com>
Message-ID: <debff5a81001242236u4fb153f3j6cfee6550429573f@mail.gmail.com>

I have two X-Lite logged in FreeSWITCH that could call , monitor presence
data and also send text messages to each others. So as I'm able to have
every step of those operations via events mechanism, I expect to see MESSAGE
event in case of sending text messages between them.

appreciate your help
-- afshin

On Mon, Jan 25, 2010 at 12:35 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org>wrote:

> Which events you don't get? From which channel in which circumstances?
> (I mean what you do and what do you expect?)
>
> -giovanni
>
> On Sun, Jan 24, 2010 at 9:59 PM, afshin afzali <a.afzali2003 at gmail.com>
> wrote:
> > Hi,
> >
> > As you say, I've already done and unfortunately did not get the message
> > events although other events are fired as expected :(
> >
> > -- afshin
> >
> > On Sun, Jan 24, 2010 at 7:18 PM, Giovanni Maruzzelli <
> gmaruzz at celliax.org>
> > wrote:
> >>
> >> you subscribe to them as MESSAGE events
> >>
> >> eg, from a telnet session:
> >>
> >> telnet localhost 8021
> >> auth ClueCon
> >> events plain message
> >>
> >> then those events will show up in your telnet session.
> >> -gm
> >>
> >> On Sun, Jan 24, 2010 at 4:39 PM, afshin afzali <a.afzali2003 at gmail.com>
> >> wrote:
> >> > Hi,
> >> >
> >> > It seems that the chat messages don't fire via events by default and
> >> > just
> >> > exchange between parties.
> >> > Is it true? Is it possible to enable those via events?
> >> >
> >> > appreciate all,
> >> > -- afshin
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From abhishek.dixit at nagarro.com  Sun Jan 24 20:47:29 2010
From: abhishek.dixit at nagarro.com (Abhishek Dixit)
Date: Mon, 25 Jan 2010 10:17:29 +0530
Subject: [Freeswitch-users] Configuring duration of conference
Message-ID: <000001ca9d79$81aa0f50$84fe2df0$@dixit@nagarro.com>

Hi,

 

I am working on FreeSWITCH Version 1.0.4 (14460).

I am trying to set up a conference room which will allow conferencing for
limited duration. This means that when parties connect to this conference
room they can remain connected only till a specified duration. After this
duration the conference should end with announcement.

I have read wiki for mod_conference and for dialplan and studied IVR
conferencing example using javascript as well.

But I am unable to find a way to specify duration for the conference using
dialplan and javascript application.

 

I am planning now to use mod_event_socket interface to end conference after
a specified duration.

 

Please tell me if there can be a way to configure duration of conference
using dialplan configurations or javascript application.

 

Thanks

 

Abhishek Dixit

 

 

 

 

 

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From jingwei.yang at gmail.com  Sun Jan 24 23:54:28 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 25 Jan 2010 15:54:28 +0800
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <ebbc2d551001241547j7e11d4e4se093ce5a60e008b5@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
	<1264092620.14614.73.camel@luna.tc.commsmundi.com>
	<13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
	<ebbc2d551001211721n51cb5c54m996e0d494842c383@mail.gmail.com>
	<13529f9d1001211806o35fc3435j93b71bbb2ede3028@mail.gmail.com>
	<13529f9d1001222200p44afd96ema21729d0038e89ba@mail.gmail.com>
	<ebbc2d551001241547j7e11d4e4se093ce5a60e008b5@mail.gmail.com>
Message-ID: <13529f9d1001242354q18e8b05fla3cc37de466f0767@mail.gmail.com>

Thanks Jo?o. In 01_fifo.xml, the agent uses the action below to sign up.

<action application="set" data="result=${fifo_member(add ACD$1
{fifo_member_wait=nowait}loopback/1${user_name}/default/XML )"/>

The variable ${user_name} was empty when I tried both skypiax and dingaling.


2010/1/25 Jo?o Mesquita <jmesquita at freeswitch.org>

> What user_name? I don't understand that statement.
>
> I think that you can always use
> http://wiki.freeswitch.org/wiki/Channel_Variables#api_hangup_hook
>
> Regards,
> Jo?o Mesquita
> FSComm Developer
>
>
> On Sat, Jan 23, 2010 at 4:00 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>
>> Hi Jo?o, do you know how to sign the agent off automatically when either
>> party hangs up the call?
>>
>> Here's how I originate the call to the agent and sign him up in ACD1:
>>
>> originate skypiax/ANY/jingwei.yang 6*1
>>
>> However, I found the user_name property is empty. May I know how it is
>> set?
>>
>>
>> Thanks and best regards,
>> -Jingwei
>>
>> On Fri, Jan 22, 2010 at 10:06 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>>
>>> Hi Jo?o, thanks for the reply. I'll try it out.
>>>
>>> Regards,
>>> -Jingwei
>>>
>>> 2010/1/22 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>
>>> Jingwei, check my contrib dir. I think it may help you with one FIFO
>>>> since we are able there to sign in and sign off dynamic agents as well as
>>>> customize what we do when the FIFO makes a call to them.
>>>>
>>>> Regards,
>>>> Jo?o Mesquita
>>>> FSComm Developer
>>>>
>>>>
>>>> On Thu, Jan 21, 2010 at 10:05 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>>>>
>>>>> Thanks for the reply. All the agents are dynamic and I can't predefine
>>>>> them in the config file.
>>>>>
>>>>> Regards,
>>>>> -Jingwei
>>>>>
>>>>>
>>>>> On Fri, Jan 22, 2010 at 12:50 AM, Fran?ois Delawarde <
>>>>> fdelawarde at wirelessmundi.com> wrote:
>>>>>
>>>>>> Why do you need 2 fifos? You could have callback agents connected to
>>>>>> the
>>>>>> fifo and send incoming calls there, mod_fifo will do the rest.
>>>>>>
>>>>>> To configure agents for callback:
>>>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>>>>>>
>>>>>> To place a call into a fifo:
>>>>>>
>>>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue
>>>>>>
>>>>>> Fran?ois.
>>>>>>
>>>>>> On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
>>>>>> > Sorry about the confusion, I'm just trying to think over all the
>>>>>> > abnormal situations before the implementation and hope the flow has
>>>>>> no
>>>>>> > design flaws.
>>>>>> >
>>>>>> > Client A is parked in Queue 1, multiple consumers will be ringed to
>>>>>> > answer him. And once the first one is connected, all the rest will
>>>>>> > hang up. This is the targeted function.
>>>>>> >
>>>>>> > To achieve this, I'm considering to originate a call to each
>>>>>> consumer
>>>>>> > and put the calls in Queue 2. Then intercept client A to the first
>>>>>> > element of Queue 2.
>>>>>> >
>>>>>> > I'm not sure if it's the usual or the best way. If you feel not,
>>>>>> > please don't hesitate to correct me. Any thoughts are warmly
>>>>>> > appreciated.
>>>>>> >
>>>>>> > On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com>
>>>>>> wrote:
>>>>>> >         what no errors mean? how do you originate calls to
>>>>>> consumers?
>>>>>> >         I don't
>>>>>> >         understand your scenario.
>>>>>> >
>>>>>> >         2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
>>>>>> >
>>>>>> >         > Hi All,
>>>>>> >         >
>>>>>> >         > Please advise whether the following flow makes sense.
>>>>>> >         >
>>>>>> >         > 1. Client A calls in and parked in Queue 1
>>>>>> >         > 2. Originate calls to several consumers simultaneously and
>>>>>> >         park them in
>>>>>> >         > Queue 2
>>>>>> >         > 3. Intercept A's call to the first consumer of Queue 2
>>>>>> >         >
>>>>>> >         > Basically I want A's call picked up by the first among
>>>>>> many
>>>>>> >         consumers with
>>>>>> >         > no errors. Please let me know whether I'm on the right
>>>>>> >         track.
>>>>>> >         >
>>>>>> >         > Thanks and best regards,
>>>>>> >         > -Jingwei
>>>>>> >         >
>>>>>> >
>>>>>> >
>>>>>> >         _______________________________________________
>>>>>> >         FreeSWITCH-users mailing list
>>>>>> >         FreeSWITCH-users at lists.freeswitch.org
>>>>>> >
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >         UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >         http://www.freeswitch.org
>>>>>> >
>>>>>> > _______________________________________________
>>>>>> > FreeSWITCH-users mailing list
>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> > http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lei.tlfly at gmail.com  Mon Jan 25 00:16:10 2010
From: lei.tlfly at gmail.com (Lei Tang)
Date: Mon, 25 Jan 2010 16:16:10 +0800
Subject: [Freeswitch-users] Question about bridge_answer_timeout variable
Message-ID: <50c41b4e1001250016v6633eaf1r7e4c10a656d649f6@mail.gmail.com>

Hi all, I'm using freeswitch-1.0.5pre9
When I set bridge_answer_timeout variable, the call is hangup by fs even the
callee has answered the call. I  try to read the source code, found the
cause is in
file switch_ivr_bridge.c  audio_bridge_thread function, It seems  in the
thread loop, ans_a flag is not updated, So when   bridge_answer_timeout is
set, FS will think the channel is still not answered and hangup the call
when timeout.  I tried add "ans_a = switch_channel_test_flag(chan_a,
CF_ANSWERED);  "  in  "for(;;)"  loop, it seem ok now.
Does someone known something about this problem?  Or it's a known bug of
freeswitch?

BTW, my scenario is as follow:

1. A call B in FS
2.FS set bridge_answer_timeout to 30 and bridge the call to B
3.B answer the call
4.after 30 secs, FS hangup call, the cause is "ALLOTTED_TIMEOUT"


-- 
Lei.Tang
lei.tlfly at gmail.com
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From dujinfang at gmail.com  Mon Jan 25 00:28:13 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 25 Jan 2010 16:28:13 +0800
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <13529f9d1001242354q18e8b05fla3cc37de466f0767@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
	<1264092620.14614.73.camel@luna.tc.commsmundi.com>
	<13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
	<ebbc2d551001211721n51cb5c54m996e0d494842c383@mail.gmail.com>
	<13529f9d1001211806o35fc3435j93b71bbb2ede3028@mail.gmail.com>
	<13529f9d1001222200p44afd96ema21729d0038e89ba@mail.gmail.com>
	<ebbc2d551001241547j7e11d4e4se093ce5a60e008b5@mail.gmail.com>
	<13529f9d1001242354q18e8b05fla3cc37de466f0767@mail.gmail.com>
Message-ID: <23f91031001250028t46e5ee06w264aeb2a6c69fdd4@mail.gmail.com>

try to use <action application="info" ...  to see all the available params

2010/1/25 Jingwei Yang <jingwei.yang at gmail.com>:
> Thanks Jo?o. In 01_fifo.xml, the agent uses the action below to sign up.
>
> <action application="set" data="result=${fifo_member(add ACD$1
> {fifo_member_wait=nowait}loopback/1${user_name}/default/XML )"/>
>
> The variable ${user_name} was empty when I tried both skypiax and dingaling.
>
> 2010/1/25 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> What user_name? I don't understand that statement.
>> I think that you can always use
>> http://wiki.freeswitch.org/wiki/Channel_Variables#api_hangup_hook
>> Regards,
>> Jo?o Mesquita
>> FSComm Developer
>>
>>
>> On Sat, Jan 23, 2010 at 4:00 AM, Jingwei Yang <jingwei.yang at gmail.com>
>> wrote:
>>>
>>> Hi Jo?o, do you know how to sign the agent off automatically when either
>>> party hangs up the call?
>>>
>>> Here's how I originate the call to the agent and sign him up in ACD1:
>>>
>>> originate skypiax/ANY/jingwei.yang 6*1
>>>
>>> However, I found the user_name property is empty. May I know how it is
>>> set?
>>>
>>> Thanks and best regards,
>>> -Jingwei
>>>
>>> On Fri, Jan 22, 2010 at 10:06 AM, Jingwei Yang <jingwei.yang at gmail.com>
>>> wrote:
>>>>
>>>> Hi Jo?o, thanks for the reply. I'll try it out.
>>>>
>>>> Regards,
>>>> -Jingwei
>>>>
>>>> 2010/1/22 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>>>
>>>>> Jingwei, check my contrib dir. I think it may help you with one FIFO
>>>>> since we are able there to sign in and sign off dynamic agents as well as
>>>>> customize what we do when the FIFO makes a call to them.
>>>>>
>>>>> Regards,
>>>>> Jo?o Mesquita
>>>>> FSComm Developer
>>>>>
>>>>> On Thu, Jan 21, 2010 at 10:05 PM, Jingwei Yang <jingwei.yang at gmail.com>
>>>>> wrote:
>>>>>>
>>>>>> Thanks for the reply. All the agents are dynamic and I can't predefine
>>>>>> them in the config file.
>>>>>>
>>>>>> Regards,
>>>>>> -Jingwei
>>>>>>
>>>>>> On Fri, Jan 22, 2010 at 12:50 AM, Fran?ois Delawarde
>>>>>> <fdelawarde at wirelessmundi.com> wrote:
>>>>>>>
>>>>>>> Why do you need 2 fifos? You could have callback agents connected to
>>>>>>> the
>>>>>>> fifo and send incoming calls there, mod_fifo will do the rest.
>>>>>>>
>>>>>>> To configure agents for callback:
>>>>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>>>>>>>
>>>>>>> To place a call into a fifo:
>>>>>>>
>>>>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue
>>>>>>>
>>>>>>> Fran?ois.
>>>>>>>
>>>>>>> On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
>>>>>>> > Sorry about the confusion, I'm just trying to think over all the
>>>>>>> > abnormal situations before the implementation and hope the flow has
>>>>>>> > no
>>>>>>> > design flaws.
>>>>>>> >
>>>>>>> > Client A is parked in Queue 1, multiple consumers will be ringed to
>>>>>>> > answer him. And once the first one is connected, all the rest will
>>>>>>> > hang up. This is the targeted function.
>>>>>>> >
>>>>>>> > To achieve this, I'm considering to originate a call to each
>>>>>>> > consumer
>>>>>>> > and put the calls in Queue 2. Then intercept client A to the first
>>>>>>> > element of Queue 2.
>>>>>>> >
>>>>>>> > I'm not sure if it's the usual or the best way. If you feel not,
>>>>>>> > please don't hesitate to correct me. Any thoughts are warmly
>>>>>>> > appreciated.
>>>>>>> >
>>>>>>> > On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com>
>>>>>>> > wrote:
>>>>>>> > ? ? ? ? what no errors mean? how do you originate calls to
>>>>>>> > consumers?
>>>>>>> > ? ? ? ? I don't
>>>>>>> > ? ? ? ? understand your scenario.
>>>>>>> >
>>>>>>> > ? ? ? ? 2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
>>>>>>> >
>>>>>>> > ? ? ? ? > Hi All,
>>>>>>> > ? ? ? ? >
>>>>>>> > ? ? ? ? > Please advise whether the following flow makes sense.
>>>>>>> > ? ? ? ? >
>>>>>>> > ? ? ? ? > 1. Client A calls in and parked in Queue 1
>>>>>>> > ? ? ? ? > 2. Originate calls to several consumers simultaneously
>>>>>>> > and
>>>>>>> > ? ? ? ? park them in
>>>>>>> > ? ? ? ? > Queue 2
>>>>>>> > ? ? ? ? > 3. Intercept A's call to the first consumer of Queue 2
>>>>>>> > ? ? ? ? >
>>>>>>> > ? ? ? ? > Basically I want A's call picked up by the first among
>>>>>>> > many
>>>>>>> > ? ? ? ? consumers with
>>>>>>> > ? ? ? ? > no errors. Please let me know whether I'm on the right
>>>>>>> > ? ? ? ? track.
>>>>>>> > ? ? ? ? >
>>>>>>> > ? ? ? ? > Thanks and best regards,
>>>>>>> > ? ? ? ? > -Jingwei
>>>>>>> > ? ? ? ? >
>>>>>>> >
>>>>>>> >
>>>>>>> > ? ? ? ? _______________________________________________
>>>>>>> > ? ? ? ? FreeSWITCH-users mailing list
>>>>>>> > ? ? ? ? FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >
>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >
>>>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> > ? ? ? ? http://www.freeswitch.org
>>>>>>> >
>>>>>>> > _______________________________________________
>>>>>>> > FreeSWITCH-users mailing list
>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >
>>>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> > http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mike at jerris.com  Mon Jan 25 00:59:43 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 25 Jan 2010 03:59:43 -0500
Subject: [Freeswitch-users] Module multicast fail
In-Reply-To: <4B5D377D.8050408@savion.huji.ac.il>
References: <4B56F0F9.9090808@savion.huji.ac.il>
	<2A8BE7A9-EC50-47CD-9FE8-172BF0F97DB6@freeswitch.org>
	<4B5D377D.8050408@savion.huji.ac.il>
Message-ID: <85B99B6D-E4E2-457D-B643-4168ADA364A3@jerris.com>

This means it was unable to joint he multicast group you specified in the "address" in your conf file for the module.

Mie

On Jan 25, 2010, at 1:17 AM, Eli Hayun wrote:
> Hi
> I run the latest version and still getting an error
> 
> 2010-01-25 07:42:53.449509 [ERR] mod_event_multicast.c:410 Multicast Error^M
> 2010-01-25 07:42:53.449761 [CRIT] switch_loadable_module.c:872 Error
> Loading module /freeswitch-1.0.5_20100120-0400/mod/mod_event_multicast.so^M
> **Module load routine returned an error**^M
> 2010-01-25 07:42:53.450424 [CONSOLE] switch_loadable_module.c:890
> Successfully Loaded [mod_event_socket]^M



From mike at jerris.com  Mon Jan 25 01:03:28 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 25 Jan 2010 04:03:28 -0500
Subject: [Freeswitch-users] mod_radius_cdr module load error
In-Reply-To: <BAY141-W1AF84CAAD730644C7A09790610@phx.gbl>
References: <BAY141-W1AF84CAAD730644C7A09790610@phx.gbl>
Message-ID: <24760B4B-040C-46B7-8AD3-A12F6C0AE140@jerris.com>

the issue is 

"Open of mod_radius_cdr.conf failed"

You need a configuration file for that module that is not there.

Mie

On Jan 23, 2010, at 2:08 AM, satish patel wrote:

> Hi All,
> 
> I am following this wiki http://wiki.freeswitch.org/wiki/Mod_radius_cdr to hook up freeradius with freeswitch but i am getting following error in log
> 
> 2010-01-23 01:56:25.717201 [ERR] mod_radius_cdr.c:662 Open of mod_radius_cdr.conf failed
> 2010-01-23 01:56:25.717225 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_radius_cdr.so
> **Module load routine returned an error**

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From mike at jerris.com  Mon Jan 25 01:04:56 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 25 Jan 2010 04:04:56 -0500
Subject: [Freeswitch-users] Freeswitch process hangs, losses connection.
In-Reply-To: <1264355459996-4449929.post@n2.nabble.com>
References: <1264355459996-4449929.post@n2.nabble.com>
Message-ID: <FFC27F5A-E1AF-427C-954D-C37EF6D8862B@jerris.com>

Have you looked at all a the freeswitch logs?  Try tuning them up to debug, maybe adding some sort of monitoring to see when it is down to see what happened in the logs about that time.
Mike

On Jan 24, 2010, at 12:50 PM, AFalcon wrote:

> 
> Hi I am new here but have successfully configured Freeswitch to run on Snow
> Leopard. I have configured Snow Leopard so that the computer will never go
> to sleep. My issue though is that Freeswitch losses it connection at some
> point while running. When I call in I get a busy signal or a message saying
> the phone number is not in operation. 
> 
> To get around this I wrote a launchd process that restarts Freeswitch and my
> softphone every 4 hours. 
> 
> I don't like this and am wondering how to go about trouble shooting the
> issue of Freeswitch losing connectivity after a period of a few hours. 



From mike at jerris.com  Mon Jan 25 01:05:54 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 25 Jan 2010 04:05:54 -0500
Subject: [Freeswitch-users] Configuring duration of conference
In-Reply-To: <000001ca9d79$81aa0f50$84fe2df0$@dixit@nagarro.com>
References: <000001ca9d79$81aa0f50$84fe2df0$@dixit@nagarro.com>
Message-ID: <438CA6A7-1F44-499C-98D0-BE739B3A71AF@jerris.com>

sched_hangup
sched_broadcast


On Jan 24, 2010, at 11:47 PM, Abhishek Dixit wrote:

> Hi,
>  
> I am working on FreeSWITCH Version 1.0.4 (14460).
> I am trying to set up a conference room which will allow conferencing for limited duration. This means that when parties connect to this conference room they can remain connected only till a specified duration. After this duration the conference should end with announcement.
> I have read wiki for mod_conference and for dialplan and studied IVR conferencing example using javascript as well.
> But I am unable to find a way to specify duration for the conference using dialplan and javascript application.
>  
> I am planning now to use mod_event_socket interface to end conference after a specified duration.
>  
> Please tell me if there can be a way to configure duration of conference using dialplan configurations or javascript application.

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From mike at jerris.com  Mon Jan 25 01:07:04 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 25 Jan 2010 04:07:04 -0500
Subject: [Freeswitch-users] Question about bridge_answer_timeout variable
In-Reply-To: <50c41b4e1001250016v6633eaf1r7e4c10a656d649f6@mail.gmail.com>
References: <50c41b4e1001250016v6633eaf1r7e4c10a656d649f6@mail.gmail.com>
Message-ID: <504AA55B-102A-4755-BFB8-205B40DB3172@jerris.com>

Check if this is an issue in latest svn trunk.  I suspect it is already fixed.

Mike

On Jan 25, 2010, at 3:16 AM, Lei Tang wrote:

> Hi all, I'm using freeswitch-1.0.5pre9
> When I set bridge_answer_timeout variable, the call is hangup by fs even the callee has answered the call. I  try to read the source code, found the cause is in 
> file switch_ivr_bridge.c  audio_bridge_thread function, It seems  in the thread loop, ans_a flag is not updated, So when   bridge_answer_timeout is set, FS will think the channel is still not answered and hangup the call when timeout.  I tried add "ans_a = switch_channel_test_flag(chan_a, CF_ANSWERED);  "  in  "for(;;)"  loop, it seem ok now. 
> Does someone known something about this problem?  Or it's a known bug of freeswitch?
> 
> BTW, my scenario is as follow:
> 
> 1. A call B in FS
> 2.FS set bridge_answer_timeout to 30 and bridge the call to B
> 3.B answer the call
> 4.after 30 secs, FS hangup call, the cause is "ALLOTTED_TIMEOUT"



From irmatov at gmail.com  Mon Jan 25 01:13:21 2010
From: irmatov at gmail.com (Timur Irmatov)
Date: Mon, 25 Jan 2010 14:13:21 +0500
Subject: [Freeswitch-users] mod_erlang_event: disconnects
In-Reply-To: <20100122154658.GC25693@hijacked.us>
References: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>
	<20100121134241.GD1036@hijacked.us>
	<241d382f1001220522g27f67ef5p5ba86ebb3afe63c@mail.gmail.com>
	<20100122154658.GC25693@hijacked.us>
Message-ID: <241d382f1001250113q2885a27dxebdf90ee836f337@mail.gmail.com>

Hi, Andrew!

On Fri, Jan 22, 2010 at 8:46 PM, Andrew Thompson <andrew at hijacked.us> wrote:
> Give this patch a shot:
>
> http://eagle.bsd.st/~andrew/erlang_session_fix.diff
>
> And see if it makes a difference.

I have just installed this patch. Thank you. I will let you know the results.


-- 
Timur Irmatov, xmpp:irmatov at jabber.ru


From jingwei.yang at gmail.com  Mon Jan 25 01:13:21 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 25 Jan 2010 17:13:21 +0800
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <23f91031001250028t46e5ee06w264aeb2a6c69fdd4@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
	<1264092620.14614.73.camel@luna.tc.commsmundi.com>
	<13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
	<ebbc2d551001211721n51cb5c54m996e0d494842c383@mail.gmail.com>
	<13529f9d1001211806o35fc3435j93b71bbb2ede3028@mail.gmail.com>
	<13529f9d1001222200p44afd96ema21729d0038e89ba@mail.gmail.com>
	<ebbc2d551001241547j7e11d4e4se093ce5a60e008b5@mail.gmail.com>
	<13529f9d1001242354q18e8b05fla3cc37de466f0767@mail.gmail.com>
	<23f91031001250028t46e5ee06w264aeb2a6c69fdd4@mail.gmail.com>
Message-ID: <13529f9d1001250113p63dacecbx946f0457875eaaa3@mail.gmail.com>

I see.. thanks!

On Mon, Jan 25, 2010 at 4:28 PM, Seven Du <dujinfang at gmail.com> wrote:

> try to use <action application="info" ...  to see all the available params
>
> 2010/1/25 Jingwei Yang <jingwei.yang at gmail.com>:
> > Thanks Jo?o. In 01_fifo.xml, the agent uses the action below to sign up.
> >
> > <action application="set" data="result=${fifo_member(add ACD$1
> > {fifo_member_wait=nowait}loopback/1${user_name}/default/XML )"/>
> >
> > The variable ${user_name} was empty when I tried both skypiax and
> dingaling.
> >
> > 2010/1/25 Jo?o Mesquita <jmesquita at freeswitch.org>
> >>
> >> What user_name? I don't understand that statement.
> >> I think that you can always use
> >> http://wiki.freeswitch.org/wiki/Channel_Variables#api_hangup_hook
> >> Regards,
> >> Jo?o Mesquita
> >> FSComm Developer
> >>
> >>
> >> On Sat, Jan 23, 2010 at 4:00 AM, Jingwei Yang <jingwei.yang at gmail.com>
> >> wrote:
> >>>
> >>> Hi Jo?o, do you know how to sign the agent off automatically when
> either
> >>> party hangs up the call?
> >>>
> >>> Here's how I originate the call to the agent and sign him up in ACD1:
> >>>
> >>> originate skypiax/ANY/jingwei.yang 6*1
> >>>
> >>> However, I found the user_name property is empty. May I know how it is
> >>> set?
> >>>
> >>> Thanks and best regards,
> >>> -Jingwei
> >>>
> >>> On Fri, Jan 22, 2010 at 10:06 AM, Jingwei Yang <jingwei.yang at gmail.com
> >
> >>> wrote:
> >>>>
> >>>> Hi Jo?o, thanks for the reply. I'll try it out.
> >>>>
> >>>> Regards,
> >>>> -Jingwei
> >>>>
> >>>> 2010/1/22 Jo?o Mesquita <jmesquita at freeswitch.org>
> >>>>>
> >>>>> Jingwei, check my contrib dir. I think it may help you with one FIFO
> >>>>> since we are able there to sign in and sign off dynamic agents as
> well as
> >>>>> customize what we do when the FIFO makes a call to them.
> >>>>>
> >>>>> Regards,
> >>>>> Jo?o Mesquita
> >>>>> FSComm Developer
> >>>>>
> >>>>> On Thu, Jan 21, 2010 at 10:05 PM, Jingwei Yang <
> jingwei.yang at gmail.com>
> >>>>> wrote:
> >>>>>>
> >>>>>> Thanks for the reply. All the agents are dynamic and I can't
> predefine
> >>>>>> them in the config file.
> >>>>>>
> >>>>>> Regards,
> >>>>>> -Jingwei
> >>>>>>
> >>>>>> On Fri, Jan 22, 2010 at 12:50 AM, Fran?ois Delawarde
> >>>>>> <fdelawarde at wirelessmundi.com> wrote:
> >>>>>>>
> >>>>>>> Why do you need 2 fifos? You could have callback agents connected
> to
> >>>>>>> the
> >>>>>>> fifo and send incoming calls there, mod_fifo will do the rest.
> >>>>>>>
> >>>>>>> To configure agents for callback:
> >>>>>>>
> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
> >>>>>>>
> >>>>>>> To place a call into a fifo:
> >>>>>>>
> >>>>>>>
> http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue
> >>>>>>>
> >>>>>>> Fran?ois.
> >>>>>>>
> >>>>>>> On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
> >>>>>>> > Sorry about the confusion, I'm just trying to think over all the
> >>>>>>> > abnormal situations before the implementation and hope the flow
> has
> >>>>>>> > no
> >>>>>>> > design flaws.
> >>>>>>> >
> >>>>>>> > Client A is parked in Queue 1, multiple consumers will be ringed
> to
> >>>>>>> > answer him. And once the first one is connected, all the rest
> will
> >>>>>>> > hang up. This is the targeted function.
> >>>>>>> >
> >>>>>>> > To achieve this, I'm considering to originate a call to each
> >>>>>>> > consumer
> >>>>>>> > and put the calls in Queue 2. Then intercept client A to the
> first
> >>>>>>> > element of Queue 2.
> >>>>>>> >
> >>>>>>> > I'm not sure if it's the usual or the best way. If you feel not,
> >>>>>>> > please don't hesitate to correct me. Any thoughts are warmly
> >>>>>>> > appreciated.
> >>>>>>> >
> >>>>>>> > On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com>
> >>>>>>> > wrote:
> >>>>>>> >         what no errors mean? how do you originate calls to
> >>>>>>> > consumers?
> >>>>>>> >         I don't
> >>>>>>> >         understand your scenario.
> >>>>>>> >
> >>>>>>> >         2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
> >>>>>>> >
> >>>>>>> >         > Hi All,
> >>>>>>> >         >
> >>>>>>> >         > Please advise whether the following flow makes sense.
> >>>>>>> >         >
> >>>>>>> >         > 1. Client A calls in and parked in Queue 1
> >>>>>>> >         > 2. Originate calls to several consumers simultaneously
> >>>>>>> > and
> >>>>>>> >         park them in
> >>>>>>> >         > Queue 2
> >>>>>>> >         > 3. Intercept A's call to the first consumer of Queue 2
> >>>>>>> >         >
> >>>>>>> >         > Basically I want A's call picked up by the first among
> >>>>>>> > many
> >>>>>>> >         consumers with
> >>>>>>> >         > no errors. Please let me know whether I'm on the right
> >>>>>>> >         track.
> >>>>>>> >         >
> >>>>>>> >         > Thanks and best regards,
> >>>>>>> >         > -Jingwei
> >>>>>>> >         >
> >>>>>>> >
> >>>>>>> >
> >>>>>>> >         _______________________________________________
> >>>>>>> >         FreeSWITCH-users mailing list
> >>>>>>> >         FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> >
> >>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> >
> >>>>>>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> >         http://www.freeswitch.org
> >>>>>>> >
> >>>>>>> > _______________________________________________
> >>>>>>> > FreeSWITCH-users mailing list
> >>>>>>> > FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> >
> >>>>>>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> > http://www.freeswitch.org
> >>>>>>>
> >>>>>>>
> >>>>>>> _______________________________________________
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>>
> >>>>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> http://www.freeswitch.org
> >>>>>>
> >>>>>>
> >>>>>> _______________________________________________
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>
> >>>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>> http://www.freeswitch.org
> >>>>>>
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>>
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From michal.zubac at comgate.cz  Mon Jan 25 04:26:38 2010
From: michal.zubac at comgate.cz (=?ISO-8859-2?Q?Michal_Zub=E1=E8?=)
Date: Mon, 25 Jan 2010 13:26:38 +0100
Subject: [Freeswitch-users] sangoma_prid & freeswitch openzap - conflict
Message-ID: <4B5D8DFE.30904@comgate.cz>

Hi.

I'm just curious. Is sangoma_prid neccessary for Freeswitch to work with
E1 (PRI) line? (wanpipe & openzap mode)
I stopped sangoma_prid because, when I try to start Freeswitch, openzap
yells that it cannot open D-channel (/dev/wanpipe1_if16). It is already
used by sangoma_prid.

But PRI calls are behaving strangely for me. Maybe this is the cause.
How can I resolve this conflict?

Thanks for advice. It's possible, that I am doing some newbie mistake.

Michal Zubac


From elihayun at gmail.com  Mon Jan 25 04:42:42 2010
From: elihayun at gmail.com (Eli Hayun)
Date: Mon, 25 Jan 2010 14:42:42 +0200
Subject: [Freeswitch-users] Module multicast fail
In-Reply-To: <85B99B6D-E4E2-457D-B643-4168ADA364A3@jerris.com>
References: <4B56F0F9.9090808@savion.huji.ac.il>	<2A8BE7A9-EC50-47CD-9FE8-172BF0F97DB6@freeswitch.org>	<4B5D377D.8050408@savion.huji.ac.il>
	<85B99B6D-E4E2-457D-B643-4168ADA364A3@jerris.com>
Message-ID: <4B5D91C2.1030707@savion.huji.ac.il>

On 01/25/2010 10:59 AM, Michael Jerris wrote:
> This means it was unable to joint he multicast group you specified in the "address" in your conf file for the module.
>
> Mie
>
> On Jan 25, 2010, at 1:17 AM, Eli Hayun wrote:
>   
>> Hi
>> I run the latest version and still getting an error
>>
>> 2010-01-25 07:42:53.449509 [ERR] mod_event_multicast.c:410 Multicast Error^M
>> 2010-01-25 07:42:53.449761 [CRIT] switch_loadable_module.c:872 Error
>> Loading module /freeswitch-1.0.5_20100120-0400/mod/mod_event_multicast.so^M
>> **Module load routine returned an error**^M
>> 2010-01-25 07:42:53.450424 [CONSOLE] switch_loadable_module.c:890
>> Successfully Loaded [mod_event_socket]^M
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
Hi
And How do I solve this problem? I put the address of my server and it
gave me this error. I tried to stay with the default, same problem.
I want to be able to notify one FS server of all the activities of
onother FS server

Thanks
Eli



From satish_lx at hotmail.com  Mon Jan 25 05:12:08 2010
From: satish_lx at hotmail.com (Satish Patel)
Date: Mon, 25 Jan 2010 08:12:08 -0500
Subject: [Freeswitch-users] Freeswitch billing
Message-ID: <BLU0-SMTP61C041045C3AA7CC15F147905F0@phx.gbl>

I'm planing to intigrate billing fuctionality with freeswitch us there  
any thing available which I can use ?

Mod_nibble is available but is there any GUI for billing ?

Thanks,
Satish


From wasim at convergence.pk  Mon Jan 25 05:51:11 2010
From: wasim at convergence.pk (Wasim Baig)
Date: Mon, 25 Jan 2010 18:51:11 +0500
Subject: [Freeswitch-users] Freeswitch billing
In-Reply-To: <BLU0-SMTP61C041045C3AA7CC15F147905F0@phx.gbl>
References: <BLU0-SMTP61C041045C3AA7CC15F147905F0@phx.gbl>
Message-ID: <b8ad2a5b1001250551t641c53b9s182627f967e904af@mail.gmail.com>

On Mon, Jan 25, 2010 at 6:12 PM, Satish Patel <satish_lx at hotmail.com> wrote:

> I'm planing to intigrate billing fuctionality with freeswitch us there
> any thing available which I can use ?
>
> Mod_nibble is available but is there any GUI for billing ?
>

try astpp, it has a working calling card setup

-- 
wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
peace be upon you ...
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From brian at freeswitch.org  Mon Jan 25 06:37:13 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 25 Jan 2010 08:37:13 -0600
Subject: [Freeswitch-users] Module multicast fail
In-Reply-To: <4B5D91C2.1030707@savion.huji.ac.il>
References: <4B56F0F9.9090808@savion.huji.ac.il>	<2A8BE7A9-EC50-47CD-9FE8-172BF0F97DB6@freeswitch.org>	<4B5D377D.8050408@savion.huji.ac.il>
	<85B99B6D-E4E2-457D-B643-4168ADA364A3@jerris.com>
	<4B5D91C2.1030707@savion.huji.ac.il>
Message-ID: <0075CC82-4653-446C-8F96-89A0AF08B84E@freeswitch.org>

The address of your server isn't a mulitcast address.  Restore the defaults.

/b

On Jan 25, 2010, at 6:42 AM, Eli Hayun wrote:

> Hi
> And How do I solve this problem? I put the address of my server and it
> gave me this error. I tried to stay with the default, same problem.
> I want to be able to notify one FS server of all the activities of
> onother FS server
> 
> Thanks
> Eli



From brian at freeswitch.org  Mon Jan 25 06:41:24 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 25 Jan 2010 08:41:24 -0600
Subject: [Freeswitch-users] sangoma_prid & freeswitch openzap - conflict
In-Reply-To: <4B5D8DFE.30904@comgate.cz>
References: <4B5D8DFE.30904@comgate.cz>
Message-ID: <6B96C690-1E6C-4C9E-9B5F-B23DA1354B48@freeswitch.org>

If you are using PRID you do not configure D channels at all.  Sangoma PRID will use those already.

/b

On Jan 25, 2010, at 6:26 AM, Michal Zub?? wrote:

> Hi.
> 
> I'm just curious. Is sangoma_prid neccessary for Freeswitch to work with
> E1 (PRI) line? (wanpipe & openzap mode)
> I stopped sangoma_prid because, when I try to start Freeswitch, openzap
> yells that it cannot open D-channel (/dev/wanpipe1_if16). It is already
> used by sangoma_prid.
> 
> But PRI calls are behaving strangely for me. Maybe this is the cause.
> How can I resolve this conflict?
> 
> Thanks for advice. It's possible, that I am doing some newbie mistake.
> 
> Michal Zubac



From tjardick at vanderkraan.net  Mon Jan 25 07:21:22 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Mon, 25 Jan 2010 16:21:22 +0100
Subject: [Freeswitch-users] Compile error sofia on Mac OS X
In-Reply-To: <9AEDA6E3-E875-4E63-A4A5-9457317C1D1A@freeswitch.org>
References: <F026A1E8-211C-444E-B6FA-AAFCDBE806C5@vanderkraan.net>
	<9AEDA6E3-E875-4E63-A4A5-9457317C1D1A@freeswitch.org>
Message-ID: <D081C5BA-C32F-463C-A5F2-BC38AF438E05@vanderkraan.net>

Hi Brian,

Just little message to confirm it worked now.

Thanks for the quick fix!

Regards,

Tjardick


On 24 Jan 2010, at 22:15, Brian West wrote:

> Update this has been fixed now.
>
> Thanks,
> Brian
>
> On Jan 24, 2010, at 2:51 PM, Tjardick van der Kraan wrote:
>
>> Hi,
>>
>> I'm trying to compile freeswitch on my MacBook pro to have a local  
>> dev instance, but i run in to the following compile error during  
>> the make:
>>
>> Compiling mod_sofia.c ...
>> cc1: warnings being treated as errors
>> mod_sofia.c: In function 'sofia_receive_message':
>> mod_sofia.c:1446: warning: 'from_host' may be used uninitialized in  
>> this function
>> mod_sofia.c:1446: warning: 'from_user' may be used uninitialized in  
>> this function
>> make[5]: *** [mod_sofia_la-mod_sofia.lo] Error 1
>> make[4]: *** [all] Error 2
>> make[3]: *** [mod_sofia-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>>
>>
>> It's an MBP running Leopard version 10.5.8
>>
>> Any help would be appreciated.
>>
>> Kind regards,
>>
>> Tjardick
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From yehavi.bourvine at gmail.com  Mon Jan 25 07:38:25 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 25 Jan 2010 17:38:25 +0200
Subject: [Freeswitch-users] Sending CDRs to both file and SQL?
Message-ID: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com>

Hello,

  At present we send all our CDRs to a flat file using Asterisk's format
(template "asterisk" in cdr_csv.conf.xml). This file is used by our billing
software.

  For an interim period I would like to send the CDRs to both file and MySQL
database (until I finish writing script to retreive the CDRs from the
database to a file). Is it possible to send the CDRs to both?

                       Thanks, __Yehavi:
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From christian.loeschenkohl at xpirio.com  Mon Jan 25 07:52:25 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 25 Jan 2010 16:52:25 +0100
Subject: [Freeswitch-users] call drops on unanswered UPDATE messages after
	about 30s
Message-ID: <4B5DBE39.7020101@xpirio.com>

hello

we do see some new (interfering) behavior after updating to trunk (revision 16456)
the call gets up normal

- normal invite to sip endpoint (OpenCom 130, OpenCom X320)
- we get 100 trying back
- we get 180 ringing back + rtp
- we get 200 ok
- our freeswitch sends an ack message
- after that freeswitch starts sending UPDATE messages
   none of these UPDATE message is answered by the sip endpoint, so the call gets dropped
   after about 30s.

how can this UPDATE messages be disabled.
i didn't find any option for that.

this is also not nat problem (official ip on the sip device)

please advise

br

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From anthony.minessale at gmail.com  Mon Jan 25 07:56:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 25 Jan 2010 09:56:14 -0600
Subject: [Freeswitch-users] Sending CDRs to both file and SQL?
In-Reply-To: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com>
References: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com>
Message-ID: <191c3a031001250756y9ea5211hf01d2839cbf9bd74@mail.gmail.com>

we leave that exercise to the user, there is no module to write cdr's direct
to a db.


On Mon, Jan 25, 2010 at 9:38 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Hello,
>
>   At present we send all our CDRs to a flat file using Asterisk's format
> (template "asterisk" in cdr_csv.conf.xml). This file is used by our billing
> software.
>
>   For an interim period I would like to send the CDRs to both file and
> MySQL database (until I finish writing script to retreive the CDRs from the
> database to a file). Is it possible to send the CDRs to both?
>
>                        Thanks, __Yehavi:
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From yehavi.bourvine at gmail.com  Mon Jan 25 07:56:43 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 25 Jan 2010 17:56:43 +0200
Subject: [Freeswitch-users] Guide of creating national fonts for Polycom
	phones
Message-ID: <d361ca7b1001250756s490d1924iaba80921a13eb3ce@mail.gmail.com>

Hello,

  We had to add Hebrew labels support for Polycom phones. The way we did it
is described now on the wiki at "polycom configuration" page. The process
for other languages should be quite identical.

  We did not bother with right-to-lef issues, as these cannot be dealt with
without Polycom's help.

                               Regards, __Yehavi:
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From fvillarroel at yahoo.com  Mon Jan 25 08:01:53 2010
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Mon, 25 Jan 2010 08:01:53 -0800 (PST)
Subject: [Freeswitch-users] CDR Gateways
Message-ID: <961072.63349.qm@web34305.mail.mud.yahoo.com>

Dear All.

I have defined various gateways in ~/sip-profiles/external

My questions is if a gateway named foo that send calls from diferents IP address like x.x.x.x and x.x.x.y to my FS. How i can doing a group for both ip address where i can doing later a sql like accountcode=foo?

In this moment if i need know the traffic of customer foo, i should doing two differents cdr, one for every ip address.

It's fine or i can doing of different way?

I hope anyone could me explain how i can doing. 

my gateway foo.xml

<include>
    <gateway name="foo">
      <param name="username" value="foo"/>
      <param name="password" value="foo"/>
      <param name="realm" value="x.x.x.x"/>
      <param name="register" value="false"/>
      <param name="context" value="public"/>
    </gateway>
</include>

foo1.xml

<include>
    <gateway name="foo1">
      <param name="username" value="foo1"/>
      <param name="password" value="foo1"/>
      <param name="realm" value="x.x.x.y"/>
      <param name="register" value="false"/>
      <param name="context" value="public"/>
    </gateway>
</include>

Both gateways foo and foo1 are the same customer

my cdr_csv.conf.xml

<configuration name="cdr_csv.conf" description="CDR CSV Format">
  <settings>
    <!-- 'cdr-csv' will always be appended to log-base -->
    <!--<param name="log-base" value="/var/log"/>-->
    <param name="default-template" value="sql"/>
    <!-- This is like the info app but after the call is hung up -->
    <!--<param name="debug" value="true"/>-->
    <param name="rotate-on-hup" value="false"/>
    <!-- may be a b or ab -->
    <param name="legs" value="ab"/>
  </settings>
  <templates>
        <template name="sql">INSERT INTO v_cdr (v_id, caller_id_name, caller_id_number, destination_number, context, start_stamp, answer_stamp, end_stamp,
duration, billsec, hangup_cause, uuid, bleg_uuid, accountcode, read_codec, write_codec, remote_media_ip, network_addr) VALUES ("{v_id}", "${caller_id_name}"
,"${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}"
,"${uuid}","${bleg_uuid}", "${accountcode}", "${read_codec}", "${write_codec}", "${remote_media_ip}", "${network_addr}");</template>

The argument accountcode on my database is Blank or None for all records of gateways.

Regards.



      


From anthony.minessale at gmail.com  Mon Jan 25 08:03:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 25 Jan 2010 10:03:45 -0600
Subject: [Freeswitch-users] call drops on unanswered UPDATE messages
	after about 30s
In-Reply-To: <4B5DBE39.7020101@xpirio.com>
References: <4B5DBE39.7020101@xpirio.com>
Message-ID: <191c3a031001250803m1177a0f2w853346a17e9be0fd@mail.gmail.com>

you must have polycoms,
if you are running a stable firmware on them the updates should be accepted
and replied to by your phone.
otherwise add the global variable ignore_display_updates=true


2010/1/25 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> hello
>
> we do see some new (interfering) behavior after updating to trunk (revision
> 16456)
> the call gets up normal
>
> - normal invite to sip endpoint (OpenCom 130, OpenCom X320)
> - we get 100 trying back
> - we get 180 ringing back + rtp
> - we get 200 ok
> - our freeswitch sends an ack message
> - after that freeswitch starts sending UPDATE messages
>   none of these UPDATE message is answered by the sip endpoint, so the call
> gets dropped
>   after about 30s.
>
> how can this UPDATE messages be disabled.
> i didn't find any option for that.
>
> this is also not nat problem (official ip on the sip device)
>
> please advise
>
> br
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From christian.loeschenkohl at xpirio.com  Mon Jan 25 08:21:44 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 25 Jan 2010 17:21:44 +0100
Subject: [Freeswitch-users] call drops on unanswered UPDATE
 messages	after about 30s
In-Reply-To: <191c3a031001250803m1177a0f2w853346a17e9be0fd@mail.gmail.com>
References: <4B5DBE39.7020101@xpirio.com>
	<191c3a031001250803m1177a0f2w853346a17e9be0fd@mail.gmail.com>
Message-ID: <4B5DC518.5080309@xpirio.com>

thank you for you quick reply

these devices are sip pbx kind-of-do-it-all (sip, dect, isdn, analog ...)
see also http://www.aastra.com/cps/rde/xchg/04/hs.xsl/15668.htm

i also posted a call setup done right now with ignore_display_updates=true and i
see no difference - please note the many update packages send from our freeswitch (92.63.208.24)
after 200 ok.

http://pastebin.freeswitch.org/11934

br

On 2010-01-25 17:03, Anthony Minessale wrote:

> you must have polycoms,
> if you are running a stable firmware on them the updates should be
> accepted and replied to by your phone.
> otherwise add the global variable ignore_display_updates=true
>
>
> 2010/1/25 Christian L?schenkohl <christian.loeschenkohl at xpirio.com
> <mailto:christian.loeschenkohl at xpirio.com>>
>
>     hello
>
>     we do see some new (interfering) behavior after updating to trunk
>     (revision 16456)
>     the call gets up normal
>
>     - normal invite to sip endpoint (OpenCom 130, OpenCom X320)
>     - we get 100 trying back
>     - we get 180 ringing back + rtp
>     - we get 200 ok
>     - our freeswitch sends an ack message
>     - after that freeswitch starts sending UPDATE messages
>        none of these UPDATE message is answered by the sip endpoint, so
>     the call gets dropped
>        after about 30s.
>
>     how can this UPDATE messages be disabled.
>     i didn't find any option for that.
>
>     this is also not nat problem (official ip on the sip device)
>
>     please advise
>
>     br
>
>     --
>     Ing. Christian L?schenkohl
>     Technische Leitung, Forschung & Entwicklung VoIP
>
>     xpirio
>     Telekommunikation & Service GmbH
>     Lakeside B04
>     9020 Klagenfurt
>     Austria
>
>     T  +43 (0) 5 77 11 - 1000
>     F  +43 (0) 5 77 11 - 1002
>     E christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From brian at freeswitch.org  Mon Jan 25 08:27:34 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 25 Jan 2010 10:27:34 -0600
Subject: [Freeswitch-users] call drops on unanswered UPDATE
	messages	after about 30s
In-Reply-To: <4B5DC518.5080309@xpirio.com>
References: <4B5DBE39.7020101@xpirio.com>
	<191c3a031001250803m1177a0f2w853346a17e9be0fd@mail.gmail.com>
	<4B5DC518.5080309@xpirio.com>
Message-ID: <C737B7E7-1BBA-4B5C-868C-DCED4E667E22@freeswitch.org>

global_setvar ignore_display_updates=true

/b

On Jan 25, 2010, at 10:21 AM, Christian L?schenkohl wrote:

> thank you for you quick reply
> 
> these devices are sip pbx kind-of-do-it-all (sip, dect, isdn, analog ...)
> see also http://www.aastra.com/cps/rde/xchg/04/hs.xsl/15668.htm
> 
> i also posted a call setup done right now with ignore_display_updates=true and i
> see no difference - please note the many update packages send from our freeswitch (92.63.208.24)
> after 200 ok.
> 
> http://pastebin.freeswitch.org/11934
> 
> br

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From christian.loeschenkohl at xpirio.com  Mon Jan 25 08:59:11 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 25 Jan 2010 17:59:11 +0100
Subject: [Freeswitch-users] call drops on unanswered
 UPDATE	messages	after about 30s
In-Reply-To: <C737B7E7-1BBA-4B5C-868C-DCED4E667E22@freeswitch.org>
References: <4B5DBE39.7020101@xpirio.com>	<191c3a031001250803m1177a0f2w853346a17e9be0fd@mail.gmail.com>	<4B5DC518.5080309@xpirio.com>
	<C737B7E7-1BBA-4B5C-868C-DCED4E667E22@freeswitch.org>
Message-ID: <4B5DCDDF.6050305@xpirio.com>

thank you, i works now

---
for documentation

would conf/vars.xml or conf/autoload_configs/switch.conf.xml be the right place for this setting

like
<X-PRE-PROCESS cmd="set" data="ignore_display_updates=true"/>
or
<param name="ignore_display_updates" value="true"/>

so i can put it in the wiki

br

On 2010-01-25 17:27, Brian West wrote:

> global_setvar ignore_display_updates=true
>
> /b
>
> On Jan 25, 2010, at 10:21 AM, Christian L?schenkohl wrote:
>
>> thank you for you quick reply
>>
>> these devices are sip pbx kind-of-do-it-all (sip, dect, isdn, analog ...)
>> see also http://www.aastra.com/cps/rde/xchg/04/hs.xsl/15668.htm
>>
>> i also posted a call setup done right now with
>> ignore_display_updates=true and i
>> see no difference - please note the many update packages send from our
>> freeswitch (92.63.208.24)
>> after 200 ok.
>>
>> http://pastebin.freeswitch.org/11934
>>
>> br
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From brian at freeswitch.org  Mon Jan 25 09:10:38 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 25 Jan 2010 11:10:38 -0600
Subject: [Freeswitch-users] call drops on unanswered
	UPDATE	messages	after about 30s
In-Reply-To: <4B5DCDDF.6050305@xpirio.com>
References: <4B5DBE39.7020101@xpirio.com>	<191c3a031001250803m1177a0f2w853346a17e9be0fd@mail.gmail.com>	<4B5DC518.5080309@xpirio.com>
	<C737B7E7-1BBA-4B5C-868C-DCED4E667E22@freeswitch.org>
	<4B5DCDDF.6050305@xpirio.com>
Message-ID: <0B75CBBB-4D09-4988-A1CD-3DCF04066C8E@freeswitch.org>

see vars.xml you'll see others like this... also you're in bypass media mode.  Didn't mention that eh?

/b

On Jan 25, 2010, at 10:59 AM, Christian L?schenkohl wrote:

> <X-PRE-PROCESS cmd="set" data="ignore_display_updates=true"/>



From testeador01 at gmail.com  Mon Jan 25 09:11:06 2010
From: testeador01 at gmail.com (Milena)
Date: Mon, 25 Jan 2010 12:11:06 -0500
Subject: [Freeswitch-users] making calls through an FXO gw without knowing
	the IP
Message-ID: <c29b930b1001250911vfdc4cb7t30f0fbc2e36d0a92@mail.gmail.com>

Hello everyone,

I need some help in my dialplan, I have an FXO gateway and i can
receive calls from it,

now, if I assign a static IP on it, i can make calls through the fxo
gateway using this:
                 <action application="bridge"
data="sofia/<profile>/<destination>@<gw_IP>"/>

I want to know if it is possible to make calls through this gateway if
it has a variable IP, and how to do it if so;
the gateway uses SIP accounts to register to FS so i think it might be
possible to route the call using one of the registered extension
numbers but I am not sure how so I need some pointers,

Anyone has ideas or knows how to do this?, help is greatly appreciated!


From christian.loeschenkohl at xpirio.com  Mon Jan 25 09:18:29 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 25 Jan 2010 18:18:29 +0100
Subject: [Freeswitch-users] call drops on unanswered
 UPDATE	messages	after about 30s
In-Reply-To: <0B75CBBB-4D09-4988-A1CD-3DCF04066C8E@freeswitch.org>
References: <4B5DBE39.7020101@xpirio.com>	<191c3a031001250803m1177a0f2w853346a17e9be0fd@mail.gmail.com>	<4B5DC518.5080309@xpirio.com>
	<C737B7E7-1BBA-4B5C-868C-DCED4E667E22@freeswitch.org>
	<4B5DCDDF.6050305@xpirio.com>
	<0B75CBBB-4D09-4988-A1CD-3DCF04066C8E@freeswitch.org>
Message-ID: <4B5DD265.7030500@xpirio.com>

yes, caught :-)
we do use bypass media mode (because of t.38)

yes, i also think vars.xml is a good place for this - will put it in the wiki there

br

On 2010-01-25 18:10, Brian West wrote:

> see vars.xml you'll see others like this... also you're in bypass media mode.  Didn't mention that eh?
>
> /b
>
> On Jan 25, 2010, at 10:59 AM, Christian L?schenkohl wrote:
>
>> <X-PRE-PROCESS cmd="set" data="ignore_display_updates=true"/>
>


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From testeador01 at gmail.com  Mon Jan 25 09:20:21 2010
From: testeador01 at gmail.com (Milena)
Date: Mon, 25 Jan 2010 12:20:21 -0500
Subject: [Freeswitch-users] Sending CDRs to both file and SQL?
In-Reply-To: <191c3a031001250756y9ea5211hf01d2839cbf9bd74@mail.gmail.com>
References: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com>
	<191c3a031001250756y9ea5211hf01d2839cbf9bd74@mail.gmail.com>
Message-ID: <c29b930b1001250920g5999d707i1caa38298795cbae@mail.gmail.com>

hello Yehavi,

This is what the wiki suggests anyways, in case you didn't read this yet:
http://wiki.freeswitch.org/wiki/Mod_cdr_csv#Example_Perl_Script_for_CDR_into_Mysql


2010/1/25 Anthony Minessale <anthony.minessale at gmail.com>:
> we leave that exercise to the user, there is no module to write cdr's
direct
> to a db.
>
>
> On Mon, Jan 25, 2010 at 9:38 AM, Yehavi Bourvine <
yehavi.bourvine at gmail.com>
> wrote:
>>
>> Hello,
>>
>>   At present we send all our CDRs to a flat file using Asterisk's format
>> (template "asterisk" in cdr_csv.conf.xml). This file is used by our
billing
>> software.
>>
>>   For an interim period I would like to send the CDRs to both file and
>> MySQL database (until I finish writing script to retreive the CDRs from
the
>> database to a file). Is it possible to send the CDRs to both?
>>
>>                        Thanks, __Yehavi:
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Jan 25 09:20:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 25 Jan 2010 11:20:30 -0600
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <9881D312-67D1-40D3-B169-A178202F4E6C@jerris.com>
References: <984278.36075.qm@web33504.mail.mud.yahoo.com>
	<9881D312-67D1-40D3-B169-A178202F4E6C@jerris.com>
Message-ID: <191c3a031001250920y5e622c0ewa7a80b9fe5799388@mail.gmail.com>

its possible your string hits the parser more than once.
try using 4 \

\\\\sound


On Sun, Jan 24, 2010 at 4:03 AM, Michael Jerris <mike at jerris.com> wrote:

> As noted on that bug, you should be able to either use \\ or / for the path
> separator there and it should work.
>
> Mike
>
> On Jan 22, 2010, at 9:18 AM, Diego Toro wrote:
>
> > Hi, with svn r16440 the problem persists, I creted a jira report
> http://jira.freeswitch.org/browse/LBSNDF-8 this is a minor issue, but
> activing playback delimiter no audio file can be played. On FS the audio
> files are placed in the \sound\ directory, building the path on Windows
> would be \sound '\s' which is replaced by 'ound'.
> >
> > Thank you
> >
> > Diego Toro
> > http://lacarretade.blogspot.com/
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From vfclists at gmail.com  Mon Jan 25 09:36:42 2010
From: vfclists at gmail.com (vfclists)
Date: Mon, 25 Jan 2010 09:36:42 -0800 (PST)
Subject: [Freeswitch-users] Where can I set the IP address for Freeswitch to
 bind to?
Message-ID: <27308006.post@talk.nabble.com>


I have a rather complicated but necessary IP setup. 

I have a VPN running on the system, and some of the IP address a bridged
with other network devices, and Freeswitch is not quite sure which one to
bind to.

It has an IP for the local network, one for the VPN and one for the gateway,
but it appears to bind to the gateway IP or the VPN when the VPN comes up. I
want it bind to the local IP, and let the VPN do its normal stuff on the
outbound,

I edited vars.xml and set local ip but it doesn't seem to be working.

  <X-PRE-PROCESS cmd="set" data="domain=192.168.1.41"/>
  <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>

Is the vars.xml the right place to do it?

I take it that vars.xml is where you change stuff in the default settings?

Thanks

/vfclists

-- 
View this message in context: http://old.nabble.com/Where-can-I-set-the-IP-address-for-Freeswitch-to-bind-to--tp27308006p27308006.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From brian at freeswitch.org  Mon Jan 25 10:07:55 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 25 Jan 2010 12:07:55 -0600
Subject: [Freeswitch-users] Where can I set the IP address for
	Freeswitch to bind to?
In-Reply-To: <27308006.post@talk.nabble.com>
References: <27308006.post@talk.nabble.com>
Message-ID: <4F1F4A28-19BC-4EC2-BCA2-C656354B3071@freeswitch.org>

open up sip_profiles/*.xml and put the IP in there for rtp-ip and sip-ip

/b

On Jan 25, 2010, at 11:36 AM, vfclists wrote:

> 
> I have a rather complicated but necessary IP setup. 
> 
> I have a VPN running on the system, and some of the IP address a bridged
> with other network devices, and Freeswitch is not quite sure which one to
> bind to.
> 
> It has an IP for the local network, one for the VPN and one for the gateway,
> but it appears to bind to the gateway IP or the VPN when the VPN comes up. I
> want it bind to the local IP, and let the VPN do its normal stuff on the
> outbound,
> 
> I edited vars.xml and set local ip but it doesn't seem to be working.
> 
>  <X-PRE-PROCESS cmd="set" data="domain=192.168.1.41"/>
>  <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
> 
> Is the vars.xml the right place to do it?
> 
> I take it that vars.xml is where you change stuff in the default settings?
> 
> Thanks
> 
> /vfclists



From mgg at giagnocavo.net  Mon Jan 25 10:08:50 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Mon, 25 Jan 2010 13:08:50 -0500
Subject: [Freeswitch-users] Where can I set the IP address for
 Freeswitch to bind to?
In-Reply-To: <27308006.post@talk.nabble.com>
References: <27308006.post@talk.nabble.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C67032C749445@mse17be1.mse17.exchange.ms>

That'll be in the per-profile settings (sofia).
-Michael

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of vfclists
Sent: Monday, January 25, 2010 10:37 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Where can I set the IP address for Freeswitch to bind to?


I have a rather complicated but necessary IP setup. 

I have a VPN running on the system, and some of the IP address a bridged with other network devices, and Freeswitch is not quite sure which one to bind to.

It has an IP for the local network, one for the VPN and one for the gateway, but it appears to bind to the gateway IP or the VPN when the VPN comes up. I want it bind to the local IP, and let the VPN do its normal stuff on the outbound,

I edited vars.xml and set local ip but it doesn't seem to be working.

  <X-PRE-PROCESS cmd="set" data="domain=192.168.1.41"/>
  <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>

Is the vars.xml the right place to do it?

I take it that vars.xml is where you change stuff in the default settings?

Thanks

/vfclists

--
View this message in context: http://old.nabble.com/Where-can-I-set-the-IP-address-for-Freeswitch-to-bind-to--tp27308006p27308006.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From sos at sokhapkin.dyndns.org  Mon Jan 25 10:09:22 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 25 Jan 2010 13:09:22 -0500
Subject: [Freeswitch-users] Where can I set the IP address for
	Freeswitch to bind to?
In-Reply-To: <27308006.post@talk.nabble.com>
References: <27308006.post@talk.nabble.com>
Message-ID: <201001251309.22400.sos@sokhapkin.dyndns.org>

You need to set IP in the  profile settings, rtp-ip and sip-ip parameters.

On Monday 25 January 2010, vfclists wrote:
> I have a rather complicated but necessary IP setup.
>
> I have a VPN running on the system, and some of the IP address a bridged
> with other network devices, and Freeswitch is not quite sure which one to
> bind to.
>
> It has an IP for the local network, one for the VPN and one for the
> gateway, but it appears to bind to the gateway IP or the VPN when the VPN
> comes up. I want it bind to the local IP, and let the VPN do its normal
> stuff on the outbound,
>
> I edited vars.xml and set local ip but it doesn't seem to be working.
>
>   <X-PRE-PROCESS cmd="set" data="domain=192.168.1.41"/>
>   <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
>
> Is the vars.xml the right place to do it?
>
> I take it that vars.xml is where you change stuff in the default settings?
>
> Thanks
>
> /vfclists




From yehavi.bourvine at gmail.com  Mon Jan 25 10:11:46 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 25 Jan 2010 20:11:46 +0200
Subject: [Freeswitch-users] Sending CDRs to both file and SQL?
In-Reply-To: <c29b930b1001250920g5999d707i1caa38298795cbae@mail.gmail.com>
References: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com>
	<191c3a031001250756y9ea5211hf01d2839cbf9bd74@mail.gmail.com>
	<c29b930b1001250920g5999d707i1caa38298795cbae@mail.gmail.com>
Message-ID: <d361ca7b1001251011q32120d50l5f14a98f33fcee29@mail.gmail.com>

Thanks. I've seen that. Will try to implement this.

                          Thanks, __Yehavi:

2010/1/25 Milena <testeador01 at gmail.com>

> hello Yehavi,
>
> This is what the wiki suggests anyways, in case you didn't read this yet:
>
> http://wiki.freeswitch.org/wiki/Mod_cdr_csv#Example_Perl_Script_for_CDR_into_Mysql
>
>
> 2010/1/25 Anthony Minessale <anthony.minessale at gmail.com>:
>
> > we leave that exercise to the user, there is no module to write cdr's
> direct
> > to a db.
> >
> >
> > On Mon, Jan 25, 2010 at 9:38 AM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com>
> > wrote:
> >>
> >> Hello,
> >>
> >>   At present we send all our CDRs to a flat file using Asterisk's format
> >> (template "asterisk" in cdr_csv.conf.xml). This file is used by our
> billing
> >> software.
> >>
> >>   For an interim period I would like to send the CDRs to both file and
> >> MySQL database (until I finish writing script to retreive the CDRs from
> the
> >> database to a file). Is it possible to send the CDRs to both?
> >>
> >>                        Thanks, __Yehavi:
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From info at daccii.it  Mon Jan 25 10:15:33 2010
From: info at daccii.it (Daniele Salvatore Albano)
Date: Mon, 25 Jan 2010 19:15:33 +0100
Subject: [Freeswitch-users] making calls through an FXO gw without
 knowing the IP
In-Reply-To: <c29b930b1001250911vfdc4cb7t30f0fbc2e36d0a92@mail.gmail.com>
References: <c29b930b1001250911vfdc4cb7t30f0fbc2e36d0a92@mail.gmail.com>
Message-ID: <4B5DDFC5.7000402@daccii.it>

I don't know if it's possible to do using plain xml, but you can try 
writing some javascript/lua/what-you-want code to acquire parameters 
from the registration and use them to do the call

Milena ha scritto:
> Hello everyone,
>
> I need some help in my dialplan, I have an FXO gateway and i can
> receive calls from it,
>
> now, if I assign a static IP on it, i can make calls through the fxo
> gateway using this:
>                  <action application="bridge"
> data="sofia/<profile>/<destination>@<gw_IP>"/>
>
> I want to know if it is possible to make calls through this gateway if
> it has a variable IP, and how to do it if so;
> the gateway uses SIP accounts to register to FS so i think it might be
> possible to route the call using one of the registered extension
> numbers but I am not sure how so I need some pointers,
>
> Anyone has ideas or knows how to do this?, help is greatly appreciated!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From lon at kickasspixels.com  Mon Jan 25 10:22:35 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Mon, 25 Jan 2010 10:22:35 -0800
Subject: [Freeswitch-users] Sending CDRs to both file and SQL?
In-Reply-To: <d361ca7b1001251011q32120d50l5f14a98f33fcee29@mail.gmail.com>
References: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com> 
	<191c3a031001250756y9ea5211hf01d2839cbf9bd74@mail.gmail.com> 
	<c29b930b1001250920g5999d707i1caa38298795cbae@mail.gmail.com> 
	<d361ca7b1001251011q32120d50l5f14a98f33fcee29@mail.gmail.com>
Message-ID: <5d3e0dc61001251022h6c98722ag8b59fb12ce2d8ee8@mail.gmail.com>

For what its worth, I wrote a simple Sinatra-based web service to
accept the CDRs over HTTP. It then queues them for processing into a
database, syslog and/or file.

Its easy to load balance and lets me throttle CDR processing so I
don't put excessive load on the database server at peak times.

Lon


From jerry.richards at teotech.com  Mon Jan 25 10:30:47 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 25 Jan 2010 10:30:47 -0800
Subject: [Freeswitch-users] Presence Change Distribution
In-Reply-To: <2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com>
References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com>
	<2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com>
Message-ID: <26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com>

Okay, I notice that if I reduce the Presence SUBSCRIBE Expires duration
(from 3600 seconds to 60 seconds), then the delay between PUBLISH's and
NOTIFY's is reduced, but FS still waits nearly 45 seconds to send the
NOTIFY's after it receives a PUBLISH.
 
Can a change be made in FS so that NOTIFYs are sent as a direct result of
receipt of the PUBLISH message?  And not tied to the SUBSCRIBE expiration?
I really don't want to configure all my phones to re-subscribe every 30 or
15 seconds.
 
Thanks and Best Regards,
Jerry
 


  _____  

From: RobertT [mailto:siniypin at gmail.com] 
Sent: Tuesday, December 29, 2009 12:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


You can try to reduce your registration time. 
I for one made my client apps send PUBLISH message every minute in addition
to reduced registration time.

Regards, Robert.


2009/12/28 Jerry Richards <jerry.richards at teotech.com>


Is there a setting to control how fast FS distributes presence changes to
subscribers?  Currently, it appears to take several minutes before I see
presence changes.  I would like to see them almost instantaneously, if
possible.

Thanks and Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



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From anthony.minessale at gmail.com  Mon Jan 25 11:04:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 25 Jan 2010 13:04:57 -0600
Subject: [Freeswitch-users] Presence Change Distribution
In-Reply-To: <26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com>
References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com>
	<2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com>
	<26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com>
Message-ID: <191c3a031001251104p55ba7009g9381841f7de56d65@mail.gmail.com>

the notify will be instant after the publish
the notify you see are not triggered by the publish or they would be
instant.

Same drill, turn on presence debugging in sofia.conf.xml
and look at the sql stmts and see why


On Mon, Jan 25, 2010 at 12:30 PM, Jerry Richards <jerry.richards at teotech.com
> wrote:

>  Okay, I notice that if I reduce the Presence SUBSCRIBE Expires duration
> (from 3600 seconds to 60 seconds), then the delay between PUBLISH's and
> NOTIFY's is reduced, but FS still waits nearly 45 seconds to send the
> NOTIFY's after it receives a PUBLISH.
>
> Can a change be made in FS so that NOTIFYs are sent as a direct result of
> receipt of the PUBLISH message?  And not tied to the SUBSCRIBE expiration?
> I really don't want to configure all my phones to re-subscribe every 30 or
> 15 seconds.
>
> Thanks and Best Regards,
> Jerry
>
>
>  ------------------------------
> *From:* RobertT [mailto:siniypin at gmail.com]
> *Sent:* Tuesday, December 29, 2009 12:02 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>
> You can try to reduce your registration time.
> I for one made my client apps send PUBLISH message every minute in addition
> to reduced registration time.
>
> Regards, Robert.
>
> 2009/12/28 Jerry Richards <jerry.richards at teotech.com>
>
>> Is there a setting to control how fast FS distributes presence changes to
>> subscribers?  Currently, it appears to take several minutes before I see
>> presence changes.  I would like to see them almost instantaneously, if
>> possible.
>>
>> Thanks and Best Regards,
>> Jerry
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From dftoro at yahoo.com  Mon Jan 25 11:53:07 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Mon, 25 Jan 2010 11:53:07 -0800 (PST)
Subject: [Freeswitch-users] How to get chat message via event
In-Reply-To: <7b197bef1001241305o3ee9bc9cq7decdc412f90575c@mail.gmail.com>
Message-ID: <561160.39572.qm@web33504.mail.mud.yahoo.com>

Hi, try with ESL  (libs/esl).


Diego Toro
http://lacarretade.blogspot.com/


--- On Sun, 1/24/10, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:

> From: Giovanni Maruzzelli <gmaruzz at celliax.org>
> Subject: Re: [Freeswitch-users] How to get chat message via event
> To: freeswitch-users at lists.freeswitch.org
> Date: Sunday, January 24, 2010, 4:05 PM
> Which events you don't get? From
> which channel in which circumstances?
> (I mean what you do and what do you expect?)
> 
> -giovanni
> 
> On Sun, Jan 24, 2010 at 9:59 PM, afshin afzali <a.afzali2003 at gmail.com>
> wrote:
> > Hi,
> >
> > As you say, I've already done and unfortunately did
> not get the message
> > events although other events are fired as expected :(
> >
> > -- afshin
> >
> > On Sun, Jan 24, 2010 at 7:18 PM, Giovanni Maruzzelli
> <gmaruzz at celliax.org>
> > wrote:
> >>
> >> you subscribe to them as MESSAGE events
> >>
> >> eg, from a telnet session:
> >>
> >> telnet localhost 8021
> >> auth ClueCon
> >> events plain message
> >>
> >> then those events will show up in your telnet
> session.
> >> -gm
> >>
> >> On Sun, Jan 24, 2010 at 4:39 PM, afshin afzali
> <a.afzali2003 at gmail.com>
> >> wrote:
> >> > Hi,
> >> >
> >> > It seems that the chat messages don't fire
> via events by default and
> >> > just
> >> > exchange between parties.
> >> > Is it true? Is it possible to enable those
> via events?
> >> >
> >> > appreciate all,
> >> > -- afshin
> >> >
> >> >
> _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From christian.loeschenkohl at xpirio.com  Mon Jan 25 12:58:48 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 25 Jan 2010 21:58:48 +0100
Subject: [Freeswitch-users] little hangup problem - prepaid application
Message-ID: <4B5E0608.3070001@xpirio.com>

hello

i try to implement a prepaid application and mod_nibblebill isn't my first choice.
the implementation is quite ok so far. now my problem:

- the call setup is normal (a little scripting)
- the calls get ended by sched_hangup (max duration of calls)
- then i would like to execute a script with api_hangup_hook
   in this last script i would also like to use some channel vars (a-number, call duration
   after answer) but i can't do that

i try to use an outbound event socket script but i couldn't get the variables/information
i need to calculate the amount charge

any hint here

br

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From sos at sokhapkin.dyndns.org  Mon Jan 25 13:20:35 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 25 Jan 2010 16:20:35 -0500
Subject: [Freeswitch-users] little hangup problem - prepaid application
In-Reply-To: <4B5E0608.3070001@xpirio.com>
References: <4B5E0608.3070001@xpirio.com>
Message-ID: <201001251620.35308.sos@sokhapkin.dyndns.org>

To my experience most of "interesting" billing-related variables are created 
not when channel is hung up, but later, when channel enters REPORTING state. 
I use mod_cdr_csv to access these variables:

cdr_csv.conf.xml:

<param name="default-template" value="lua"/>

<template name="lua">${lua(onhangup.lua)}</template>

onhangup.lua script accesses required variables like "billsec", does final 
calculations and writes CDR to DB.


On Monday 25 January 2010, Christian L?schenkohl wrote:
> hello
>
> i try to implement a prepaid application and mod_nibblebill isn't my first
> choice. the implementation is quite ok so far. now my problem:
>
> - the call setup is normal (a little scripting)
> - the calls get ended by sched_hangup (max duration of calls)
> - then i would like to execute a script with api_hangup_hook
>    in this last script i would also like to use some channel vars
> (a-number, call duration after answer) but i can't do that
>
> i try to use an outbound event socket script but i couldn't get the
> variables/information i need to calculate the amount charge
>
> any hint here
>
> br




From mike at van.lammeren.net  Mon Jan 25 14:02:27 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Mon, 25 Jan 2010 17:02:27 -0500
Subject: [Freeswitch-users] making calls through an FXO gw without
	knowing the IP
In-Reply-To: <4B5DDFC5.7000402@daccii.it>
References: <c29b930b1001250911vfdc4cb7t30f0fbc2e36d0a92@mail.gmail.com>
	<4B5DDFC5.7000402@daccii.it>
Message-ID: <5d2828f1001251402w28a5e962td9970fab6e1ff3e8@mail.gmail.com>

Hi Milena!

A device with an IP address has either been assigned a static IP by a
person, or assigned a dynamic IP by, probably, a DHCP server. If your FXO
gateway has been assigned a dynamic IP, then before you can figure out how
to get that value in the configuration, you need to determine what that IP
is.

How do you know the IP assigned to your FXO gateway? Does it implement the
UPnP standard?

Just out of curiosity, why not let it have a static IP address?


Mike van Lammeren


On Mon, Jan 25, 2010 at 1:15 PM, Daniele Salvatore Albano <info at daccii.it>wrote:

> I don't know if it's possible to do using plain xml, but you can try
> writing some javascript/lua/what-you-want code to acquire parameters
> from the registration and use them to do the call
>
> Milena ha scritto:
> > Hello everyone,
> >
> > I need some help in my dialplan, I have an FXO gateway and i can
> > receive calls from it,
> >
> > now, if I assign a static IP on it, i can make calls through the fxo
> > gateway using this:
> >                  <action application="bridge"
> > data="sofia/<profile>/<destination>@<gw_IP>"/>
> >
> > I want to know if it is possible to make calls through this gateway if
> > it has a variable IP, and how to do it if so;
> > the gateway uses SIP accounts to register to FS so i think it might be
> > possible to route the call using one of the registered extension
> > numbers but I am not sure how so I need some pointers,
> >
> > Anyone has ideas or knows how to do this?, help is greatly appreciated!
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Mon Jan 25 14:09:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 25 Jan 2010 16:09:06 -0600
Subject: [Freeswitch-users] making calls through an FXO gw without
	knowing the IP
In-Reply-To: <4B5DDFC5.7000402@daccii.it>
References: <c29b930b1001250911vfdc4cb7t30f0fbc2e36d0a92@mail.gmail.com>
	<4B5DDFC5.7000402@daccii.it>
Message-ID: <191c3a031001251409u50976653ib88f586f7aab9e1b@mail.gmail.com>

Either:
sofa/<profile>/<ext gateway regged as>

of if domain was different:

sofa/<profile>/<ext gateway regged as>%<domain gateway regged as>


On Mon, Jan 25, 2010 at 12:15 PM, Daniele Salvatore Albano
<info at daccii.it>wrote:

> I don't know if it's possible to do using plain xml, but you can try
> writing some javascript/lua/what-you-want code to acquire parameters
> from the registration and use them to do the call
>
> Milena ha scritto:
> > Hello everyone,
> >
> > I need some help in my dialplan, I have an FXO gateway and i can
> > receive calls from it,
> >
> > now, if I assign a static IP on it, i can make calls through the fxo
> > gateway using this:
> >                  <action application="bridge"
> > data="sofia/<profile>/<destination>@<gw_IP>"/>
> >
> > I want to know if it is possible to make calls through this gateway if
> > it has a variable IP, and how to do it if so;
> > the gateway uses SIP accounts to register to FS so i think it might be
> > possible to route the call using one of the registered extension
> > numbers but I am not sure how so I need some pointers,
> >
> > Anyone has ideas or knows how to do this?, help is greatly appreciated!
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From nazim.agabekov at gmail.com  Mon Jan 25 14:31:12 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Tue, 26 Jan 2010 02:31:12 +0400
Subject: [Freeswitch-users] Sending CDRs to both file and SQL?
In-Reply-To: <5d3e0dc61001251022h6c98722ag8b59fb12ce2d8ee8@mail.gmail.com>
References: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com>
	<191c3a031001250756y9ea5211hf01d2839cbf9bd74@mail.gmail.com>
	<c29b930b1001250920g5999d707i1caa38298795cbae@mail.gmail.com>
	<d361ca7b1001251011q32120d50l5f14a98f33fcee29@mail.gmail.com>
	<5d3e0dc61001251022h6c98722ag8b59fb12ce2d8ee8@mail.gmail.com>
Message-ID: <4B5E1BB0.2020104@gmail.com>

I have another implementation based on fastcgi and libxml2. Source could
be easily modified to log into the file as well.

http://blog.buta-tech.com/?p=1

On 01/25/2010 10:22 PM, Lon Baker wrote:
> For what its worth, I wrote a simple Sinatra-based web service to
> accept the CDRs over HTTP. It then queues them for processing into a
> database, syslog and/or file.
>
> Its easy to load balance and lets me throttle CDR processing so I
> don't put excessive load on the database server at peak times.
>
> Lon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From m.sobkow at marketelsystems.com  Mon Jan 25 17:49:41 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 25 Jan 2010 19:49:41 -0600
Subject: [Freeswitch-users] Has anyone had Freeswitch crash on loading
	mod_voicemail?
Message-ID: <4B5E4A35.3060803@marketelsystems.com>

I tried a "load mod_voicemail" in fs_cli, hoping to see what 
configuration section it requested from Erlang, but instead of loading 
the module, Freeswitch crashed without any error messages.  SVN 15188 
built on Ubuntu Hardy 32-bit.


From vfclists at googlemail.com  Mon Jan 25 17:13:50 2010
From: vfclists at googlemail.com (Frank Church)
Date: Tue, 26 Jan 2010 01:13:50 +0000
Subject: [Freeswitch-users] little hangup problem - prepaid application
In-Reply-To: <201001251620.35308.sos@sokhapkin.dyndns.org>
References: <4B5E0608.3070001@xpirio.com>
	<201001251620.35308.sos@sokhapkin.dyndns.org>
Message-ID: <7abab2411001251713p31e542b2xf6204ff13c556d03@mail.gmail.com>

I am new to Freeswitch and I am interested in how it works. When the
record is sent to the lua program what format is it sent in?

Are the details sent like parameters on the command line like

script_lua -calleridname=XXXXXX --caller_id_number=XXXXXXX
--start_stamp=XXXXXX ...?

On looking at the template as well, how is the SQL configured?

<template name="sql">INSERT INTO cdr VALUES
("${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}",
"${accountcode}");</template>

Is it a matter of configuring with the

2010/1/25 Sergey Okhapkin <sos at sokhapkin.dyndns.org>:
> To my experience most of "interesting" billing-related variables are created
> not when channel is hung up, but later, when channel enters REPORTING state.
> I use mod_cdr_csv to access these variables:
>
> cdr_csv.conf.xml:
>
> <param name="default-template" value="lua"/>
>
> <template name="lua">${lua(onhangup.lua)}</template>
>
> onhangup.lua script accesses required variables like "billsec", does final
> calculations and writes CDR to DB.
>
>
> On Monday 25 January 2010, Christian L?schenkohl wrote:
>> hello
>>
>> i try to implement a prepaid application and mod_nibblebill isn't my first
>> choice. the implementation is quite ok so far. now my problem:
>>
>> - the call setup is normal (a little scripting)
>> - the calls get ended by sched_hangup (max duration of calls)
>> - then i would like to execute a script with api_hangup_hook
>> ? ?in this last script i would also like to use some channel vars
>> (a-number, call duration after answer) but i can't do that
>>
>> i try to use an outbound event socket script but i couldn't get the
>> variables/information i need to calculate the amount charge
>>
>> any hint here
>>
>> br
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Frank Church

=======================
<a href="http://devblog.brahmancreations.com">devblog.brahmancreations.com</a>


From mike at jerris.com  Mon Jan 25 19:26:25 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 25 Jan 2010 22:26:25 -0500
Subject: [Freeswitch-users] Has anyone had Freeswitch crash on loading
	mod_voicemail?
In-Reply-To: <4B5E4A35.3060803@marketelsystems.com>
References: <4B5E4A35.3060803@marketelsystems.com>
Message-ID: <189DAD7E-77B3-4A09-AA32-1375554A2C0C@jerris.com>

Please confirm this is still the case in svn trunk and if so, report a bug with a backtrace to http://jira.freeswitch.org.

Mike

On Jan 25, 2010, at 8:49 PM, Mark Sobkow wrote:

> I tried a "load mod_voicemail" in fs_cli, hoping to see what 
> configuration section it requested from Erlang, but instead of loading 
> the module, Freeswitch crashed without any error messages.  SVN 15188 
> built on Ubuntu Hardy 32-bit.



From sos at sokhapkin.dyndns.org  Mon Jan 25 19:34:07 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 25 Jan 2010 22:34:07 -0500
Subject: [Freeswitch-users] little hangup problem - prepaid application
In-Reply-To: <7abab2411001251713p31e542b2xf6204ff13c556d03@mail.gmail.com>
References: <4B5E0608.3070001@xpirio.com>
	<201001251620.35308.sos@sokhapkin.dyndns.org>
	<7abab2411001251713p31e542b2xf6204ff13c556d03@mail.gmail.com>
Message-ID: <201001252234.08162.sos@sokhapkin.dyndns.org>

No record is sent to the script, but the script has access to all channel 
variables.

On Monday 25 January 2010, Frank Church wrote:
> I am new to Freeswitch and I am interested in how it works. When the
> record is sent to the lua program what format is it sent in?
>
> Are the details sent like parameters on the command line like
>
> script_lua -calleridname=XXXXXX --caller_id_number=XXXXXXX
> --start_stamp=XXXXXX ...?
>
> On looking at the template as well, how is the SQL configured?
>
> <template name="sql">INSERT INTO cdr VALUES
> ("${caller_id_name}","${caller_id_number}","${destination_number}","${conte
>xt}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${bill
>sec}","${hangup_cause}","${uuid}","${bleg_uuid}",
> "${accountcode}");</template>
>
> Is it a matter of configuring with the
>
> 2010/1/25 Sergey Okhapkin <sos at sokhapkin.dyndns.org>:
> > To my experience most of "interesting" billing-related variables are
> > created not when channel is hung up, but later, when channel enters
> > REPORTING state. I use mod_cdr_csv to access these variables:
> >
> > cdr_csv.conf.xml:
> >
> > <param name="default-template" value="lua"/>
> >
> > <template name="lua">${lua(onhangup.lua)}</template>
> >
> > onhangup.lua script accesses required variables like "billsec", does
> > final calculations and writes CDR to DB.
> >
> > On Monday 25 January 2010, Christian L?schenkohl wrote:
> >> hello
> >>
> >> i try to implement a prepaid application and mod_nibblebill isn't my
> >> first choice. the implementation is quite ok so far. now my problem:
> >>
> >> - the call setup is normal (a little scripting)
> >> - the calls get ended by sched_hangup (max duration of calls)
> >> - then i would like to execute a script with api_hangup_hook
> >> ? ?in this last script i would also like to use some channel vars
> >> (a-number, call duration after answer) but i can't do that
> >>
> >> i try to use an outbound event socket script but i couldn't get the
> >> variables/information i need to calculate the amount charge
> >>
> >> any hint here
> >>
> >> br
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From sos at sokhapkin.dyndns.org  Mon Jan 25 19:41:22 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 25 Jan 2010 22:41:22 -0500
Subject: [Freeswitch-users] IAX2 Support Removed.
In-Reply-To: <0C2714A2-355A-42C7-B589-F0704D436607@freeswitch.org>
References: <0C2714A2-355A-42C7-B589-F0704D436607@freeswitch.org>
Message-ID: <201001252241.22488.sos@sokhapkin.dyndns.org>

Where is "unsupported" directory in SVN if I need to build mod_iax?

On Friday 22 January 2010, Brian West wrote:
> Due to lack of support for the libiax2 being updated to support the newer
> protocol changes and the lack of interest from anyone willing to actually
> work on it.  I have moved mod_iax to unsupported where it will stay until
> someone steps up to rewrite a new IAX2 lib.
>
> Thanks,
> Brian
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Mon Jan 25 19:58:39 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 25 Jan 2010 21:58:39 -0600
Subject: [Freeswitch-users] IAX2 Support Removed.
In-Reply-To: <201001252241.22488.sos@sokhapkin.dyndns.org>
References: <0C2714A2-355A-42C7-B589-F0704D436607@freeswitch.org>
	<201001252241.22488.sos@sokhapkin.dyndns.org>
Message-ID: <C67C1F33-9462-489A-9904-4B4B1BBCB1A6@freeswitch.org>

Not recommend it will crash if someone calls it with a newer iax lib...

http://svn.freeswitch.org/svn/unsupported/

/b

On Jan 25, 2010, at 9:41 PM, Sergey Okhapkin wrote:

> Where is "unsupported" directory in SVN if I need to build mod_iax?



From vfclists at googlemail.com  Mon Jan 25 20:34:07 2010
From: vfclists at googlemail.com (Frank Church)
Date: Tue, 26 Jan 2010 04:34:07 +0000
Subject: [Freeswitch-users] little hangup problem - prepaid application
In-Reply-To: <201001252234.08162.sos@sokhapkin.dyndns.org>
References: <4B5E0608.3070001@xpirio.com>
	<201001251620.35308.sos@sokhapkin.dyndns.org>
	<7abab2411001251713p31e542b2xf6204ff13c556d03@mail.gmail.com>
	<201001252234.08162.sos@sokhapkin.dyndns.org>
Message-ID: <7abab2411001252034kaf90e64g97b5d5c8f7d65e20@mail.gmail.com>

That is new to me, does that mean that all the languages linked in
with Freeswitch have access to the events and variables in FS at all
times?

Can you link me to the documenation that describes this part in more
detail and some examples?

2010/1/26 Sergey Okhapkin <sos at sokhapkin.dyndns.org>:
> No record is sent to the script, but the script has access to all channel
> variables.
>
> On Monday 25 January 2010, Frank Church wrote:
>> I am new to Freeswitch and I am interested in how it works. When the
>> record is sent to the lua program what format is it sent in?
>>
>> Are the details sent like parameters on the command line like
>>
>> script_lua -calleridname=XXXXXX --caller_id_number=XXXXXXX
>> --start_stamp=XXXXXX ...?
>>
>> On looking at the template as well, how is the SQL configured?
>>
>> <template name="sql">INSERT INTO cdr VALUES
>> ("${caller_id_name}","${caller_id_number}","${destination_number}","${conte
>>xt}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${bill
>>sec}","${hangup_cause}","${uuid}","${bleg_uuid}",
>> "${accountcode}");</template>
>>
>> Is it a matter of configuring with the
>>
>> 2010/1/25 Sergey Okhapkin <sos at sokhapkin.dyndns.org>:
>> > To my experience most of "interesting" billing-related variables are
>> > created not when channel is hung up, but later, when channel enters
>> > REPORTING state. I use mod_cdr_csv to access these variables:
>> >
>> > cdr_csv.conf.xml:
>> >
>> > <param name="default-template" value="lua"/>
>> >
>> > <template name="lua">${lua(onhangup.lua)}</template>
>> >
>> > onhangup.lua script accesses required variables like "billsec", does
>> > final calculations and writes CDR to DB.
>> >
>> > On Monday 25 January 2010, Christian L?schenkohl wrote:
>> >> hello
>> >>
>> >> i try to implement a prepaid application and mod_nibblebill isn't my
>> >> first choice. the implementation is quite ok so far. now my problem:
>> >>
>> >> - the call setup is normal (a little scripting)
>> >> - the calls get ended by sched_hangup (max duration of calls)
>> >> - then i would like to execute a script with api_hangup_hook
>> >> ? ?in this last script i would also like to use some channel vars
>> >> (a-number, call duration after answer) but i can't do that
>> >>
>> >> i try to use an outbound event socket script but i couldn't get the
>> >> variables/information i need to calculate the amount charge
>> >>
>> >> any hint here
>> >>
>> >> br
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Frank Church

=======================
<a href="http://devblog.brahmancreations.com">devblog.brahmancreations.com</a>


From a.afzali2003 at gmail.com  Mon Jan 25 23:51:02 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Tue, 26 Jan 2010 11:21:02 +0330
Subject: [Freeswitch-users] How to get chat message via event
In-Reply-To: <561160.39572.qm@web33504.mail.mud.yahoo.com>
References: <7b197bef1001241305o3ee9bc9cq7decdc412f90575c@mail.gmail.com>
	<561160.39572.qm@web33504.mail.mud.yahoo.com>
Message-ID: <debff5a81001252351g1befbe63m8d6c9b97dfa31cf5@mail.gmail.com>

As I see in the code, exchanging chat messages don't fire by events at least
if there is not any sip session between parties.

--afshin

On Mon, Jan 25, 2010 at 11:23 PM, Diego Toro <dftoro at yahoo.com> wrote:

> Hi, try with ESL  (libs/esl).
>
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
> --- On Sun, 1/24/10, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>
> > From: Giovanni Maruzzelli <gmaruzz at celliax.org>
> > Subject: Re: [Freeswitch-users] How to get chat message via event
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Sunday, January 24, 2010, 4:05 PM
> > Which events you don't get? From
> > which channel in which circumstances?
> > (I mean what you do and what do you expect?)
> >
> > -giovanni
> >
> > On Sun, Jan 24, 2010 at 9:59 PM, afshin afzali <a.afzali2003 at gmail.com>
> > wrote:
> > > Hi,
> > >
> > > As you say, I've already done and unfortunately did
> > not get the message
> > > events although other events are fired as expected :(
> > >
> > > -- afshin
> > >
> > > On Sun, Jan 24, 2010 at 7:18 PM, Giovanni Maruzzelli
> > <gmaruzz at celliax.org>
> > > wrote:
> > >>
> > >> you subscribe to them as MESSAGE events
> > >>
> > >> eg, from a telnet session:
> > >>
> > >> telnet localhost 8021
> > >> auth ClueCon
> > >> events plain message
> > >>
> > >> then those events will show up in your telnet
> > session.
> > >> -gm
> > >>
> > >> On Sun, Jan 24, 2010 at 4:39 PM, afshin afzali
> > <a.afzali2003 at gmail.com>
> > >> wrote:
> > >> > Hi,
> > >> >
> > >> > It seems that the chat messages don't fire
> > via events by default and
> > >> > just
> > >> > exchange between parties.
> > >> > Is it true? Is it possible to enable those
> > via events?
> > >> >
> > >> > appreciate all,
> > >> > -- afshin
> > >> >
> > >> >
> > _______________________________________________
> > >> > FreeSWITCH-users mailing list
> > >> > FreeSWITCH-users at lists.freeswitch.org
> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> > http://www.freeswitch.org
> > >> >
> > >> >
> > >>
> > >>
> > >>
> > >> --
> > >> Sincerely,
> > >>
> > >> Giovanni Maruzzelli
> > >> Cell : +39-347-2665618
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> >
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.villasmil.work at gmail.com  Tue Jan 26 00:14:10 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 26 Jan 2010 09:14:10 +0100
Subject: [Freeswitch-users] Sending CDRs to both file and SQL?
In-Reply-To: <4B5E1BB0.2020104@gmail.com>
References: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com>
	<191c3a031001250756y9ea5211hf01d2839cbf9bd74@mail.gmail.com>
	<c29b930b1001250920g5999d707i1caa38298795cbae@mail.gmail.com>
	<d361ca7b1001251011q32120d50l5f14a98f33fcee29@mail.gmail.com>
	<5d3e0dc61001251022h6c98722ag8b59fb12ce2d8ee8@mail.gmail.com>
	<4B5E1BB0.2020104@gmail.com>
Message-ID: <9853f4ff1001260014q7f68206ek2783b509b1e9dc5a@mail.gmail.com>

xml_cdr is perfect for that, why not use it?

On Mon, Jan 25, 2010 at 11:31 PM, Nazim Agabekov
<nazim.agabekov at gmail.com> wrote:
> I have another implementation based on fastcgi and libxml2. Source could
> be easily modified to log into the file as well.
>
> http://blog.buta-tech.com/?p=1
>
> On 01/25/2010 10:22 PM, Lon Baker wrote:
>> For what its worth, I wrote a simple Sinatra-based web service to
>> accept the CDRs over HTTP. It then queues them for processing into a
>> database, syslog and/or file.
>>
>> Its easy to load balance and lets me throttle CDR processing so I
>> don't put excessive load on the database server at peak times.
>>
>> Lon
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From irmatov at gmail.com  Tue Jan 26 01:37:18 2010
From: irmatov at gmail.com (Timur Irmatov)
Date: Tue, 26 Jan 2010 14:37:18 +0500
Subject: [Freeswitch-users] mod_erlang_event: disconnects
In-Reply-To: <20100122154658.GC25693@hijacked.us>
References: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>
	<20100121134241.GD1036@hijacked.us>
	<241d382f1001220522g27f67ef5p5ba86ebb3afe63c@mail.gmail.com>
	<20100122154658.GC25693@hijacked.us>
Message-ID: <241d382f1001260137o535b9e3boe479160378a4a747@mail.gmail.com>

Hi, Andrew!


On Fri, Jan 22, 2010 at 8:46 PM, Andrew Thompson <andrew at hijacked.us> wrote:
> Give this patch a shot:
>
> http://eagle.bsd.st/~andrew/erlang_session_fix.diff
>
> And see if it makes a difference.

24 hours passed since I have installed this patch. Seems to be working
fine - I haven't seen a single disconnect between FreeSWITCH and my
application.

Thank you very much! I owe you a beer, if you ever to visit Tashkent,
Uzbekistan.. :-)


-- 
Timur Irmatov, xmpp:irmatov at jabber.ru


From michal.zubac at comgate.cz  Tue Jan 26 02:59:12 2010
From: michal.zubac at comgate.cz (=?UTF-8?B?TWljaGFsIFp1YsOhxI0=?=)
Date: Tue, 26 Jan 2010 11:59:12 +0100
Subject: [Freeswitch-users] sangoma_prid & freeswitch openzap - conflict
In-Reply-To: <6B96C690-1E6C-4C9E-9B5F-B23DA1354B48@freeswitch.org>
References: <4B5D8DFE.30904@comgate.cz>
	<6B96C690-1E6C-4C9E-9B5F-B23DA1354B48@freeswitch.org>
Message-ID: <4B5ECB00.2070800@comgate.cz>

I've finally figured it out.
I have to use sangoma_prid and (ozmod_sangoma_boost) <boost_spans>
configuration in openzap.conf.xml.

Now  we have problems with sangoma_prid version (I assume), because
Freeswitch logs following messages:
[CRIT] sangoma_boost_client.c:356 Invalid Boost Version 100  Expecting 101
[CRIT] sangoma_boost_client.c:370 Critical Error: PQ Invalid Event
lenght from boost rxlen=23 evsz=1031

Our versions:
sangoma_prid: = Sangoma PRI Protocol Stack Daemon       =
virtual sangoma_prid: = Version: 1.25                           =
virtual sangoma_prid: = Date: Dec 22 2009                       =
virtual sangoma_prid: = Wanpipe Release: wanpipe-3.5.8.11          =
FreeSwitch from SVN trunk (r16509)

I assume, we have to wait for new drivers from Sangoma according to post
<mike at jerris.com> Re: [Freeswitch-users] Need Help to setup freeswitch
with sangoma card [Sun, 24 Jan 2010 05:13:32 -0500]

Mighq

Brian West napsal(a):
> If you are using PRID you do not configure D channels at all.  Sangoma PRID will use those already.
>
> /b
>
> On Jan 25, 2010, at 6:26 AM, Michal Zub?? wrote:
>
>   
>> Hi.
>>
>> I'm just curious. Is sangoma_prid neccessary for Freeswitch to work with
>> E1 (PRI) line? (wanpipe & openzap mode)
>> I stopped sangoma_prid because, when I try to start Freeswitch, openzap
>> yells that it cannot open D-channel (/dev/wanpipe1_if16). It is already
>> used by sangoma_prid.
>>
>> But PRI calls are behaving strangely for me. Maybe this is the cause.
>> How can I resolve this conflict?
>>
>> Thanks for advice. It's possible, that I am doing some newbie mistake.
>>
>> Michal Zubac
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From nazim.agabekov at gmail.com  Tue Jan 26 03:12:39 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Tue, 26 Jan 2010 15:12:39 +0400
Subject: [Freeswitch-users] Sending CDRs to both file and SQL?
In-Reply-To: <9853f4ff1001260014q7f68206ek2783b509b1e9dc5a@mail.gmail.com>
References: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com>	<191c3a031001250756y9ea5211hf01d2839cbf9bd74@mail.gmail.com>	<c29b930b1001250920g5999d707i1caa38298795cbae@mail.gmail.com>	<d361ca7b1001251011q32120d50l5f14a98f33fcee29@mail.gmail.com>	<5d3e0dc61001251022h6c98722ag8b59fb12ce2d8ee8@mail.gmail.com>	<4B5E1BB0.2020104@gmail.com>
	<9853f4ff1001260014q7f68206ek2783b509b1e9dc5a@mail.gmail.com>
Message-ID: <4B5ECE27.5090603@gmail.com>

We do use it for cdr generation, for parsing every one has it's own recipe.
In my opinion php is an easiest way to parse:
http://www.0xdecafbad.com/?p=28
Personally, I prefer fastcgi and C.

On 01/26/2010 12:14 PM, David Villasmil wrote:
> xml_cdr is perfect for that, why not use it?
>
> On Mon, Jan 25, 2010 at 11:31 PM, Nazim Agabekov
> <nazim.agabekov at gmail.com> wrote:
>   
>> I have another implementation based on fastcgi and libxml2. Source could
>> be easily modified to log into the file as well.
>>
>> http://blog.buta-tech.com/?p=1
>>
>> On 01/25/2010 10:22 PM, Lon Baker wrote:
>>     
>>> For what its worth, I wrote a simple Sinatra-based web service to
>>> accept the CDRs over HTTP. It then queues them for processing into a
>>> database, syslog and/or file.
>>>
>>> Its easy to load balance and lets me throttle CDR processing so I
>>> don't put excessive load on the database server at peak times.
>>>
>>> Lon
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From null at invalid.name  Tue Jan 26 06:14:25 2010
From: null at invalid.name (Dan Lane)
Date: Tue, 26 Jan 2010 14:14:25 +0000
Subject: [Freeswitch-users] Conference talk detection
Message-ID: <b042b23b1001260614ie185801h6fe871607afe6615@mail.gmail.com>

Hi,

I have a conference bridge with a simple web interface that shows who
is talking using the event api. Unfortunately the users don't like the
way the first few ms of speech after silence is cut off. Presumably
this is an unavoidable side-effect of cutting out audio when someone
isn't speaking.

Is there any way to have a conference where all audio is sent to the
conference but start/stop talking events are still generated?
Currently setting energy-level to 0 disables start/stop talking events
:(

Cheers,
Dan


From moises.silva at gmail.com  Tue Jan 26 06:18:17 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 26 Jan 2010 09:18:17 -0500
Subject: [Freeswitch-users] sangoma_prid & freeswitch openzap - conflict
In-Reply-To: <4B5ECB00.2070800@comgate.cz>
References: <4B5D8DFE.30904@comgate.cz>
	<6B96C690-1E6C-4C9E-9B5F-B23DA1354B48@freeswitch.org>
	<4B5ECB00.2070800@comgate.cz>
Message-ID: <c4d05cbe1001260618o11516ecax1642fd8510184b9@mail.gmail.com>

On Tue, Jan 26, 2010 at 5:59 AM, Michal Zub?? <michal.zubac at comgate.cz>wrote:

> Now  we have problems with sangoma_prid version (I assume), because
> Freeswitch logs following messages:
> [CRIT] sangoma_boost_client.c:356 Invalid Boost Version 100  Expecting 101
> [CRIT] sangoma_boost_client.c:370 Critical Error: PQ Invalid Event
> lenght from boost rxlen=23 evsz=1031
>
> I assume, we have to wait for new drivers from Sangoma according to post
> <mike at jerris.com> Re: [Freeswitch-users] Need Help to setup freeswitch
> with sangoma card [Sun, 24 Jan 2010 05:13:32 -0500]
>
> Mighq
>

That is correct. We apologize for the inconvenience. The waiting shouldn't
be long.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From rob4manhere at gmail.com  Tue Jan 26 06:30:18 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 26 Jan 2010 08:30:18 -0600
Subject: [Freeswitch-users] Conference talk detection
In-Reply-To: <b042b23b1001260614ie185801h6fe871607afe6615@mail.gmail.com>
References: <b042b23b1001260614ie185801h6fe871607afe6615@mail.gmail.com>
Message-ID: <2C9414EA-0F35-47AA-8B9D-754B693B5D89@gmail.com>

Hi Dan,

What happens when you set the energy-level to something small, such as  
10?

Rob

On Jan 26, 2010, at 8:14 AM, Dan Lane wrote:

> Hi,
>
> I have a conference bridge with a simple web interface that shows who
> is talking using the event api. Unfortunately the users don't like the
> way the first few ms of speech after silence is cut off. Presumably
> this is an unavoidable side-effect of cutting out audio when someone
> isn't speaking.
>
> Is there any way to have a conference where all audio is sent to the
> conference but start/stop talking events are still generated?
> Currently setting energy-level to 0 disables start/stop talking events
> :(
>
> Cheers,
> Dan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From null at invalid.name  Tue Jan 26 07:03:23 2010
From: null at invalid.name (Dan Lane)
Date: Tue, 26 Jan 2010 15:03:23 +0000
Subject: [Freeswitch-users] Conference talk detection
In-Reply-To: <2C9414EA-0F35-47AA-8B9D-754B693B5D89@gmail.com>
References: <b042b23b1001260614ie185801h6fe871607afe6615@mail.gmail.com>
	<2C9414EA-0F35-47AA-8B9D-754B693B5D89@gmail.com>
Message-ID: <b042b23b1001260703r42744b64o8c2196562777b05b@mail.gmail.com>

The same thing, with the energy level set to anything other than 0 the
first few ms of audio is not sent to the conference. Presumably this
is by design as this is the amount of time noise has to be made for
before it's detected as speech and audio is sent to the conference;
including the audio that wasn't sent to the conference would introduce
delay.

What I'd like to be able to do is generate start/stop talk events when
people are talking but without starting or stopping the audio stream.


On Tue, Jan 26, 2010 at 2:30 PM, Rob Forman <rob4manhere at gmail.com> wrote:
> Hi Dan,
>
> What happens when you set the energy-level to something small, such as
> 10?
>
> Rob
>
> On Jan 26, 2010, at 8:14 AM, Dan Lane wrote:
>
>> Hi,
>>
>> I have a conference bridge with a simple web interface that shows who
>> is talking using the event api. Unfortunately the users don't like the
>> way the first few ms of speech after silence is cut off. Presumably
>> this is an unavoidable side-effect of cutting out audio when someone
>> isn't speaking.
>>
>> Is there any way to have a conference where all audio is sent to the
>> conference but start/stop talking events are still generated?
>> Currently setting energy-level to 0 disables start/stop talking events
>> :(
>>
>> Cheers,
>> Dan
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From fvillarroel at yahoo.com  Tue Jan 26 07:15:58 2010
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Tue, 26 Jan 2010 07:15:58 -0800 (PST)
Subject: [Freeswitch-users] default-template
Message-ID: <590217.53161.qm@web34308.mail.mud.yahoo.com>

Dear All.

It?s possible define the default-template in each external gateways, like this:

<include>
    <gateway name="foo">
      <param name="username" value="foo"/>
      <param name="password" value="foo"/>
      <param name="realm" value="190.XXX.XX.XXX"/>
      <param name="register" value="false"/>
      <param name="context" value="public"/>
      <param name="default-template" value="sql1"/>
    </gateway>
</include>

<include>
    <gateway name="foo1">
      <param name="username" value="foo1"/>
      <param name="password" value="foo1"/>
      <param name="realm" value="190.XXX.XX.YYY"/>
      <param name="register" value="false"/>
      <param name="context" value="public"/>
      <param name="default-template" value="sql2"/>
    </gateway>
</include>



Regards


      


From anthony.minessale at gmail.com  Tue Jan 26 07:22:29 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Jan 2010 09:22:29 -0600
Subject: [Freeswitch-users] Conference talk detection
In-Reply-To: <b042b23b1001260703r42744b64o8c2196562777b05b@mail.gmail.com>
References: <b042b23b1001260614ie185801h6fe871607afe6615@mail.gmail.com>
	<2C9414EA-0F35-47AA-8B9D-754B693B5D89@gmail.com>
	<b042b23b1001260703r42744b64o8c2196562777b05b@mail.gmail.com>
Message-ID: <191c3a031001260722u25949cabj973e4c8ced4eb7a8@mail.gmail.com>

We use a conference with energy detection on 12+ hours a day and I don't
recall losing any of the audio.
The instant you breach the level that same packet is sent, and you have to
have several consecutive packets below the level to stop.

My guess is you are talking to something lame like a Sonus who is resetting
its jitterbuffer when you start talking again.

try editing your profile and adding this param

<param name="conference-flags" value="waste-bandwidth"/>

This will make the conference send rtp to the other side of the call even
when you are not talking which typically soothes unruly RTP devices.




On Tue, Jan 26, 2010 at 9:03 AM, Dan Lane <null at invalid.name> wrote:

> The same thing, with the energy level set to anything other than 0 the
> first few ms of audio is not sent to the conference. Presumably this
> is by design as this is the amount of time noise has to be made for
> before it's detected as speech and audio is sent to the conference;
> including the audio that wasn't sent to the conference would introduce
> delay.
>
> What I'd like to be able to do is generate start/stop talk events when
> people are talking but without starting or stopping the audio stream.
>
>
> On Tue, Jan 26, 2010 at 2:30 PM, Rob Forman <rob4manhere at gmail.com> wrote:
> > Hi Dan,
> >
> > What happens when you set the energy-level to something small, such as
> > 10?
> >
> > Rob
> >
> > On Jan 26, 2010, at 8:14 AM, Dan Lane wrote:
> >
> >> Hi,
> >>
> >> I have a conference bridge with a simple web interface that shows who
> >> is talking using the event api. Unfortunately the users don't like the
> >> way the first few ms of speech after silence is cut off. Presumably
> >> this is an unavoidable side-effect of cutting out audio when someone
> >> isn't speaking.
> >>
> >> Is there any way to have a conference where all audio is sent to the
> >> conference but start/stop talking events are still generated?
> >> Currently setting energy-level to 0 disables start/stop talking events
> >> :(
> >>
> >> Cheers,
> >> Dan
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From zolotov at altron.ua  Tue Jan 26 07:28:12 2010
From: zolotov at altron.ua (Evgeniy Zolotov)
Date: Tue, 26 Jan 2010 17:28:12 +0200
Subject: [Freeswitch-users] Conference talk detection
In-Reply-To: <b042b23b1001260703r42744b64o8c2196562777b05b@mail.gmail.com>
References: <b042b23b1001260614ie185801h6fe871607afe6615@mail.gmail.com>	<2C9414EA-0F35-47AA-8B9D-754B693B5D89@gmail.com>
	<b042b23b1001260703r42744b64o8c2196562777b05b@mail.gmail.com>
Message-ID: <4B5F0A0C.2060808@altron.ua>

  Try to set flag 'waste' on conference profile.

Dan Lane ?????:
> The same thing, with the energy level set to anything other than 0 the
> first few ms of audio is not sent to the conference. Presumably this
> is by design as this is the amount of time noise has to be made for
> before it's detected as speech and audio is sent to the conference;
> including the audio that wasn't sent to the conference would introduce
> delay.
>
> What I'd like to be able to do is generate start/stop talk events when
> people are talking but without starting or stopping the audio stream.
>
>
> On Tue, Jan 26, 2010 at 2:30 PM, Rob Forman <rob4manhere at gmail.com> wrote:
>   
>> Hi Dan,
>>
>> What happens when you set the energy-level to something small, such as
>> 10?
>>
>> Rob
>>
>> On Jan 26, 2010, at 8:14 AM, Dan Lane wrote:
>>
>>     
>>> Hi,
>>>
>>> I have a conference bridge with a simple web interface that shows who
>>> is talking using the event api. Unfortunately the users don't like the
>>> way the first few ms of speech after silence is cut off. Presumably
>>> this is an unavoidable side-effect of cutting out audio when someone
>>> isn't speaking.
>>>
>>> Is there any way to have a conference where all audio is sent to the
>>> conference but start/stop talking events are still generated?
>>> Currently setting energy-level to 0 disables start/stop talking events
>>> :(
>>>
>>> Cheers,
>>> Dan
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From andrew at hijacked.us  Tue Jan 26 07:28:49 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Tue, 26 Jan 2010 10:28:49 -0500
Subject: [Freeswitch-users] mod_erlang_event: disconnects
In-Reply-To: <241d382f1001260137o535b9e3boe479160378a4a747@mail.gmail.com>
References: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>
	<20100121134241.GD1036@hijacked.us>
	<241d382f1001220522g27f67ef5p5ba86ebb3afe63c@mail.gmail.com>
	<20100122154658.GC25693@hijacked.us>
	<241d382f1001260137o535b9e3boe479160378a4a747@mail.gmail.com>
Message-ID: <20100126152849.GG6569@hijacked.us>

On Tue, Jan 26, 2010 at 02:37:18PM +0500, Timur Irmatov wrote:
> 24 hours passed since I have installed this patch. Seems to be working
> fine - I haven't seen a single disconnect between FreeSWITCH and my
> application.
> 
Okay, I'll commit it to trunk then. Thanks for the report.

> Thank you very much! I owe you a beer, if you ever to visit Tashkent,
> Uzbekistan.. :-)
> 
I'll keep that in mind :)

Andrew


From anthony.minessale at gmail.com  Tue Jan 26 07:30:15 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Jan 2010 09:30:15 -0600
Subject: [Freeswitch-users] How to get chat message via event
In-Reply-To: <debff5a81001252351g1befbe63m8d6c9b97dfa31cf5@mail.gmail.com>
References: <7b197bef1001241305o3ee9bc9cq7decdc412f90575c@mail.gmail.com>
	<561160.39572.qm@web33504.mail.mud.yahoo.com>
	<debff5a81001252351g1befbe63m8d6c9b97dfa31cf5@mail.gmail.com>
Message-ID: <191c3a031001260730o7450b95ene02e1b56ccd182b0@mail.gmail.com>

if you are directly controlling a session "myevents <session>" or it was an
outbound socket call
you can send the command "divert_events" to get the events on event_socket
if you are in an embedded script you can get them from the input callback

On Tue, Jan 26, 2010 at 1:51 AM, afshin afzali <a.afzali2003 at gmail.com>wrote:

> As I see in the code, exchanging chat messages don't fire by events at
> least if there is not any sip session between parties.
>
> --afshin
>
> On Mon, Jan 25, 2010 at 11:23 PM, Diego Toro <dftoro at yahoo.com> wrote:
>
>> Hi, try with ESL  (libs/esl).
>>
>>
>> Diego Toro
>> http://lacarretade.blogspot.com/
>>
>>
>> --- On Sun, 1/24/10, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>
>> > From: Giovanni Maruzzelli <gmaruzz at celliax.org>
>> > Subject: Re: [Freeswitch-users] How to get chat message via event
>> > To: freeswitch-users at lists.freeswitch.org
>> > Date: Sunday, January 24, 2010, 4:05 PM
>> > Which events you don't get? From
>> > which channel in which circumstances?
>> > (I mean what you do and what do you expect?)
>> >
>> > -giovanni
>> >
>> > On Sun, Jan 24, 2010 at 9:59 PM, afshin afzali <a.afzali2003 at gmail.com>
>> > wrote:
>> > > Hi,
>> > >
>> > > As you say, I've already done and unfortunately did
>> > not get the message
>> > > events although other events are fired as expected :(
>> > >
>> > > -- afshin
>> > >
>> > > On Sun, Jan 24, 2010 at 7:18 PM, Giovanni Maruzzelli
>> > <gmaruzz at celliax.org>
>> > > wrote:
>> > >>
>> > >> you subscribe to them as MESSAGE events
>> > >>
>> > >> eg, from a telnet session:
>> > >>
>> > >> telnet localhost 8021
>> > >> auth ClueCon
>> > >> events plain message
>> > >>
>> > >> then those events will show up in your telnet
>> > session.
>> > >> -gm
>> > >>
>> > >> On Sun, Jan 24, 2010 at 4:39 PM, afshin afzali
>> > <a.afzali2003 at gmail.com>
>> > >> wrote:
>> > >> > Hi,
>> > >> >
>> > >> > It seems that the chat messages don't fire
>> > via events by default and
>> > >> > just
>> > >> > exchange between parties.
>> > >> > Is it true? Is it possible to enable those
>> > via events?
>> > >> >
>> > >> > appreciate all,
>> > >> > -- afshin
>> > >> >
>> > >> >
>> > _______________________________________________
>> > >> > FreeSWITCH-users mailing list
>> > >> > FreeSWITCH-users at lists.freeswitch.org
>> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > >> > http://www.freeswitch.org
>> > >> >
>> > >> >
>> > >>
>> > >>
>> > >>
>> > >> --
>> > >> Sincerely,
>> > >>
>> > >> Giovanni Maruzzelli
>> > >> Cell : +39-347-2665618
>> > >>
>> > >> _______________________________________________
>> > >> FreeSWITCH-users mailing list
>> > >> FreeSWITCH-users at lists.freeswitch.org
>> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > >> http://www.freeswitch.org
>> > >
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> > >
>> >
>> >
>> >
>> > --
>> > Sincerely,
>> >
>> > Giovanni Maruzzelli
>> > Cell : +39-347-2665618
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From freeswitch-users at digitaldan.com  Tue Jan 26 07:34:39 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Tue, 26 Jan 2010 08:34:39 -0700 (MST)
Subject: [Freeswitch-users] Lua: io.popen/read blocking
	in	other	incoming calls
In-Reply-To: <15818864.1.1264519951723.JavaMail.root@zimbra>
Message-ID: <10147006.4.1264520079191.JavaMail.root@zimbra>

Thanks, I'm using os.execute now and piping the output to a temp file. It's not the most elagant solution, but it works. Call volume is light so I don't think there will be any scalability issues. I'll take a look at the sites you mention and see if I can find something a little bit less hacky. 

Thanks again. 
Dan- 

----- Original Message ----- 
From: "Fernando Gregianin Testa" <testa at voicetechnology.com.br> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Sunday, January 24, 2010 12:21:51 PM 
Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls 

You may consider use lua socket.http package as an alternative to popen+wget. Check: 
https://web.tecgraf.puc-rio.br/luasocket/ 
http://www.tecgraf.puc-rio.br/~diego/professional/luasocket/http.html 


Maybe you can be interested also in http://github.com/fertesta/restinlua 




Em 19/01/2010, ?s 21:17, Dan escreveu: 




I would, but I need to post a a wav file that gets recorded, I didn't see a way to supply the location of a file to use as the post data. It looks like you have to url encode the data in the script and pass it all in the call. 


Thanks 
Dan- 
----- Original Message ----- 
From: "Rupa Schomaker" < rupa at rupa.com > 
To: "freeswitch-users" < freeswitch-users at lists.freeswitch.org > 
Sent: Tuesday, January 19, 2010 4:06:36 PM 
Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls 




On Tue, Jan 19, 2010 at 3:03 PM, Dan < freeswitch-users at digitaldan.com > wrote: 




My lua script is calling wget through lua's io.popen to send and receive data from a web service. While the f:read to wget is running, other incoming calls will block on the same io.popen call until the first call closes the pipe (with f:close()). 


You might want to look at the api that mod_curl exposes to do what you want. No need to do an expensive system call just to call a webservice. 
-- 
-Rupa 

_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 
_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 



Fernando Gregianin Testa 
testa at voicetechnology.com.br 
+55 11 35882166 




_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 
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From dftoro at yahoo.com  Tue Jan 26 07:55:35 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 26 Jan 2010 07:55:35 -0800 (PST)
Subject: [Freeswitch-users] compilation error on fs_cli (Windows) (latest
	version)
Message-ID: <571042.39380.qm@web33502.mail.mud.yahoo.com>

Hi, I have compilation error "error C2220" on fs_cli project on Windows using VS2008.

FS: latest version (2010/01/26)
VS: VS2008
SO: Windows 7

VS2008 Error log:

Error	1	error C2220: warning treated as error - no 'object' file generated	g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c	106	fs_cli

Warning	2	warning C6385: Invalid data: accessing 'global_profile->console_fnkeys', the readable size is '48' bytes, but '-4' bytes might be read: Lines: 86, 88, 90	g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c	90	fs_cli

Warning	3	warning C6246: Local declaration of 'p' hides declaration of the same name in outer scope. For additional information, see previous declaration at line '844' of 'g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c': Lines: 844	g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c	895	fs_cli

Warning	4	warning C6011: Dereferencing NULL pointer 'cursor': Lines: 839, 840, 841, 842, 843, 844, 846, 849, 853, 857, 868, 870, 871, 884	g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c	884	fs_cli


Thank you

Diego Toro
http://lacarretade.blogspot.com/


      


From dftoro at yahoo.com  Tue Jan 26 07:58:23 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 26 Jan 2010 07:58:23 -0800 (PST)
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <191c3a031001250920y5e622c0ewa7a80b9fe5799388@mail.gmail.com>
Message-ID: <99859.15231.qm@web33505.mail.mud.yahoo.com>

Hi, using \\\\ the is changed also when there is a match with an escape character (\s,\n...)

Thank you

Diego Toro
http://lacarretade.blogspot.com/


--- On Mon, 1/25/10, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] mutiple playback files (unescape_char) Windows
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, January 25, 2010, 12:20 PM
> its possible your string hits the parser
> more than once.
> try using 4 \ 
> 
> \\\\sound
> 
> 
> On Sun, Jan 24, 2010 at 4:03 AM,
> Michael Jerris <mike at jerris.com>
> wrote:
> 
> As noted on that bug, you should be
> able to either use \\ or / for the path separator
> there and it should work.
> 
> 
> 
> 
> Mike
> 
> 
> 
> On Jan 22, 2010, at 9:18 AM, Diego Toro wrote:
> 
> 
> 
> > Hi, with svn r16440 the problem persists, I creted a
> jira report http://jira.freeswitch.org/browse/LBSNDF-8
> this is a minor issue, but activing playback delimiter no
> audio file can be played. On FS the audio files are placed
> in the \sound\ directory, building the path on
> Windows would be \sound '\s' which is
> replaced by 'ound'.
> 
> 
> >
> 
> > Thank you
> 
> >
> 
> > Diego Toro
> 
> > http://lacarretade.blogspot.com/
> 
> >
> 
> 
> 
> 
> 
> _______________________________________________
> 
> FreeSWITCH-users mailing list
> 
> FreeSWITCH-users at lists.freeswitch.org
> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> 
> IRC: irc.freenode.net
> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> 
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From anthony.minessale at gmail.com  Tue Jan 26 08:27:12 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Jan 2010 10:27:12 -0600
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <99859.15231.qm@web33505.mail.mud.yahoo.com>
References: <191c3a031001250920y5e622c0ewa7a80b9fe5799388@mail.gmail.com>
	<99859.15231.qm@web33505.mail.mud.yahoo.com>
Message-ID: <191c3a031001260827j544e1414oaec56f527e116eea@mail.gmail.com>

I didn't understand that

On Tue, Jan 26, 2010 at 9:58 AM, Diego Toro <dftoro at yahoo.com> wrote:

> Hi, using \\\\ the is changed also when there is a match with an escape
> character (\s,\n...)
>
> Thank you
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
> --- On Mon, 1/25/10, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>
> > From: Anthony Minessale <anthony.minessale at gmail.com>
> > Subject: Re: [Freeswitch-users] mutiple playback files (unescape_char)
> Windows
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Monday, January 25, 2010, 12:20 PM
> > its possible your string hits the parser
> > more than once.
> > try using 4 \
> >
> > \\\\sound
> >
> >
> > On Sun, Jan 24, 2010 at 4:03 AM,
> > Michael Jerris <mike at jerris.com>
> > wrote:
> >
> > As noted on that bug, you should be
> > able to either use \\ or / for the path separator
> > there and it should work.
> >
> >
> >
> >
> > Mike
> >
> >
> >
> > On Jan 22, 2010, at 9:18 AM, Diego Toro wrote:
> >
> >
> >
> > > Hi, with svn r16440 the problem persists, I creted a
> > jira report http://jira.freeswitch.org/browse/LBSNDF-8
> > this is a minor issue, but activing playback delimiter no
> > audio file can be played. On FS the audio files are placed
> > in the \sound\ directory, building the path on
> > Windows would be \sound '\s' which is
> > replaced by 'ound'.
> >
> >
> > >
> >
> > > Thank you
> >
> > >
> >
> > > Diego Toro
> >
> > > http://lacarretade.blogspot.com/
> >
> > >
> >
> >
> >
> >
> >
> > _______________________________________________
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net
> > #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> >
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> >
> > -----Inline Attachment Follows-----
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From jeff at jefflenk.com  Tue Jan 26 08:42:54 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 26 Jan 2010 10:42:54 -0600
Subject: [Freeswitch-users] compilation error on fs_cli (Windows)
 (latest	version)
In-Reply-To: <571042.39380.qm@web33502.mail.mud.yahoo.com>
References: <571042.39380.qm@web33502.mail.mud.yahoo.com>
Message-ID: <COL117-W44C42225C0FBBE6BAF2813B95E0@phx.gbl>


I dont see this - do a rebuild all. Are there more errors before these?


 
> Date: Tue, 26 Jan 2010 07:55:35 -0800
> From: dftoro at yahoo.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] compilation error on fs_cli (Windows) (latest version)
> 
> Hi, I have compilation error "error C2220" on fs_cli project on Windows using VS2008.
> 
> FS: latest version (2010/01/26)
> VS: VS2008
> SO: Windows 7
> 
> VS2008 Error log:
> 
> Error 1 error C2220: warning treated as error - no 'object' file generated g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c 106 fs_cli
> 
> Warning 2 warning C6385: Invalid data: accessing 'global_profile->console_fnkeys', the readable size is '48' bytes, but '-4' bytes might be read: Lines: 86, 88, 90 g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c 90 fs_cli
> 
> Warning 3 warning C6246: Local declaration of 'p' hides declaration of the same name in outer scope. For additional information, see previous declaration at line '844' of 'g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c': Lines: 844 g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c 895 fs_cli
> 
> Warning 4 warning C6011: Dereferencing NULL pointer 'cursor': Lines: 839, 840, 841, 842, 843, 844, 846, 849, 853, 857, 868, 870, 871, 884 g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c 884 fs_cli
> 
> 
> Thank you
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
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From mrene_lists at avgs.ca  Tue Jan 26 08:45:49 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 26 Jan 2010 11:45:49 -0500
Subject: [Freeswitch-users] compilation error on fs_cli (Windows)
	(latest version)
In-Reply-To: <571042.39380.qm@web33502.mail.mud.yahoo.com>
References: <571042.39380.qm@web33502.mail.mud.yahoo.com>
Message-ID: <3072AE90-DDEA-4FD0-9B4B-140051730073@avgs.ca>

Looks like the code analyzer is running, this is normally turned off  
when you do a normal build, turn it off and try again.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 26-Jan-10, at 10:55 AM, Diego Toro wrote:

> Hi, I have compilation error "error C2220" on fs_cli project on  
> Windows using VS2008.
>
> FS: latest version (2010/01/26)
> VS: VS2008
> SO: Windows 7
>
> VS2008 Error log:
>
> Error	1	error C2220: warning treated as error - no 'object' file  
> generated	g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl 
> \fs_cli.c	106	fs_cli
>
> Warning	2	warning C6385: Invalid data: accessing 'global_profile- 
> >console_fnkeys', the readable size is '48' bytes, but '-4' bytes  
> might be read: Lines: 86, 88, 90	g:\ftp\incoming\fs 
> \freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c	90	fs_cli
>
> Warning	3	warning C6246: Local declaration of 'p' hides declaration  
> of the same name in outer scope. For additional information, see  
> previous declaration at line '844' of 'g:\ftp\incoming\fs 
> \freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c': Lines: 844	g:\ftp 
> \incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c	895	 
> fs_cli
>
> Warning	4	warning C6011: Dereferencing NULL pointer 'cursor': Lines:  
> 839, 840, 841, 842, 843, 844, 846, 849, 853, 857, 868, 870, 871, 884	 
> g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c	 
> 884	fs_cli
>
>
> Thank you
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From null at invalid.name  Tue Jan 26 08:56:48 2010
From: null at invalid.name (Dan Lane)
Date: Tue, 26 Jan 2010 16:56:48 +0000
Subject: [Freeswitch-users] Conference talk detection
In-Reply-To: <191c3a031001260722u25949cabj973e4c8ced4eb7a8@mail.gmail.com>
References: <b042b23b1001260614ie185801h6fe871607afe6615@mail.gmail.com>
	<2C9414EA-0F35-47AA-8B9D-754B693B5D89@gmail.com>
	<b042b23b1001260703r42744b64o8c2196562777b05b@mail.gmail.com>
	<191c3a031001260722u25949cabj973e4c8ced4eb7a8@mail.gmail.com>
Message-ID: <b042b23b1001260856x21a4c938v619cfe6eb7255561@mail.gmail.com>

On Tue, Jan 26, 2010 at 3:22 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> We use a conference with energy detection on 12+ hours a day and I don't
> recall losing any of the audio.
> The instant you breach the level that same packet is sent, and you have to
> have several consecutive packets below the level to stop.

Interesting could you post the conferences.xml config for that
particular conference? I have yet to find a way of using energy
detection that doesn't miss the first few ms of sound, for example the
word "Testing" often comes out as "esting" perhaps with the tail end
of the T coming across.

It's not a problem when everyone is a native english speaker but when
on a conference containing people with a poor grasp of english or
people who reply with lots of quick utterances such as "yup"

> My guess is you are talking to something lame like a Sonus who is resetting
> its jitterbuffer when you start talking again.

In production it's hitting Cisco gateways but I've experienced this in
a lab environment consisting of a mixture of Snom 360, 820 and Polycom
IP6000 devices.

What would be great is to have start/stop talk events while still
always sending audio from participants to the conference


From anthony.minessale at gmail.com  Tue Jan 26 09:53:55 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Jan 2010 11:53:55 -0600
Subject: [Freeswitch-users] Conference talk detection
In-Reply-To: <b042b23b1001260856x21a4c938v619cfe6eb7255561@mail.gmail.com>
References: <b042b23b1001260614ie185801h6fe871607afe6615@mail.gmail.com>
	<2C9414EA-0F35-47AA-8B9D-754B693B5D89@gmail.com>
	<b042b23b1001260703r42744b64o8c2196562777b05b@mail.gmail.com>
	<191c3a031001260722u25949cabj973e4c8ced4eb7a8@mail.gmail.com>
	<b042b23b1001260856x21a4c938v619cfe6eb7255561@mail.gmail.com>
Message-ID: <191c3a031001260953p4846a541m74da287d1d33897d@mail.gmail.com>

try the param i mentioned and gave the specific example for.


On Tue, Jan 26, 2010 at 10:56 AM, Dan Lane <null at invalid.name> wrote:

> On Tue, Jan 26, 2010 at 3:22 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > We use a conference with energy detection on 12+ hours a day and I don't
> > recall losing any of the audio.
> > The instant you breach the level that same packet is sent, and you have
> to
> > have several consecutive packets below the level to stop.
>
> Interesting could you post the conferences.xml config for that
> particular conference? I have yet to find a way of using energy
> detection that doesn't miss the first few ms of sound, for example the
> word "Testing" often comes out as "esting" perhaps with the tail end
> of the T coming across.
>
> It's not a problem when everyone is a native english speaker but when
> on a conference containing people with a poor grasp of english or
> people who reply with lots of quick utterances such as "yup"
>
> > My guess is you are talking to something lame like a Sonus who is
> resetting
> > its jitterbuffer when you start talking again.
>
> In production it's hitting Cisco gateways but I've experienced this in
> a lab environment consisting of a mixture of Snom 360, 820 and Polycom
> IP6000 devices.
>
> What would be great is to have start/stop talk events while still
> always sending audio from participants to the conference
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jerry.richards at teotech.com  Tue Jan 26 13:02:08 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 26 Jan 2010 13:02:08 -0800
Subject: [Freeswitch-users] Presence Change Distribution
In-Reply-To: <191c3a031001251104p55ba7009g9381841f7de56d65@mail.gmail.com>
References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com><2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com><26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com>
	<191c3a031001251104p55ba7009g9381841f7de56d65@mail.gmail.com>
Message-ID: <BE6F5600B8E54EE297963A8B08F70E5E@greyhawk.tonecommander.com>

Okay, I setup my FS to force SUBSCRIBE Expires to 3600 seconds.  Then I
captured a FS console trace of a Bria softphone changing it's presence state
from 'Busy' to 'Away' (see http://pastebin.freeswitch.org/11953) and
observed that the subscribing Bria softphone did not update to 'Away'.  At
the same time, I executed the sqlite3 app and pasted each of the 3 SQL
select statements I saw in the FS console log, and pasted them below.  I'm
new to sqlite3.  Do you see what my issue is?
 
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'internal' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
sip|5382|192.168.72.79|5401|192.168.72.79|presence|"5382 on 79"
<sip:5382 at 192.168.72.150:34672>|ZTQ2ZWQwZGRlZjRiNTdkYTJjNGM5NTgzOWIyNmIwZmU.
|"5382 on 79" <sip:5382 at 192.168.72.79>;tag=68bb4eb6|SIP/2.0/UDP
192.168.72.150:34672;branch=z9hG4bK-d8754z-eafc60166305eaef-1---d8754z-;rpor
t=34672|1264546204|Teo Softphone release 2.5.4 stamp
55958||internal|Away|away|192.168.72.79|Away|away
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name =
'internal-ipv6' or sip_subscriptions.presence_hosts !=
sip_subscriptions.sub_to_host);
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'external' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
sqlite>

Thanks and Best Regards,
Jerry


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Monday, January 25, 2010 11:05 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


the notify will be instant after the publish
the notify you see are not triggered by the publish or they would be
instant.

Same drill, turn on presence debugging in sofia.conf.xml 
and look at the sql stmts and see why



On Mon, Jan 25, 2010 at 12:30 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:


Okay, I notice that if I reduce the Presence SUBSCRIBE Expires duration
(from 3600 seconds to 60 seconds), then the delay between PUBLISH's and
NOTIFY's is reduced, but FS still waits nearly 45 seconds to send the
NOTIFY's after it receives a PUBLISH.
 
Can a change be made in FS so that NOTIFYs are sent as a direct result of
receipt of the PUBLISH message?  And not tied to the SUBSCRIBE expiration?
I really don't want to configure all my phones to re-subscribe every 30 or
15 seconds.
 
Thanks and Best Regards,
Jerry
 


  _____  

From: RobertT [mailto:siniypin at gmail.com] 
Sent: Tuesday, December 29, 2009 12:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


You can try to reduce your registration time. 
I for one made my client apps send PUBLISH message every minute in addition
to reduced registration time.

Regards, Robert.


2009/12/28 Jerry Richards <jerry.richards at teotech.com>


Is there a setting to control how fast FS distributes presence changes to
subscribers?  Currently, it appears to take several minutes before I see
presence changes.  I would like to see them almost instantaneously, if
possible.

Thanks and Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


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From dftoro at yahoo.com  Tue Jan 26 13:02:55 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 26 Jan 2010 13:02:55 -0800 (PST)
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <191c3a031001260827j544e1414oaec56f527e116eea@mail.gmail.com>
Message-ID: <955182.59161.qm@web33504.mail.mud.yahoo.com>

Hi, sorry, I explain better. Using \\\\ is also changed when path matches a character such as \s,\n... My alternative on Windows is to use '/' like path separator.


Diego Toro
http://lacarretade.blogspot.com/


--- On Tue, 1/26/10, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] mutiple playback files (unescape_char) Windows
> To: freeswitch-users at lists.freeswitch.org
> Date: Tuesday, January 26, 2010, 11:27 AM
> I didn't understand that 
> 
> On Tue, Jan 26, 2010 at 9:58 AM,
> Diego Toro <dftoro at yahoo.com>
> wrote:
> 
> Hi, using \\\\ the is changed also when
> there is a match with an escape character
> (\s,\n...)
> 
> 
> 
> Thank you
> 
> 
> 
> Diego Toro
> 
> http://lacarretade.blogspot.com/
> 
> 
> 
> 
> 
> --- On Mon, 1/25/10, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
> 
> 
> 
> > From: Anthony Minessale <anthony.minessale at gmail.com>
> 
> > Subject: Re: [Freeswitch-users]
> mutiple playback files (unescape_char) Windows
> 
> > To: freeswitch-users at lists.freeswitch.org
> 
> > Date: Monday, January 25, 2010, 12:20 PM
> 
> > its possible your
> string hits the parser
> 
> > more than once.
> 
> > try using 4 \
> 
> >
> 
> > \\\\sound
> 
> >
> 
> >
> 
> > On Sun, Jan 24, 2010 at 4:03 AM,
> 
> > Michael Jerris <mike at jerris.com>
> 
> > wrote:
> 
> >
> 
> > As noted on that bug, you should be
> 
> > able to either use \\ or / for the path
> separator
> 
> > there and it should work.
> 
> >
> 
> >
> 
> >
> 
> >
> 
> > Mike
> 
> >
> 
> >
> 
> >
> 
> > On Jan 22, 2010, at 9:18 AM, Diego Toro wrote:
> 
> >
> 
> >
> 
> >
> 
> > > Hi, with svn r16440 the problem persists, I
> creted a
> 
> > jira report http://jira.freeswitch.org/browse/LBSNDF-8
> 
> > this is a minor issue, but activing playback delimiter
> no
> 
> > audio file can be played. On FS the audio files are
> placed
> 
> > in the \sound\ directory, building the path
> on
> 
> > Windows would be \sound '\s' which is
> 
> > replaced by 'ound'.
> 
> >
> 
> >
> 
> > >
> 
> >
> 
> > > Thank you
> 
> >
> 
> > >
> 
> >
> 
> > > Diego Toro
> 
> >
> 
> > > http://lacarretade.blogspot.com/
> 
> >
> 
> > >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> > _______________________________________________
> 
> >
> 
> > FreeSWITCH-users mailing list
> 
> >
> 
> > FreeSWITCH-users at lists.freeswitch.org
> 
> >
> 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> >
> 
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> >
> 
> > http://www.freeswitch.org
> 
> >
> 
> >
> 
> >
> 
> >
> 
> > --
> 
> > Anthony Minessale II
> 
> >
> 
> > FreeSWITCH http://www.freeswitch.org/
> 
> > ClueCon http://www.cluecon.com/
> 
> >
> 
> > Twitter: http://twitter.com/FreeSWITCH_wire
> 
> >
> 
> > AIM: anthm
> 
> > MSN:anthony_minessale at hotmail.com
> 
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> 
> >
> 
> > IRC: irc.freenode.net
> 
> > #freeswitch
> 
> >
> 
> > FreeSWITCH Developer Conference
> 
> > sip:888 at conference.freeswitch.org
> 
> > iax:guest at conference.freeswitch.org/888
> 
> >
> 
> > googletalk:conf+888 at conference.freeswitch.org
> 
> > pstn:+19193869900
> 
> >
> 
> >
> 
> > -----Inline Attachment Follows-----
> 
> >
> 
> > _______________________________________________
> 
> > FreeSWITCH-users mailing list
> 
> > FreeSWITCH-users at lists.freeswitch.org
> 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> > http://www.freeswitch.org
> 
> >
> 
> 
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> 
> FreeSWITCH-users mailing list
> 
> FreeSWITCH-users at lists.freeswitch.org
> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> 
> IRC: irc.freenode.net
> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> 
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From camilin2212 at hotmail.com  Tue Jan 26 13:10:48 2010
From: camilin2212 at hotmail.com (juan camilo ospina quintero)
Date: Tue, 26 Jan 2010 16:10:48 -0500
Subject: [Freeswitch-users] Inbound sip invite from external gateway
Message-ID: <BAY143-W25E4C7D5F62FD42D7BA37BA15E0@phx.gbl>


hi to all

im already do the integration with. Freeswitch sends invite messages to sailfin, in sailfin there is a sip
servlet that acts as a proxy, this means it receives the invite from extension1000 and send the invite back
to freeswitch at extension 1001, but i get the freeswitch messages go to sailfin, but i dont get freeswitch
to understand sailfin messages.

there is my configuration for sending messages and for receiving messages

In /freeswitch/conf/dialplan/default.xml

    <extension name="redirect">
      <condition field="network_addr" expression="^192\.168\.2\.9$"/>    
      <condition field="destination_number" expression="^1(\d+)$">
          <action application="bridge" data="sofia/external/$0 at 127.0.0.1:5070"/>
      </condition>
    </extension>

this works fine, it redirects the messages to sailfin in 127.0.0.1


In /freeswitch/conf/dialplan/public.xml

    <extension name="receive">
      <condition field="network_addr" expression="^127\.0\.0\.1$"/>    
      <condition field="destination_number" expression="^1(\d+)$">
          <action application="bridge" data="sofia/internal/$0 at 192.168.2.9:5060"/>
      </condition>
    </extension>

this doesnt work, i also use <action application="transfer" data="$1001 XML default/> instead <action application="bridge" data="sofia/internal/$0 at 192.168.2.9:5060"/>

but still doesnt work, the invite that sailfin sends appears in the freeswitch console, but the 1001 extension doesnt get it
 		 	   		  
_________________________________________________________________
Keep your friends updated?even when you?re not signed in.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010
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From anthony.minessale at gmail.com  Tue Jan 26 13:21:54 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Jan 2010 15:21:54 -0600
Subject: [Freeswitch-users] Presence Change Distribution
In-Reply-To: <BE6F5600B8E54EE297963A8B08F70E5E@greyhawk.tonecommander.com>
References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com>
	<2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com>
	<26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com>
	<191c3a031001251104p55ba7009g9381841f7de56d65@mail.gmail.com>
	<BE6F5600B8E54EE297963A8B08F70E5E@greyhawk.tonecommander.com>
Message-ID: <191c3a031001261321v2e8ea21cm7da19e01a11b59f9@mail.gmail.com>

its sending a notify to them right away (line 174 of your PB)
the xml in the notify we send looks the same as what they sent except one
thing

They send: <rpid:away/>
We send: <rpid:busy/>

everybody who implements this seems to have their own idea of what to say
here.

This crazy xml presence crap is pure garbage so maybe that's it.



On Tue, Jan 26, 2010 at 3:02 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  Okay, I setup my FS to force SUBSCRIBE Expires to 3600 seconds.  Then I
> captured a FS console trace of a Bria softphone changing it's presence state
> from 'Busy' to 'Away' (see http://pastebin.freeswitch.org/11953) and
> observed that the subscribing Bria softphone did not update to 'Away'.  At
> the same time, I executed the sqlite3 app and pasted each of the 3 SQL
> select statements I saw in the FS console log, and pasted them below.  I'm
> new to sqlite3.  Do you see what my issue is?
>
> sqlite>
> sqlite> select
> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
> from sip_subscriptions left join sip_presence on
> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
> sip_subscriptions.sub_to_host=sip_presence.sip_host and
> sip_subscriptions.profile_name=sip_presence.profile_name) where
> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
> like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'internal' or
> sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
> sip|5382|192.168.72.79|5401|192.168.72.79|presence|"5382 on 79" <
> sip:5382 at 192.168.72.150:34672>|ZTQ2ZWQwZGRlZjRiNTdkYTJjNGM5NTgzOWIyNmIwZmU.|"5382
> on 79" <sip:5382 at 192.168.72.79 <sip%3A5382 at 192.168.72.79>>;tag=68bb4eb6|SIP/2.0/UDP
> 192.168.72.150:34672;branch=z9hG4bK-d8754z-eafc60166305eaef-1---d8754z-;rport=34672|1264546204|Teo
> Softphone release 2.5.4 stamp
> 55958||internal|Away|away|192.168.72.79|Away|away
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite> select
> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
> from sip_subscriptions left join sip_presence on
> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
> sip_subscriptions.sub_to_host=sip_presence.sip_host and
> sip_subscriptions.profile_name=sip_presence.profile_name) where
> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
> like '%192.168.72.79%') and (sip_subscriptions.profile_name =
> 'internal-ipv6' or sip_subscriptions.presence_hosts !=
> sip_subscriptions.sub_to_host);
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite> select
> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
> from sip_subscriptions left join sip_presence on
> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
> sip_subscriptions.sub_to_host=sip_presence.sip_host and
> sip_subscriptions.profile_name=sip_presence.profile_name) where
> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
> like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'external' or
> sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
> sqlite>
> Thanks and Best Regards,
> Jerry
>
>  ------------------------------
> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
> *Sent:* Monday, January 25, 2010 11:05 AM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>
> the notify will be instant after the publish
> the notify you see are not triggered by the publish or they would be
> instant.
>
> Same drill, turn on presence debugging in sofia.conf.xml
> and look at the sql stmts and see why
>
>
> On Mon, Jan 25, 2010 at 12:30 PM, Jerry Richards <
> jerry.richards at teotech.com> wrote:
>
>>  Okay, I notice that if I reduce the Presence SUBSCRIBE Expires duration
>> (from 3600 seconds to 60 seconds), then the delay between PUBLISH's and
>> NOTIFY's is reduced, but FS still waits nearly 45 seconds to send the
>> NOTIFY's after it receives a PUBLISH.
>>
>> Can a change be made in FS so that NOTIFYs are sent as a direct result of
>> receipt of the PUBLISH message?  And not tied to the SUBSCRIBE expiration?
>> I really don't want to configure all my phones to re-subscribe every 30 or
>> 15 seconds.
>>
>> Thanks and Best Regards,
>> Jerry
>>
>>
>>  ------------------------------
>> *From:* RobertT [mailto:siniypin at gmail.com]
>> *Sent:* Tuesday, December 29, 2009 12:02 AM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>>
>>   You can try to reduce your registration time.
>> I for one made my client apps send PUBLISH message every minute in
>> addition to reduced registration time.
>>
>> Regards, Robert.
>>
>> 2009/12/28 Jerry Richards <jerry.richards at teotech.com>
>>
>>> Is there a setting to control how fast FS distributes presence changes to
>>> subscribers?  Currently, it appears to take several minutes before I see
>>> presence changes.  I would like to see them almost instantaneously, if
>>> possible.
>>>
>>> Thanks and Best Regards,
>>> Jerry
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Tue Jan 26 14:26:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Jan 2010 16:26:51 -0600
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <955182.59161.qm@web33504.mail.mud.yahoo.com>
References: <191c3a031001260827j544e1414oaec56f527e116eea@mail.gmail.com>
	<955182.59161.qm@web33504.mail.mud.yahoo.com>
Message-ID: <191c3a031001261426h7fd87e1fpf95824788d639557@mail.gmail.com>

please update again and try 4 slashes

you need 4 because the expand vars on the data="" will eat the 4 down to 2
then the splitter on ! will turn \\s into \s





On Tue, Jan 26, 2010 at 3:02 PM, Diego Toro <dftoro at yahoo.com> wrote:

> Hi, sorry, I explain better. Using \\\\ is also changed when path matches a
> character such as \s,\n... My alternative on Windows is to use '/' like path
> separator.
>
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
> --- On Tue, 1/26/10, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>
> > From: Anthony Minessale <anthony.minessale at gmail.com>
> > Subject: Re: [Freeswitch-users] mutiple playback files (unescape_char)
> Windows
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Tuesday, January 26, 2010, 11:27 AM
> > I didn't understand that
> >
> > On Tue, Jan 26, 2010 at 9:58 AM,
> > Diego Toro <dftoro at yahoo.com>
> > wrote:
> >
> > Hi, using \\\\ the is changed also when
> > there is a match with an escape character
> > (\s,\n...)
> >
> >
> >
> > Thank you
> >
> >
> >
> > Diego Toro
> >
> > http://lacarretade.blogspot.com/
> >
> >
> >
> >
> >
> > --- On Mon, 1/25/10, Anthony Minessale <anthony.minessale at gmail.com>
> > wrote:
> >
> >
> >
> > > From: Anthony Minessale <anthony.minessale at gmail.com>
> >
> > > Subject: Re: [Freeswitch-users]
> > mutiple playback files (unescape_char) Windows
> >
> > > To: freeswitch-users at lists.freeswitch.org
> >
> > > Date: Monday, January 25, 2010, 12:20 PM
> >
> > > its possible your
> > string hits the parser
> >
> > > more than once.
> >
> > > try using 4 \
> >
> > >
> >
> > > \\\\sound
> >
> > >
> >
> > >
> >
> > > On Sun, Jan 24, 2010 at 4:03 AM,
> >
> > > Michael Jerris <mike at jerris.com>
> >
> > > wrote:
> >
> > >
> >
> > > As noted on that bug, you should be
> >
> > > able to either use \\ or / for the path
> > separator
> >
> > > there and it should work.
> >
> > >
> >
> > >
> >
> > >
> >
> > >
> >
> > > Mike
> >
> > >
> >
> > >
> >
> > >
> >
> > > On Jan 22, 2010, at 9:18 AM, Diego Toro wrote:
> >
> > >
> >
> > >
> >
> > >
> >
> > > > Hi, with svn r16440 the problem persists, I
> > creted a
> >
> > > jira report http://jira.freeswitch.org/browse/LBSNDF-8
> >
> > > this is a minor issue, but activing playback delimiter
> > no
> >
> > > audio file can be played. On FS the audio files are
> > placed
> >
> > > in the \sound\ directory, building the path
> > on
> >
> > > Windows would be \sound '\s' which is
> >
> > > replaced by 'ound'.
> >
> > >
> >
> > >
> >
> > > >
> >
> > >
> >
> > > > Thank you
> >
> > >
> >
> > > >
> >
> > >
> >
> > > > Diego Toro
> >
> > >
> >
> > > > http://lacarretade.blogspot.com/
> >
> > >
> >
> > > >
> >
> > >
> >
> > >
> >
> > >
> >
> > >
> >
> > >
> >
> > > _______________________________________________
> >
> > >
> >
> > > FreeSWITCH-users mailing list
> >
> > >
> >
> > > FreeSWITCH-users at lists.freeswitch.org
> >
> > >
> >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >
> >
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >
> >
> > > http://www.freeswitch.org
> >
> > >
> >
> > >
> >
> > >
> >
> > >
> >
> > > --
> >
> > > Anthony Minessale II
> >
> > >
> >
> > > FreeSWITCH http://www.freeswitch.org/
> >
> > > ClueCon http://www.cluecon.com/
> >
> > >
> >
> > > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > >
> >
> > > AIM: anthm
> >
> > > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >
> > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >
> > >
> >
> > > IRC: irc.freenode.net
> >
> > > #freeswitch
> >
> > >
> >
> > > FreeSWITCH Developer Conference
> >
> > > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >
> > > iax:guest at conference.freeswitch.org/888
> >
> > >
> >
> > > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >
> > > pstn:+19193869900
> >
> > >
> >
> > >
> >
> > > -----Inline Attachment Follows-----
> >
> > >
> >
> > > _______________________________________________
> >
> > > FreeSWITCH-users mailing list
> >
> > > FreeSWITCH-users at lists.freeswitch.org
> >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > http://www.freeswitch.org
> >
> > >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net
> > #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> >
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> >
> > -----Inline Attachment Follows-----
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jcasale at activenetwerx.com  Tue Jan 26 14:37:55 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 26 Jan 2010 22:37:55 +0000
Subject: [Freeswitch-users] Trouble w/ incoming calls from gateway
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90A4907@prato.activenetwerx.local>

Trying to track down what changed and I am sort of baffled.

I had an SPA3102's FXO port setup as a UA (1004) as I couldn't figure out
how to set it up as "gateway" from fs' perspective. Everything was working fine.

Now an incoming call gets dropped with this:
Hangup sofia/internal/XXXxxxXXXX at 192.168.13.1 [CS_NEW] [INCOMPATIBLE_DESTINATION]

Its set to dial ext 2000 (a group call) and being that its setup as a UA, 1004,
how does this develop the line above?

Thanks,
jlc


From sos at sokhapkin.dyndns.org  Tue Jan 26 14:48:57 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 26 Jan 2010 17:48:57 -0500
Subject: [Freeswitch-users] Trouble w/ incoming calls from gateway
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E90A4907@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E90A4907@prato.activenetwerx.local>
Message-ID: <201001261748.57503.sos@sokhapkin.dyndns.org>

INCOMPATIBLE_DESTINATION usually means codec-related problems. Perhaps SPA 
offers G729 codec only and the codec name in SPA settings is set to G729a?

On Tuesday 26 January 2010, Joseph L. Casale wrote:
> Trying to track down what changed and I am sort of baffled.
>
> I had an SPA3102's FXO port setup as a UA (1004) as I couldn't figure out
> how to set it up as "gateway" from fs' perspective. Everything was working
> fine.
>
> Now an incoming call gets dropped with this:
> Hangup sofia/internal/XXXxxxXXXX at 192.168.13.1 [CS_NEW]
> [INCOMPATIBLE_DESTINATION]
>
> Its set to dial ext 2000 (a group call) and being that its setup as a UA,
> 1004, how does this develop the line above?
>
> Thanks,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From msc at freeswitch.org  Tue Jan 26 14:49:11 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 26 Jan 2010 14:49:11 -0800
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <1b46b4e81001221322w5da04799s2ea86000c5c4a9a@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
	<e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
	<5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>
	<5d2828f1001201406p4154b98ald1af1c5c25f59337@mail.gmail.com>
	<1b46b4e81001210335l42baef16r8a2952aa5b92f6e6@mail.gmail.com>
	<87f2f3b91001211412h7aa0a84ageadb9557b869ca01@mail.gmail.com>
	<1b46b4e81001221322w5da04799s2ea86000c5c4a9a@mail.gmail.com>
Message-ID: <87f2f3b91001261449x3401f48eibae516d2b7abc8d4@mail.gmail.com>

Thanks for your contributions! They are much appreciated.
-MC

On Fri, Jan 22, 2010 at 1:22 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> No problem, here it is:
>
> - http://wiki.freeswitch.org/wiki/Call_retry_based_on_hangup_cause
>
> It is linked from your reference (
> http://wiki.freeswitch.org/wiki/Mod_lua#More_Samples).
>
> Sorry I didn't do it early, I hadn't seen your email.
>
> I also added another, more complete, example here (also linked):
>
> - http://wiki.freeswitch.org/wiki/Bridging_two_calls_with_retry
>
>
>
> On Thu, Jan 21, 2010 at 7:12 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Thu, Jan 21, 2010 at 3:35 AM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>
>>>
>>> On Wed, Jan 20, 2010 at 7:06 PM, Mike van Lammeren <
>>> mike at van.lammeren.net> wrote:
>>>
>>>> So, I've been reading about early media in the wiki, and have made a
>>>> little progress, which leads to more questions.
>>>>
>>>> I understand now why a call is considered connected before one person
>>>> has picked up the phone. I am also able to get my script to wait for the
>>>> phone to be picked up, by setting the ignore_early_media variable when
>>>> starting a new session, like this:
>>>>
>>>> customerSession =
>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/example.com/"
>>>> .. customerPhoneNumber)
>>>>
>>>>
>>>> After that line, the script waits for the other phone to be picked up.
>>>>
>>>> However, now I wonder what to do with calls that don't complete, get
>>>> busy signals, etc.
>>>>
>>>> What do people do in this case? The only related example I can find on
>>>> the web is for a javascript dialer, which doesn't address any of these
>>>> cases.
>>>>
>>>
>>>
>>> I guess it depends on what you want to do. For example I have a lua
>>> script very similar to what you describe, although there is no confirmation
>>> involved. Depending on the hangup cause the session gets, it might try
>>> redialing with a different gateway, try again or just hangup.
>>>
>>> Take a look here http://wiki.freeswitch.org/wiki/Hangup_causes to see
>>> what each hangup cause means. You don't need to have a special case for all
>>> of them, only the ones you are interested in.
>>>
>>> Here's an example in code which retries a call depending on the hangup
>>> cause. It retries max_retries1 times and alternates between 2 different
>>> gateways:
>>>
>>> session1 = null;
>>> max_retries1 = 3;
>>> retries = 0;
>>> ostr = "";
>>> repeat
>>>         retries = retries + 1;
>>>         if (retries % 2) then ostr = originate_str1;
>>>         else ostr = originate_str12; end
>>>         freeswitch.consoleLog("notice", "*********** Dialing Leg1: " ..
>>> ostr .. " - Try: "..retries.." ***********\n");
>>>         session1 = freeswitch.Session(ostr);
>>>         local hcause = session1:hangupCause();
>>>         freeswitch.consoleLog("notice", "*********** Leg1: " .. hcause ..
>>> " - Try: "..retries.." ***********\n");
>>> until not ((hcause == 'NO_ROUTE_DESTINATION' or hcause ==
>>> 'RECOVERY_ON_TIMER_EXPIRE' or hcause == 'INCOMPATIBLE_DESTINATION' or hcause
>>> == 'CALL_REJECTED' or hcause == 'NORMAL_TEMPORARY_FAILURE') and (retries <
>>> max_retriesl1))
>>>
>>>
>>> Note: originate_str1 and originate_str2 are two different dial strings
>>> for 2 different gateways.
>>>
>>>
>> Nicolas,
>>
>> This is really nice. Would you be willing to add this script and a brief
>> explanation to the wiki? You could create a whole new page and just link to
>> it from here: http://wiki.freeswitch.org/wiki/Mod_lua#More_Samples
>>
>> If you have any questions please let me know!
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jcasale at activenetwerx.com  Tue Jan 26 15:04:25 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 26 Jan 2010 23:04:25 +0000
Subject: [Freeswitch-users] Trouble w/ incoming calls from gateway
In-Reply-To: <201001261748.57503.sos@sokhapkin.dyndns.org>
References: <CA5A491E9DEFBE4CB777DE97E21575E90A4907@prato.activenetwerx.local>
	<201001261748.57503.sos@sokhapkin.dyndns.org>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90A49C7@prato.activenetwerx.local>

>INCOMPATIBLE_DESTINATION usually means codec-related problems. Perhaps SPA 
>offers G729 codec only and the codec name in SPA settings is set to G729a?

I think you narrowed it down:) After following the cisco docs for days to get
rid of echo unsuccessfully, we were horsing around and changed the codec from
G711u to G726-40 and the echo disappeared. Blank stares were met with a shrug.
The firmware was updated but no powerdown was done and a test yielded it still
working so we left it. The only change was a powerdown recently.

The output leading up to the hangup is:

2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [G726-40:96:8000:20]/[G7221:115:32000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [G726-40:96:8000:20]/[G7221:107:16000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [G726-40:96:8000:20]/[G722:9:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [G726-40:96:8000:20]/[PCMU:0:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [G726-40:96:8000:20]/[PCMA:8:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [G726-40:96:8000:20]/[GSM:3:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [NSE:100:8000:20]/[G7221:115:32000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [NSE:100:8000:20]/[G7221:107:16000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [NSE:100:8000:20]/[G722:9:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [NSE:100:8000:20]/[PCMU:0:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [NSE:100:8000:20]/[PCMA:8:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [NSE:100:8000:20]/[GSM:3:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3286 Set 2833 dtmf payload to 101
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [telephone-event:101:8000:20]/[G7221:115:32000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [telephone-event:101:8000:20]/[G7221:107:16000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [telephone-event:101:8000:20]/[G722:9:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [telephone-event:101:8000:20]/[PCMU:0:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [telephone-event:101:8000:20]/[GSM:3:8000:20]

So if I gather from this correctly, fs doesn't even have a codec to match against it?

Thanks a ton for the insight Sergey,
jlc


From sos at sokhapkin.dyndns.org  Tue Jan 26 15:22:26 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 26 Jan 2010 18:22:26 -0500
Subject: [Freeswitch-users] Trouble w/ incoming calls from gateway
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E90A49C7@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E90A4907@prato.activenetwerx.local>
	<201001261748.57503.sos@sokhapkin.dyndns.org>
	<CA5A491E9DEFBE4CB777DE97E21575E90A49C7@prato.activenetwerx.local>
Message-ID: <201001261822.26304.sos@sokhapkin.dyndns.org>

Do you have G726-40 enabled in global_codec_prefs in vars.xml?

On Tuesday 26 January 2010, Joseph L. Casale wrote:
> >INCOMPATIBLE_DESTINATION usually means codec-related problems. Perhaps SPA
> >offers G729 codec only and the codec name in SPA settings is set to G729a?
>
> I think you narrowed it down:) After following the cisco docs for days to
> get rid of echo unsuccessfully, we were horsing around and changed the
> codec from G711u to G726-40 and the echo disappeared. Blank stares were met
> with a shrug. The firmware was updated but no powerdown was done and a test
> yielded it still working so we left it. The only change was a powerdown
> recently.
>
> The output leading up to the hangup is:
>
> 2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [G726-40:96:8000:20]/[G7221:115:32000:20] 2010-01-26 16:04:36.682392
> [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [G726-40:96:8000:20]/[G7221:107:16000:20] 2010-01-26 16:04:36.682392
> [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [G726-40:96:8000:20]/[G722:9:8000:20] 2010-01-26 16:04:36.682392 [DEBUG]
> sofia_glue.c:3330 Audio Codec Compare [G726-40:96:8000:20]/[PCMU:0:8000:20]
> 2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [G726-40:96:8000:20]/[PCMA:8:8000:20] 2010-01-26 16:04:36.682392 [DEBUG]
> sofia_glue.c:3330 Audio Codec Compare [G726-40:96:8000:20]/[GSM:3:8000:20]
> 2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [NSE:100:8000:20]/[G7221:115:32000:20] 2010-01-26 16:04:36.682392 [DEBUG]
> sofia_glue.c:3330 Audio Codec Compare
> [NSE:100:8000:20]/[G7221:107:16000:20] 2010-01-26 16:04:36.682392 [DEBUG]
> sofia_glue.c:3330 Audio Codec Compare [NSE:100:8000:20]/[G722:9:8000:20]
> 2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [NSE:100:8000:20]/[PCMU:0:8000:20] 2010-01-26 16:04:36.682392 [DEBUG]
> sofia_glue.c:3330 Audio Codec Compare [NSE:100:8000:20]/[PCMA:8:8000:20]
> 2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [NSE:100:8000:20]/[GSM:3:8000:20] 2010-01-26 16:04:36.682392 [DEBUG]
> sofia_glue.c:3286 Set 2833 dtmf payload to 101 2010-01-26 16:04:36.682392
> [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [telephone-event:101:8000:20]/[G7221:115:32000:20] 2010-01-26
> 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [telephone-event:101:8000:20]/[G7221:107:16000:20] 2010-01-26
> 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [telephone-event:101:8000:20]/[G722:9:8000:20] 2010-01-26 16:04:36.682392
> [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [telephone-event:101:8000:20]/[PCMU:0:8000:20] 2010-01-26 16:04:36.682392
> [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [telephone-event:101:8000:20]/[PCMA:8:8000:20] 2010-01-26 16:04:36.682392
> [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [telephone-event:101:8000:20]/[GSM:3:8000:20]
>
> So if I gather from this correctly, fs doesn't even have a codec to match
> against it?
>
> Thanks a ton for the insight Sergey,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jcasale at activenetwerx.com  Tue Jan 26 15:21:27 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 26 Jan 2010 23:21:27 +0000
Subject: [Freeswitch-users] Trouble w/ incoming calls from gateway
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E90A49C7@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E90A4907@prato.activenetwerx.local>
	<201001261748.57503.sos@sokhapkin.dyndns.org>
	<CA5A491E9DEFBE4CB777DE97E21575E90A49C7@prato.activenetwerx.local>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90A4A0B@prato.activenetwerx.local>

>So if I gather from this correctly, fs doesn't even have a codec to match against it?

Should have read the wiki before replying to that, from the codecs page and
my vars.xml I see it doesn't have that codec configured.

So are there any optimal codecs to choose for this device? I see there is
not an explicitly named module for 726, is that part of the default codecs?

Thanks,
jlc



From jcasale at activenetwerx.com  Tue Jan 26 15:40:00 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 26 Jan 2010 23:40:00 +0000
Subject: [Freeswitch-users] Trouble w/ incoming calls from gateway
In-Reply-To: <201001261822.26304.sos@sokhapkin.dyndns.org>
References: <CA5A491E9DEFBE4CB777DE97E21575E90A4907@prato.activenetwerx.local>
	<201001261748.57503.sos@sokhapkin.dyndns.org>
	<CA5A491E9DEFBE4CB777DE97E21575E90A49C7@prato.activenetwerx.local>
	<201001261822.26304.sos@sokhapkin.dyndns.org>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90A4A6F@prato.activenetwerx.local>

>Do you have G726-40 enabled in global_codec_prefs in vars.xml?

Now I do, but that produced an unbearable call? This 3102 seems
to be a pain, lots of forum posts related to this device hack on
it. Given I have a new TDM410p card, I wish I could get this working
reliably as when it does it works well. Oh well...

Thanks!
jlc


From msc at freeswitch.org  Tue Jan 26 15:54:22 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 26 Jan 2010 15:54:22 -0800
Subject: [Freeswitch-users] FreeSWITCH 1.0.5 and Buying Dinner For The Dev
	Team
Message-ID: <87f2f3b91001261554m2bfbbc45q21fc41c7d0a43b8@mail.gmail.com>

Hello all,

The FreeSWITCH development team is planning to meet in one place during the
week of February 8 to release version 1.0.5! We would like to invite
everyone to show their appreciation by donating a few dollars to help pay
for dinner for the development team. More information is available here:

http://www.freeswitch.org/node/234

Let's all show the team how much we appreciate them by giving them a
well-deserved dinner!

-Michael
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From camilin2212 at hotmail.com  Tue Jan 26 17:30:24 2010
From: camilin2212 at hotmail.com (juan camilo ospina quintero)
Date: Tue, 26 Jan 2010 20:30:24 -0500
Subject: [Freeswitch-users] External Profile Problem
Message-ID: <BAY143-W23698E1AF0CEBDBC04EFBDA15D0@phx.gbl>


Hi, 

im trying to establish a simple conference using freeswitch and sailfin, sailfin is
and application server that works with SipSevlets.
the all thing works as follow.

two softphone register with freeswitch, extension 1000 and 1001
1000 sends and invite to 1001, this invite goes to sailfin, i use this

    <extension name="redirect">
      <condition field="network_addr" expression="^192\.168\.2\.9$"/>    
      <condition field="destination_number" expression="^1(\d+)$">
    <action application="bridge" data="sofia/external/$0 at 127.0.0.1:5070"/>
      </condition>
    </extension>

this is in /freeswitch/conf/dialplan/default.xml


this far all goes well, the servlet receives the invite, and sends back the invite to freeswitch, i put this

    <extension name="receive">
      <condition field="network_addr" expression="^192\.168\.2\.9$"/>
      <condition field="destination_number" expression="^1(\d+)$">
        <action application="transfer" data="$0 XML default"/>
      </condition>
    </extension>

in /freeswitch/conf/dialplan/public.xml, but freeswitch returns this


------------------------------------------------------------------------
2010-01-26 20:14:29.512927 [NOTICE] switch_channel.c:602 New Channel sofia/external/1000 at 192.168.2.9 [5177e93a-0ae1-11df-afc9-db39c681a2f1]
2010-01-26 20:14:29.512927 [NOTICE] sofia.c:3527 Hangup sofia/external/1000 at 192.168.2.9 [CS_NEW] [MANDATORY_IE_MISSING]
send 632 bytes to udp/[192.168.2.9]:5070 at 01:14:29.517927:
   ------------------------------------------------------------------------
   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/UDP 192.168.153.1:5070;branch=z9hG4bKdaacdd64d693615c451ab9db43f9c71c2626;received=192.168.2.9
   From: <sip:1000 at 192.168.2.9>;tag=g4xfbi12-3
   To: <sip:1001 at 192.168.2.9:5080>;tag=4r91165pvcycB
   Call-Id: 192.168.153.1_3_3990383226484831353
   Cseq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING"
   Content-Length: 0
   
   ------------------------------------------------------------------------
2010-01-26 20:14:29.525646 [NOTICE] switch_core_session.c:1086 Session 9 (sofia/external/1000 at 192.168.2.9) Ended
2010-01-26 20:14:29.525646 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/1000 at 192.168.2.9 [CS_DESTROY]

i dont understand why i doesnt work if in public.xml, i tell to transfer the call if extension starts with 1 and the caller ip address is 192.168.2.9

please if someone can help me
 		 	   		  
_________________________________________________________________
Windows Live: Keep your friends up to date with what you do online.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_1:092010
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From frank at carmickle.com  Tue Jan 26 19:09:29 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Tue, 26 Jan 2010 22:09:29 -0500
Subject: [Freeswitch-users] External Profile Problem
In-Reply-To: <BAY143-W23698E1AF0CEBDBC04EFBDA15D0@phx.gbl>
References: <BAY143-W23698E1AF0CEBDBC04EFBDA15D0@phx.gbl>
Message-ID: <20100127030929.GC3841@base.carmickle.com>

On Tue, Jan 26, juan camilo ospina quintero wrote:
> 
> Hi, 
> 
> im trying to establish a simple conference using freeswitch and sailfin, sailfin is
> and application server that works with SipSevlets.
> the all thing works as follow.
> 
> two softphone register with freeswitch, extension 1000 and 1001
> 1000 sends and invite to 1001, this invite goes to sailfin, i use this
> 
>     <extension name="redirect">
>       <condition field="network_addr" expression="^192\.168\.2\.9$"/>    
>       <condition field="destination_number" expression="^1(\d+)$">
>     <action application="bridge" data="sofia/external/$0 at 127.0.0.1:5070"/>

And what is the external profile listening on?  Probably not the loopback address.  Set up another profile listening on 127.0.0.1 and bridge to that.

I could be off base here because you haven't given us very much info about your freeswitch configurations.

--FC


From dftoro at yahoo.com  Tue Jan 26 19:16:24 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 26 Jan 2010 19:16:24 -0800 (PST)
Subject: [Freeswitch-users] compilation error on fs_cli (Windows)
	(latest version)
In-Reply-To: <3072AE90-DDEA-4FD0-9B4B-140051730073@avgs.ca>
Message-ID: <761142.19511.qm@web33506.mail.mud.yahoo.com>


yes, code analyzer is active. when I turn it off fs_cli project compiled fine. Before, this project compiled fine, why I need turn off analyzer code now ?

Thank you

Diego Toro
http://lacarretade.blogspot.com/


--- On Tue, 1/26/10, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> From: Mathieu Rene <mrene_lists at avgs.ca>
> Subject: Re: [Freeswitch-users] compilation error on fs_cli (Windows) (latest version)
> To: freeswitch-users at lists.freeswitch.org
> Date: Tuesday, January 26, 2010, 11:45 AM
> Looks like the code analyzer is
> running, this is normally turned off? 
> when you do a normal build, turn it off and try again.
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 
> 
> 
> On 26-Jan-10, at 10:55 AM, Diego Toro wrote:
> 
> > Hi, I have compilation error "error C2220" on fs_cli
> project on? 
> > Windows using VS2008.
> >
> > FS: latest version (2010/01/26)
> > VS: VS2008
> > SO: Windows 7
> >
> > VS2008 Error log:
> >
> > Error??? 1??? error
> C2220: warning treated as error - no 'object' file? 
> > generated???
> g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl 
> > \fs_cli.c??? 106???
> fs_cli
> >
> > Warning??? 2??? warning
> C6385: Invalid data: accessing 'global_profile- 
> > >console_fnkeys', the readable size is '48' bytes,
> but '-4' bytes? 
> > might be read: Lines: 86, 88, 90???
> g:\ftp\incoming\fs 
> >
> \freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c???
> 90??? fs_cli
> >
> > Warning??? 3??? warning
> C6246: Local declaration of 'p' hides declaration? 
> > of the same name in outer scope. For additional
> information, see? 
> > previous declaration at line '844' of
> 'g:\ftp\incoming\fs 
> > \freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c':
> Lines: 844??? g:\ftp 
> >
> \incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c???
> 895?????
> > fs_cli
> >
> > Warning??? 4??? warning
> C6011: Dereferencing NULL pointer 'cursor': Lines:? 
> > 839, 840, 841, 842, 843, 844, 846, 849, 853, 857, 868,
> 870, 871, 884?????
> >
> g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c?????
> > 884??? fs_cli
> >
> >
> > Thank you
> >
> > Diego Toro
> > http://lacarretade.blogspot.com/
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From airsignal at wavecable.com  Tue Jan 26 00:57:34 2010
From: airsignal at wavecable.com (Airsignal)
Date: Tue, 26 Jan 2010 00:57:34 -0800
Subject: [Freeswitch-users] NAT keep alive
Message-ID: <8657153B18DB4AB3B17DB7A4BAAFF862@Terminal>

Good Evening:

I am trying to get my switch to send keep alives to the ata's in the field.

<param name="nat-options-ping" value="true"/>   seems to be meant for this.

where should it go?  I can little documentation discussing this...


Thanks!
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From matomoya at yahoo.co.jp  Tue Jan 26 04:07:10 2010
From: matomoya at yahoo.co.jp (=?ISO-2022-JP?B?GyRCJF4kRCRQJGkbKEIgGyRCJEgkYiRkGyhC?=)
Date: Tue, 26 Jan 2010 21:07:10 +0900 (JST)
Subject: [Freeswitch-users] mod_spidermonkey memory leak?
Message-ID: <20100126120711.92321.qmail@web10512.mail.ogk.yahoo.co.jp>

I have tested the SVN trunk 15341. 
When the following scripts are executed, it seems to do the memory leak. 
Is the mistake found in the following scripts?
Or Is there a problem in mod_spidermonkey?

-- test script --
session.execute("ring_ready");
session.answer();
session.setVariable("ringback", "%(1000, 2000, 440, 460)");

var bleg = new Session();
var sound_wav = "sounds/test01.wav";
var sound_leg = "both";
var op = "api_on_answer=uuid_broadcast "+session.uuid+" "+sound_wav+"
"+sound_leg;
var ret = bleg.originate(session, "{"+op+"}" +
"sofia/gateway/profile0_gateway1/1000");
if(!session.ready()){
  return;
}


if(!ret){ // bleg not answered.
  var sound_wav = "sounds/test02.wav";
  session.streamFile(sound_wav);
  if(session.ready()){
    session.hangup();
  }
  return;
}

if(bleg.ready()){
  bridge(session, bleg);
}



From codeghar at gmail.com  Tue Jan 26 17:38:17 2010
From: codeghar at gmail.com (Code Ghar)
Date: Tue, 26 Jan 2010 19:38:17 -0600
Subject: [Freeswitch-users] Replace Internal IP with External IP in From
	Header
Message-ID: <4571ff701001261738w4f51b33dqee1d19b8d0e2236d@mail.gmail.com>

I followed the example in Freeswitch behind NAT (
http://wiki.freeswitch.org/wiki/NAT_Traversal#Freeswitch_behind_NAT). In the
Contact header of invite sent to an external gateway, I see
sip:extension at ExternalIP:port but in the From header I see
sip:extension at InternalIP. How can I change the From header of SIP message so
that it displays the external IP instead of internal IP? The reason for
doing this is that the external gateway authenticates and authorizes call
based on the IP in From header. They expect an external IP and not an
internal IP.
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From matsubara_tomoya at intec.co.jp  Tue Jan 26 19:35:33 2010
From: matsubara_tomoya at intec.co.jp (Tomoya Matsubara)
Date: Wed, 27 Jan 2010 12:35:33 +0900
Subject: [Freeswitch-users] Question about javascript
Message-ID: <20100127123533.602850e1.matsubara_tomoya@intec.co.jp>

Hello,

When the following scripts were tested, it seems to do the memory leak. 
Please teach when there is a problem in this script. 

-- test script --
session.execute("ring_ready");
session.answer();
session.setVariable("ringback", "%(1000, 2000, 440, 460)");

var bleg = new Session();
var sound_wav = "sounds/test01.wav";
var sound_leg = "both";
var op = "api_on_answer=uuid_broadcast "+session.uuid+" "+sound_wav+" "+sound_leg;
var ret = bleg.originate(session, "{"+op+"}" + "sofia/gateway/profile0_gateway1/1000");
if(!session.ready()){
  return;
}

if(!ret){ // bleg not answered.
  var sound_wav = "sounds/test02.wav";
  session.streamFile(sound_wav);
  if(session.ready()){
    session.hangup();
  }
  return;
}

if(bleg.ready()){
  bridge(session, bleg);
}


From j4szczur at gmail.com  Tue Jan 26 15:03:13 2010
From: j4szczur at gmail.com (michal kalinowski)
Date: Wed, 27 Jan 2010 00:03:13 +0100
Subject: [Freeswitch-users] mod_fax ECM
Message-ID: <7c74f5761001261503g457879f9x1447ee8dab1e9104@mail.gmail.com>

Hello,

What mean ECM and for what is used ?

I found in my fax.conf parameter:
<param name="use-ecm"           value="true"/>

BR,
Micha?


From camilin2212 at hotmail.com  Tue Jan 26 20:06:12 2010
From: camilin2212 at hotmail.com (juan camilo ospina quintero)
Date: Tue, 26 Jan 2010 23:06:12 -0500
Subject: [Freeswitch-users] External Profile Problem
In-Reply-To: <20100127030929.GC3841@base.carmickle.com>
References: <BAY143-W23698E1AF0CEBDBC04EFBDA15D0@phx.gbl>,
	<20100127030929.GC3841@base.carmickle.com>
Message-ID: <BAY143-W20A3102CC2FF36EF1C170FA15D0@phx.gbl>


Hi

This works fine

<extension name="redirect">
 <condition field="network_addr" expression="^192\.168\.2\.9$"/>    
 <condition field="destination_number" expression="^1(\d+)$">
 <action application="bridge" data="sofia/external/$0 at 127.0.0.1:5070"/>
</extension>

this redirects from freeswitch to sailfin (127.0.0.1:5070), and is in default.xml, in the dialplan.

the problem is this

    <extension name="receive">
      <condition field="network_addr" expression="^127\.0\.0\.1$"/>
      <condition field="destination_number" expression="^1(\d+)$">
        <action application="transfer" data="$0 XML default"/>
      </condition>
    </extension>
this doesnt work, this configuration can be found in public.xml in the dialplan, the idea of
this is that when a sip invite comes from sailfin (127.0.0.1) transfer the invite to the destination number

the both configurations above are the only configuration i have change from  the default instalation of 
freeswitch.

i would like to have some hep with this thanks

here is the trace log again

2010-01-26 20:14:29.512927 [NOTICE] switch_channel.c:602 New Channel sofia/external/1000 at 192.168.2.9 [5177e93a-0ae1-11df-afc9-db39c681a2f1]
2010-01-26 20:14:29.512927 [NOTICE] sofia.c:3527 Hangup sofia/external/1000 at 192.168.2.9 [CS_NEW] [MANDATORY_IE_MISSING]
send 632 bytes to udp/[192.168.2.9]:5070 at 01:14:29.517927:
   ------------------------------------------------------------------------
   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/UDP 192.168.153.1:5070;branch=z9hG4bKdaacdd64d693615c451ab9db43f9c71c2626;received=192.168.2.9
   From: <sip:1000 at 192.168.2.9>;tag=g4xfbi12-3
   To: <sip:1001 at 192.168.2.9:5080>;tag=4r91165pvcycB
   Call-Id: 192.168.153.1_3_3990383226484831353
   Cseq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING"
   Content-Length: 0
   
   ------------------------------------------------------------------------
2010-01-26 20:14:29.525646 [NOTICE] switch_core_session.c:1086 Session 9 (sofia/external/1000 at 192.168.2.9) Ended
2010-01-26 20:14:29.525646 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/1000 at 192.168.2.9 [CS_DESTROY]



> Date: Tue, 26 Jan 2010 22:09:29 -0500
> From: frank at carmickle.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] External Profile Problem
> 
> On Tue, Jan 26, juan camilo ospina quintero wrote:
> > 
> > Hi, 
> > 
> > im trying to establish a simple conference using freeswitch and sailfin, sailfin is
> > and application server that works with SipSevlets.
> > the all thing works as follow.
> > 
> > two softphone register with freeswitch, extension 1000 and 1001
> > 1000 sends and invite to 1001, this invite goes to sailfin, i use this
> > 
> >     <extension name="redirect">
> >       <condition field="network_addr" expression="^192\.168\.2\.9$"/>    
> >       <condition field="destination_number" expression="^1(\d+)$">
> >     <action application="bridge" data="sofia/external/$0 at 127.0.0.1:5070"/>
> 
> And what is the external profile listening on?  Probably not the loopback address.  Set up another profile listening on 127.0.0.1 and bridge to that.
> 
> I could be off base here because you haven't given us very much info about your freeswitch configurations.
> 
> --FC
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_3:092010
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From jcasale at activenetwerx.com  Tue Jan 26 20:51:13 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 27 Jan 2010 04:51:13 +0000
Subject: [Freeswitch-users] mod_fax ECM
In-Reply-To: <7c74f5761001261503g457879f9x1447ee8dab1e9104@mail.gmail.com>
References: <7c74f5761001261503g457879f9x1447ee8dab1e9104@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90A4F15@prato.activenetwerx.local>

>What mean ECM and for what is used ?
>
>I found in my fax.conf parameter:
><param name="use-ecm"           value="true"/>

Error correction mode, first hit on google -> "fax ecm" :)


From freeswitch-users at digitaldan.com  Tue Jan 26 21:15:37 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Tue, 26 Jan 2010 22:15:37 -0700 (MST)
Subject: [Freeswitch-users] FreeSWITCH 1.0.5 and Buying Dinner For The
 Dev	Team
In-Reply-To: <87f2f3b91001261554m2bfbbc45q21fc41c7d0a43b8@mail.gmail.com>
Message-ID: <14340419.35.1264569337594.JavaMail.root@zimbra>

bon appetit! 
----- Original Message ----- 
From: "Michael Collins" <msc at freeswitch.org> 
To: freeswitch-users at lists.freeswitch.org, freeswitch-dev at lists.freeswitch.org 
Sent: Tuesday, January 26, 2010 4:54:22 PM 
Subject: [Freeswitch-users] FreeSWITCH 1.0.5 and Buying Dinner For The Dev Team 

Hello all, 

The FreeSWITCH development team is planning to meet in one place during the week of February 8 to release version 1.0.5! We would like to invite everyone to show their appreciation by donating a few dollars to help pay for dinner for the development team. More information is available here: 

http://www.freeswitch.org/node/234 

Let's all show the team how much we appreciate them by giving them a well-deserved dinner! 

-Michael 

_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 
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From jason at jasonjgw.net  Tue Jan 26 22:28:22 2010
From: jason at jasonjgw.net (Jason White)
Date: Wed, 27 Jan 2010 17:28:22 +1100
Subject: [Freeswitch-users] External Profile Problem
In-Reply-To: <BAY143-W20A3102CC2FF36EF1C170FA15D0@phx.gbl>
References: <BAY143-W23698E1AF0CEBDBC04EFBDA15D0@phx.gbl>
	<20100127030929.GC3841@base.carmickle.com>
	<BAY143-W20A3102CC2FF36EF1C170FA15D0@phx.gbl>
Message-ID: <20100127062822.GA27365@jdc.jasonjgw.net>

juan camilo ospina quintero <camilin2212 at hotmail.com> wrote:
> 2010-01-26 20:14:29.512927 [NOTICE] switch_channel.c:602 New Channel sofia/external/1000 at 192.168.2.9 [5177e93a-0ae1-11df-afc9-db39c681a2f1]
> 2010-01-26 20:14:29.512927 [NOTICE] sofia.c:3527 Hangup sofia/external/1000 at 192.168.2.9 [CS_NEW] [MANDATORY_IE_MISSING]

I haven't been following the thread, but the above error is usually an
authentication problem, for example, the destination isn't accepting the call.
If the destination is a FreeSWITCH instance, make sure that auth-calls is set
appropriately and that the ACLs aren't causing any problems.



From Prometheus001 at gmx.net  Wed Jan 27 02:14:07 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 27 Jan 2010 11:14:07 +0100
Subject: [Freeswitch-users] Wrong RTP port submitted?
Message-ID: <4B6011EF.6090706@gmx.net>

I have defined the rtp port range for 12000-12100 in switch.conf.xml.
However Freeswitch is offering a port 48320 in the invite message. The
result is, that the incoming RTP stream is blocked by the firewall (I
can see a reject for UDP 48320).
Any hint how to solve this?

Best regards Peter

See config and invite message:
    <!--RTP port range -->
    <param name="rtp-start-port" value="12000"/>-->
    <param name="rtp-end-port" value="12100"/>-->

Invite:
INVITE sip:027xxxxxxxx at sip.itsp.de SIP/2.0.
Via: SIP/2.0/UDP 217.24.xx.xxx:5080;rport;branch=z9hG4bKjD923NvctXaFm.
Max-Forwards: 69.
From: "0608xxxxxxx"
<sip:4918xxxxxxxxxxxx at sip.itsp.de;transport=udp>;tag=0Kp4tvU44UmXp.
To: <sip:027xxxxxxxx at sip.itsp.de>.
Call-ID: 30c86b94-85ca-122d-f88e-080027e51f59.
CSeq: 126174137 INVITE.
Contact: <sip:gw+tgnet3 at 217.24.xx.xxx:5080;transport=udp;gw=tgnet3>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16032.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 320.
P-Key-Flags: keys="3".
X-FS-Support: update_display.
Remote-Party-ID: "0608xxxxxxx"
<sip:0608xxxxxxx at sip.itsp.de>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1264536651 1264536652 IN IP4 217.24.xx.xxx.
s=FreeSWITCH.
c=IN IP4 217.24.xx.xxx.
t=0 0.
m=audio 48320 RTP/AVP 8 0 98 3 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:98 SPEEX/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


From codecomplete at free.fr  Wed Jan 27 04:38:20 2010
From: codecomplete at free.fr (Fred-145)
Date: Wed, 27 Jan 2010 04:38:20 -0800 (PST)
Subject: [Freeswitch-users] Equivalent to Asterisk's "directrtpsetup=yes"?
Message-ID: <27338355.post@talk.nabble.com>


Hello

Thanks to "directrtpsetup=yes", it appears that Asterisk can act as an SIP
proxy and a PBX at the same time, ie. RTP packets will flow directly between
the two SIP end-points with Asterisk still being able to provide PBX
services like call transfer, MoH, etc.
The point is to lower latency for packets to reach their final destination
and lower CPU load on the Freeswitch server.

Does Freeswitch offer the same feature, or must RTP packets always go
through the Freeswitch servers?

Thank you.
-- 
View this message in context: http://old.nabble.com/Equivalent-to-Asterisk%27s-%22directrtpsetup%3Dyes%22--tp27338355p27338355.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From codecomplete at free.fr  Wed Jan 27 04:38:50 2010
From: codecomplete at free.fr (Fred-145)
Date: Wed, 27 Jan 2010 04:38:50 -0800 (PST)
Subject: [Freeswitch-users] Equivalent to Asterisk's "directrtpsetup=yes"?
Message-ID: <27338355.post@talk.nabble.com>


Hello

Thanks to "directrtpsetup=yes", it appears that Asterisk can act as an SIP
proxy and a PBX at the same time, ie. RTP packets will flow directly between
the two SIP end-points with Asterisk still being able to provide PBX
services like call transfer, MoH, etc.
The point is lower latency for packets to reach their final destination and
lower CPU load on the Freeswitch server.

Does Freeswitch offer the same feature, or must RTP packets always go
through the Freeswitch servers?

Thank you.
-- 
View this message in context: http://old.nabble.com/Equivalent-to-Asterisk%27s-%22directrtpsetup%3Dyes%22--tp27338355p27338355.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From Russell.Mosemann at cune.org  Wed Jan 27 04:47:28 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Wed, 27 Jan 2010 06:47:28 -0600
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
In-Reply-To: <27338355.post@talk.nabble.com>
References: <27338355.post@talk.nabble.com>
Message-ID: <9BB5BC82B9F54466ACFA5BA610669FD7@cune.pri>

Fred-145 was wondering:

> Does Freeswitch offer the same feature, or must RTP packets always go
> through the Freeswitch servers?

http://wiki.freeswitch.org/wiki/Bypass_Media

> Thank you.

You're welcome.

--
Russell Mosemann



From rob4manhere at gmail.com  Wed Jan 27 04:49:43 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Wed, 27 Jan 2010 06:49:43 -0600
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
In-Reply-To: <27338355.post@talk.nabble.com>
References: <27338355.post@talk.nabble.com>
Message-ID: <C70CDDCF-AB2E-4D10-A4EC-0E6205193A31@gmail.com>

Hi Fred,

Check out bypass_media mode:
http://wiki.freeswitch.org/wiki/Bypass_Media

Cheers,
Rob

On Jan 27, 2010, at 6:38 AM, Fred-145 wrote:

>
> Hello
>
> Thanks to "directrtpsetup=yes", it appears that Asterisk can act as  
> an SIP
> proxy and a PBX at the same time, ie. RTP packets will flow directly  
> between
> the two SIP end-points with Asterisk still being able to provide PBX
> services like call transfer, MoH, etc.
> The point is lower latency for packets to reach their final  
> destination and
> lower CPU load on the Freeswitch server.
>
> Does Freeswitch offer the same feature, or must RTP packets always go
> through the Freeswitch servers?
>
> Thank you.
> -- 
> View this message in context: http://old.nabble.com/Equivalent-to-Asterisk%27s-%22directrtpsetup%3Dyes%22--tp27338355p27338355.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From sos at sokhapkin.dyndns.org  Wed Jan 27 04:50:16 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 27 Jan 2010 07:50:16 -0500
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
In-Reply-To: <27338355.post@talk.nabble.com>
References: <27338355.post@talk.nabble.com>
Message-ID: <201001270750.16772.sos@sokhapkin.dyndns.org>

set bypass_media=true

On Wednesday 27 January 2010, Fred-145 wrote:
> Hello
>
> Thanks to "directrtpsetup=yes", it appears that Asterisk can act as an SIP
> proxy and a PBX at the same time, ie. RTP packets will flow directly
> between the two SIP end-points with Asterisk still being able to provide
> PBX services like call transfer, MoH, etc.
> The point is lower latency for packets to reach their final destination and
> lower CPU load on the Freeswitch server.
>
> Does Freeswitch offer the same feature, or must RTP packets always go
> through the Freeswitch servers?
>
> Thank you.




From mcampbellsmith at gmail.com  Wed Jan 27 05:06:52 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 28 Jan 2010 00:06:52 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
Message-ID: <33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>

Thanks guys.  I have this working except for one user who is
registered like this:

Call-ID:        2ff39277-fd9a6ab1 at 10.0.0.1
User:           2000 at 192.168.1.120
Contact:        2000
<sip:2000 at 124.xxx.xxx.xxx:5075;transport=tls;fs_nat=yes;fs_path=sip%3A2000%40124.xxx.xxx.xxx%3A10281%3Btransport%3Dtls>
Agent:          Linksys/SPA3102-5.1.10(GW)
Status:         Registered(TLS-NAT)(unknown) EXP(2010-01-28 00:29:34)
Host:           freeswitch
IP:             124.xxx.xxx.xxx
Port:           10281
Auth-User:      2000
Auth-Realm:     mydns.dyndns.org
MWI-Account:    2000 at 192.168.1.120

When I do the following commands via the telnet socket, no notify
command is sent to user 2000:

sendevent NOTIFY
profile: internal
content-type: application/simple-message-summary
event-string: reboot_now
user: 2000
host: 192.168.1.120
content-length: 0

However, if I do exactly the same thing with user 2001 it works.  2001
is registered as:

Contact:        2001 <sip:2001 at 124.yyy.yyy.yyy:5066;transport=tls>

Any ideas why that would be?

On Mon, Jan 25, 2010 at 11:50 AM, Ognjen Seslija <oseslija at gmail.com> wrote:
> The phone is asking FS to authenticate prior then accepting a NOTIFY from
> it.
> The authentication of notify's from spa endpoints work (afaik) only with
> Linksys SPA9000 PBX , and FS doesn't have the code for that for obvious
> reasons.
> If you have SPA9000 maybe you can collect SIP traces.
>
> Ognjen
>
>
>
>
>
> On Mon, Jan 25, 2010 at 1:29 AM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>>
>> Hi Ognjen,
>>
>> Thanks for the tip on the resync under the EXT tab. ?It now works
>> using mod_event_socket and the following:
>>
>> sendevent NOTIFY
>> profile: internal
>> event-string: resync
>> user: 1000
>> host: 192.168.1.121
>> content-type: application/simple-message-summary
>>
>> However, if AUTH is required, why does FS send the wrong information to
>> the SPA?
>>
>> On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija <oseslija at gmail.com>
>> wrote:
>> > You? should not authenticate those NOTIFYs (this will work only with
>> > SPA9000
>> > afaik). The option to change for this is in EXT tabs:
>> >
>> > Auth Resync-Reboot: No
>> >
>> > Also, FSs code will do a reboot of a phone, not resync (it sends
>> > reboot_now
>> > event). For that to work a patch is required.
>> >
>> > I've just tried to reboot my 942 (rev 16506) and it definitely works.
>> >
>> > Regards,
>> > Ognjen
>> >
>> >
>> > On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
>> > <mcampbellsmith at gmail.com> wrote:
>> >>
>> >> Thanks Brian.. this still does not work. ?Maybe I need to open a Jira?
>> >> ?Notice the username in the authorization field. ?It should be 1000.
>> >>
>> >> Cheers
>> >> Mark
>> >>
>> >> freeswitch at internal> sofia status profile internal user
>> >> 1000 at 192.168.1.120
>> >>
>> >> Registrations:
>> >>
>> >>
>> >> =================================================================================================
>> >> Call-ID: ? ? ? ?bd783b73-66877627 at 192.168.1.121
>> >> User: ? ? ? ? ? 1000 at 192.168.1.120
>> >> Contact: ? ? ? ?1000 <sip:1000 at 192.168.1.121:5060>
>> >> Agent: ? ? ? ? ?Linksys/PAP2T-5.1.6(LS)
>> >> Status: ? ? ? ? Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
>> >> Host: ? ? ? ? ? freeswitch
>> >> IP: ? ? ? ? ? ? 192.168.1.121
>> >> Port: ? ? ? ? ? 5060
>> >> Auth-User: ? ? ?1000
>> >> Auth-Realm: ? ? 192.168.1.120
>> >> MWI-Account: ? ?1000 at 192.168.1.120
>> >>
>> >>
>> >>
>> >> =================================================================================================
>> >>
>> >> freeswitch at internal> sofia profile internal flush_inbound_reg
>> >> bd783b73-66877627 at 192.168.1.121 reboot
>> >> +OK rebooting all registrations matching specified call_id
>> >>
>> >> freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060 at
>> >> 23:55:49.012627:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> >> ? Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
>> >> ? Max-Forwards: 70
>> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> ? To: <sip:1000 at 192.168.1.120>
>> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> ? CSeq: 126070338 NOTIFY
>> >> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> >> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> ? Supported: timer, precondition, path, replaces
>> >> ? Event: reboot_now
>> >> ? Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> >> include-session-description, presence.winfo, message-summary, refer
>> >> ? Subscription-State: terminated;reason=timeout
>> >> ? Content-Type: application/simple-message-summary
>> >> ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? SIP/2.0 401 Unauthorized
>> >> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> ? CSeq: 126070338 NOTIFY
>> >> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
>> >> ? Server: Linksys/PAP2T-5.1.6(LS)
>> >> ? WWW-Authenticate: Digest realm="192.168.1.120", nonce="8e54805b",
>> >> qop="auth", algorithm=md5
>> >> ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> >> ? Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
>> >> ? Max-Forwards: 70
>> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> ? To: <sip:1000 at 192.168.1.120>
>> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> ? CSeq: 126070339 NOTIFY
>> >> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> >> ? Expires: 3590
>> >> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> ? Supported: timer, precondition, path, replaces
>> >> ? Event: reboot_now
>> >> ? Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> >> include-session-description, presence.winfo, message-summary, refer
>> >> ? Subscription-State: terminated;reason=timeout
>> >> ? Authorization: Digest username="1115633124", realm="192.168.1.120",
>> >> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
>> >> uri="sip:1000 at 192.168.1.121:5060",
>> >> response="747b4d04544c84535dbbd987f2999ca7", qop=auth, nc=00000001
>> >> ? Content-Type: application/simple-message-summary
>> >> ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? SIP/2.0 401 Unauthorized
>> >> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> ? CSeq: 126070339 NOTIFY
>> >> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
>> >> ? Server: Linksys/PAP2T-5.1.6(LS)
>> >> ? WWW-Authenticate: Digest realm="192.168.1.120", nonce="5339c7ba",
>> >> qop="auth", algorithm=md5
>> >> ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >>
>> >> On Mon, Jan 25, 2010 at 4:46 AM, Brian West <brian at freeswitch.org>
>> >> wrote:
>> >> > or sofia profile xxx flush_inbound_reg callid reboot
>> >> > callid you can get from sofia status profile xxx
>> >> > /b
>> >> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
>> >> >
>> >> > Actually I just
>> >> > found?http://wiki.freeswitch.org/wiki/Mod_event_socket
>> >> >
>> >> > If I telnet to FS as described
>> >> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do I
>> >> > just need to enter somthing like:
>> >> >
>> >> > sendevent NOTIFY
>> >> > profile: internal
>> >> > event-string: resync
>> >> > user: 1000
>> >> > host: 192.168.1.121
>> >> > content-type: application/simple-message-summary
>> >> >
>> >> > where 192.168.1.121 is the ip address of one of the Linksys devices?
>> >> >
>> >> > I don't see any messages sent when I do this. ?What am I doing wrong?
>> >> >
>> >> > Thanks
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From nagalenoj at gmail.com  Wed Jan 27 05:14:47 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 27 Jan 2010 18:44:47 +0530
Subject: [Freeswitch-users] Event socket: filter delete isn't working
Message-ID: <fe67a3341001270514g2715de2ag6aa2e3b099fa7816@mail.gmail.com>

Dear friends,
   I've tried to delete the filter which I applied for an unique id. But, it
doesn't work. After executing 'filter delete', I am receiving the events
from that uuid.
   I used the command as 'filter delete unique-id
c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565'.

I did the following operations.
    Made call to the event socket.
    Registered events for all. (events plain all).
    Applied filter for the uuid. (filter unique-id
aa3cb8ea-0b2f-11df-9e84-fb15c3cd8565).
    I've got a new uuid by using create_uuid.
    Applied filter for this new uuid. (filter unique-id
c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565)
    Originated a call with that uuid.
    Now, I could receive events from both uuids. (Tested by giving DTMFs in
both end and checked unique-id in event header).
    Then, I wanted to delete a uuid from the filter. (filter delete
unique-id c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565).
    I thought, i won't receive the events from this deleted unique-id. But,
I received the dtmfs from both unique-id.

I'm using 'FreeSWITCH Version 1.0.trunk (15982)'.
-- 
Regards,
Nagalenoj H.
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From dftoro at yahoo.com  Wed Jan 27 05:27:38 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Wed, 27 Jan 2010 05:27:38 -0800 (PST)
Subject: [Freeswitch-users] External Profile Problem
In-Reply-To: <BAY143-W20A3102CC2FF36EF1C170FA15D0@phx.gbl>
Message-ID: <443888.41110.qm@web33505.mail.mud.yahoo.com>

Hi, 

<action application="transfer" data="$0 XML default"/>

You must take into account that transfer application not "transfer" a call to destination, only transfer it to a new context

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer


Diego Toro
http://lacarretade.blogspot.com/


--- On Tue, 1/26/10, juan camilo ospina quintero <camilin2212 at hotmail.com> wrote:

> From: juan camilo ospina quintero <camilin2212 at hotmail.com>
> Subject: Re: [Freeswitch-users] External Profile Problem
> To: "freeswitch" <freeswitch-users at lists.freeswitch.org>
> Date: Tuesday, January 26, 2010, 11:06 PM
> 
> 
> 
> 
>  
> Hi
> 
> This works fine
> 
> <extension name="redirect">
> ?<condition field="network_addr"
> expression="^192\.168\.2\.9$"/>   
> 
> ?<condition field="destination_number"
> expression="^1(\d+)$">
> ?<action application="bridge"
> data="sofia/external/$0 at 127.0.0.1:5070"/>
> </extension>
> 
> this redirects from freeswitch to sailfin (127.0.0.1:5070),
> and is in default.xml, in the dialplan.
> 
> the problem is this
> 
>     <extension name="receive">
>       <condition field="network_addr"
> expression="^127\.0\.0\.1$"/>
>       <condition field="destination_number"
> expression="^1(\d+)$">
>         <action application="transfer"
> data="$0 XML default"/>
>       </condition>
>     </extension>
> this doesnt work, this configuration can be found in
> public.xml in the dialplan, the idea of
> this is that when a sip invite comes from sailfin
> (127.0.0.1) transfer the invite to the destination number
> 
> the both configurations above are the only configuration i
> have change from? the default instalation of 
> freeswitch.
> 
> i would like to have some hep with this thanks
> 
> here is the trace log again
> 
> 2010-01-26 20:14:29.512927 [NOTICE]
> switch_channel.c:602 New Channel sofia/external/1000
> at 192.168.2.9 [5177e93a-0ae1-11df-afc9-db39c681a2f1]
> 2010-01-26 20:14:29.512927 [NOTICE] sofia.c:3527 Hangup
> sofia/external/1000
> at 192.168.2.9 [CS_NEW] [MANDATORY_IE_MISSING]
> send 632 bytes to udp/[192.168.2.9]:5070 at
> 01:14:29.517927:
>   
> ------------------------------------------------------------------------
>    SIP/2.0 480 Temporarily Unavailable
>    Via: SIP/2.0/UDP
> 192.168.153.1:5070;branch=z9hG4bKdaacdd64d693615c451ab9db43f9c71c2626;received=192.168.2.9
>    From: <sip:1000
> at 192.168.2.9>;tag=g4xfbi12-3
>    To: <sip:1001
> at 192.168.2.9:5080>;tag=4r91165pvcycB
>    Call-Id: 192.168.153.1_3_3990383226484831353
>    Cseq: 1 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
> MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, refer
>    Reason:
> Q.850;cause=96;text="MANDATORY_IE_MISSING"
>    Content-Length: 0
>    
>   
> ------------------------------------------------------------------------
> 2010-01-26 20:14:29.525646 [NOTICE]
> switch_core_session.c:1086 Session 9 (sofia/external/1000
> at 192.168.2.9) Ended
> 2010-01-26 20:14:29.525646 [NOTICE]
> switch_core_session.c:1088 Close Channel sofia/external/1000
> at 192.168.2.9 [CS_DESTROY]
> 
> 
> 
> > Date: Tue, 26 Jan 2010 22:09:29 -0500
> > From: frank at carmickle.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] External Profile
> Problem
> > 
> > On Tue, Jan 26, juan camilo ospina quintero wrote:
> > > 
> > > Hi, 
> > > 
> > > im trying to establish a simple conference using
> freeswitch and sailfin, sailfin is
> > > and application server that works with
> SipSevlets.
> > > the all thing works as follow.
> > > 
> > > two softphone register with freeswitch, extension
> 1000 and 1001
> > > 1000 sends and invite to 1001, this invite goes
> to sailfin, i use this
> > > 
> > >     <extension name="redirect">
> > >       <condition
> field="network_addr"
> expression="^192\.168\.2\.9$"/>   
> 
> > >       <condition
> field="destination_number"
> expression="^1(\d+)$">
> > >     <action application="bridge"
> data="sofia/external/$0 at 127.0.0.1:5070"/>
> > 
> > And what is the external profile listening on? 
> Probably not the loopback address.  Set up another profile
> listening on 127.0.0.1 and bridge to that.
> > 
> > I could be off base here because you haven't given
> us very much info about your freeswitch configurations.
> > 
> > --FC
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>  		 	   		  
> Windows Live:  Friends
> get your Flickr, Yelp, and Digg updates when they e-mail
> you. 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From Prometheus001 at gmx.net  Wed Jan 27 07:01:47 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 27 Jan 2010 16:01:47 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
Message-ID: <4B60555B.2020004@gmx.net>

I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
to all contributors, excellent work!

In order to have more than 8 channels working, I have followed the
instructions in
http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
ist Debian 5.0R3)
It compiled well however when I start snd-dummy I only have
one-way-audio and my logs show

Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
/usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
working on a machine with 250HZ kernel
Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
obsolete setsockopt SO_BSDCOMPAT
Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
suppressed
Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
obsolete setsockopt SO_BSDCOMPAT
Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
suppressed
If I reinstall alsa from deb everything sworks fine again (of course
with the current limitations).

First question: Has anybody had this issue before? How can I solve this?

Second question:
As I do not need 64 channels or more: how do I manage, that Skype
instances 9..15 use a second instance of snd-dummy as addressed in the wiki?

Best regards
Peter


From jeff at jefflenk.com  Wed Jan 27 07:07:18 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 27 Jan 2010 09:07:18 -0600
Subject: [Freeswitch-users] Polycom buddy watch
Message-ID: <COL117-W58990B36F9ECF38B0E23FEB95D0@phx.gbl>


Hello,

 

Do the Polycom buddy watch presence updates work for "away" "busy" "Dnd" etc. I am running SIP 3.1.3.

 

FS is sending the notifies for when the other phones are in use and that works fine.

 

The Polys are not sending publish events to FS at all for the extended presence states as seen with "sofia profile internal siptrace on". Is this normal?

 

Thanks

 

Jeff
 		 	   		  
_________________________________________________________________
Your E-mail and More On-the-Go. Get Windows Live Hotmail Free.
http://clk.atdmt.com/GBL/go/196390709/direct/01/
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From codecomplete at free.fr  Wed Jan 27 07:20:23 2010
From: codecomplete at free.fr (Fred-145)
Date: Wed, 27 Jan 2010 07:20:23 -0800 (PST)
Subject: [Freeswitch-users]  Investigating one-way sound?
Message-ID: <27341219.post@talk.nabble.com>


Hello

With both a PC+XLite and a Siemens A580IP phone on the same LAN, sound is OK
both ways when I call from the XLite application, but when calling from the
Siemens, I can't hear sound coming from the XLite application. FWIW, both
phones are configured to use G711a and G711u, in this order.

To investigate, what kind of error message should I pay attention to in all
the messages that scroll through the console?

Thank you.
-- 
View this message in context: http://old.nabble.com/Investigating-one-way-sound--tp27341219p27341219.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From kristian.kielhofner at gmail.com  Wed Jan 27 07:20:14 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 27 Jan 2010 10:20:14 -0500
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
In-Reply-To: <27338355.post@talk.nabble.com>
References: <27338355.post@talk.nabble.com>
Message-ID: <2d9149cd1001270720q63ec12bfof4173343a26026b8@mail.gmail.com>

On Wed, Jan 27, 2010 at 7:38 AM, Fred-145 <codecomplete at free.fr> wrote:
>
> Hello
>
> Thanks to "directrtpsetup=yes", it appears that Asterisk can act as an SIP
> proxy and a PBX at the same time, ie. RTP packets will flow directly between
> the two SIP end-points with Asterisk still being able to provide PBX
> services like call transfer, MoH, etc.
> The point is to lower latency for packets to reach their final destination
> and lower CPU load on the Freeswitch server.
>
> Does Freeswitch offer the same feature, or must RTP packets always go
> through the Freeswitch servers?

  This isn't exactly true...  If you do some research you'll find that
OEJ (the author of the Asterisk SIP channel driver) does NOT recommend
the use of directrtpsetup because its use hasn't been tested with many
scenarios (including some you describe, I'm sure).  AFAIK it's still
marked "experimental".  The last time this came up on Asterisk-users
here was the exchange:

Kevin P. Fleming wrote:
> Kristian Kielhofner wrote:
>
>
>> What version of Asterisk is this?  Last I heard (from Olle) this
>> option was very experimental and should not be used on production
>> systems.
>>
>
> He even helpfully documented it that way in the sip.conf.sample file,
> along with a list of (known) cases where it will fail, although there
> are probably plenty more.
>
>

  So Kevin Fleming agrees.

  Needless to say FreeSWITCH has bypass_media.  Last I heard
FreeSWITCH will re-INVITE itself back in the media path if you put a
call on hold (for example) but it won't go back to bypass_media when
you take the call off hold.  Anthony said it would probably take about
$500 in bounty to get that functionality.  Then again, maybe he just
decided to do anyway it because he thought it was cool.  That's been
known to happen too.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From brian at freeswitch.org  Wed Jan 27 07:22:53 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 27 Jan 2010 09:22:53 -0600
Subject: [Freeswitch-users] Polycom buddy watch
In-Reply-To: <COL117-W58990B36F9ECF38B0E23FEB95D0@phx.gbl>
References: <COL117-W58990B36F9ECF38B0E23FEB95D0@phx.gbl>
Message-ID: <3D42D858-4933-4DBC-89F7-14A83A77726B@freeswitch.org>

I'm going to guess we are missing some type of outbound subscription similar to how we do it in sofia_sla.c for the presence events.

/b

On Jan 27, 2010, at 9:07 AM, Jeff Lenk wrote:

> Hello,
>  
> Do the Polycom buddy watch presence updates work for "away" "busy" "Dnd" etc. I am running SIP 3.1.3.
>  
> FS is sending the notifies for when the other phones are in use and that works fine.
>  
> The Polys are not sending publish events to FS at all for the extended presence states as seen with "sofia profile internal siptrace on". Is this normal?
>  
> Thanks
>  
> Jeff

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From gmaruzz at celliax.org  Wed Jan 27 07:26:08 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 27 Jan 2010 16:26:08 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B60555B.2020004@gmx.net>
References: <4B60555B.2020004@gmx.net>
Message-ID: <7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com>

Ciao Peter

one instance of snd-dummy "customized" is enough for 64 instances of
skype clients, no need (and do not works) with more instances of
snd-dummy-customized.

Maybe you got the one-way problem because of kernel at 250HZ (don't
know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
8.04).

Or maybe you have to check and modify which sound devices the skype
clients are using (try to check that with snd-summy-custom loaded,
maybe with the ssh -X trick (as in the wiki page).

To load more than one snd-dummy-original (the non modified one), you
do this with the modprobe command, as in:

rmmod snd-dummy
modprobe snd-dummy enable=1,1,1

this command will enable three instances of snd-dummy original, so
you'll have three fake soundcards, and you'll have to setup each group
of 8 skype instances to use sound devices from one fake soundcard, RG:
no more than 8 skype client instances can use one instance of fake
soundcard.

Also, please update the mod_skypiax code (svn up in its directory) I
just committed some improvements.

If you have any other doubts, or need more info, don't hesitate to
write the mailing list again,

ciao for now,

-giovanni



On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
> to all contributors, excellent work!
>
> In order to have more than 8 channels working, I have followed the
> instructions in
> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
> ist Debian 5.0R3)
> It compiled well however when I start snd-dummy I only have
> one-way-audio and my logs show
>
> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
> working on a machine with 250HZ kernel
> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
> obsolete setsockopt SO_BSDCOMPAT
> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
> suppressed
> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
> obsolete setsockopt SO_BSDCOMPAT
> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
> suppressed
> If I reinstall alsa from deb everything sworks fine again (of course
> with the current limitations).
>
> First question: Has anybody had this issue before? How can I solve this?
>
> Second question:
> As I do not need 64 channels or more: how do I manage, that Skype
> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From brian at freeswitch.org  Wed Jan 27 07:30:41 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 27 Jan 2010 09:30:41 -0600
Subject: [Freeswitch-users] Investigating one-way sound?
In-Reply-To: <27341219.post@talk.nabble.com>
References: <27341219.post@talk.nabble.com>
Message-ID: <22DD670E-B920-4328-9939-56447375D5C7@freeswitch.org>

Looking at the SIP traffic, paying special attention to the SDP.  I'm going to guess right off the X-Lite is putting its public IP and since maybe your NAT router can't hair pin the media you get on way media.  If you go into the settings make sure not to be using the Globally Discovered IP and use the Local IP in the network options.

/b

On Jan 27, 2010, at 9:20 AM, Fred-145 wrote:

> To investigate, what kind of error message should I pay attention to in all
> the messages that scroll through the console?



From mouncifbb at gmail.com  Wed Jan 27 07:31:45 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Wed, 27 Jan 2010 10:31:45 -0500
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <5F37CA2E-129F-4082-B304-D50D0E1A4FAF@freeswitch.org>
References: <201001052125.06909.sos@sokhapkin.dyndns.org>
	<4B4417B3.9090807@aastral.net>
	<201001060620.13735.sos@sokhapkin.dyndns.org>
	<8ccbff061001060859p59aa6b5bw6f52275650f9138@mail.gmail.com>
	<ebbc2d551001061416l25c17143gafe62de35cc6dee0@mail.gmail.com>
	<5F37CA2E-129F-4082-B304-D50D0E1A4FAF@freeswitch.org>
Message-ID: <ade741d61001270731w4925d05mb5f3e87e8dec94c9@mail.gmail.com>

yeah it will be nice if mod_nibble can do call minimum and rounding, same
way opensource a2billing doing for asterisk.
I think without this feature the module will not be useful for business use.
a common bill minimum/increments should in a form of: 6/6 for US , 6/30 for
A-Z and 60/60 for Mexico.
1/1 type is not offered by Tier1 Telecom providers.

BTW Freeswitch Rocks Rocks!!!!! It's The best IP telephony App I ever used!!



On Wed, Jan 6, 2010 at 5:33 PM, Brian West <brian at freeswitch.org> wrote:

> Or better yet take over the maint. of the module if its making you money
> give back by providing some help to the project... its the ultimate way to
> give back.
>
> /b
>
> On Jan 6, 2010, at 4:16 PM, Jo?o Mesquita wrote:
>
> > Why can't someone just sponsor some love to the module? The author has
> his email on the header or open a Jira asking stuff so we have nibblebill
> more mature.
> >
> > Jo?o Mesquita
> > FreeSWITCH? Solutions
> > t: +1 (646) 4959927
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From camilin2212 at hotmail.com  Wed Jan 27 07:48:56 2010
From: camilin2212 at hotmail.com (juan camilo ospina quintero)
Date: Wed, 27 Jan 2010 10:48:56 -0500
Subject: [Freeswitch-users] External Profile Problem
In-Reply-To: <443888.41110.qm@web33505.mail.mud.yahoo.com>
References: <BAY143-W20A3102CC2FF36EF1C170FA15D0@phx.gbl>,
	<443888.41110.qm@web33505.mail.mud.yahoo.com>
Message-ID: <BAY143-W267C2F235ED9FCFA619FA1A15D0@phx.gbl>


hi thanks

sorry but i dont really understand what a context is.

so, when i put <action application="transfer" data="$0 XML default"/>
what does it really does, what it means that transfer to new context,


bye

> Date: Wed, 27 Jan 2010 05:27:38 -0800
> From: dftoro at yahoo.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] External Profile Problem
> 
> Hi, 
> 
> <action application="transfer" data="$0 XML default"/>
> 
> You must take into account that transfer application not "transfer" a call to destination, only transfer it to a new context
> 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer
> 
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 
> 
> --- On Tue, 1/26/10, juan camilo ospina quintero <camilin2212 at hotmail.com> wrote:
> 
> > From: juan camilo ospina quintero <camilin2212 at hotmail.com>
> > Subject: Re: [Freeswitch-users] External Profile Problem
> > To: "freeswitch" <freeswitch-users at lists.freeswitch.org>
> > Date: Tuesday, January 26, 2010, 11:06 PM
> > 
> > 
> > 
> > 
> >  
> > Hi
> > 
> > This works fine
> > 
> > <extension name="redirect">
> >  <condition field="network_addr"
> > expression="^192\.168\.2\.9$"/>   
> > 
> >  <condition field="destination_number"
> > expression="^1(\d+)$">
> >  <action application="bridge"
> > data="sofia/external/$0 at 127.0.0.1:5070"/>
> > </extension>
> > 
> > this redirects from freeswitch to sailfin (127.0.0.1:5070),
> > and is in default.xml, in the dialplan.
> > 
> > the problem is this
> > 
> >     <extension name="receive">
> >       <condition field="network_addr"
> > expression="^127\.0\.0\.1$"/>
> >       <condition field="destination_number"
> > expression="^1(\d+)$">
> >         <action application="transfer"
> > data="$0 XML default"/>
> >       </condition>
> >     </extension>
> > this doesnt work, this configuration can be found in
> > public.xml in the dialplan, the idea of
> > this is that when a sip invite comes from sailfin
> > (127.0.0.1) transfer the invite to the destination number
> > 
> > the both configurations above are the only configuration i
> > have change from  the default instalation of 
> > freeswitch.
> > 
> > i would like to have some hep with this thanks
> > 
> > here is the trace log again
> > 
> > 2010-01-26 20:14:29.512927 [NOTICE]
> > switch_channel.c:602 New Channel sofia/external/1000
> > at 192.168.2.9 [5177e93a-0ae1-11df-afc9-db39c681a2f1]
> > 2010-01-26 20:14:29.512927 [NOTICE] sofia.c:3527 Hangup
> > sofia/external/1000
> > at 192.168.2.9 [CS_NEW] [MANDATORY_IE_MISSING]
> > send 632 bytes to udp/[192.168.2.9]:5070 at
> > 01:14:29.517927:
> >   
> > ------------------------------------------------------------------------
> >    SIP/2.0 480 Temporarily Unavailable
> >    Via: SIP/2.0/UDP
> > 192.168.153.1:5070;branch=z9hG4bKdaacdd64d693615c451ab9db43f9c71c2626;received=192.168.2.9
> >    From: <sip:1000
> > at 192.168.2.9>;tag=g4xfbi12-3
> >    To: <sip:1001
> > at 192.168.2.9:5080>;tag=4r91165pvcycB
> >    Call-Id: 192.168.153.1_3_3990383226484831353
> >    Cseq: 1 INVITE
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
> >    Accept: application/sdp
> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
> > MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
> >    Supported: timer, precondition, path, replaces
> >    Allow-Events: talk, refer
> >    Reason:
> > Q.850;cause=96;text="MANDATORY_IE_MISSING"
> >    Content-Length: 0
> >    
> >   
> > ------------------------------------------------------------------------
> > 2010-01-26 20:14:29.525646 [NOTICE]
> > switch_core_session.c:1086 Session 9 (sofia/external/1000
> > at 192.168.2.9) Ended
> > 2010-01-26 20:14:29.525646 [NOTICE]
> > switch_core_session.c:1088 Close Channel sofia/external/1000
> > at 192.168.2.9 [CS_DESTROY]
> > 
> > 
> > 
> > > Date: Tue, 26 Jan 2010 22:09:29 -0500
> > > From: frank at carmickle.com
> > > To: freeswitch-users at lists.freeswitch.org
> > > Subject: Re: [Freeswitch-users] External Profile
> > Problem
> > > 
> > > On Tue, Jan 26, juan camilo ospina quintero wrote:
> > > > 
> > > > Hi, 
> > > > 
> > > > im trying to establish a simple conference using
> > freeswitch and sailfin, sailfin is
> > > > and application server that works with
> > SipSevlets.
> > > > the all thing works as follow.
> > > > 
> > > > two softphone register with freeswitch, extension
> > 1000 and 1001
> > > > 1000 sends and invite to 1001, this invite goes
> > to sailfin, i use this
> > > > 
> > > >     <extension name="redirect">
> > > >       <condition
> > field="network_addr"
> > expression="^192\.168\.2\.9$"/>   
> > 
> > > >       <condition
> > field="destination_number"
> > expression="^1(\d+)$">
> > > >     <action application="bridge"
> > data="sofia/external/$0 at 127.0.0.1:5070"/>
> > > 
> > > And what is the external profile listening on? 
> > Probably not the loopback address.  Set up another profile
> > listening on 127.0.0.1 and bridge to that.
> > > 
> > > I could be off base here because you haven't given
> > us very much info about your freeswitch configurations.
> > > 
> > > --FC
> > > 
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >  		 	   		  
> > Windows Live:  Friends
> > get your Flickr, Yelp, and Digg updates when they e-mail
> > you. 
> > 
> > -----Inline Attachment Follows-----
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> 
> 
>       
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
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From gmaruzz at celliax.org  Wed Jan 27 08:23:15 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 27 Jan 2010 17:23:15 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com>
Message-ID: <7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com>

This warning is harmless:

Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
obsolete setsockopt SO_BSDCOMPAT

On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> Ciao Peter
>
> one instance of snd-dummy "customized" is enough for 64 instances of
> skype clients, no need (and do not works) with more instances of
> snd-dummy-customized.
>
> Maybe you got the one-way problem because of kernel at 250HZ (don't
> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
> 8.04).
>
> Or maybe you have to check and modify which sound devices the skype
> clients are using (try to check that with snd-summy-custom loaded,
> maybe with the ssh -X trick (as in the wiki page).
>
> To load more than one snd-dummy-original (the non modified one), you
> do this with the modprobe command, as in:
>
> rmmod snd-dummy
> modprobe snd-dummy enable=1,1,1
>
> this command will enable three instances of snd-dummy original, so
> you'll have three fake soundcards, and you'll have to setup each group
> of 8 skype instances to use sound devices from one fake soundcard, RG:
> no more than 8 skype client instances can use one instance of fake
> soundcard.
>
> Also, please update the mod_skypiax code (svn up in its directory) I
> just committed some improvements.
>
> If you have any other doubts, or need more info, don't hesitate to
> write the mailing list again,
>
> ciao for now,
>
> -giovanni
>
>
>
> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>> to all contributors, excellent work!
>>
>> In order to have more than 8 channels working, I have followed the
>> instructions in
>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>> ist Debian 5.0R3)
>> It compiled well however when I start snd-dummy I only have
>> one-way-audio and my logs show
>>
>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>> working on a machine with 250HZ kernel
>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>> obsolete setsockopt SO_BSDCOMPAT
>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>> suppressed
>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>> obsolete setsockopt SO_BSDCOMPAT
>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>> suppressed
>> If I reinstall alsa from deb everything sworks fine again (of course
>> with the current limitations).
>>
>> First question: Has anybody had this issue before? How can I solve this?
>>
>> Second question:
>> As I do not need 64 channels or more: how do I manage, that Skype
>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>
>> Best regards
>> Peter
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From anthony.minessale at gmail.com  Wed Jan 27 08:45:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 10:45:56 -0600
Subject: [Freeswitch-users] mod_spidermonkey memory leak?
In-Reply-To: <20100126120711.92321.qmail@web10512.mail.ogk.yahoo.co.jp>
References: <20100126120711.92321.qmail@web10512.mail.ogk.yahoo.co.jp>
Message-ID: <191c3a031001270845v20f30563w64d7ce9b0510f21a@mail.gmail.com>

if you suspect a memory leak
can you please update to the latest SVN trunk (over 1000 revs higher than
yours)
then reproduce the problem under valgrind and send the report in a jira
http://jira.freeswitch.org

valgrind --tool=memcheck --log-file=vg.log --leak-check=full
--leak-resolution=high --show-reachable=yes
/usr/local/freeswitch/bin/freeswitch -vg



2010/1/26 ???? ??? <matomoya at yahoo.co.jp>

> I have tested the SVN trunk 15341.
> When the following scripts are executed, it seems to do the memory leak.
> Is the mistake found in the following scripts?
> Or Is there a problem in mod_spidermonkey?
>
> -- test script --
> session.execute("ring_ready");
> session.answer();
> session.setVariable("ringback", "%(1000, 2000, 440, 460)");
>
> var bleg = new Session();
> var sound_wav = "sounds/test01.wav";
> var sound_leg = "both";
> var op = "api_on_answer=uuid_broadcast "+session.uuid+" "+sound_wav+"
> "+sound_leg;
> var ret = bleg.originate(session, "{"+op+"}" +
> "sofia/gateway/profile0_gateway1/1000");
> if(!session.ready()){
>  return;
> }
>
>
> if(!ret){ // bleg not answered.
>  var sound_wav = "sounds/test02.wav";
>  session.streamFile(sound_wav);
>  if(session.ready()){
>    session.hangup();
>  }
>  return;
> }
>
> if(bleg.ready()){
>  bridge(session, bleg);
> }
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From Prometheus001 at gmx.net  Wed Jan 27 08:58:16 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 27 Jan 2010 17:58:16 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com>
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com>
Message-ID: <4B6070A8.6050607@gmx.net>

Thanks Giovanni,

I think there may be the problem, that I have 2 sound devices now:
- Dummy PCM (hw0:0) (this is from debian install)
- Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
since I compiled alsa newly)

I tried both, but both do not work. How do I get rid of the old alsa device?
By the way: I uninstalled Alsa before I installed the new driver
(apt-get remove alsa-utils alsa-base).

Best regards
Peter


Giovanni Maruzzelli schrieb:
> This warning is harmless:
>
> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
> obsolete setsockopt SO_BSDCOMPAT
>
> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
> <gmaruzz at celliax.org> wrote:
>   
>> Ciao Peter
>>
>> one instance of snd-dummy "customized" is enough for 64 instances of
>> skype clients, no need (and do not works) with more instances of
>> snd-dummy-customized.
>>
>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>> 8.04).
>>
>> Or maybe you have to check and modify which sound devices the skype
>> clients are using (try to check that with snd-summy-custom loaded,
>> maybe with the ssh -X trick (as in the wiki page).
>>
>> To load more than one snd-dummy-original (the non modified one), you
>> do this with the modprobe command, as in:
>>
>> rmmod snd-dummy
>> modprobe snd-dummy enable=1,1,1
>>
>> this command will enable three instances of snd-dummy original, so
>> you'll have three fake soundcards, and you'll have to setup each group
>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>> no more than 8 skype client instances can use one instance of fake
>> soundcard.
>>
>> Also, please update the mod_skypiax code (svn up in its directory) I
>> just committed some improvements.
>>
>> If you have any other doubts, or need more info, don't hesitate to
>> write the mailing list again,
>>
>> ciao for now,
>>
>> -giovanni
>>
>>
>>
>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>     
>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>> to all contributors, excellent work!
>>>
>>> In order to have more than 8 channels working, I have followed the
>>> instructions in
>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>> ist Debian 5.0R3)
>>> It compiled well however when I start snd-dummy I only have
>>> one-way-audio and my logs show
>>>
>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>> working on a machine with 250HZ kernel
>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>> obsolete setsockopt SO_BSDCOMPAT
>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>> suppressed
>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>> obsolete setsockopt SO_BSDCOMPAT
>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>> suppressed
>>> If I reinstall alsa from deb everything sworks fine again (of course
>>> with the current limitations).
>>>
>>> First question: Has anybody had this issue before? How can I solve this?
>>>
>>> Second question:
>>> As I do not need 64 channels or more: how do I manage, that Skype
>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>
>>> Best regards
>>> Peter
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>>     
>
>
>
>   


From gmaruzz at celliax.org  Wed Jan 27 09:07:54 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 27 Jan 2010 18:07:54 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B6070A8.6050607@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com> 
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com> 
	<4B6070A8.6050607@gmx.net>
Message-ID: <7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com>

I don't think you got two snd-dummy loaded (but maybe yes)
what's the output of:

aplay -l

?

If instead you are referring to the choices that skype clients offers
you in the "set audio devices" window, choose Dummy PCM (hw0:0)

Eg: not the "default", but the "hardware" one


On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Thanks Giovanni,
>
> I think there may be the problem, that I have 2 sound devices now:
> - Dummy PCM (hw0:0) (this is from debian install)
> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
> since I compiled alsa newly)
>
> I tried both, but both do not work. How do I get rid of the old alsa device?
> By the way: I uninstalled Alsa before I installed the new driver
> (apt-get remove alsa-utils alsa-base).
>
> Best regards
> Peter
>
>
> Giovanni Maruzzelli schrieb:
>> This warning is harmless:
>>
>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>> obsolete setsockopt SO_BSDCOMPAT
>>
>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>> <gmaruzz at celliax.org> wrote:
>>
>>> Ciao Peter
>>>
>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>> skype clients, no need (and do not works) with more instances of
>>> snd-dummy-customized.
>>>
>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>> 8.04).
>>>
>>> Or maybe you have to check and modify which sound devices the skype
>>> clients are using (try to check that with snd-summy-custom loaded,
>>> maybe with the ssh -X trick (as in the wiki page).
>>>
>>> To load more than one snd-dummy-original (the non modified one), you
>>> do this with the modprobe command, as in:
>>>
>>> rmmod snd-dummy
>>> modprobe snd-dummy enable=1,1,1
>>>
>>> this command will enable three instances of snd-dummy original, so
>>> you'll have three fake soundcards, and you'll have to setup each group
>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>> no more than 8 skype client instances can use one instance of fake
>>> soundcard.
>>>
>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>> just committed some improvements.
>>>
>>> If you have any other doubts, or need more info, don't hesitate to
>>> write the mailing list again,
>>>
>>> ciao for now,
>>>
>>> -giovanni
>>>
>>>
>>>
>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>> to all contributors, excellent work!
>>>>
>>>> In order to have more than 8 channels working, I have followed the
>>>> instructions in
>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>> ist Debian 5.0R3)
>>>> It compiled well however when I start snd-dummy I only have
>>>> one-way-audio and my logs show
>>>>
>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>> working on a machine with 250HZ kernel
>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>> obsolete setsockopt SO_BSDCOMPAT
>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>> suppressed
>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>> obsolete setsockopt SO_BSDCOMPAT
>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>> suppressed
>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>> with the current limitations).
>>>>
>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>
>>>> Second question:
>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>>
>>
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From jeff at jefflenk.com  Wed Jan 27 09:11:00 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 27 Jan 2010 11:11:00 -0600
Subject: [Freeswitch-users] Polycom buddy watch
In-Reply-To: <3D42D858-4933-4DBC-89F7-14A83A77726B@freeswitch.org>
References: <COL117-W58990B36F9ECF38B0E23FEB95D0@phx.gbl>,
	<3D42D858-4933-4DBC-89F7-14A83A77726B@freeswitch.org>
Message-ID: <COL117-W31E21D3B622F7F39375F70B95D0@phx.gbl>


Thanks Brian,

 

I was just wondering if anyone else sees the same thing so I can eliminate whether I have it configured wrong.

 

Do others use the Buddy watch feature with Polycom and have the "Busy" "DND" or "Away" extended statuses working?

 

-Jeff
 


From: brian at freeswitch.org
Date: Wed, 27 Jan 2010 09:22:53 -0600
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom buddy watch

I'm going to guess we are missing some type of outbound subscription similar to how we do it in sofia_sla.c for the presence events.


/b



On Jan 27, 2010, at 9:07 AM, Jeff Lenk wrote:
Hello,
 
Do the Polycom buddy watch presence updates work for "away" "busy" "Dnd" etc. I am running SIP 3.1.3.
 
FS is sending the notifies for when the other phones are in use and that works fine.
 
The Polys are not sending publish events to FS at all for the extended presence states as seen with "sofia profile internal siptrace on". Is this normal?
 
Thanks
 
Jeff
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with powerful SPAM protection.
http://clk.atdmt.com/GBL/go/196390707/direct/01/
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From jcasale at activenetwerx.com  Wed Jan 27 09:19:47 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 27 Jan 2010 17:19:47 +0000
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90B9E04@prato.activenetwerx.local>

Anyone running this with their latest 2.2.1 release successfully
using Digium analog TDM cards? Is it known to work fine, or possibly
not been tested yet?

Thanks,
jlc


From freeswitch at aastral.net  Wed Jan 27 09:31:11 2010
From: freeswitch at aastral.net (Bill W)
Date: Wed, 27 Jan 2010 12:31:11 -0500
Subject: [Freeswitch-users] Strategies for reliably detecting nat on B-leg?
Message-ID: <4B60785F.6030505@aastral.net>

Hey All,

I know FreeSWITCH has the agressive-nat-detection parameter for sofia 
configs which will detect NAT on an incoming call.  So we know if the 
A-leg is natted.  The question is, are there any reliable ways to detect 
nat at the destination before bridging that call?

One could assume that if the destination is a gateway, then it would be 
okay to bypass-media.

Also, one could check the sofia registry and assume that if an endpoint 
is registered to FreeSWITCH that it is NATted and therefore no point in 
trying to bypass media.

But neither of these options seems 100% reliable.

Thoughts?  Suggestions?

Thanks!
Bill W.


From Prometheus001 at gmx.net  Wed Jan 27 09:35:00 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 27 Jan 2010 18:35:00 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com>
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com>
	<4B6070A8.6050607@gmx.net>
	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com>
Message-ID: <4B607944.4040700@gmx.net>

Her's the output:

skype:~# aplay -l
bash: aplay: command not found

Giovanni Maruzzelli schrieb:
> I don't think you got two snd-dummy loaded (but maybe yes)
> what's the output of:
>
> aplay -l
>
> ?
>
> If instead you are referring to the choices that skype clients offers
> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>
> Eg: not the "default", but the "hardware" one
>
>
> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> Thanks Giovanni,
>>
>> I think there may be the problem, that I have 2 sound devices now:
>> - Dummy PCM (hw0:0) (this is from debian install)
>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>> since I compiled alsa newly)
>>
>> I tried both, but both do not work. How do I get rid of the old alsa device?
>> By the way: I uninstalled Alsa before I installed the new driver
>> (apt-get remove alsa-utils alsa-base).
>>
>> Best regards
>> Peter
>>
>>
>> Giovanni Maruzzelli schrieb:
>>     
>>> This warning is harmless:
>>>
>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>> obsolete setsockopt SO_BSDCOMPAT
>>>
>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>> <gmaruzz at celliax.org> wrote:
>>>
>>>       
>>>> Ciao Peter
>>>>
>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>> skype clients, no need (and do not works) with more instances of
>>>> snd-dummy-customized.
>>>>
>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>> 8.04).
>>>>
>>>> Or maybe you have to check and modify which sound devices the skype
>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>
>>>> To load more than one snd-dummy-original (the non modified one), you
>>>> do this with the modprobe command, as in:
>>>>
>>>> rmmod snd-dummy
>>>> modprobe snd-dummy enable=1,1,1
>>>>
>>>> this command will enable three instances of snd-dummy original, so
>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>> no more than 8 skype client instances can use one instance of fake
>>>> soundcard.
>>>>
>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>> just committed some improvements.
>>>>
>>>> If you have any other doubts, or need more info, don't hesitate to
>>>> write the mailing list again,
>>>>
>>>> ciao for now,
>>>>
>>>> -giovanni
>>>>
>>>>
>>>>
>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>
>>>>         
>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>> to all contributors, excellent work!
>>>>>
>>>>> In order to have more than 8 channels working, I have followed the
>>>>> instructions in
>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>> ist Debian 5.0R3)
>>>>> It compiled well however when I start snd-dummy I only have
>>>>> one-way-audio and my logs show
>>>>>
>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>> working on a machine with 250HZ kernel
>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>> suppressed
>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>> suppressed
>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>> with the current limitations).
>>>>>
>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>
>>>>> Second question:
>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>           
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>>
>>>>         
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   


From moises.silva at gmail.com  Wed Jan 27 09:46:15 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 27 Jan 2010 12:46:15 -0500
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E90B9E04@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E90B9E04@prato.activenetwerx.local>
Message-ID: <c4d05cbe1001270946r39532b7al8290f64442415f4@mail.gmail.com>

On Wed, Jan 27, 2010 at 12:19 PM, Joseph L. Casale <
jcasale at activenetwerx.com> wrote:

> Anyone running this with their latest 2.2.1 release successfully
> using Digium analog TDM cards? Is it known to work fine, or possibly
> not been tested yet?


To my knowledge, neither one. The best answer I can give you is "it should
work", the dahdi driver API do not change often and previous versions of
DAHDI were tested just fine.

 --
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From brian at freeswitch.org  Wed Jan 27 09:50:59 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 27 Jan 2010 11:50:59 -0600
Subject: [Freeswitch-users] Strategies for reliably detecting nat on
	B-leg?
In-Reply-To: <4B60785F.6030505@aastral.net>
References: <4B60785F.6030505@aastral.net>
Message-ID: <FB139E6B-F685-491C-A672-16844878200E@freeswitch.org>

update to trunk.  and don't use agressive-nat, set local-network-acl, set the ext-rtp-ip and ext-sip-ip to autonat:x.x.x.x or if you're behind a natpmp or upnp router set it to auto-nat.  

It should just work.  Again you have no real way to know if the far end client never lies to you.  Which it should never do anyway.  Endpoints should know how to traverse their own nat and not leave it up to the registrar to figure it out.

/b

On Jan 27, 2010, at 11:31 AM, Bill W wrote:

> Thoughts?  Suggestions?



From Russell.Mosemann at cune.org  Wed Jan 27 10:00:58 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Wed, 27 Jan 2010 18:00:58 -0000
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E90B9E04@prato.activenetwerx.local>
Message-ID: <20100127180058.29470216DEA@cuneorg-email.cune.pri>

"Joseph L. Casale" <jcasale at activenetwerx.com> said:

> Anyone running this with their latest 2.2.1 release successfully
> using Digium analog TDM cards? Is it known to work fine, or possibly
> not been tested yet?

It's working OK here. We're actually using 2.2.1RC2 with a TE110P in a VZ
environment.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From Prometheus001 at gmx.net  Wed Jan 27 10:04:04 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 27 Jan 2010 19:04:04 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <4B607944.4040700@gmx.net>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com>
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com>
	<4B6070A8.6050607@gmx.net>
	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com>
	<4B607944.4040700@gmx.net>
Message-ID: <4B608014.4030902@gmx.net>

I installed alsa-utile,

now I get:

skype:/var/cache/apt/archives# aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
  Subdevices: 127/128
  Subdevice #0: subdevice #0
  Subdevice #1: subdevice #1
  Subdevice #2: subdevice #2
  Subdevice #3: subdevice #3
  Subdevice #4: subdevice #4
  Subdevice #5: subdevice #5
  Subdevice #6: subdevice #6
  Subdevice #7: subdevice #7
  Subdevice #8: subdevice #8
  Subdevice #9: subdevice #9
  Subdevice #10: subdevice #10
  Subdevice #11: subdevice #11
  Subdevice #12: subdevice #12
  Subdevice #13: subdevice #13
  Subdevice #14: subdevice #14
  Subdevice #15: subdevice #15
  Subdevice #16: subdevice #16
  Subdevice #17: subdevice #17
  Subdevice #18: subdevice #18
  Subdevice #19: subdevice #19
  Subdevice #20: subdevice #20
  Subdevice #21: subdevice #21
  Subdevice #22: subdevice #22
  Subdevice #23: subdevice #23
  Subdevice #24: subdevice #24
  Subdevice #25: subdevice #25
  Subdevice #26: subdevice #26
  Subdevice #27: subdevice #27
  Subdevice #28: subdevice #28
  Subdevice #29: subdevice #29
  Subdevice #30: subdevice #30
  Subdevice #31: subdevice #31
  Subdevice #32: subdevice #32
  Subdevice #33: subdevice #33
  Subdevice #34: subdevice #34
  Subdevice #35: subdevice #35
  Subdevice #36: subdevice #36
  Subdevice #37: subdevice #37
  Subdevice #38: subdevice #38
  Subdevice #39: subdevice #39
  Subdevice #40: subdevice #40
  Subdevice #41: subdevice #41
  Subdevice #42: subdevice #42
  Subdevice #43: subdevice #43
  Subdevice #44: subdevice #44
  Subdevice #45: subdevice #45
  Subdevice #46: subdevice #46
  Subdevice #47: subdevice #47
  Subdevice #48: subdevice #48
  Subdevice #49: subdevice #49
  Subdevice #50: subdevice #50
  Subdevice #51: subdevice #51
  Subdevice #52: subdevice #52
  Subdevice #53: subdevice #53
  Subdevice #54: subdevice #54
  Subdevice #55: subdevice #55
  Subdevice #56: subdevice #56
  Subdevice #57: subdevice #57
  Subdevice #58: subdevice #58
  Subdevice #59: subdevice #59
  Subdevice #60: subdevice #60
  Subdevice #61: subdevice #61
  Subdevice #62: subdevice #62
  Subdevice #63: subdevice #63
  Subdevice #64: subdevice #64
  Subdevice #65: subdevice #65
  Subdevice #66: subdevice #66
  Subdevice #67: subdevice #67
  Subdevice #68: subdevice #68
  Subdevice #69: subdevice #69
  Subdevice #70: subdevice #70
  Subdevice #71: subdevice #71
  Subdevice #72: subdevice #72
  Subdevice #73: subdevice #73
  Subdevice #74: subdevice #74
  Subdevice #75: subdevice #75
  Subdevice #76: subdevice #76
  Subdevice #77: subdevice #77
  Subdevice #78: subdevice #78
  Subdevice #79: subdevice #79
  Subdevice #80: subdevice #80
  Subdevice #81: subdevice #81
  Subdevice #82: subdevice #82
  Subdevice #83: subdevice #83
  Subdevice #84: subdevice #84
  Subdevice #85: subdevice #85
  Subdevice #86: subdevice #86
  Subdevice #87: subdevice #87
  Subdevice #88: subdevice #88
  Subdevice #89: subdevice #89
  Subdevice #90: subdevice #90
  Subdevice #91: subdevice #91
  Subdevice #92: subdevice #92
  Subdevice #93: subdevice #93
  Subdevice #94: subdevice #94
  Subdevice #95: subdevice #95
  Subdevice #96: subdevice #96
  Subdevice #97: subdevice #97
  Subdevice #98: subdevice #98
  Subdevice #99: subdevice #99
  Subdevice #100: subdevice #100
  Subdevice #101: subdevice #101
  Subdevice #102: subdevice #102
  Subdevice #103: subdevice #103
  Subdevice #104: subdevice #104
  Subdevice #105: subdevice #105
  Subdevice #106: subdevice #106
  Subdevice #107: subdevice #107
  Subdevice #108: subdevice #108
  Subdevice #109: subdevice #109
  Subdevice #110: subdevice #110
  Subdevice #111: subdevice #111
  Subdevice #112: subdevice #112
  Subdevice #113: subdevice #113
  Subdevice #114: subdevice #114
  Subdevice #115: subdevice #115
  Subdevice #116: subdevice #116
  Subdevice #117: subdevice #117
  Subdevice #118: subdevice #118
  Subdevice #119: subdevice #119
  Subdevice #120: subdevice #120
  Subdevice #121: subdevice #121
  Subdevice #122: subdevice #122
  Subdevice #123: subdevice #123
  Subdevice #124: subdevice #124
  Subdevice #125: subdevice #125
  Subdevice #126: subdevice #126
  Subdevice #127: subdevice #127


Peter P GMX schrieb:
> Her's the output:
>
> skype:~# aplay -l
> bash: aplay: command not found
>
> Giovanni Maruzzelli schrieb:
>   
>> I don't think you got two snd-dummy loaded (but maybe yes)
>> what's the output of:
>>
>> aplay -l
>>
>> ?
>>
>> If instead you are referring to the choices that skype clients offers
>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>
>> Eg: not the "default", but the "hardware" one
>>
>>
>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>   
>>     
>>> Thanks Giovanni,
>>>
>>> I think there may be the problem, that I have 2 sound devices now:
>>> - Dummy PCM (hw0:0) (this is from debian install)
>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>> since I compiled alsa newly)
>>>
>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>> By the way: I uninstalled Alsa before I installed the new driver
>>> (apt-get remove alsa-utils alsa-base).
>>>
>>> Best regards
>>> Peter
>>>
>>>
>>> Giovanni Maruzzelli schrieb:
>>>     
>>>       
>>>> This warning is harmless:
>>>>
>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>
>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>> <gmaruzz at celliax.org> wrote:
>>>>
>>>>       
>>>>         
>>>>> Ciao Peter
>>>>>
>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>> skype clients, no need (and do not works) with more instances of
>>>>> snd-dummy-customized.
>>>>>
>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>> 8.04).
>>>>>
>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>
>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>> do this with the modprobe command, as in:
>>>>>
>>>>> rmmod snd-dummy
>>>>> modprobe snd-dummy enable=1,1,1
>>>>>
>>>>> this command will enable three instances of snd-dummy original, so
>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>> no more than 8 skype client instances can use one instance of fake
>>>>> soundcard.
>>>>>
>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>> just committed some improvements.
>>>>>
>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>> write the mailing list again,
>>>>>
>>>>> ciao for now,
>>>>>
>>>>> -giovanni
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>
>>>>>         
>>>>>           
>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>> to all contributors, excellent work!
>>>>>>
>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>> instructions in
>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>> ist Debian 5.0R3)
>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>> one-way-audio and my logs show
>>>>>>
>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>> working on a machine with 250HZ kernel
>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>> suppressed
>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>> suppressed
>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>> with the current limitations).
>>>>>>
>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>
>>>>>> Second question:
>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>
>>>>>> Best regards
>>>>>> Peter
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>           
>>>>>>             
>>>>> --
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> Cell : +39-347-2665618
>>>>>
>>>>>
>>>>>         
>>>>>           
>>>>       
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>     
>>>       
>>
>>   
>>     
>
>   


From anthony.minessale at gmail.com  Wed Jan 27 10:05:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 12:05:10 -0600
Subject: [Freeswitch-users] Strategies for reliably detecting nat on
	B-leg?
In-Reply-To: <FB139E6B-F685-491C-A672-16844878200E@freeswitch.org>
References: <4B60785F.6030505@aastral.net>
	<FB139E6B-F685-491C-A672-16844878200E@freeswitch.org>
Message-ID: <191c3a031001271005t62631e77sf6d9ca406054ba00@mail.gmail.com>

also you can set
sip_sticky_contact=true
channel var which will make that session turn on nat lock in the b leg so
they can't change the contact to a nat addr

add it in {} to your dial string like

{sip_sticky_contact=true}sofia/internal/foo at bar.com




On Wed, Jan 27, 2010 at 11:50 AM, Brian West <brian at freeswitch.org> wrote:

> update to trunk.  and don't use agressive-nat, set local-network-acl, set
> the ext-rtp-ip and ext-sip-ip to autonat:x.x.x.x or if you're behind a
> natpmp or upnp router set it to auto-nat.
>
> It should just work.  Again you have no real way to know if the far end
> client never lies to you.  Which it should never do anyway.  Endpoints
> should know how to traverse their own nat and not leave it up to the
> registrar to figure it out.
>
> /b
>
> On Jan 27, 2010, at 11:31 AM, Bill W wrote:
>
> > Thoughts?  Suggestions?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From freeswitch at aastral.net  Wed Jan 27 10:44:23 2010
From: freeswitch at aastral.net (Bill W)
Date: Wed, 27 Jan 2010 13:44:23 -0500
Subject: [Freeswitch-users] Strategies for reliably detecting nat
	on	B-leg?
In-Reply-To: <191c3a031001271005t62631e77sf6d9ca406054ba00@mail.gmail.com>
References: <4B60785F.6030505@aastral.net>	<FB139E6B-F685-491C-A672-16844878200E@freeswitch.org>
	<191c3a031001271005t62631e77sf6d9ca406054ba00@mail.gmail.com>
Message-ID: <4B608987.9090606@aastral.net>

Thanks for the reply!

Just to make sure we're on the same page,  my FreeSWITCH sesrver has a 
public IP, and I'm trying to bypass media whenever possible to reduce my 
bandwidth usage.   My concern is trying to bypass media when one of the 
remote endpoints (b-leg) is behind NAT (since I can reliably detect nat 
with aggressive-nat on the A-leg).

Isn't local-network-acl and autonat:x.x.x.x for FS behind nat?

Does this new information change your responses?

Thanks again!
Bill



Anthony Minessale wrote:
> also you can set
> sip_sticky_contact=true
> channel var which will make that session turn on nat lock in the b leg 
> so they can't change the contact to a nat addr
> 
> add it in {} to your dial string like
> 
> {sip_sticky_contact=true}sofia/internal/foo at bar.com <mailto:foo at bar.com>
> 
> 
> 
> 
> On Wed, Jan 27, 2010 at 11:50 AM, Brian West <brian at freeswitch.org 
> <mailto:brian at freeswitch.org>> wrote:
> 
>     update to trunk.  and don't use agressive-nat, set
>     local-network-acl, set the ext-rtp-ip and ext-sip-ip to
>     autonat:x.x.x.x or if you're behind a natpmp or upnp router set it
>     to auto-nat.
> 
>     It should just work.  Again you have no real way to know if the far
>     end client never lies to you.  Which it should never do anyway.
>      Endpoints should know how to traverse their own nat and not leave
>     it up to the registrar to figure it out.
> 
>     /b
> 
>     On Jan 27, 2010, at 11:31 AM, Bill W wrote:
> 
>      > Thoughts?  Suggestions?
> 
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From nicolas at medularis.com  Wed Jan 27 10:49:19 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Wed, 27 Jan 2010 15:49:19 -0300
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <87f2f3b91001261449x3401f48eibae516d2b7abc8d4@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
	<e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
	<5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>
	<5d2828f1001201406p4154b98ald1af1c5c25f59337@mail.gmail.com>
	<1b46b4e81001210335l42baef16r8a2952aa5b92f6e6@mail.gmail.com>
	<87f2f3b91001211412h7aa0a84ageadb9557b869ca01@mail.gmail.com>
	<1b46b4e81001221322w5da04799s2ea86000c5c4a9a@mail.gmail.com>
	<87f2f3b91001261449x3401f48eibae516d2b7abc8d4@mail.gmail.com>
Message-ID: <1b46b4e81001271049u527be075n3439028e916181af@mail.gmail.com>

On the contrary, thank you and the whole FreeSWITCH team. Your software and
support are awesome, thank you very much!

Here's something else, a little guide on creating a simple click to call
application with FreeSWITCH:

- Part I: http://www.guayal.com/how-to-bridge-two-calls-with-freeswitch

- Part II: http://www.guayal.com/how-to-create-a-basic-click-to-call-app

I still have to write Part III and see if I keep going.


On Tue, Jan 26, 2010 at 7:49 PM, Michael Collins <msc at freeswitch.org> wrote:

> Thanks for your contributions! They are much appreciated.
> -MC
>
>
> On Fri, Jan 22, 2010 at 1:22 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> No problem, here it is:
>>
>> - http://wiki.freeswitch.org/wiki/Call_retry_based_on_hangup_cause
>>
>> It is linked from your reference (
>> http://wiki.freeswitch.org/wiki/Mod_lua#More_Samples).
>>
>> Sorry I didn't do it early, I hadn't seen your email.
>>
>> I also added another, more complete, example here (also linked):
>>
>> - http://wiki.freeswitch.org/wiki/Bridging_two_calls_with_retry
>>
>>
>>
>> On Thu, Jan 21, 2010 at 7:12 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Thu, Jan 21, 2010 at 3:35 AM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>>
>>>>
>>>> On Wed, Jan 20, 2010 at 7:06 PM, Mike van Lammeren <
>>>> mike at van.lammeren.net> wrote:
>>>>
>>>>> So, I've been reading about early media in the wiki, and have made a
>>>>> little progress, which leads to more questions.
>>>>>
>>>>> I understand now why a call is considered connected before one person
>>>>> has picked up the phone. I am also able to get my script to wait for the
>>>>> phone to be picked up, by setting the ignore_early_media variable when
>>>>> starting a new session, like this:
>>>>>
>>>>> customerSession =
>>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/
>>>>> example.com/" .. customerPhoneNumber)
>>>>>
>>>>>
>>>>> After that line, the script waits for the other phone to be picked up.
>>>>>
>>>>> However, now I wonder what to do with calls that don't complete, get
>>>>> busy signals, etc.
>>>>>
>>>>> What do people do in this case? The only related example I can find on
>>>>> the web is for a javascript dialer, which doesn't address any of these
>>>>> cases.
>>>>>
>>>>
>>>>
>>>> I guess it depends on what you want to do. For example I have a lua
>>>> script very similar to what you describe, although there is no confirmation
>>>> involved. Depending on the hangup cause the session gets, it might try
>>>> redialing with a different gateway, try again or just hangup.
>>>>
>>>> Take a look here http://wiki.freeswitch.org/wiki/Hangup_causes to see
>>>> what each hangup cause means. You don't need to have a special case for all
>>>> of them, only the ones you are interested in.
>>>>
>>>> Here's an example in code which retries a call depending on the hangup
>>>> cause. It retries max_retries1 times and alternates between 2 different
>>>> gateways:
>>>>
>>>> session1 = null;
>>>> max_retries1 = 3;
>>>> retries = 0;
>>>> ostr = "";
>>>> repeat
>>>>         retries = retries + 1;
>>>>         if (retries % 2) then ostr = originate_str1;
>>>>         else ostr = originate_str12; end
>>>>         freeswitch.consoleLog("notice", "*********** Dialing Leg1: " ..
>>>> ostr .. " - Try: "..retries.." ***********\n");
>>>>         session1 = freeswitch.Session(ostr);
>>>>         local hcause = session1:hangupCause();
>>>>         freeswitch.consoleLog("notice", "*********** Leg1: " .. hcause
>>>> .. " - Try: "..retries.." ***********\n");
>>>> until not ((hcause == 'NO_ROUTE_DESTINATION' or hcause ==
>>>> 'RECOVERY_ON_TIMER_EXPIRE' or hcause == 'INCOMPATIBLE_DESTINATION' or hcause
>>>> == 'CALL_REJECTED' or hcause == 'NORMAL_TEMPORARY_FAILURE') and (retries <
>>>> max_retriesl1))
>>>>
>>>>
>>>> Note: originate_str1 and originate_str2 are two different dial strings
>>>> for 2 different gateways.
>>>>
>>>>
>>> Nicolas,
>>>
>>> This is really nice. Would you be willing to add this script and a brief
>>> explanation to the wiki? You could create a whole new page and just link to
>>> it from here: http://wiki.freeswitch.org/wiki/Mod_lua#More_Samples
>>>
>>> If you have any questions please let me know!
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Jan 27 10:50:53 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 27 Jan 2010 12:50:53 -0600
Subject: [Freeswitch-users] Strategies for reliably detecting nat on
	B-leg?
In-Reply-To: <4B608987.9090606@aastral.net>
References: <4B60785F.6030505@aastral.net>	<FB139E6B-F685-491C-A672-16844878200E@freeswitch.org>
	<191c3a031001271005t62631e77sf6d9ca406054ba00@mail.gmail.com>
	<4B608987.9090606@aastral.net>
Message-ID: <3206F734-534B-478B-87A7-A51C2B999E03@freeswitch.org>


On Jan 27, 2010, at 12:44 PM, Bill W wrote:

> 
> Isn't local-network-acl and autonat:x.x.x.x for FS behind nat?

Yes Yes... sorry my misunderstanding.  ;)

> 
> Does this new information change your responses?
> 
> Thanks again!
> Bill



From jcasale at activenetwerx.com  Wed Jan 27 10:54:05 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 27 Jan 2010 18:54:05 +0000
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
In-Reply-To: <20100127180058.29470216DEA@cuneorg-email.cune.pri>
References: <CA5A491E9DEFBE4CB777DE97E21575E90B9E04@prato.activenetwerx.local>
	<20100127180058.29470216DEA@cuneorg-email.cune.pri>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90BC47F@prato.activenetwerx.local>

>It's working OK here. We're actually using 2.2.1RC2 with a TE110P in a VZ
>environment.

Russell,
Appreciate you letting me know this. I had some trouble getting 2.1.0 working
with my TDM410P, not sure how oz handles the difference between these two cards
but I am going to try again.

I was getting all sorts of intermittent behavior of calls working/not working?
CID working, then not working... That was on CentOS 5.4 though before you guys
informed me of the issues, so it likely had nothing to do with dahdi then. I
plan to give this another go now with my 5.3 setup.

Thanks a lot!
jlc


From jonas.gauffin at gmail.com  Wed Jan 27 11:26:00 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Wed, 27 Jan 2010 20:26:00 +0100
Subject: [Freeswitch-users] Call hanging up after 30 minutes (exactly)
Message-ID: <d021d3be1001271126y7a40e209t13e3d5b71af0d1a4@mail.gmail.com>

Hello,

I have problems with calls being hung up after 30 minutes.
I do not know if the problem is with FreeSWITCH or my sip provider.

I've got a log here: http://pastebin.freeswitch.org/11962

Can you please point me in the right direction?

Also, why is a second invite sent after 15 minutes?

Regards,
  Jonas
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From brian at freeswitch.org  Wed Jan 27 11:40:01 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 27 Jan 2010 13:40:01 -0600
Subject: [Freeswitch-users] Call hanging up after 30 minutes (exactly)
In-Reply-To: <d021d3be1001271126y7a40e209t13e3d5b71af0d1a4@mail.gmail.com>
References: <d021d3be1001271126y7a40e209t13e3d5b71af0d1a4@mail.gmail.com>
Message-ID: <ADD964DC-4B88-4DFA-A330-D391BE786F78@freeswitch.org>

Its a session timer.

/b

On Jan 27, 2010, at 1:26 PM, Jonas Gauffin wrote:

> Also, why is a second invite sent after 15 minutes?



From jcasale at activenetwerx.com  Wed Jan 27 11:48:10 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 27 Jan 2010 19:48:10 +0000
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
In-Reply-To: <20100127180058.29470216DEA@cuneorg-email.cune.pri>
References: <CA5A491E9DEFBE4CB777DE97E21575E90B9E04@prato.activenetwerx.local>
	<20100127180058.29470216DEA@cuneorg-email.cune.pri>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90BD7F4@prato.activenetwerx.local>

>It's working OK here. We're actually using 2.2.1RC2 with a TE110P in a VZ
>environment.

Russell,
On that note, what OS are you on?



From jerry.richards at teotech.com  Wed Jan 27 11:56:53 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 27 Jan 2010 11:56:53 -0800
Subject: [Freeswitch-users] Presence Change Distribution
In-Reply-To: <191c3a031001261321v2e8ea21cm7da19e01a11b59f9@mail.gmail.com>
References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com><2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com><26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com><191c3a031001251104p55ba7009g9381841f7de56d65@mail.gmail.com><BE6F5600B8E54EE297963A8B08F70E5E@greyhawk.tonecommander.com>
	<191c3a031001261321v2e8ea21cm7da19e01a11b59f9@mail.gmail.com>
Message-ID: <591B9C113F064880993543272B16ADF3@greyhawk.tonecommander.com>

There are two places in the XML body that are diffierent:
 
FS Rcvd PUBLISH has:  <rpid:away/> and <dm:note>Away</dm:note>
FS Sent NOTIFY   has:  <rpid:busy/> and <dm:note>Busy</dm:note>
 
This behavior (above) is why I'm not seeing the published presence at the
subscribing softphone.  FS should be sending the new Away status in the
NOTIFY message.
 
I did notice there is an "[ERR] sofia_presnece.c:674" FS log between the
PUBLISH and the NOTIFY (please see Line 89 of
http://pastebin.freeswitch.org/11953).  Line 674 is in the
sofia_presence_event_thread_run() function where it calls
switch_mutex_unlock(mod_sofia_globals.mutex).  Do you think this [ERR] is
related to why FS sends the previous status and not updated status?
 
Thanks And Best Regards,
Jerry
 

  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Tuesday, January 26, 2010 1:22 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


its sending a notify to them right away (line 174 of your PB)
the xml in the notify we send looks the same as what they sent except one
thing 

They send: <rpid:away/>
We send: <rpid:busy/>

everybody who implements this seems to have their own idea of what to say
here.

This crazy xml presence crap is pure garbage so maybe that's it.

 


On Tue, Jan 26, 2010 at 3:02 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


Okay, I setup my FS to force SUBSCRIBE Expires to 3600 seconds.  Then I
captured a FS console trace of a Bria softphone changing it's presence state
from 'Busy' to 'Away' (see http://pastebin.freeswitch.org/11953) and
observed that the subscribing Bria softphone did not update to 'Away'.  At
the same time, I executed the sqlite3 app and pasted each of the 3 SQL
select statements I saw in the FS console log, and pasted them below.  I'm
new to sqlite3.  Do you see what my issue is?
 
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'internal' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
sip|5382|192.168.72.79|5401|192.168.72.79|presence|"5382 on 79"
<sip:5382 at 192.168.72.150:34672>|ZTQ2ZWQwZGRlZjRiNTdkYTJjNGM5NTgzOWIyNmIwZmU.
|"5382 on 79" <sip:5382 at 192.168.72.79 <mailto:sip%3A5382 at 192.168.72.79>
>;tag=68bb4eb6|SIP/2.0/UDP
192.168.72.150:34672;branch=z9hG4bK-d8754z-eafc60166305eaef-1---d8754z-;rpor
t=34672|1264546204|Teo Softphone release 2.5.4 stamp
55958||internal|Away|away|192.168.72.79|Away|away
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name =
'internal-ipv6' or sip_subscriptions.presence_hosts !=
sip_subscriptions.sub_to_host);
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'external' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
sqlite>

Thanks and Best Regards,
Jerry


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Monday, January 25, 2010 11:05 AM 

To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


the notify will be instant after the publish
the notify you see are not triggered by the publish or they would be
instant.

Same drill, turn on presence debugging in sofia.conf.xml 
and look at the sql stmts and see why



On Mon, Jan 25, 2010 at 12:30 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:


Okay, I notice that if I reduce the Presence SUBSCRIBE Expires duration
(from 3600 seconds to 60 seconds), then the delay between PUBLISH's and
NOTIFY's is reduced, but FS still waits nearly 45 seconds to send the
NOTIFY's after it receives a PUBLISH.
 
Can a change be made in FS so that NOTIFYs are sent as a direct result of
receipt of the PUBLISH message?  And not tied to the SUBSCRIBE expiration?
I really don't want to configure all my phones to re-subscribe every 30 or
15 seconds.
 
Thanks and Best Regards,
Jerry
 


  _____  

From: RobertT [mailto:siniypin at gmail.com] 
Sent: Tuesday, December 29, 2009 12:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


You can try to reduce your registration time. 
I for one made my client apps send PUBLISH message every minute in addition
to reduced registration time.

Regards, Robert.


2009/12/28 Jerry Richards <jerry.richards at teotech.com>


Is there a setting to control how fast FS distributes presence changes to
subscribers?  Currently, it appears to take several minutes before I see
presence changes.  I would like to see them almost instantaneously, if
possible.

Thanks and Best Regards,
Jerry


_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


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From jonas.gauffin at gmail.com  Wed Jan 27 11:58:37 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Wed, 27 Jan 2010 20:58:37 +0100
Subject: [Freeswitch-users] Call hanging up after 30 minutes (exactly)
In-Reply-To: <ADD964DC-4B88-4DFA-A330-D391BE786F78@freeswitch.org>
References: <d021d3be1001271126y7a40e209t13e3d5b71af0d1a4@mail.gmail.com>
	<ADD964DC-4B88-4DFA-A330-D391BE786F78@freeswitch.org>
Message-ID: <d021d3be1001271158h352b9ffcn4da299003aeb0f87@mail.gmail.com>

Ok. And it's my sip provider and not FreeSWITCH that do not refresh
properly?


On Wed, Jan 27, 2010 at 8:40 PM, Brian West <brian at freeswitch.org> wrote:

> Its a session timer.
>
> /b
>
> On Jan 27, 2010, at 1:26 PM, Jonas Gauffin wrote:
>
> > Also, why is a second invite sent after 15 minutes?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From troy at tlainvestments.com  Wed Jan 27 12:05:08 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Wed, 27 Jan 2010 13:05:08 -0700
Subject: [Freeswitch-users] Call Dropping with SIP 503 status
In-Reply-To: <ADD964DC-4B88-4DFA-A330-D391BE786F78@freeswitch.org>
References: <d021d3be1001271126y7a40e209t13e3d5b71af0d1a4@mail.gmail.com>
	<ADD964DC-4B88-4DFA-A330-D391BE786F78@freeswitch.org>
Message-ID: <BD4CD56A-D488-4CCD-955D-0A838CB3CFD1@tlainvestments.com>

We are experiencing an odd issue.  We have many calls that don't drop, but some do after being up a minute or two.  

The reason code is NORMAL_TEMPORARY_FAILURE and the sip status that is triggering that is 503 (Service Unavailable).  With only one or two calls up at a time, I don't think it's a session limit issue (set to 1000).

Here is the console log from just before the 503 status - any help is greatly appreciated!

2010-01-27 12:49:11.879251 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
2010-01-27 12:49:11.899334 [INFO] sofia.c:597 Update Callee ID to "400" <400>
2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [ready][200]
2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4011 Duplicate SDP
v=0
o=- 1264621687 1264621687 IN IP4 192.168.0.46
s=Polycom IP Phone
c=IN IP4 192.168.0.46
t=0 0
a=sendrecv
m=audio 2222 RTP/AVP 0 8 18 127
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000

2010-01-27 12:50:06.068999 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [ready][200]
2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4011 Duplicate SDP
v=0
o=- 1264621687 1264621687 IN IP4 192.168.0.46
s=Polycom IP Phone
c=IN IP4 192.168.0.46
t=0 0
a=sendrecv
m=audio 2222 RTP/AVP 0 8 18 127
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000

2010-01-27 12:51:05.259614 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [ready][200]
2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4011 Duplicate SDP
v=0
o=- 1264621687 1264621687 IN IP4 192.168.0.46
s=Polycom IP Phone
c=IN IP4 192.168.0.46
t=0 0
a=sendrecv
m=audio 2222 RTP/AVP 0 8 18 127
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000

2010-01-27 12:52:05.369138 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [ready][200]
2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4011 Duplicate SDP
v=0
o=- 1264621687 1264621687 IN IP4 192.168.0.46
s=Polycom IP Phone
c=IN IP4 192.168.0.46
t=0 0
a=sendrecv
m=audio 2222 RTP/AVP 0 8 18 127
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000

2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [terminating][503]
2010-01-27 12:53:09.599069 [NOTICE] sofia.c:4647 Hangup sofia/internal/400 at 192.168.0.31 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
2010-01-27 12:53:09.599069 [DEBUG] switch_ivr_bridge.c:466 sofia/internal/400 at 192.168.0.31 ending bridge by request from write function

	



From gmaruzz at celliax.org  Wed Jan 27 12:15:30 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 27 Jan 2010 21:15:30 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B608014.4030902@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com> 
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com> 
	<4B6070A8.6050607@gmx.net>
	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com> 
	<4B607944.4040700@gmx.net> <4B608014.4030902@gmx.net>
Message-ID: <7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>

good, so you have only one sound device, the right one.

Use the one with hw:0 in the window that skype gives you to set sound devices

-gm

On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> I installed alsa-utile,
>
> now I get:
>
> skype:/var/cache/apt/archives# aplay -l
> **** List of PLAYBACK Hardware Devices ****
> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
> ?Subdevices: 127/128
> ?Subdevice #0: subdevice #0
> ?Subdevice #1: subdevice #1
> ?Subdevice #2: subdevice #2
> ?Subdevice #3: subdevice #3
> ?Subdevice #4: subdevice #4
> ?Subdevice #5: subdevice #5
> ?Subdevice #6: subdevice #6
> ?Subdevice #7: subdevice #7
> ?Subdevice #8: subdevice #8
> ?Subdevice #9: subdevice #9
> ?Subdevice #10: subdevice #10
> ?Subdevice #11: subdevice #11
> ?Subdevice #12: subdevice #12
> ?Subdevice #13: subdevice #13
> ?Subdevice #14: subdevice #14
> ?Subdevice #15: subdevice #15
> ?Subdevice #16: subdevice #16
> ?Subdevice #17: subdevice #17
> ?Subdevice #18: subdevice #18
> ?Subdevice #19: subdevice #19
> ?Subdevice #20: subdevice #20
> ?Subdevice #21: subdevice #21
> ?Subdevice #22: subdevice #22
> ?Subdevice #23: subdevice #23
> ?Subdevice #24: subdevice #24
> ?Subdevice #25: subdevice #25
> ?Subdevice #26: subdevice #26
> ?Subdevice #27: subdevice #27
> ?Subdevice #28: subdevice #28
> ?Subdevice #29: subdevice #29
> ?Subdevice #30: subdevice #30
> ?Subdevice #31: subdevice #31
> ?Subdevice #32: subdevice #32
> ?Subdevice #33: subdevice #33
> ?Subdevice #34: subdevice #34
> ?Subdevice #35: subdevice #35
> ?Subdevice #36: subdevice #36
> ?Subdevice #37: subdevice #37
> ?Subdevice #38: subdevice #38
> ?Subdevice #39: subdevice #39
> ?Subdevice #40: subdevice #40
> ?Subdevice #41: subdevice #41
> ?Subdevice #42: subdevice #42
> ?Subdevice #43: subdevice #43
> ?Subdevice #44: subdevice #44
> ?Subdevice #45: subdevice #45
> ?Subdevice #46: subdevice #46
> ?Subdevice #47: subdevice #47
> ?Subdevice #48: subdevice #48
> ?Subdevice #49: subdevice #49
> ?Subdevice #50: subdevice #50
> ?Subdevice #51: subdevice #51
> ?Subdevice #52: subdevice #52
> ?Subdevice #53: subdevice #53
> ?Subdevice #54: subdevice #54
> ?Subdevice #55: subdevice #55
> ?Subdevice #56: subdevice #56
> ?Subdevice #57: subdevice #57
> ?Subdevice #58: subdevice #58
> ?Subdevice #59: subdevice #59
> ?Subdevice #60: subdevice #60
> ?Subdevice #61: subdevice #61
> ?Subdevice #62: subdevice #62
> ?Subdevice #63: subdevice #63
> ?Subdevice #64: subdevice #64
> ?Subdevice #65: subdevice #65
> ?Subdevice #66: subdevice #66
> ?Subdevice #67: subdevice #67
> ?Subdevice #68: subdevice #68
> ?Subdevice #69: subdevice #69
> ?Subdevice #70: subdevice #70
> ?Subdevice #71: subdevice #71
> ?Subdevice #72: subdevice #72
> ?Subdevice #73: subdevice #73
> ?Subdevice #74: subdevice #74
> ?Subdevice #75: subdevice #75
> ?Subdevice #76: subdevice #76
> ?Subdevice #77: subdevice #77
> ?Subdevice #78: subdevice #78
> ?Subdevice #79: subdevice #79
> ?Subdevice #80: subdevice #80
> ?Subdevice #81: subdevice #81
> ?Subdevice #82: subdevice #82
> ?Subdevice #83: subdevice #83
> ?Subdevice #84: subdevice #84
> ?Subdevice #85: subdevice #85
> ?Subdevice #86: subdevice #86
> ?Subdevice #87: subdevice #87
> ?Subdevice #88: subdevice #88
> ?Subdevice #89: subdevice #89
> ?Subdevice #90: subdevice #90
> ?Subdevice #91: subdevice #91
> ?Subdevice #92: subdevice #92
> ?Subdevice #93: subdevice #93
> ?Subdevice #94: subdevice #94
> ?Subdevice #95: subdevice #95
> ?Subdevice #96: subdevice #96
> ?Subdevice #97: subdevice #97
> ?Subdevice #98: subdevice #98
> ?Subdevice #99: subdevice #99
> ?Subdevice #100: subdevice #100
> ?Subdevice #101: subdevice #101
> ?Subdevice #102: subdevice #102
> ?Subdevice #103: subdevice #103
> ?Subdevice #104: subdevice #104
> ?Subdevice #105: subdevice #105
> ?Subdevice #106: subdevice #106
> ?Subdevice #107: subdevice #107
> ?Subdevice #108: subdevice #108
> ?Subdevice #109: subdevice #109
> ?Subdevice #110: subdevice #110
> ?Subdevice #111: subdevice #111
> ?Subdevice #112: subdevice #112
> ?Subdevice #113: subdevice #113
> ?Subdevice #114: subdevice #114
> ?Subdevice #115: subdevice #115
> ?Subdevice #116: subdevice #116
> ?Subdevice #117: subdevice #117
> ?Subdevice #118: subdevice #118
> ?Subdevice #119: subdevice #119
> ?Subdevice #120: subdevice #120
> ?Subdevice #121: subdevice #121
> ?Subdevice #122: subdevice #122
> ?Subdevice #123: subdevice #123
> ?Subdevice #124: subdevice #124
> ?Subdevice #125: subdevice #125
> ?Subdevice #126: subdevice #126
> ?Subdevice #127: subdevice #127
>
>
> Peter P GMX schrieb:
>> Her's the output:
>>
>> skype:~# aplay -l
>> bash: aplay: command not found
>>
>> Giovanni Maruzzelli schrieb:
>>
>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>> what's the output of:
>>>
>>> aplay -l
>>>
>>> ?
>>>
>>> If instead you are referring to the choices that skype clients offers
>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>
>>> Eg: not the "default", but the "hardware" one
>>>
>>>
>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>
>>>> Thanks Giovanni,
>>>>
>>>> I think there may be the problem, that I have 2 sound devices now:
>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>> since I compiled alsa newly)
>>>>
>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>> (apt-get remove alsa-utils alsa-base).
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>>
>>>> Giovanni Maruzzelli schrieb:
>>>>
>>>>
>>>>> This warning is harmless:
>>>>>
>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>
>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>> <gmaruzz at celliax.org> wrote:
>>>>>
>>>>>
>>>>>
>>>>>> Ciao Peter
>>>>>>
>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>> snd-dummy-customized.
>>>>>>
>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>> 8.04).
>>>>>>
>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>
>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>> do this with the modprobe command, as in:
>>>>>>
>>>>>> rmmod snd-dummy
>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>
>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>> soundcard.
>>>>>>
>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>> just committed some improvements.
>>>>>>
>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>> write the mailing list again,
>>>>>>
>>>>>> ciao for now,
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>
>>>>>>
>>>>>>
>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>> to all contributors, excellent work!
>>>>>>>
>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>> instructions in
>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>> ist Debian 5.0R3)
>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>> one-way-audio and my logs show
>>>>>>>
>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>> working on a machine with 250HZ kernel
>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>> suppressed
>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>> suppressed
>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>> with the current limitations).
>>>>>>>
>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>
>>>>>>> Second question:
>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>
>>>>>>> Best regards
>>>>>>> Peter
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>> --
>>>>>> Sincerely,
>>>>>>
>>>>>> Giovanni Maruzzelli
>>>>>> Cell : +39-347-2665618
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>
>>>
>>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From Russell.Mosemann at cune.org  Wed Jan 27 13:08:36 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Wed, 27 Jan 2010 21:08:36 -0000
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E90BD7F4@prato.activenetwerx.local>
Message-ID: <20100127210836.5ED5D2E3B2D@cuneorg-email.cune.pri>

"Joseph L. Casale" <jcasale at activenetwerx.com> said:

> On that note, what OS are you on?

Debian 5.0.3

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From anthony.minessale at gmail.com  Wed Jan 27 13:08:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 15:08:38 -0600
Subject: [Freeswitch-users] Presence Change Distribution
In-Reply-To: <591B9C113F064880993543272B16ADF3@greyhawk.tonecommander.com>
References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com>
	<2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com>
	<26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com>
	<191c3a031001251104p55ba7009g9381841f7de56d65@mail.gmail.com>
	<BE6F5600B8E54EE297963A8B08F70E5E@greyhawk.tonecommander.com>
	<191c3a031001261321v2e8ea21cm7da19e01a11b59f9@mail.gmail.com>
	<591B9C113F064880993543272B16ADF3@greyhawk.tonecommander.com>
Message-ID: <191c3a031001271308l5c0c4eedw925e7660fbc2069d@mail.gmail.com>

Try latest trunk.
I tried forcing the db update in real-time to avoid a race on the event.





On Wed, Jan 27, 2010 at 1:56 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  There are two places in the XML body that are diffierent:
>
> FS Rcvd PUBLISH has:  <rpid:away/> and <dm:note>Away</dm:note>
> FS Sent NOTIFY   has:  <rpid:busy/> and <dm:note>Busy</dm:note>
>
> This behavior (above) is why I'm not seeing the published presence at the
> subscribing softphone.  FS should be sending the new Away status in the
> NOTIFY message.
>
> I did notice there is an "[ERR] sofia_presnece.c:674" FS log between the
> PUBLISH and the NOTIFY (please see Line 89 of
> http://pastebin.freeswitch.org/11953).  Line 674 is in the
> sofia_presence_event_thread_run() function where it calls
> switch_mutex_unlock(mod_sofia_globals.mutex).  Do you think this [ERR] is
> related to why FS sends the previous status and not updated status?
>
> Thanks And Best Regards,
> Jerry
>
>
>  ------------------------------
> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
> *Sent:* Tuesday, January 26, 2010 1:22 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>
> its sending a notify to them right away (line 174 of your PB)
> the xml in the notify we send looks the same as what they sent except one
> thing
>
> They send: <rpid:away/>
> We send: <rpid:busy/>
>
> everybody who implements this seems to have their own idea of what to say
> here.
>
> This crazy xml presence crap is pure garbage so maybe that's it.
>
>
>
> On Tue, Jan 26, 2010 at 3:02 PM, Jerry Richards <
> jerry.richards at teotech.com> wrote:
>
>>  Okay, I setup my FS to force SUBSCRIBE Expires to 3600 seconds.  Then I
>> captured a FS console trace of a Bria softphone changing it's presence state
>> from 'Busy' to 'Away' (see http://pastebin.freeswitch.org/11953) and
>> observed that the subscribing Bria softphone did not update to 'Away'.  At
>> the same time, I executed the sqlite3 app and pasted each of the 3 SQL
>> select statements I saw in the FS console log, and pasted them below.  I'm
>> new to sqlite3.  Do you see what my issue is?
>>
>> sqlite>
>> sqlite> select
>> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
>> from sip_subscriptions left join sip_presence on
>> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
>> sip_subscriptions.sub_to_host=sip_presence.sip_host and
>> sip_subscriptions.profile_name=sip_presence.profile_name) where
>> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
>> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
>> like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'internal' or
>> sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
>> sip|5382|192.168.72.79|5401|192.168.72.79|presence|"5382 on 79" <
>> sip:5382 at 192.168.72.150:34672>|ZTQ2ZWQwZGRlZjRiNTdkYTJjNGM5NTgzOWIyNmIwZmU.|"5382
>> on 79" <sip:5382 at 192.168.72.79 <sip%3A5382 at 192.168.72.79>>;tag=68bb4eb6|SIP/2.0/UDP
>> 192.168.72.150:34672;branch=z9hG4bK-d8754z-eafc60166305eaef-1---d8754z-;rport=34672|1264546204|Teo
>> Softphone release 2.5.4 stamp
>> 55958||internal|Away|away|192.168.72.79|Away|away
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite> select
>> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
>> from sip_subscriptions left join sip_presence on
>> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
>> sip_subscriptions.sub_to_host=sip_presence.sip_host and
>> sip_subscriptions.profile_name=sip_presence.profile_name) where
>> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
>> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
>> like '%192.168.72.79%') and (sip_subscriptions.profile_name =
>> 'internal-ipv6' or sip_subscriptions.presence_hosts !=
>> sip_subscriptions.sub_to_host);
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite> select
>> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
>> from sip_subscriptions left join sip_presence on
>> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
>> sip_subscriptions.sub_to_host=sip_presence.sip_host and
>> sip_subscriptions.profile_name=sip_presence.profile_name) where
>> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
>> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
>> like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'external' or
>> sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
>> sqlite>
>>  Thanks and Best Regards,
>> Jerry
>>
>>  ------------------------------
>> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
>> *Sent:* Monday, January 25, 2010 11:05 AM
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>>
>>   the notify will be instant after the publish
>> the notify you see are not triggered by the publish or they would be
>> instant.
>>
>> Same drill, turn on presence debugging in sofia.conf.xml
>> and look at the sql stmts and see why
>>
>>
>> On Mon, Jan 25, 2010 at 12:30 PM, Jerry Richards <
>> jerry.richards at teotech.com> wrote:
>>
>>>  Okay, I notice that if I reduce the Presence SUBSCRIBE Expires duration
>>> (from 3600 seconds to 60 seconds), then the delay between PUBLISH's and
>>> NOTIFY's is reduced, but FS still waits nearly 45 seconds to send the
>>> NOTIFY's after it receives a PUBLISH.
>>>
>>> Can a change be made in FS so that NOTIFYs are sent as a direct result of
>>> receipt of the PUBLISH message?  And not tied to the SUBSCRIBE expiration?
>>> I really don't want to configure all my phones to re-subscribe every 30 or
>>> 15 seconds.
>>>
>>> Thanks and Best Regards,
>>> Jerry
>>>
>>>
>>>  ------------------------------
>>> *From:* RobertT [mailto:siniypin at gmail.com]
>>> *Sent:* Tuesday, December 29, 2009 12:02 AM
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>>>
>>>   You can try to reduce your registration time.
>>> I for one made my client apps send PUBLISH message every minute in
>>> addition to reduced registration time.
>>>
>>> Regards, Robert.
>>>
>>> 2009/12/28 Jerry Richards <jerry.richards at teotech.com>
>>>
>>>> Is there a setting to control how fast FS distributes presence changes
>>>> to
>>>> subscribers?  Currently, it appears to take several minutes before I see
>>>> presence changes.  I would like to see them almost instantaneously, if
>>>> possible.
>>>>
>>>> Thanks and Best Regards,
>>>> Jerry
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Jan 27 14:31:04 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 16:31:04 -0600
Subject: [Freeswitch-users] Call Dropping with SIP 503 status
In-Reply-To: <BD4CD56A-D488-4CCD-955D-0A838CB3CFD1@tlainvestments.com>
References: <d021d3be1001271126y7a40e209t13e3d5b71af0d1a4@mail.gmail.com>
	<ADD964DC-4B88-4DFA-A330-D391BE786F78@freeswitch.org>
	<BD4CD56A-D488-4CCD-955D-0A838CB3CFD1@tlainvestments.com>
Message-ID: <191c3a031001271431re6cfad2w1389e96db4e6c1a0@mail.gmail.com>

try turning on sip trace as well to see the sip traffic

sofia profile internal siptrace on (from cli)
probably its something that said it could do session timers but was lying


On Wed, Jan 27, 2010 at 2:05 PM, Troy Anderson <troy at tlainvestments.com>wrote:

> We are experiencing an odd issue.  We have many calls that don't drop, but
> some do after being up a minute or two.
>
> The reason code is NORMAL_TEMPORARY_FAILURE and the sip status that is
> triggering that is 503 (Service Unavailable).  With only one or two calls up
> at a time, I don't think it's a session limit issue (set to 1000).
>
> Here is the console log from just before the 503 status - any help is
> greatly appreciated!
>
> 2010-01-27 12:49:11.879251 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:49:11.899334 [INFO] sofia.c:597 Update Callee ID to "400"
> <400>
> 2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [ready][200]
> 2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4011 Duplicate SDP
> v=0
> o=- 1264621687 1264621687 IN IP4 192.168.0.46
> s=Polycom IP Phone
> c=IN IP4 192.168.0.46
> t=0 0
> a=sendrecv
> m=audio 2222 RTP/AVP 0 8 18 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
>
> 2010-01-27 12:50:06.068999 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [ready][200]
> 2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4011 Duplicate SDP
> v=0
> o=- 1264621687 1264621687 IN IP4 192.168.0.46
> s=Polycom IP Phone
> c=IN IP4 192.168.0.46
> t=0 0
> a=sendrecv
> m=audio 2222 RTP/AVP 0 8 18 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
>
> 2010-01-27 12:51:05.259614 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [ready][200]
> 2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4011 Duplicate SDP
> v=0
> o=- 1264621687 1264621687 IN IP4 192.168.0.46
> s=Polycom IP Phone
> c=IN IP4 192.168.0.46
> t=0 0
> a=sendrecv
> m=audio 2222 RTP/AVP 0 8 18 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
>
> 2010-01-27 12:52:05.369138 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [ready][200]
> 2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4011 Duplicate SDP
> v=0
> o=- 1264621687 1264621687 IN IP4 192.168.0.46
> s=Polycom IP Phone
> c=IN IP4 192.168.0.46
> t=0 0
> a=sendrecv
> m=audio 2222 RTP/AVP 0 8 18 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
>
> 2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [terminating][503]
> 2010-01-27 12:53:09.599069 [NOTICE] sofia.c:4647 Hangup sofia/internal/
> 400 at 192.168.0.31 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
> 2010-01-27 12:53:09.599069 [DEBUG] switch_ivr_bridge.c:466 sofia/internal/
> 400 at 192.168.0.31 ending bridge by request from write function
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From wiltingtree at gmail.com  Wed Jan 27 16:15:48 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Wed, 27 Jan 2010 19:15:48 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
Message-ID: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>

Hi, I followed the instructions in the Lua documentation for setting up
luasql, but when I try to run my script I get:

2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module
'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
        /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment prot
after reloc: Permission denied
stack traceback:
        [C]: ?
        [C]: in function 'require'
        /usr/local/freeswitch/scripts/l.lua:2: in main chunk

I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.

I changed the permissions for mysql.so and for my script to 777, so I'm not
sure where the permission problem could be.

I'd appreciate any suggestions.

Thanks,
Adam
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From anthony.minessale at gmail.com  Wed Jan 27 17:02:44 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 19:02:44 -0600
Subject: [Freeswitch-users] Event socket: filter delete isn't working
In-Reply-To: <fe67a3341001270514g2715de2ag6aa2e3b099fa7816@mail.gmail.com>
References: <fe67a3341001270514g2715de2ag6aa2e3b099fa7816@mail.gmail.com>
Message-ID: <191c3a031001271702l3e77c952lc8aba35bacbf3d58@mail.gmail.com>

in the future please report issues to jira http://jira.freeswitch.org

please try svn trunk 16527 or higher

This was not a bug but I made it work the way you describe since it made
sense.

you should have done

filter delete unique-id

which would have delete all the unique-id filters that was the only option


you should be able to now say

filter delete unique-id <value>

To delete entry with specific value

or

filter delete unique-id

to delete all entries with matching key





On Wed, Jan 27, 2010 at 7:14 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Dear friends,
>    I've tried to delete the filter which I applied for an unique id. But,
> it doesn't work. After executing 'filter delete', I am receiving the events
> from that uuid.
>    I used the command as 'filter delete unique-id
> c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565'.
>
> I did the following operations.
>     Made call to the event socket.
>     Registered events for all. (events plain all).
>     Applied filter for the uuid. (filter unique-id
> aa3cb8ea-0b2f-11df-9e84-fb15c3cd8565).
>     I've got a new uuid by using create_uuid.
>     Applied filter for this new uuid. (filter unique-id
> c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565)
>     Originated a call with that uuid.
>     Now, I could receive events from both uuids. (Tested by giving DTMFs in
> both end and checked unique-id in event header).
>     Then, I wanted to delete a uuid from the filter. (filter delete
> unique-id c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565).
>     I thought, i won't receive the events from this deleted unique-id. But,
> I received the dtmfs from both unique-id.
>
> I'm using 'FreeSWITCH Version 1.0.trunk (15982)'.
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From david.villasmil.work at gmail.com  Wed Jan 27 17:09:33 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 28 Jan 2010 02:09:33 +0100
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
Message-ID: <9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>

I got the same error, my script was working with no problems before an
update to trunk.

David

On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com> wrote:
> Hi, I followed the instructions in the Lua documentation for setting up
> luasql, but when I try to run my script I get:
> 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module
> 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
> ?? ? ? ?/usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment prot
> after reloc: Permission denied
> stack traceback:
> ?? ? ? ?[C]: ?
> ?? ? ? ?[C]: in function 'require'
> ?? ? ? ?/usr/local/freeswitch/scripts/l.lua:2: in main chunk
> I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
> I changed the permissions for mysql.so and for my script to 777, so I'm not
> sure where the permission problem could be.
> I'd appreciate any suggestions.
> Thanks,
> Adam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Wed Jan 27 17:09:41 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 19:09:41 -0600
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
Message-ID: <191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>

user and host have to match too

On Wed, Jan 27, 2010 at 7:06 AM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Thanks guys.  I have this working except for one user who is
> registered like this:
>
> Call-ID:        2ff39277-fd9a6ab1 at 10.0.0.1
> User:           2000 at 192.168.1.120
> Contact:        2000
> <sip:2000 at 124.xxx.xxx.xxx
> :5075;transport=tls;fs_nat=yes;fs_path=sip%3A2000%40124.xxx.xxx.xxx%3A10281%3Btransport%3Dtls>
> Agent:          Linksys/SPA3102-5.1.10(GW)
> Status:         Registered(TLS-NAT)(unknown) EXP(2010-01-28 00:29:34)
> Host:           freeswitch
> IP:             124.xxx.xxx.xxx
> Port:           10281
> Auth-User:      2000
> Auth-Realm:     mydns.dyndns.org
> MWI-Account:    2000 at 192.168.1.120
>
> When I do the following commands via the telnet socket, no notify
> command is sent to user 2000:
>
> sendevent NOTIFY
> profile: internal
> content-type: application/simple-message-summary
> event-string: reboot_now
> user: 2000
> host: 192.168.1.120
> content-length: 0
>
> However, if I do exactly the same thing with user 2001 it works.  2001
> is registered as:
>
> Contact:        2001 <sip:2001 at 124.yyy.yyy.yyy:5066;transport=tls>
>
> Any ideas why that would be?
>
> On Mon, Jan 25, 2010 at 11:50 AM, Ognjen Seslija <oseslija at gmail.com>
> wrote:
> > The phone is asking FS to authenticate prior then accepting a NOTIFY from
> > it.
> > The authentication of notify's from spa endpoints work (afaik) only with
> > Linksys SPA9000 PBX , and FS doesn't have the code for that for obvious
> > reasons.
> > If you have SPA9000 maybe you can collect SIP traces.
> >
> > Ognjen
> >
> >
> >
> >
> >
> > On Mon, Jan 25, 2010 at 1:29 AM, Mark Campbell-Smith
> > <mcampbellsmith at gmail.com> wrote:
> >>
> >> Hi Ognjen,
> >>
> >> Thanks for the tip on the resync under the EXT tab.  It now works
> >> using mod_event_socket and the following:
> >>
> >> sendevent NOTIFY
> >> profile: internal
> >> event-string: resync
> >> user: 1000
> >> host: 192.168.1.121
> >> content-type: application/simple-message-summary
> >>
> >> However, if AUTH is required, why does FS send the wrong information to
> >> the SPA?
> >>
> >> On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija <oseslija at gmail.com>
> >> wrote:
> >> > You  should not authenticate those NOTIFYs (this will work only with
> >> > SPA9000
> >> > afaik). The option to change for this is in EXT tabs:
> >> >
> >> > Auth Resync-Reboot: No
> >> >
> >> > Also, FSs code will do a reboot of a phone, not resync (it sends
> >> > reboot_now
> >> > event). For that to work a patch is required.
> >> >
> >> > I've just tried to reboot my 942 (rev 16506) and it definitely works.
> >> >
> >> > Regards,
> >> > Ognjen
> >> >
> >> >
> >> > On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
> >> > <mcampbellsmith at gmail.com> wrote:
> >> >>
> >> >> Thanks Brian.. this still does not work.  Maybe I need to open a
> Jira?
> >> >>  Notice the username in the authorization field.  It should be 1000.
> >> >>
> >> >> Cheers
> >> >> Mark
> >> >>
> >> >> freeswitch at internal> sofia status profile internal user
> >> >> 1000 at 192.168.1.120
> >> >>
> >> >> Registrations:
> >> >>
> >> >>
> >> >>
> =================================================================================================
> >> >> Call-ID:        bd783b73-66877627 at 192.168.1.121
> >> >> User:           1000 at 192.168.1.120
> >> >> Contact:        1000 <sip:1000 at 192.168.1.121:5060>
> >> >> Agent:          Linksys/PAP2T-5.1.6(LS)
> >> >> Status:         Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
> >> >> Host:           freeswitch
> >> >> IP:             192.168.1.121
> >> >> Port:           5060
> >> >> Auth-User:      1000
> >> >> Auth-Realm:     192.168.1.120
> >> >> MWI-Account:    1000 at 192.168.1.120
> >> >>
> >> >>
> >> >>
> >> >>
> =================================================================================================
> >> >>
> >> >> freeswitch at internal> sofia profile internal flush_inbound_reg
> >> >> bd783b73-66877627 at 192.168.1.121 reboot
> >> >> +OK rebooting all registrations matching specified call_id
> >> >>
> >> >> freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060 at
> >> >> 23:55:49.012627:
> >> >>
> >> >>
> ------------------------------------------------------------------------
> >> >>   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
> >> >>   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
> >> >>   Max-Forwards: 70
> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >>   CSeq: 126070338 NOTIFY
> >> >>   Contact: <sip:mod_sofia at 192.168.1.120:5060>
> >> >>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
> >> >>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >>   Supported: timer, precondition, path, replaces
> >> >>   Event: reboot_now
> >> >>   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >> >> include-session-description, presence.winfo, message-summary, refer
> >> >>   Subscription-State: terminated;reason=timeout
> >> >>   Content-Type: application/simple-message-summary
> >> >>   Content-Length: 0
> >> >>
> >> >>
> >> >>
> ------------------------------------------------------------------------
> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
> >> >>
> >> >>
> ------------------------------------------------------------------------
> >> >>   SIP/2.0 401 Unauthorized
> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >>   CSeq: 126070338 NOTIFY
> >> >>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
> >> >>   Server: Linksys/PAP2T-5.1.6(LS)
> >> >>   WWW-Authenticate: Digest realm="192.168.1.120", nonce="8e54805b",
> >> >> qop="auth", algorithm=md5
> >> >>   Content-Length: 0
> >> >>
> >> >>
> >> >>
> ------------------------------------------------------------------------
> >> >> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
> >> >>
> >> >>
> ------------------------------------------------------------------------
> >> >>   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
> >> >>   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
> >> >>   Max-Forwards: 70
> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >>   CSeq: 126070339 NOTIFY
> >> >>   Contact: <sip:mod_sofia at 192.168.1.120:5060>
> >> >>   Expires: 3590
> >> >>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
> >> >>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >>   Supported: timer, precondition, path, replaces
> >> >>   Event: reboot_now
> >> >>   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >> >> include-session-description, presence.winfo, message-summary, refer
> >> >>   Subscription-State: terminated;reason=timeout
> >> >>   Authorization: Digest username="1115633124", realm="192.168.1.120",
> >> >> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
> >> >> uri="sip:1000 at 192.168.1.121:5060",
> >> >> response="747b4d04544c84535dbbd987f2999ca7", qop=auth, nc=00000001
> >> >>   Content-Type: application/simple-message-summary
> >> >>   Content-Length: 0
> >> >>
> >> >>
> >> >>
> ------------------------------------------------------------------------
> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
> >> >>
> >> >>
> ------------------------------------------------------------------------
> >> >>   SIP/2.0 401 Unauthorized
> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >>   CSeq: 126070339 NOTIFY
> >> >>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
> >> >>   Server: Linksys/PAP2T-5.1.6(LS)
> >> >>   WWW-Authenticate: Digest realm="192.168.1.120", nonce="5339c7ba",
> >> >> qop="auth", algorithm=md5
> >> >>   Content-Length: 0
> >> >>
> >> >>
> >> >>
> ------------------------------------------------------------------------
> >> >>
> >> >> On Mon, Jan 25, 2010 at 4:46 AM, Brian West <brian at freeswitch.org>
> >> >> wrote:
> >> >> > or sofia profile xxx flush_inbound_reg callid reboot
> >> >> > callid you can get from sofia status profile xxx
> >> >> > /b
> >> >> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
> >> >> >
> >> >> > Actually I just
> >> >> > found http://wiki.freeswitch.org/wiki/Mod_event_socket
> >> >> >
> >> >> > If I telnet to FS as described
> >> >> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do
> I
> >> >> > just need to enter somthing like:
> >> >> >
> >> >> > sendevent NOTIFY
> >> >> > profile: internal
> >> >> > event-string: resync
> >> >> > user: 1000
> >> >> > host: 192.168.1.121
> >> >> > content-type: application/simple-message-summary
> >> >> >
> >> >> > where 192.168.1.121 is the ip address of one of the Linksys
> devices?
> >> >> >
> >> >> > I don't see any messages sent when I do this.  What am I doing
> wrong?
> >> >> >
> >> >> > Thanks
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at van.lammeren.net  Wed Jan 27 17:13:36 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Wed, 27 Jan 2010 20:13:36 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
Message-ID: <5d2828f1001271713x5a86d8cjc11e5609bddd5b43@mail.gmail.com>

For me, the mysql.so library didn't work until I ran ldconfig on the
directory that contained it.

Mike van Lammeren


On Wed, Jan 27, 2010 at 7:15 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Hi, I followed the instructions in the Lua documentation for setting up
> luasql, but when I try to run my script I get:
>
> 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module
> 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
>         /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment prot
> after reloc: Permission denied
> stack traceback:
>         [C]: ?
>         [C]: in function 'require'
>         /usr/local/freeswitch/scripts/l.lua:2: in main chunk
>
> I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
>
> I changed the permissions for mysql.so and for my script to 777, so I'm not
> sure where the permission problem could be.
>
> I'd appreciate any suggestions.
>
> Thanks,
> Adam
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Wed Jan 27 17:26:28 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 28 Jan 2010 12:26:28 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
Message-ID: <33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>

Thanks Anthony,

I think user matches (ie the extension 2000 or 2001).   What should
host be?  In the sofia printout, it says 'freeswitch' (freeswitch has
ip address 192.168.1.120).

However, if I try to use 'freeswitch' as the host for user 2001,
nothing is sent.  But using 192.168.1.120 does.

If I do exactly the same thing for 2000, the NOTIFY message is not
sent.  Are there logs I can send to show you or any ideas what I am
doing wrong?



On Thu, Jan 28, 2010 at 12:09 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> user and host have to match too
>
> On Wed, Jan 27, 2010 at 7:06 AM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>>
>> Thanks guys. ?I have this working except for one user who is
>> registered like this:
>>
>> Call-ID: ? ? ? ?2ff39277-fd9a6ab1 at 10.0.0.1
>> User: ? ? ? ? ? 2000 at 192.168.1.120
>> Contact: ? ? ? ?2000
>>
>> <sip:2000 at 124.xxx.xxx.xxx:5075;transport=tls;fs_nat=yes;fs_path=sip%3A2000%40124.xxx.xxx.xxx%3A10281%3Btransport%3Dtls>
>> Agent: ? ? ? ? ?Linksys/SPA3102-5.1.10(GW)
>> Status: ? ? ? ? Registered(TLS-NAT)(unknown) EXP(2010-01-28 00:29:34)
>> Host: ? ? ? ? ? freeswitch
>> IP: ? ? ? ? ? ? 124.xxx.xxx.xxx
>> Port: ? ? ? ? ? 10281
>> Auth-User: ? ? ?2000
>> Auth-Realm: ? ? mydns.dyndns.org
>> MWI-Account: ? ?2000 at 192.168.1.120
>>
>> When I do the following commands via the telnet socket, no notify
>> command is sent to user 2000:
>>
>> sendevent NOTIFY
>> profile: internal
>> content-type: application/simple-message-summary
>> event-string: reboot_now
>> user: 2000
>> host: 192.168.1.120
>> content-length: 0
>>
>> However, if I do exactly the same thing with user 2001 it works. ?2001
>> is registered as:
>>
>> Contact: ? ? ? ?2001 <sip:2001 at 124.yyy.yyy.yyy:5066;transport=tls>
>>
>> Any ideas why that would be?
>>
>> On Mon, Jan 25, 2010 at 11:50 AM, Ognjen Seslija <oseslija at gmail.com>
>> wrote:
>> > The phone is asking FS to authenticate prior then accepting a NOTIFY
>> > from
>> > it.
>> > The authentication of notify's from spa endpoints work (afaik) only with
>> > Linksys SPA9000 PBX , and FS doesn't have the code for that for obvious
>> > reasons.
>> > If you have SPA9000 maybe you can collect SIP traces.
>> >
>> > Ognjen
>> >
>> >
>> >
>> >
>> >
>> > On Mon, Jan 25, 2010 at 1:29 AM, Mark Campbell-Smith
>> > <mcampbellsmith at gmail.com> wrote:
>> >>
>> >> Hi Ognjen,
>> >>
>> >> Thanks for the tip on the resync under the EXT tab. ?It now works
>> >> using mod_event_socket and the following:
>> >>
>> >> sendevent NOTIFY
>> >> profile: internal
>> >> event-string: resync
>> >> user: 1000
>> >> host: 192.168.1.121
>> >> content-type: application/simple-message-summary
>> >>
>> >> However, if AUTH is required, why does FS send the wrong information to
>> >> the SPA?
>> >>
>> >> On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija <oseslija at gmail.com>
>> >> wrote:
>> >> > You? should not authenticate those NOTIFYs (this will work only with
>> >> > SPA9000
>> >> > afaik). The option to change for this is in EXT tabs:
>> >> >
>> >> > Auth Resync-Reboot: No
>> >> >
>> >> > Also, FSs code will do a reboot of a phone, not resync (it sends
>> >> > reboot_now
>> >> > event). For that to work a patch is required.
>> >> >
>> >> > I've just tried to reboot my 942 (rev 16506) and it definitely works.
>> >> >
>> >> > Regards,
>> >> > Ognjen
>> >> >
>> >> >
>> >> > On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
>> >> > <mcampbellsmith at gmail.com> wrote:
>> >> >>
>> >> >> Thanks Brian.. this still does not work. ?Maybe I need to open a
>> >> >> Jira?
>> >> >> ?Notice the username in the authorization field. ?It should be 1000.
>> >> >>
>> >> >> Cheers
>> >> >> Mark
>> >> >>
>> >> >> freeswitch at internal> sofia status profile internal user
>> >> >> 1000 at 192.168.1.120
>> >> >>
>> >> >> Registrations:
>> >> >>
>> >> >>
>> >> >>
>> >> >> =================================================================================================
>> >> >> Call-ID: ? ? ? ?bd783b73-66877627 at 192.168.1.121
>> >> >> User: ? ? ? ? ? 1000 at 192.168.1.120
>> >> >> Contact: ? ? ? ?1000 <sip:1000 at 192.168.1.121:5060>
>> >> >> Agent: ? ? ? ? ?Linksys/PAP2T-5.1.6(LS)
>> >> >> Status: ? ? ? ? Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
>> >> >> Host: ? ? ? ? ? freeswitch
>> >> >> IP: ? ? ? ? ? ? 192.168.1.121
>> >> >> Port: ? ? ? ? ? 5060
>> >> >> Auth-User: ? ? ?1000
>> >> >> Auth-Realm: ? ? 192.168.1.120
>> >> >> MWI-Account: ? ?1000 at 192.168.1.120
>> >> >>
>> >> >>
>> >> >>
>> >> >>
>> >> >> =================================================================================================
>> >> >>
>> >> >> freeswitch at internal> sofia profile internal flush_inbound_reg
>> >> >> bd783b73-66877627 at 192.168.1.121 reboot
>> >> >> +OK rebooting all registrations matching specified call_id
>> >> >>
>> >> >> freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060 at
>> >> >> 23:55:49.012627:
>> >> >>
>> >> >>
>> >> >> ------------------------------------------------------------------------
>> >> >> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> >> >> ? Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
>> >> >> ? Max-Forwards: 70
>> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> ? To: <sip:1000 at 192.168.1.120>
>> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> ? CSeq: 126070338 NOTIFY
>> >> >> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> >> >> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> >> >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> >> ? Supported: timer, precondition, path, replaces
>> >> >> ? Event: reboot_now
>> >> >> ? Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> >> >> include-session-description, presence.winfo, message-summary, refer
>> >> >> ? Subscription-State: terminated;reason=timeout
>> >> >> ? Content-Type: application/simple-message-summary
>> >> >> ? Content-Length: 0
>> >> >>
>> >> >>
>> >> >>
>> >> >> ------------------------------------------------------------------------
>> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
>> >> >>
>> >> >>
>> >> >> ------------------------------------------------------------------------
>> >> >> ? SIP/2.0 401 Unauthorized
>> >> >> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> ? CSeq: 126070338 NOTIFY
>> >> >> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
>> >> >> ? Server: Linksys/PAP2T-5.1.6(LS)
>> >> >> ? WWW-Authenticate: Digest realm="192.168.1.120", nonce="8e54805b",
>> >> >> qop="auth", algorithm=md5
>> >> >> ? Content-Length: 0
>> >> >>
>> >> >>
>> >> >>
>> >> >> ------------------------------------------------------------------------
>> >> >> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
>> >> >>
>> >> >>
>> >> >> ------------------------------------------------------------------------
>> >> >> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> >> >> ? Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
>> >> >> ? Max-Forwards: 70
>> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> ? To: <sip:1000 at 192.168.1.120>
>> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> ? CSeq: 126070339 NOTIFY
>> >> >> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> >> >> ? Expires: 3590
>> >> >> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> >> >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> >> ? Supported: timer, precondition, path, replaces
>> >> >> ? Event: reboot_now
>> >> >> ? Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> >> >> include-session-description, presence.winfo, message-summary, refer
>> >> >> ? Subscription-State: terminated;reason=timeout
>> >> >> ? Authorization: Digest username="1115633124",
>> >> >> realm="192.168.1.120",
>> >> >> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
>> >> >> uri="sip:1000 at 192.168.1.121:5060",
>> >> >> response="747b4d04544c84535dbbd987f2999ca7", qop=auth, nc=00000001
>> >> >> ? Content-Type: application/simple-message-summary
>> >> >> ? Content-Length: 0
>> >> >>
>> >> >>
>> >> >>
>> >> >> ------------------------------------------------------------------------
>> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
>> >> >>
>> >> >>
>> >> >> ------------------------------------------------------------------------
>> >> >> ? SIP/2.0 401 Unauthorized
>> >> >> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> ? CSeq: 126070339 NOTIFY
>> >> >> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
>> >> >> ? Server: Linksys/PAP2T-5.1.6(LS)
>> >> >> ? WWW-Authenticate: Digest realm="192.168.1.120", nonce="5339c7ba",
>> >> >> qop="auth", algorithm=md5
>> >> >> ? Content-Length: 0
>> >> >>
>> >> >>
>> >> >>
>> >> >> ------------------------------------------------------------------------
>> >> >>
>> >> >> On Mon, Jan 25, 2010 at 4:46 AM, Brian West <brian at freeswitch.org>
>> >> >> wrote:
>> >> >> > or sofia profile xxx flush_inbound_reg callid reboot
>> >> >> > callid you can get from sofia status profile xxx
>> >> >> > /b
>> >> >> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
>> >> >> >
>> >> >> > Actually I just
>> >> >> > found?http://wiki.freeswitch.org/wiki/Mod_event_socket
>> >> >> >
>> >> >> > If I telnet to FS as described
>> >> >> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do
>> >> >> > I
>> >> >> > just need to enter somthing like:
>> >> >> >
>> >> >> > sendevent NOTIFY
>> >> >> > profile: internal
>> >> >> > event-string: resync
>> >> >> > user: 1000
>> >> >> > host: 192.168.1.121
>> >> >> > content-type: application/simple-message-summary
>> >> >> >
>> >> >> > where 192.168.1.121 is the ip address of one of the Linksys
>> >> >> > devices?
>> >> >> >
>> >> >> > I don't see any messages sent when I do this. ?What am I doing
>> >> >> > wrong?
>> >> >> >
>> >> >> > Thanks
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Wed Jan 27 17:41:36 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 19:41:36 -0600
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
Message-ID: <191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>

the host is not resolved it has to be an exact string match with the host
that is in the db.
if you want to normalize it set force-reg-domain and force-reg-db-domain to
the same val


On Wed, Jan 27, 2010 at 7:26 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Thanks Anthony,
>
> I think user matches (ie the extension 2000 or 2001).   What should
> host be?  In the sofia printout, it says 'freeswitch' (freeswitch has
> ip address 192.168.1.120).
>
> However, if I try to use 'freeswitch' as the host for user 2001,
> nothing is sent.  But using 192.168.1.120 does.
>
> If I do exactly the same thing for 2000, the NOTIFY message is not
> sent.  Are there logs I can send to show you or any ideas what I am
> doing wrong?
>
>
>
> On Thu, Jan 28, 2010 at 12:09 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > user and host have to match too
> >
> > On Wed, Jan 27, 2010 at 7:06 AM, Mark Campbell-Smith
> > <mcampbellsmith at gmail.com> wrote:
> >>
> >> Thanks guys.  I have this working except for one user who is
> >> registered like this:
> >>
> >> Call-ID:        2ff39277-fd9a6ab1 at 10.0.0.1
> >> User:           2000 at 192.168.1.120
> >> Contact:        2000
> >>
> >> <sip:2000 at 124.xxx.xxx.xxx
> :5075;transport=tls;fs_nat=yes;fs_path=sip%3A2000%40124.xxx.xxx.xxx%3A10281%3Btransport%3Dtls>
> >> Agent:          Linksys/SPA3102-5.1.10(GW)
> >> Status:         Registered(TLS-NAT)(unknown) EXP(2010-01-28 00:29:34)
> >> Host:           freeswitch
> >> IP:             124.xxx.xxx.xxx
> >> Port:           10281
> >> Auth-User:      2000
> >> Auth-Realm:     mydns.dyndns.org
> >> MWI-Account:    2000 at 192.168.1.120
> >>
> >> When I do the following commands via the telnet socket, no notify
> >> command is sent to user 2000:
> >>
> >> sendevent NOTIFY
> >> profile: internal
> >> content-type: application/simple-message-summary
> >> event-string: reboot_now
> >> user: 2000
> >> host: 192.168.1.120
> >> content-length: 0
> >>
> >> However, if I do exactly the same thing with user 2001 it works.  2001
> >> is registered as:
> >>
> >> Contact:        2001 <sip:2001 at 124.yyy.yyy.yyy:5066;transport=tls>
> >>
> >> Any ideas why that would be?
> >>
> >> On Mon, Jan 25, 2010 at 11:50 AM, Ognjen Seslija <oseslija at gmail.com>
> >> wrote:
> >> > The phone is asking FS to authenticate prior then accepting a NOTIFY
> >> > from
> >> > it.
> >> > The authentication of notify's from spa endpoints work (afaik) only
> with
> >> > Linksys SPA9000 PBX , and FS doesn't have the code for that for
> obvious
> >> > reasons.
> >> > If you have SPA9000 maybe you can collect SIP traces.
> >> >
> >> > Ognjen
> >> >
> >> >
> >> >
> >> >
> >> >
> >> > On Mon, Jan 25, 2010 at 1:29 AM, Mark Campbell-Smith
> >> > <mcampbellsmith at gmail.com> wrote:
> >> >>
> >> >> Hi Ognjen,
> >> >>
> >> >> Thanks for the tip on the resync under the EXT tab.  It now works
> >> >> using mod_event_socket and the following:
> >> >>
> >> >> sendevent NOTIFY
> >> >> profile: internal
> >> >> event-string: resync
> >> >> user: 1000
> >> >> host: 192.168.1.121
> >> >> content-type: application/simple-message-summary
> >> >>
> >> >> However, if AUTH is required, why does FS send the wrong information
> to
> >> >> the SPA?
> >> >>
> >> >> On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija <oseslija at gmail.com
> >
> >> >> wrote:
> >> >> > You  should not authenticate those NOTIFYs (this will work only
> with
> >> >> > SPA9000
> >> >> > afaik). The option to change for this is in EXT tabs:
> >> >> >
> >> >> > Auth Resync-Reboot: No
> >> >> >
> >> >> > Also, FSs code will do a reboot of a phone, not resync (it sends
> >> >> > reboot_now
> >> >> > event). For that to work a patch is required.
> >> >> >
> >> >> > I've just tried to reboot my 942 (rev 16506) and it definitely
> works.
> >> >> >
> >> >> > Regards,
> >> >> > Ognjen
> >> >> >
> >> >> >
> >> >> > On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
> >> >> > <mcampbellsmith at gmail.com> wrote:
> >> >> >>
> >> >> >> Thanks Brian.. this still does not work.  Maybe I need to open a
> >> >> >> Jira?
> >> >> >>  Notice the username in the authorization field.  It should be
> 1000.
> >> >> >>
> >> >> >> Cheers
> >> >> >> Mark
> >> >> >>
> >> >> >> freeswitch at internal> sofia status profile internal user
> >> >> >> 1000 at 192.168.1.120
> >> >> >>
> >> >> >> Registrations:
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> =================================================================================================
> >> >> >> Call-ID:        bd783b73-66877627 at 192.168.1.121
> >> >> >> User:           1000 at 192.168.1.120
> >> >> >> Contact:        1000 <sip:1000 at 192.168.1.121:5060>
> >> >> >> Agent:          Linksys/PAP2T-5.1.6(LS)
> >> >> >> Status:         Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
> >> >> >> Host:           freeswitch
> >> >> >> IP:             192.168.1.121
> >> >> >> Port:           5060
> >> >> >> Auth-User:      1000
> >> >> >> Auth-Realm:     192.168.1.120
> >> >> >> MWI-Account:    1000 at 192.168.1.120
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> =================================================================================================
> >> >> >>
> >> >> >> freeswitch at internal> sofia profile internal flush_inbound_reg
> >> >> >> bd783b73-66877627 at 192.168.1.121 reboot
> >> >> >> +OK rebooting all registrations matching specified call_id
> >> >> >>
> >> >> >> freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060
> at
> >> >> >> 23:55:49.012627:
> >> >> >>
> >> >> >>
> >> >> >>
> ------------------------------------------------------------------------
> >> >> >>   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
> >> >> >>   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
> >> >> >>   Max-Forwards: 70
> >> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
> >> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >> >>   CSeq: 126070338 NOTIFY
> >> >> >>   Contact: <sip:mod_sofia at 192.168.1.120:5060>
> >> >> >>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
> >> >> >>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >> >>   Supported: timer, precondition, path, replaces
> >> >> >>   Event: reboot_now
> >> >> >>   Allow-Events: talk, presence, dialog, line-seize, call-info,
> sla,
> >> >> >> include-session-description, presence.winfo, message-summary,
> refer
> >> >> >>   Subscription-State: terminated;reason=timeout
> >> >> >>   Content-Type: application/simple-message-summary
> >> >> >>   Content-Length: 0
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
> >> >> >>
> >> >> >>
> >> >> >>
> ------------------------------------------------------------------------
> >> >> >>   SIP/2.0 401 Unauthorized
> >> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
> >> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >> >>   CSeq: 126070338 NOTIFY
> >> >> >>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
> >> >> >>   Server: Linksys/PAP2T-5.1.6(LS)
> >> >> >>   WWW-Authenticate: Digest realm="192.168.1.120",
> nonce="8e54805b",
> >> >> >> qop="auth", algorithm=md5
> >> >> >>   Content-Length: 0
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
> >> >> >>
> >> >> >>
> >> >> >>
> ------------------------------------------------------------------------
> >> >> >>   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
> >> >> >>   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
> >> >> >>   Max-Forwards: 70
> >> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
> >> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >> >>   CSeq: 126070339 NOTIFY
> >> >> >>   Contact: <sip:mod_sofia at 192.168.1.120:5060>
> >> >> >>   Expires: 3590
> >> >> >>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
> >> >> >>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >> >>   Supported: timer, precondition, path, replaces
> >> >> >>   Event: reboot_now
> >> >> >>   Allow-Events: talk, presence, dialog, line-seize, call-info,
> sla,
> >> >> >> include-session-description, presence.winfo, message-summary,
> refer
> >> >> >>   Subscription-State: terminated;reason=timeout
> >> >> >>   Authorization: Digest username="1115633124",
> >> >> >> realm="192.168.1.120",
> >> >> >> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
> >> >> >> uri="sip:1000 at 192.168.1.121:5060",
> >> >> >> response="747b4d04544c84535dbbd987f2999ca7", qop=auth, nc=00000001
> >> >> >>   Content-Type: application/simple-message-summary
> >> >> >>   Content-Length: 0
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
> >> >> >>
> >> >> >>
> >> >> >>
> ------------------------------------------------------------------------
> >> >> >>   SIP/2.0 401 Unauthorized
> >> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
> >> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >> >>   CSeq: 126070339 NOTIFY
> >> >> >>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
> >> >> >>   Server: Linksys/PAP2T-5.1.6(LS)
> >> >> >>   WWW-Authenticate: Digest realm="192.168.1.120",
> nonce="5339c7ba",
> >> >> >> qop="auth", algorithm=md5
> >> >> >>   Content-Length: 0
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> ------------------------------------------------------------------------
> >> >> >>
> >> >> >> On Mon, Jan 25, 2010 at 4:46 AM, Brian West <brian at freeswitch.org
> >
> >> >> >> wrote:
> >> >> >> > or sofia profile xxx flush_inbound_reg callid reboot
> >> >> >> > callid you can get from sofia status profile xxx
> >> >> >> > /b
> >> >> >> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
> >> >> >> >
> >> >> >> > Actually I just
> >> >> >> > found http://wiki.freeswitch.org/wiki/Mod_event_socket
> >> >> >> >
> >> >> >> > If I telnet to FS as described
> >> >> >> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client,
> do
> >> >> >> > I
> >> >> >> > just need to enter somthing like:
> >> >> >> >
> >> >> >> > sendevent NOTIFY
> >> >> >> > profile: internal
> >> >> >> > event-string: resync
> >> >> >> > user: 1000
> >> >> >> > host: 192.168.1.121
> >> >> >> > content-type: application/simple-message-summary
> >> >> >> >
> >> >> >> > where 192.168.1.121 is the ip address of one of the Linksys
> >> >> >> > devices?
> >> >> >> >
> >> >> >> > I don't see any messages sent when I do this.  What am I doing
> >> >> >> > wrong?
> >> >> >> >
> >> >> >> > Thanks
> >> >> >> >
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jcasale at activenetwerx.com  Wed Jan 27 18:09:06 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Thu, 28 Jan 2010 02:09:06 +0000
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
In-Reply-To: <20100127210836.5ED5D2E3B2D@cuneorg-email.cune.pri>
References: <CA5A491E9DEFBE4CB777DE97E21575E90BD7F4@prato.activenetwerx.local>
	<20100127210836.5ED5D2E3B2D@cuneorg-email.cune.pri>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90BF91C@prato.activenetwerx.local>

>Debian 5.0.3

Well, given the time I had tonight, I tried on my CentOS 5.3 box.
The incoming log is the first block, and an outgoing log is the
second block at http://pastebin.freeswitch.org/11965

When I call in, I can hear it get answered, as I play the wav file
I hear the tone go very low, but no sound.

When I try to call out, nothing happens?

Is there anything in the log that might standout from your perspective?

Thanks everyone!
jlc


From mcampbellsmith at gmail.com  Wed Jan 27 18:34:14 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 28 Jan 2010 13:34:14 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
Message-ID: <33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>

If it works with 2001 doesn't that mean I am using the correct host?

Both 2001 and 2000 register with exactly the same data, except
username and password ....



On Thu, Jan 28, 2010 at 12:41 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> the host is not resolved it has to be an exact string match with the host
> that is in the db.
> if you want to normalize it set force-reg-domain and force-reg-db-domain to
> the same val
>
>
> On Wed, Jan 27, 2010 at 7:26 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>>
>> Thanks Anthony,
>>
>> I think user matches (ie the extension 2000 or 2001). ? What should
>> host be? ?In the sofia printout, it says 'freeswitch' (freeswitch has
>> ip address 192.168.1.120).
>>
>> However, if I try to use 'freeswitch' as the host for user 2001,
>> nothing is sent. ?But using 192.168.1.120 does.
>>
>> If I do exactly the same thing for 2000, the NOTIFY message is not
>> sent. ?Are there logs I can send to show you or any ideas what I am
>> doing wrong?
>>
>>
>>
>> On Thu, Jan 28, 2010 at 12:09 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > user and host have to match too
>> >
>> > On Wed, Jan 27, 2010 at 7:06 AM, Mark Campbell-Smith
>> > <mcampbellsmith at gmail.com> wrote:
>> >>
>> >> Thanks guys. ?I have this working except for one user who is
>> >> registered like this:
>> >>
>> >> Call-ID: ? ? ? ?2ff39277-fd9a6ab1 at 10.0.0.1
>> >> User: ? ? ? ? ? 2000 at 192.168.1.120
>> >> Contact: ? ? ? ?2000
>> >>
>> >>
>> >> <sip:2000 at 124.xxx.xxx.xxx:5075;transport=tls;fs_nat=yes;fs_path=sip%3A2000%40124.xxx.xxx.xxx%3A10281%3Btransport%3Dtls>
>> >> Agent: ? ? ? ? ?Linksys/SPA3102-5.1.10(GW)
>> >> Status: ? ? ? ? Registered(TLS-NAT)(unknown) EXP(2010-01-28 00:29:34)
>> >> Host: ? ? ? ? ? freeswitch
>> >> IP: ? ? ? ? ? ? 124.xxx.xxx.xxx
>> >> Port: ? ? ? ? ? 10281
>> >> Auth-User: ? ? ?2000
>> >> Auth-Realm: ? ? mydns.dyndns.org
>> >> MWI-Account: ? ?2000 at 192.168.1.120
>> >>
>> >> When I do the following commands via the telnet socket, no notify
>> >> command is sent to user 2000:
>> >>
>> >> sendevent NOTIFY
>> >> profile: internal
>> >> content-type: application/simple-message-summary
>> >> event-string: reboot_now
>> >> user: 2000
>> >> host: 192.168.1.120
>> >> content-length: 0
>> >>
>> >> However, if I do exactly the same thing with user 2001 it works. ?2001
>> >> is registered as:
>> >>
>> >> Contact: ? ? ? ?2001 <sip:2001 at 124.yyy.yyy.yyy:5066;transport=tls>
>> >>
>> >> Any ideas why that would be?
>> >>
>> >> On Mon, Jan 25, 2010 at 11:50 AM, Ognjen Seslija <oseslija at gmail.com>
>> >> wrote:
>> >> > The phone is asking FS to authenticate prior then accepting a NOTIFY
>> >> > from
>> >> > it.
>> >> > The authentication of notify's from spa endpoints work (afaik) only
>> >> > with
>> >> > Linksys SPA9000 PBX , and FS doesn't have the code for that for
>> >> > obvious
>> >> > reasons.
>> >> > If you have SPA9000 maybe you can collect SIP traces.
>> >> >
>> >> > Ognjen
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > On Mon, Jan 25, 2010 at 1:29 AM, Mark Campbell-Smith
>> >> > <mcampbellsmith at gmail.com> wrote:
>> >> >>
>> >> >> Hi Ognjen,
>> >> >>
>> >> >> Thanks for the tip on the resync under the EXT tab. ?It now works
>> >> >> using mod_event_socket and the following:
>> >> >>
>> >> >> sendevent NOTIFY
>> >> >> profile: internal
>> >> >> event-string: resync
>> >> >> user: 1000
>> >> >> host: 192.168.1.121
>> >> >> content-type: application/simple-message-summary
>> >> >>
>> >> >> However, if AUTH is required, why does FS send the wrong information
>> >> >> to
>> >> >> the SPA?
>> >> >>
>> >> >> On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija
>> >> >> <oseslija at gmail.com>
>> >> >> wrote:
>> >> >> > You? should not authenticate those NOTIFYs (this will work only
>> >> >> > with
>> >> >> > SPA9000
>> >> >> > afaik). The option to change for this is in EXT tabs:
>> >> >> >
>> >> >> > Auth Resync-Reboot: No
>> >> >> >
>> >> >> > Also, FSs code will do a reboot of a phone, not resync (it sends
>> >> >> > reboot_now
>> >> >> > event). For that to work a patch is required.
>> >> >> >
>> >> >> > I've just tried to reboot my 942 (rev 16506) and it definitely
>> >> >> > works.
>> >> >> >
>> >> >> > Regards,
>> >> >> > Ognjen
>> >> >> >
>> >> >> >
>> >> >> > On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
>> >> >> > <mcampbellsmith at gmail.com> wrote:
>> >> >> >>
>> >> >> >> Thanks Brian.. this still does not work. ?Maybe I need to open a
>> >> >> >> Jira?
>> >> >> >> ?Notice the username in the authorization field. ?It should be
>> >> >> >> 1000.
>> >> >> >>
>> >> >> >> Cheers
>> >> >> >> Mark
>> >> >> >>
>> >> >> >> freeswitch at internal> sofia status profile internal user
>> >> >> >> 1000 at 192.168.1.120
>> >> >> >>
>> >> >> >> Registrations:
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> =================================================================================================
>> >> >> >> Call-ID: ? ? ? ?bd783b73-66877627 at 192.168.1.121
>> >> >> >> User: ? ? ? ? ? 1000 at 192.168.1.120
>> >> >> >> Contact: ? ? ? ?1000 <sip:1000 at 192.168.1.121:5060>
>> >> >> >> Agent: ? ? ? ? ?Linksys/PAP2T-5.1.6(LS)
>> >> >> >> Status: ? ? ? ? Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
>> >> >> >> Host: ? ? ? ? ? freeswitch
>> >> >> >> IP: ? ? ? ? ? ? 192.168.1.121
>> >> >> >> Port: ? ? ? ? ? 5060
>> >> >> >> Auth-User: ? ? ?1000
>> >> >> >> Auth-Realm: ? ? 192.168.1.120
>> >> >> >> MWI-Account: ? ?1000 at 192.168.1.120
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> =================================================================================================
>> >> >> >>
>> >> >> >> freeswitch at internal> sofia profile internal flush_inbound_reg
>> >> >> >> bd783b73-66877627 at 192.168.1.121 reboot
>> >> >> >> +OK rebooting all registrations matching specified call_id
>> >> >> >>
>> >> >> >> freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060
>> >> >> >> at
>> >> >> >> 23:55:49.012627:
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> ------------------------------------------------------------------------
>> >> >> >> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> >> >> >> ? Via: SIP/2.0/UDP
>> >> >> >> 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
>> >> >> >> ? Max-Forwards: 70
>> >> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> >> ? To: <sip:1000 at 192.168.1.120>
>> >> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> >> ? CSeq: 126070338 NOTIFY
>> >> >> >> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> >> >> >> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> >> >> >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>> >> >> >> INFO,
>> >> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> >> >> ? Supported: timer, precondition, path, replaces
>> >> >> >> ? Event: reboot_now
>> >> >> >> ? Allow-Events: talk, presence, dialog, line-seize, call-info,
>> >> >> >> sla,
>> >> >> >> include-session-description, presence.winfo, message-summary,
>> >> >> >> refer
>> >> >> >> ? Subscription-State: terminated;reason=timeout
>> >> >> >> ? Content-Type: application/simple-message-summary
>> >> >> >> ? Content-Length: 0
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> ------------------------------------------------------------------------
>> >> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> ------------------------------------------------------------------------
>> >> >> >> ? SIP/2.0 401 Unauthorized
>> >> >> >> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> >> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> >> ? CSeq: 126070338 NOTIFY
>> >> >> >> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
>> >> >> >> ? Server: Linksys/PAP2T-5.1.6(LS)
>> >> >> >> ? WWW-Authenticate: Digest realm="192.168.1.120",
>> >> >> >> nonce="8e54805b",
>> >> >> >> qop="auth", algorithm=md5
>> >> >> >> ? Content-Length: 0
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> ------------------------------------------------------------------------
>> >> >> >> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> ------------------------------------------------------------------------
>> >> >> >> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> >> >> >> ? Via: SIP/2.0/UDP
>> >> >> >> 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
>> >> >> >> ? Max-Forwards: 70
>> >> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> >> ? To: <sip:1000 at 192.168.1.120>
>> >> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> >> ? CSeq: 126070339 NOTIFY
>> >> >> >> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> >> >> >> ? Expires: 3590
>> >> >> >> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> >> >> >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>> >> >> >> INFO,
>> >> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> >> >> ? Supported: timer, precondition, path, replaces
>> >> >> >> ? Event: reboot_now
>> >> >> >> ? Allow-Events: talk, presence, dialog, line-seize, call-info,
>> >> >> >> sla,
>> >> >> >> include-session-description, presence.winfo, message-summary,
>> >> >> >> refer
>> >> >> >> ? Subscription-State: terminated;reason=timeout
>> >> >> >> ? Authorization: Digest username="1115633124",
>> >> >> >> realm="192.168.1.120",
>> >> >> >> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
>> >> >> >> uri="sip:1000 at 192.168.1.121:5060",
>> >> >> >> response="747b4d04544c84535dbbd987f2999ca7", qop=auth,
>> >> >> >> nc=00000001
>> >> >> >> ? Content-Type: application/simple-message-summary
>> >> >> >> ? Content-Length: 0
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> ------------------------------------------------------------------------
>> >> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> ------------------------------------------------------------------------
>> >> >> >> ? SIP/2.0 401 Unauthorized
>> >> >> >> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> >> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> >> ? CSeq: 126070339 NOTIFY
>> >> >> >> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
>> >> >> >> ? Server: Linksys/PAP2T-5.1.6(LS)
>> >> >> >> ? WWW-Authenticate: Digest realm="192.168.1.120",
>> >> >> >> nonce="5339c7ba",
>> >> >> >> qop="auth", algorithm=md5
>> >> >> >> ? Content-Length: 0
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> ------------------------------------------------------------------------
>> >> >> >>
>> >> >> >> On Mon, Jan 25, 2010 at 4:46 AM, Brian West
>> >> >> >> <brian at freeswitch.org>
>> >> >> >> wrote:
>> >> >> >> > or sofia profile xxx flush_inbound_reg callid reboot
>> >> >> >> > callid you can get from sofia status profile xxx
>> >> >> >> > /b
>> >> >> >> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
>> >> >> >> >
>> >> >> >> > Actually I just
>> >> >> >> > found?http://wiki.freeswitch.org/wiki/Mod_event_socket
>> >> >> >> >
>> >> >> >> > If I telnet to FS as described
>> >> >> >> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client,
>> >> >> >> > do
>> >> >> >> > I
>> >> >> >> > just need to enter somthing like:
>> >> >> >> >
>> >> >> >> > sendevent NOTIFY
>> >> >> >> > profile: internal
>> >> >> >> > event-string: resync
>> >> >> >> > user: 1000
>> >> >> >> > host: 192.168.1.121
>> >> >> >> > content-type: application/simple-message-summary
>> >> >> >> >
>> >> >> >> > where 192.168.1.121 is the ip address of one of the Linksys
>> >> >> >> > devices?
>> >> >> >> >
>> >> >> >> > I don't see any messages sent when I do this. ?What am I doing
>> >> >> >> > wrong?
>> >> >> >> >
>> >> >> >> > Thanks
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > _______________________________________________
>> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> > http://www.freeswitch.org
>> >> >> >> >
>> >> >> >> >
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Wed Jan 27 19:25:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 21:25:17 -0600
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
	<33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
Message-ID: <191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>

you said one was the word freeswitch and one was an ip didn't you?


On Wed, Jan 27, 2010 at 8:34 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> If it works with 2001 doesn't that mean I am using the correct host?
>
> Both 2001 and 2000 register with exactly the same data, except
> username and password ....
>
>
>
> On Thu, Jan 28, 2010 at 12:41 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > the host is not resolved it has to be an exact string match with the host
> > that is in the db.
> > if you want to normalize it set force-reg-domain and force-reg-db-domain
> to
> > the same val
> >
> >
> > On Wed, Jan 27, 2010 at 7:26 PM, Mark Campbell-Smith
> > <mcampbellsmith at gmail.com> wrote:
> >>
> >> Thanks Anthony,
> >>
> >> I think user matches (ie the extension 2000 or 2001).   What should
> >> host be?  In the sofia printout, it says 'freeswitch' (freeswitch has
> >> ip address 192.168.1.120).
> >>
> >> However, if I try to use 'freeswitch' as the host for user 2001,
> >> nothing is sent.  But using 192.168.1.120 does.
> >>
> >> If I do exactly the same thing for 2000, the NOTIFY message is not
> >> sent.  Are there logs I can send to show you or any ideas what I am
> >> doing wrong?
> >>
> >>
> >>
> >> On Thu, Jan 28, 2010 at 12:09 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >> > user and host have to match too
> >> >
> >> > On Wed, Jan 27, 2010 at 7:06 AM, Mark Campbell-Smith
> >> > <mcampbellsmith at gmail.com> wrote:
> >> >>
> >> >> Thanks guys.  I have this working except for one user who is
> >> >> registered like this:
> >> >>
> >> >> Call-ID:        2ff39277-fd9a6ab1 at 10.0.0.1
> >> >> User:           2000 at 192.168.1.120
> >> >> Contact:        2000
> >> >>
> >> >>
> >> >> <sip:2000 at 124.xxx.xxx.xxx
> :5075;transport=tls;fs_nat=yes;fs_path=sip%3A2000%40124.xxx.xxx.xxx%3A10281%3Btransport%3Dtls>
> >> >> Agent:          Linksys/SPA3102-5.1.10(GW)
> >> >> Status:         Registered(TLS-NAT)(unknown) EXP(2010-01-28 00:29:34)
> >> >> Host:           freeswitch
> >> >> IP:             124.xxx.xxx.xxx
> >> >> Port:           10281
> >> >> Auth-User:      2000
> >> >> Auth-Realm:     mydns.dyndns.org
> >> >> MWI-Account:    2000 at 192.168.1.120
> >> >>
> >> >> When I do the following commands via the telnet socket, no notify
> >> >> command is sent to user 2000:
> >> >>
> >> >> sendevent NOTIFY
> >> >> profile: internal
> >> >> content-type: application/simple-message-summary
> >> >> event-string: reboot_now
> >> >> user: 2000
> >> >> host: 192.168.1.120
> >> >> content-length: 0
> >> >>
> >> >> However, if I do exactly the same thing with user 2001 it works.
>  2001
> >> >> is registered as:
> >> >>
> >> >> Contact:        2001 <sip:2001 at 124.yyy.yyy.yyy:5066;transport=tls>
> >> >>
> >> >> Any ideas why that would be?
> >> >>
> >> >> On Mon, Jan 25, 2010 at 11:50 AM, Ognjen Seslija <oseslija at gmail.com
> >
> >> >> wrote:
> >> >> > The phone is asking FS to authenticate prior then accepting a
> NOTIFY
> >> >> > from
> >> >> > it.
> >> >> > The authentication of notify's from spa endpoints work (afaik) only
> >> >> > with
> >> >> > Linksys SPA9000 PBX , and FS doesn't have the code for that for
> >> >> > obvious
> >> >> > reasons.
> >> >> > If you have SPA9000 maybe you can collect SIP traces.
> >> >> >
> >> >> > Ognjen
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > On Mon, Jan 25, 2010 at 1:29 AM, Mark Campbell-Smith
> >> >> > <mcampbellsmith at gmail.com> wrote:
> >> >> >>
> >> >> >> Hi Ognjen,
> >> >> >>
> >> >> >> Thanks for the tip on the resync under the EXT tab.  It now works
> >> >> >> using mod_event_socket and the following:
> >> >> >>
> >> >> >> sendevent NOTIFY
> >> >> >> profile: internal
> >> >> >> event-string: resync
> >> >> >> user: 1000
> >> >> >> host: 192.168.1.121
> >> >> >> content-type: application/simple-message-summary
> >> >> >>
> >> >> >> However, if AUTH is required, why does FS send the wrong
> information
> >> >> >> to
> >> >> >> the SPA?
> >> >> >>
> >> >> >> On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija
> >> >> >> <oseslija at gmail.com>
> >> >> >> wrote:
> >> >> >> > You  should not authenticate those NOTIFYs (this will work only
> >> >> >> > with
> >> >> >> > SPA9000
> >> >> >> > afaik). The option to change for this is in EXT tabs:
> >> >> >> >
> >> >> >> > Auth Resync-Reboot: No
> >> >> >> >
> >> >> >> > Also, FSs code will do a reboot of a phone, not resync (it sends
> >> >> >> > reboot_now
> >> >> >> > event). For that to work a patch is required.
> >> >> >> >
> >> >> >> > I've just tried to reboot my 942 (rev 16506) and it definitely
> >> >> >> > works.
> >> >> >> >
> >> >> >> > Regards,
> >> >> >> > Ognjen
> >> >> >> >
> >> >> >> >
> >> >> >> > On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
> >> >> >> > <mcampbellsmith at gmail.com> wrote:
> >> >> >> >>
> >> >> >> >> Thanks Brian.. this still does not work.  Maybe I need to open
> a
> >> >> >> >> Jira?
> >> >> >> >>  Notice the username in the authorization field.  It should be
> >> >> >> >> 1000.
> >> >> >> >>
> >> >> >> >> Cheers
> >> >> >> >> Mark
> >> >> >> >>
> >> >> >> >> freeswitch at internal> sofia status profile internal user
> >> >> >> >> 1000 at 192.168.1.120
> >> >> >> >>
> >> >> >> >> Registrations:
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> =================================================================================================
> >> >> >> >> Call-ID:        bd783b73-66877627 at 192.168.1.121
> >> >> >> >> User:           1000 at 192.168.1.120
> >> >> >> >> Contact:        1000 <sip:1000 at 192.168.1.121:5060>
> >> >> >> >> Agent:          Linksys/PAP2T-5.1.6(LS)
> >> >> >> >> Status:         Registered(UDP)(unknown) EXP(2010-01-25
> 11:25:05)
> >> >> >> >> Host:           freeswitch
> >> >> >> >> IP:             192.168.1.121
> >> >> >> >> Port:           5060
> >> >> >> >> Auth-User:      1000
> >> >> >> >> Auth-Realm:     192.168.1.120
> >> >> >> >> MWI-Account:    1000 at 192.168.1.120
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> =================================================================================================
> >> >> >> >>
> >> >> >> >> freeswitch at internal> sofia profile internal flush_inbound_reg
> >> >> >> >> bd783b73-66877627 at 192.168.1.121 reboot
> >> >> >> >> +OK rebooting all registrations matching specified call_id
> >> >> >> >>
> >> >> >> >> freeswitch at internal> send 804 bytes to
> udp/[192.168.1.121]:5060
> >> >> >> >> at
> >> >> >> >> 23:55:49.012627:
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> >>   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
> >> >> >> >>   Via: SIP/2.0/UDP
> >> >> >> >> 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
> >> >> >> >>   Max-Forwards: 70
> >> >> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
> >> >> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >> >> >>   CSeq: 126070338 NOTIFY
> >> >> >> >>   Contact: <sip:mod_sofia at 192.168.1.120:5060>
> >> >> >> >>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
> >> >> >> >>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
> >> >> >> >> INFO,
> >> >> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >> >> >>   Supported: timer, precondition, path, replaces
> >> >> >> >>   Event: reboot_now
> >> >> >> >>   Allow-Events: talk, presence, dialog, line-seize, call-info,
> >> >> >> >> sla,
> >> >> >> >> include-session-description, presence.winfo, message-summary,
> >> >> >> >> refer
> >> >> >> >>   Subscription-State: terminated;reason=timeout
> >> >> >> >>   Content-Type: application/simple-message-summary
> >> >> >> >>   Content-Length: 0
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at
> 23:55:49.045267:
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> >>   SIP/2.0 401 Unauthorized
> >> >> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
> >> >> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >> >> >>   CSeq: 126070338 NOTIFY
> >> >> >> >>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
> >> >> >> >>   Server: Linksys/PAP2T-5.1.6(LS)
> >> >> >> >>   WWW-Authenticate: Digest realm="192.168.1.120",
> >> >> >> >> nonce="8e54805b",
> >> >> >> >> qop="auth", algorithm=md5
> >> >> >> >>   Content-Length: 0
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> >> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> >>   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
> >> >> >> >>   Via: SIP/2.0/UDP
> >> >> >> >> 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
> >> >> >> >>   Max-Forwards: 70
> >> >> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
> >> >> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >> >> >>   CSeq: 126070339 NOTIFY
> >> >> >> >>   Contact: <sip:mod_sofia at 192.168.1.120:5060>
> >> >> >> >>   Expires: 3590
> >> >> >> >>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
> >> >> >> >>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
> >> >> >> >> INFO,
> >> >> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >> >> >>   Supported: timer, precondition, path, replaces
> >> >> >> >>   Event: reboot_now
> >> >> >> >>   Allow-Events: talk, presence, dialog, line-seize, call-info,
> >> >> >> >> sla,
> >> >> >> >> include-session-description, presence.winfo, message-summary,
> >> >> >> >> refer
> >> >> >> >>   Subscription-State: terminated;reason=timeout
> >> >> >> >>   Authorization: Digest username="1115633124",
> >> >> >> >> realm="192.168.1.120",
> >> >> >> >> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q",
> algorithm=MD5,
> >> >> >> >> uri="sip:1000 at 192.168.1.121:5060",
> >> >> >> >> response="747b4d04544c84535dbbd987f2999ca7", qop=auth,
> >> >> >> >> nc=00000001
> >> >> >> >>   Content-Type: application/simple-message-summary
> >> >> >> >>   Content-Length: 0
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at
> 23:55:49.086375:
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> >>   SIP/2.0 401 Unauthorized
> >> >> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
> >> >> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >> >> >>   CSeq: 126070339 NOTIFY
> >> >> >> >>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
> >> >> >> >>   Server: Linksys/PAP2T-5.1.6(LS)
> >> >> >> >>   WWW-Authenticate: Digest realm="192.168.1.120",
> >> >> >> >> nonce="5339c7ba",
> >> >> >> >> qop="auth", algorithm=md5
> >> >> >> >>   Content-Length: 0
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> >>
> >> >> >> >> On Mon, Jan 25, 2010 at 4:46 AM, Brian West
> >> >> >> >> <brian at freeswitch.org>
> >> >> >> >> wrote:
> >> >> >> >> > or sofia profile xxx flush_inbound_reg callid reboot
> >> >> >> >> > callid you can get from sofia status profile xxx
> >> >> >> >> > /b
> >> >> >> >> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
> >> >> >> >> >
> >> >> >> >> > Actually I just
> >> >> >> >> > found http://wiki.freeswitch.org/wiki/Mod_event_socket
> >> >> >> >> >
> >> >> >> >> > If I telnet to FS as described
> >> >> >> >> >
> http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client,
> >> >> >> >> > do
> >> >> >> >> > I
> >> >> >> >> > just need to enter somthing like:
> >> >> >> >> >
> >> >> >> >> > sendevent NOTIFY
> >> >> >> >> > profile: internal
> >> >> >> >> > event-string: resync
> >> >> >> >> > user: 1000
> >> >> >> >> > host: 192.168.1.121
> >> >> >> >> > content-type: application/simple-message-summary
> >> >> >> >> >
> >> >> >> >> > where 192.168.1.121 is the ip address of one of the Linksys
> >> >> >> >> > devices?
> >> >> >> >> >
> >> >> >> >> > I don't see any messages sent when I do this.  What am I
> doing
> >> >> >> >> > wrong?
> >> >> >> >> >
> >> >> >> >> > Thanks
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > _______________________________________________
> >> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> > http://www.freeswitch.org
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >>
> >> >> >> >> _______________________________________________
> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mcampbellsmith at gmail.com  Wed Jan 27 19:39:54 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 28 Jan 2010 14:39:54 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
	<33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
	<191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>
Message-ID: <33c87fa31001271939i668e9676y1f687dc182b4ca89@mail.gmail.com>

Apologies Anthony...

This is the printout from sofia.  2000 does not work sending custom
NOTIFY messages and 2001 does.  Is it that 2001 is NAT'd, the port
10281 out of range or the contact incorrect or something?

Thanks!

Call-ID:        a7c8c53f-c18596ef at 192.168.1.3
User:           2001 at 192.168.1.120
Contact:        2001 <sip:2001 at 124.yyy.yyy.yyy:5072;transport=tls>
Agent:          Linksys/SPA3102-5.1.10(GW)
Status:         Registered(TLS)(unknown) EXP(2010-01-28 12:54:31)
Host:           freeswitch
IP:             124.yyy.yyy.yyy
Port:           5072
Auth-User:      2001
Auth-Realm:     mydns.dyndns.org
MWI-Account:    2001 at 192.168.1.120

Call-ID:        2ff39277-fd9a6ab1 at 10.0.0.1
User:           2000 at 192.168.1.120
Contact:        2000
<sip:2000 at 124.xxx.xxx.xxx:5075;transport=tls;fs_nat=yes;fs_path=sip%3A2000%40124.xxx.xxx.xxx%3A10281%3Btransport%3Dtls>
Agent:          Linksys/SPA3102-5.1.10(GW)
Status:         Registered(TLS-NAT)(unknown) EXP(2010-01-28 00:29:34)
Host:           freeswitch
IP:             124.xxx.xxx.xxx
Port:           10281
Auth-User:      2000
Auth-Realm:     mydns.dyndns.org
MWI-Account:    2000 at 192.168.1.120

On Thu, Jan 28, 2010 at 2:25 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> you said one was the word freeswitch and one was an ip didn't you?
>
>
> On Wed, Jan 27, 2010 at 8:34 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>>
>> If it works with 2001 doesn't that mean I am using the correct host?
>>
>> Both 2001 and 2000 register with exactly the same data, except
>> username and password ....
>>
>>
>>
>> On Thu, Jan 28, 2010 at 12:41 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > the host is not resolved it has to be an exact string match with the
>> > host
>> > that is in the db.
>> > if you want to normalize it set force-reg-domain and force-reg-db-domain
>> > to
>> > the same val
>> >
>> >
>> > On Wed, Jan 27, 2010 at 7:26 PM, Mark Campbell-Smith
>> > <mcampbellsmith at gmail.com> wrote:
>> >>
>> >> Thanks Anthony,
>> >>
>> >> I think user matches (ie the extension 2000 or 2001). ? What should
>> >> host be? ?In the sofia printout, it says 'freeswitch' (freeswitch has
>> >> ip address 192.168.1.120).
>> >>
>> >> However, if I try to use 'freeswitch' as the host for user 2001,
>> >> nothing is sent. ?But using 192.168.1.120 does.
>> >>
>> >> If I do exactly the same thing for 2000, the NOTIFY message is not
>> >> sent. ?Are there logs I can send to show you or any ideas what I am
>> >> doing wrong?
>> >>
>> >>
>> >>
>> >> On Thu, Jan 28, 2010 at 12:09 PM, Anthony Minessale
>> >> <anthony.minessale at gmail.com> wrote:
>> >> > user and host have to match too
>> >> >
>> >> > On Wed, Jan 27, 2010 at 7:06 AM, Mark Campbell-Smith
>> >> > <mcampbellsmith at gmail.com> wrote:
>> >> >>
>> >> >> Thanks guys. ?I have this working except for one user who is
>> >> >> registered like this:
>> >> >>
>> >> >> Call-ID: ? ? ? ?2ff39277-fd9a6ab1 at 10.0.0.1
>> >> >> User: ? ? ? ? ? 2000 at 192.168.1.120
>> >> >> Contact: ? ? ? ?2000
>> >> >>
>> >> >>
>> >> >>
>> >> >> <sip:2000 at 124.xxx.xxx.xxx:5075;transport=tls;fs_nat=yes;fs_path=sip%3A2000%40124.xxx.xxx.xxx%3A10281%3Btransport%3Dtls>
>> >> >> Agent: ? ? ? ? ?Linksys/SPA3102-5.1.10(GW)
>> >> >> Status: ? ? ? ? Registered(TLS-NAT)(unknown) EXP(2010-01-28
>> >> >> 00:29:34)
>> >> >> Host: ? ? ? ? ? freeswitch
>> >> >> IP: ? ? ? ? ? ? 124.xxx.xxx.xxx
>> >> >> Port: ? ? ? ? ? 10281
>> >> >> Auth-User: ? ? ?2000
>> >> >> Auth-Realm: ? ? mydns.dyndns.org
>> >> >> MWI-Account: ? ?2000 at 192.168.1.120
>> >> >>
>> >> >> When I do the following commands via the telnet socket, no notify
>> >> >> command is sent to user 2000:
>> >> >>
>> >> >> sendevent NOTIFY
>> >> >> profile: internal
>> >> >> content-type: application/simple-message-summary
>> >> >> event-string: reboot_now
>> >> >> user: 2000
>> >> >> host: 192.168.1.120
>> >> >> content-length: 0
>> >> >>
>> >> >> However, if I do exactly the same thing with user 2001 it works.
>> >> >> ?2001
>> >> >> is registered as:
>> >> >>
>> >> >> Contact: ? ? ? ?2001 <sip:2001 at 124.yyy.yyy.yyy:5066;transport=tls>
>> >> >>
>> >> >> Any ideas why that would be?
>> >> >>
>> >> >> On Mon, Jan 25, 2010 at 11:50 AM, Ognjen Seslija
>> >> >> <oseslija at gmail.com>
>> >> >> wrote:
>> >> >> > The phone is asking FS to authenticate prior then accepting a
>> >> >> > NOTIFY
>> >> >> > from
>> >> >> > it.
>> >> >> > The authentication of notify's from spa endpoints work (afaik)
>> >> >> > only
>> >> >> > with
>> >> >> > Linksys SPA9000 PBX , and FS doesn't have the code for that for
>> >> >> > obvious
>> >> >> > reasons.
>> >> >> > If you have SPA9000 maybe you can collect SIP traces.
>> >> >> >
>> >> >> > Ognjen
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > On Mon, Jan 25, 2010 at 1:29 AM, Mark Campbell-Smith
>> >> >> > <mcampbellsmith at gmail.com> wrote:
>> >> >> >>
>> >> >> >> Hi Ognjen,
>> >> >> >>
>> >> >> >> Thanks for the tip on the resync under the EXT tab. ?It now works
>> >> >> >> using mod_event_socket and the following:
>> >> >> >>
>> >> >> >> sendevent NOTIFY
>> >> >> >> profile: internal
>> >> >> >> event-string: resync
>> >> >> >> user: 1000
>> >> >> >> host: 192.168.1.121
>> >> >> >> content-type: application/simple-message-summary
>> >> >> >>
>> >> >> >> However, if AUTH is required, why does FS send the wrong
>> >> >> >> information
>> >> >> >> to
>> >> >> >> the SPA?
>> >> >> >>
>> >> >> >> On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija
>> >> >> >> <oseslija at gmail.com>
>> >> >> >> wrote:
>> >> >> >> > You? should not authenticate those NOTIFYs (this will work only
>> >> >> >> > with
>> >> >> >> > SPA9000
>> >> >> >> > afaik). The option to change for this is in EXT tabs:
>> >> >> >> >
>> >> >> >> > Auth Resync-Reboot: No
>> >> >> >> >
>> >> >> >> > Also, FSs code will do a reboot of a phone, not resync (it
>> >> >> >> > sends
>> >> >> >> > reboot_now
>> >> >> >> > event). For that to work a patch is required.
>> >> >> >> >
>> >> >> >> > I've just tried to reboot my 942 (rev 16506) and it definitely
>> >> >> >> > works.
>> >> >> >> >
>> >> >> >> > Regards,
>> >> >> >> > Ognjen
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
>> >> >> >> > <mcampbellsmith at gmail.com> wrote:
>> >> >> >> >>
>> >> >> >> >> Thanks Brian.. this still does not work. ?Maybe I need to open
>> >> >> >> >> a
>> >> >> >> >> Jira?
>> >> >> >> >> ?Notice the username in the authorization field. ?It should be
>> >> >> >> >> 1000.
>> >> >> >> >>
>> >> >> >> >> Cheers
>> >> >> >> >> Mark
>> >> >> >> >>
>> >> >> >> >> freeswitch at internal> sofia status profile internal user
>> >> >> >> >> 1000 at 192.168.1.120
>> >> >> >> >>
>> >> >> >> >> Registrations:
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> =================================================================================================
>> >> >> >> >> Call-ID: ? ? ? ?bd783b73-66877627 at 192.168.1.121
>> >> >> >> >> User: ? ? ? ? ? 1000 at 192.168.1.120
>> >> >> >> >> Contact: ? ? ? ?1000 <sip:1000 at 192.168.1.121:5060>
>> >> >> >> >> Agent: ? ? ? ? ?Linksys/PAP2T-5.1.6(LS)
>> >> >> >> >> Status: ? ? ? ? Registered(UDP)(unknown) EXP(2010-01-25
>> >> >> >> >> 11:25:05)
>> >> >> >> >> Host: ? ? ? ? ? freeswitch
>> >> >> >> >> IP: ? ? ? ? ? ? 192.168.1.121
>> >> >> >> >> Port: ? ? ? ? ? 5060
>> >> >> >> >> Auth-User: ? ? ?1000
>> >> >> >> >> Auth-Realm: ? ? 192.168.1.120
>> >> >> >> >> MWI-Account: ? ?1000 at 192.168.1.120
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> =================================================================================================
>> >> >> >> >>
>> >> >> >> >> freeswitch at internal> sofia profile internal flush_inbound_reg
>> >> >> >> >> bd783b73-66877627 at 192.168.1.121 reboot
>> >> >> >> >> +OK rebooting all registrations matching specified call_id
>> >> >> >> >>
>> >> >> >> >> freeswitch at internal> send 804 bytes to
>> >> >> >> >> udp/[192.168.1.121]:5060
>> >> >> >> >> at
>> >> >> >> >> 23:55:49.012627:
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> ------------------------------------------------------------------------
>> >> >> >> >> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> >> >> >> >> ? Via: SIP/2.0/UDP
>> >> >> >> >> 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
>> >> >> >> >> ? Max-Forwards: 70
>> >> >> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> >> >> ? To: <sip:1000 at 192.168.1.120>
>> >> >> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> >> >> ? CSeq: 126070338 NOTIFY
>> >> >> >> >> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> >> >> >> >> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> >> >> >> >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>> >> >> >> >> INFO,
>> >> >> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> >> >> >> ? Supported: timer, precondition, path, replaces
>> >> >> >> >> ? Event: reboot_now
>> >> >> >> >> ? Allow-Events: talk, presence, dialog, line-seize, call-info,
>> >> >> >> >> sla,
>> >> >> >> >> include-session-description, presence.winfo, message-summary,
>> >> >> >> >> refer
>> >> >> >> >> ? Subscription-State: terminated;reason=timeout
>> >> >> >> >> ? Content-Type: application/simple-message-summary
>> >> >> >> >> ? Content-Length: 0
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> ------------------------------------------------------------------------
>> >> >> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at
>> >> >> >> >> 23:55:49.045267:
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> ------------------------------------------------------------------------
>> >> >> >> >> ? SIP/2.0 401 Unauthorized
>> >> >> >> >> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> >> >> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> >> >> ? CSeq: 126070338 NOTIFY
>> >> >> >> >> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
>> >> >> >> >> ? Server: Linksys/PAP2T-5.1.6(LS)
>> >> >> >> >> ? WWW-Authenticate: Digest realm="192.168.1.120",
>> >> >> >> >> nonce="8e54805b",
>> >> >> >> >> qop="auth", algorithm=md5
>> >> >> >> >> ? Content-Length: 0
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> ------------------------------------------------------------------------
>> >> >> >> >> send 1056 bytes to udp/[192.168.1.121]:5060 at
>> >> >> >> >> 23:55:49.060073:
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> ------------------------------------------------------------------------
>> >> >> >> >> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> >> >> >> >> ? Via: SIP/2.0/UDP
>> >> >> >> >> 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
>> >> >> >> >> ? Max-Forwards: 70
>> >> >> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> >> >> ? To: <sip:1000 at 192.168.1.120>
>> >> >> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> >> >> ? CSeq: 126070339 NOTIFY
>> >> >> >> >> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> >> >> >> >> ? Expires: 3590
>> >> >> >> >> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> >> >> >> >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>> >> >> >> >> INFO,
>> >> >> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> >> >> >> ? Supported: timer, precondition, path, replaces
>> >> >> >> >> ? Event: reboot_now
>> >> >> >> >> ? Allow-Events: talk, presence, dialog, line-seize, call-info,
>> >> >> >> >> sla,
>> >> >> >> >> include-session-description, presence.winfo, message-summary,
>> >> >> >> >> refer
>> >> >> >> >> ? Subscription-State: terminated;reason=timeout
>> >> >> >> >> ? Authorization: Digest username="1115633124",
>> >> >> >> >> realm="192.168.1.120",
>> >> >> >> >> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q",
>> >> >> >> >> algorithm=MD5,
>> >> >> >> >> uri="sip:1000 at 192.168.1.121:5060",
>> >> >> >> >> response="747b4d04544c84535dbbd987f2999ca7", qop=auth,
>> >> >> >> >> nc=00000001
>> >> >> >> >> ? Content-Type: application/simple-message-summary
>> >> >> >> >> ? Content-Length: 0
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> ------------------------------------------------------------------------
>> >> >> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at
>> >> >> >> >> 23:55:49.086375:
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> ------------------------------------------------------------------------
>> >> >> >> >> ? SIP/2.0 401 Unauthorized
>> >> >> >> >> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> >> >> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> >> >> ? CSeq: 126070339 NOTIFY
>> >> >> >> >> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
>> >> >> >> >> ? Server: Linksys/PAP2T-5.1.6(LS)
>> >> >> >> >> ? WWW-Authenticate: Digest realm="192.168.1.120",
>> >> >> >> >> nonce="5339c7ba",
>> >> >> >> >> qop="auth", algorithm=md5
>> >> >> >> >> ? Content-Length: 0
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> ------------------------------------------------------------------------
>> >> >> >> >>
>> >> >> >> >> On Mon, Jan 25, 2010 at 4:46 AM, Brian West
>> >> >> >> >> <brian at freeswitch.org>
>> >> >> >> >> wrote:
>> >> >> >> >> > or sofia profile xxx flush_inbound_reg callid reboot
>> >> >> >> >> > callid you can get from sofia status profile xxx
>> >> >> >> >> > /b
>> >> >> >> >> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
>> >> >> >> >> >
>> >> >> >> >> > Actually I just
>> >> >> >> >> > found?http://wiki.freeswitch.org/wiki/Mod_event_socket
>> >> >> >> >> >
>> >> >> >> >> > If I telnet to FS as described
>> >> >> >> >> >
>> >> >> >> >> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client,
>> >> >> >> >> > do
>> >> >> >> >> > I
>> >> >> >> >> > just need to enter somthing like:
>> >> >> >> >> >
>> >> >> >> >> > sendevent NOTIFY
>> >> >> >> >> > profile: internal
>> >> >> >> >> > event-string: resync
>> >> >> >> >> > user: 1000
>> >> >> >> >> > host: 192.168.1.121
>> >> >> >> >> > content-type: application/simple-message-summary
>> >> >> >> >> >
>> >> >> >> >> > where 192.168.1.121 is the ip address of one of the Linksys
>> >> >> >> >> > devices?
>> >> >> >> >> >
>> >> >> >> >> > I don't see any messages sent when I do this. ?What am I
>> >> >> >> >> > doing
>> >> >> >> >> > wrong?
>> >> >> >> >> >
>> >> >> >> >> > Thanks
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> > _______________________________________________
>> >> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> >> >
>> >> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> >> > http://www.freeswitch.org
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >>
>> >> >> >> >> _______________________________________________
>> >> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> >> http://www.freeswitch.org
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > _______________________________________________
>> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> > http://www.freeswitch.org
>> >> >> >> >
>> >> >> >> >
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> > Anthony Minessale II
>> >> >
>> >> > FreeSWITCH http://www.freeswitch.org/
>> >> > ClueCon http://www.cluecon.com/
>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >
>> >> > AIM: anthm
>> >> > MSN:anthony_minessale at hotmail.com
>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> > IRC: irc.freenode.net #freeswitch
>> >> >
>> >> > FreeSWITCH Developer Conference
>> >> > sip:888 at conference.freeswitch.org
>> >> > iax:guest at conference.freeswitch.org/888
>> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> > pstn:+19193869900
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Wed Jan 27 19:54:07 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 27 Jan 2010 21:54:07 -0600
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001271939i668e9676y1f687dc182b4ca89@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
	<33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
	<191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>
	<33c87fa31001271939i668e9676y1f687dc182b4ca89@mail.gmail.com>
Message-ID: <CF0D5382-EE44-48CB-BF14-27A443014336@freeswitch.org>

I'm suspecting the code just isn't honoring the fs_path and sending it to the right place do you have a sip trace of this?

/b

On Jan 27, 2010, at 9:39 PM, Mark Campbell-Smith wrote:

> This is the printout from sofia.  2000 does not work sending custom
> NOTIFY messages and 2001 does.  Is it that 2001 is NAT'd, the port
> 10281 out of range or the contact incorrect or something?



From mcampbellsmith at gmail.com  Wed Jan 27 20:35:47 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 28 Jan 2010 15:35:47 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <CF0D5382-EE44-48CB-BF14-27A443014336@freeswitch.org>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
	<33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
	<191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>
	<33c87fa31001271939i668e9676y1f687dc182b4ca89@mail.gmail.com>
	<CF0D5382-EE44-48CB-BF14-27A443014336@freeswitch.org>
Message-ID: <33c87fa31001272035t7c46e054t7ce2dd598fa40ebb@mail.gmail.com>

Hi Brian,

 I've previously enabled siptrace for internal profile, but I see
nothing sent and nothing received.

On Thu, Jan 28, 2010 at 2:54 PM, Brian West <brian at freeswitch.org> wrote:
> I'm suspecting the code just isn't honoring the fs_path and sending it to the right place do you have a sip trace of this?
>
> /b
>
> On Jan 27, 2010, at 9:39 PM, Mark Campbell-Smith wrote:
>
>> This is the printout from sofia. ?2000 does not work sending custom
>> NOTIFY messages and 2001 does. ?Is it that 2001 is NAT'd, the port
>> 10281 out of range or the contact incorrect or something?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From wiltingtree at gmail.com  Wed Jan 27 20:47:35 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Wed, 27 Jan 2010 23:47:35 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
Message-ID: <eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>

I tried running ldconfig on the directory containing mysql.so, but it did
not help.
So it sounds like there could be a bug in the latter versions?


On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> I got the same error, my script was working with no problems before an
> update to trunk.
>
> David
>
> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com> wrote:
> > Hi, I followed the instructions in the Lua documentation for setting up
> > luasql, but when I try to run my script I get:
> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module
> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
> >         /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment
> prot
> > after reloc: Permission denied
> > stack traceback:
> >         [C]: ?
> >         [C]: in function 'require'
> >         /usr/local/freeswitch/scripts/l.lua:2: in main chunk
> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
> > I changed the permissions for mysql.so and for my script to 777, so I'm
> not
> > sure where the permission problem could be.
> > I'd appreciate any suggestions.
> > Thanks,
> > Adam
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Wed Jan 27 21:18:51 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 28 Jan 2010 00:18:51 -0500
Subject: [Freeswitch-users] compilation error on fs_cli (Windows)
	(latest version)
In-Reply-To: <761142.19511.qm@web33506.mail.mud.yahoo.com>
References: <761142.19511.qm@web33506.mail.mud.yahoo.com>
Message-ID: <CC043BFD-5F5E-439E-AA16-EBCA411EA3C1@jerris.com>

These are real issues we need to fix.  Please open a bug on jira for these (even better with a patch to fix them).

Mie

On Jan 26, 2010, at 10:16 PM, Diego Toro wrote:

> 
> yes, code analyzer is active. when I turn it off fs_cli project compiled fine. Before, this project compiled fine, why I need turn off analyzer code now ?
> 
> Thank you
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 
> 
> --- On Tue, 1/26/10, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> 
>> From: Mathieu Rene <mrene_lists at avgs.ca>
>> Subject: Re: [Freeswitch-users] compilation error on fs_cli (Windows) (latest version)
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Tuesday, January 26, 2010, 11:45 AM
>> Looks like the code analyzer is
>> running, this is normally turned off  
>> when you do a normal build, turn it off and try again.
>> 
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>> 
>> 
>> 
>> 
>> On 26-Jan-10, at 10:55 AM, Diego Toro wrote:
>> 
>>> Hi, I have compilation error "error C2220" on fs_cli
>> project on  
>>> Windows using VS2008.
>>> 
>>> FS: latest version (2010/01/26)
>>> VS: VS2008
>>> SO: Windows 7
>>> 
>>> VS2008 Error log:
>>> 
>>> Error    1    error
>> C2220: warning treated as error - no 'object' file  
>>> generated   
>> g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl 
>>> \fs_cli.c    106   
>> fs_cli
>>> 
>>> Warning    2    warning
>> C6385: Invalid data: accessing 'global_profile- 
>>>> console_fnkeys', the readable size is '48' bytes,
>> but '-4' bytes  
>>> might be read: Lines: 86, 88, 90   
>> g:\ftp\incoming\fs 
>>> 
>> \freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c   
>> 90    fs_cli
>>> 
>>> Warning    3    warning
>> C6246: Local declaration of 'p' hides declaration  
>>> of the same name in outer scope. For additional
>> information, see  
>>> previous declaration at line '844' of
>> 'g:\ftp\incoming\fs 
>>> \freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c':
>> Lines: 844    g:\ftp 
>>> 
>> \incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c   
>> 895     
>>> fs_cli
>>> 
>>> Warning    4    warning
>> C6011: Dereferencing NULL pointer 'cursor': Lines:  
>>> 839, 840, 841, 842, 843, 844, 846, 849, 853, 857, 868,
>> 870, 871, 884     
>>> 
>> g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c     
>>> 884    fs_cli



From anthony.minessale at gmail.com  Wed Jan 27 21:25:16 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 23:25:16 -0600
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001272035t7c46e054t7ce2dd598fa40ebb@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
	<33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
	<191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>
	<33c87fa31001271939i668e9676y1f687dc182b4ca89@mail.gmail.com>
	<CF0D5382-EE44-48CB-BF14-27A443014336@freeswitch.org>
	<33c87fa31001272035t7c46e054t7ce2dd598fa40ebb@mail.gmail.com>
Message-ID: <191c3a031001272125t2b19d134lbb65dfebbb7a52b9@mail.gmail.com>

You have to look in the sql db and compare the specified vals with the ones
looked up from the event again the user and host need to match the db

On Jan 27, 2010 10:41 PM, "Mark Campbell-Smith" <mcampbellsmith at gmail.com>
wrote:

Hi Brian,

 I've previously enabled siptrace for internal profile, but I see
nothing sent and nothing received.


On Thu, Jan 28, 2010 at 2:54 PM, Brian West <brian at freeswitch.org> wrote:
> I'm suspecting the code...
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From mike at van.lammeren.net  Wed Jan 27 22:27:45 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 28 Jan 2010 01:27:45 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
Message-ID: <5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>

Have you tried running a Lua script that includes the library from outside
of FreeSWITCH? What does that do?


On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> I tried running ldconfig on the directory containing mysql.so, but it did
> not help.
> So it sounds like there could be a bug in the latter versions?
>
>
> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> I got the same error, my script was working with no problems before an
>> update to trunk.
>>
>> David
>>
>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com> wrote:
>> > Hi, I followed the instructions in the Lua documentation for setting up
>> > luasql, but when I try to run my script I get:
>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module
>> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
>> >         /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment
>> prot
>> > after reloc: Permission denied
>> > stack traceback:
>> >         [C]: ?
>> >         [C]: in function 'require'
>> >         /usr/local/freeswitch/scripts/l.lua:2: in main chunk
>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
>> > I changed the permissions for mysql.so and for my script to 777, so I'm
>> not
>> > sure where the permission problem could be.
>> > I'd appreciate any suggestions.
>> > Thanks,
>> > Adam
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ranjtech at gmail.com  Wed Jan 27 23:11:29 2010
From: ranjtech at gmail.com (RR)
Date: Thu, 28 Jan 2010 02:11:29 -0500
Subject: [Freeswitch-users] 409 Conflict When registering FS with Softswitch
Message-ID: <020c01ca9fe9$1d5952f0$580bf8d0$@com>

Gentlemen,

 

I have a probably a simple problem but I have no idea why it's occurring. I
am beyond novice/new to configuring FS and as my first try, tried to
configure it to register with our softswitch so I could have the basic
registration to happen between FS and it. However, I keep getting the "409
Conflict" message when that username is not actually registered with the
softswitch. The profile / config file under sip_profiles/external is
configured like so

 

<include>

  <gateway name="Test-Inbound">

  <param name="username" value="user1"/>-->

  <param name="password" value="user1pwd"/>-->

  <param name="realm" value="mysoftswitch.net.au"/>-->

  <param name="extension" value="1000"/>

  <param name="from-user" value="user1"/>-->

  <param name="expire-seconds" value="60"/>-->

  <param name="register" value="true"/>-->

  </gateway>-->

</include>

 

Any ideas why? And how to resolve it or what kind of traces I can send to
you guys to help me resolve it? Like I said I am so new to FS that I don't
even know what I can turn on in FS to see what's going on

 

Thanks a lot

\RR

 

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From david.villasmil.work at gmail.com  Wed Jan 27 23:43:59 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 28 Jan 2010 08:43:59 +0100
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
Message-ID: <9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>

Hello,

That works fine:

box:~# lua testdb.lua
box:~#


David

On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
<mike at van.lammeren.net> wrote:
> Have you tried running a Lua script that includes the library from outside
> of FreeSWITCH? What does that do?
>
> On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
>>
>> I tried running ldconfig on the directory containing mysql.so, but it did
>> not help.
>> So it sounds like there could be a bug in the latter versions?
>>
>> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
>> <david.villasmil.work at gmail.com> wrote:
>>>
>>> I got the same error, my script was working with no problems before an
>>> update to trunk.
>>>
>>> David
>>>
>>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com> wrote:
>>> > Hi, I followed the instructions in the Lua documentation for setting up
>>> > luasql, but when I try to run my script I get:
>>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module
>>> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
>>> > ?? ? ? ?/usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment
>>> > prot
>>> > after reloc: Permission denied
>>> > stack traceback:
>>> > ?? ? ? ?[C]: ?
>>> > ?? ? ? ?[C]: in function 'require'
>>> > ?? ? ? ?/usr/local/freeswitch/scripts/l.lua:2: in main chunk
>>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
>>> > I changed the permissions for mysql.so and for my script to 777, so I'm
>>> > not
>>> > sure where the permission problem could be.
>>> > I'd appreciate any suggestions.
>>> > Thanks,
>>> > Adam
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Wed Jan 27 23:49:56 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 28 Jan 2010 01:49:56 -0600
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
Message-ID: <1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>

Can you provide a SIP Trace?

/b

On Jan 28, 2010, at 1:11 AM, RR wrote:

> Gentlemen,
>  
> I have a probably a simple problem but I have no idea why it?s occurring. I am beyond novice/new to configuring FS and as my first try, tried to configure it to register with our softswitch so I could have the basic registration to happen between FS and it. However, I keep getting the ?409 Conflict? message when that username is not actually registered with the softswitch. The profile / config file under sip_profiles/external is configured like so
>  
> <include>
>   <gateway name="Test-Inbound">
>   <param name="username" value="user1"/>-->
>   <param name="password" value="user1pwd"/>-->
>   <param name="realm" value="mysoftswitch.net.au"/>-->
>   <param name="extension" value="1000"/>
>   <param name="from-user" value="user1"/>-->
>   <param name="expire-seconds" value="60"/>-->
>   <param name="register" value="true"/>-->
>   </gateway>-->
> </include>
>  
> Any ideas why? And how to resolve it or what kind of traces I can send to you guys to help me resolve it? Like I said I am so new to FS that I don?t even know what I can turn on in FS to see what?s going on
>  
> Thanks a lot
> \RR
>  
> 
> 
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4812 (20100128) __________
> 
> The message was checked by ESET NOD32 Antivirus.
> 
> http://www.eset.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From siniypin at gmail.com  Thu Jan 28 00:09:18 2010
From: siniypin at gmail.com (RobertT)
Date: Thu, 28 Jan 2010 11:09:18 +0300
Subject: [Freeswitch-users] Presence Change Distribution
In-Reply-To: <591B9C113F064880993543272B16ADF3@greyhawk.tonecommander.com>
References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com>
	<2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com>
	<26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com>
	<191c3a031001251104p55ba7009g9381841f7de56d65@mail.gmail.com>
	<BE6F5600B8E54EE297963A8B08F70E5E@greyhawk.tonecommander.com>
	<191c3a031001261321v2e8ea21cm7da19e01a11b59f9@mail.gmail.com>
	<591B9C113F064880993543272B16ADF3@greyhawk.tonecommander.com>
Message-ID: <2160023e1001280009v65d9b3ees78a8cb319205649a@mail.gmail.com>

You can send your own custom notes in xml. That is what I do to make
presence a little bit reliable.

Best regards, Robert.

2010/1/27 Jerry Richards <jerry.richards at teotech.com>

>  There are two places in the XML body that are diffierent:
>
> FS Rcvd PUBLISH has:  <rpid:away/> and <dm:note>Away</dm:note>
> FS Sent NOTIFY   has:  <rpid:busy/> and <dm:note>Busy</dm:note>
>
> This behavior (above) is why I'm not seeing the published presence at the
> subscribing softphone.  FS should be sending the new Away status in the
> NOTIFY message.
>
> I did notice there is an "[ERR] sofia_presnece.c:674" FS log between the
> PUBLISH and the NOTIFY (please see Line 89 of
> http://pastebin.freeswitch.org/11953).  Line 674 is in the
> sofia_presence_event_thread_run() function where it calls
> switch_mutex_unlock(mod_sofia_globals.mutex).  Do you think this [ERR] is
> related to why FS sends the previous status and not updated status?
>
> Thanks And Best Regards,
> Jerry
>
>
>  ------------------------------
> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
> *Sent:* Tuesday, January 26, 2010 1:22 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>
> its sending a notify to them right away (line 174 of your PB)
> the xml in the notify we send looks the same as what they sent except one
> thing
>
> They send: <rpid:away/>
> We send: <rpid:busy/>
>
> everybody who implements this seems to have their own idea of what to say
> here.
>
> This crazy xml presence crap is pure garbage so maybe that's it.
>
>
>
> On Tue, Jan 26, 2010 at 3:02 PM, Jerry Richards <
> jerry.richards at teotech.com> wrote:
>
>>  Okay, I setup my FS to force SUBSCRIBE Expires to 3600 seconds.  Then I
>> captured a FS console trace of a Bria softphone changing it's presence state
>> from 'Busy' to 'Away' (see http://pastebin.freeswitch.org/11953) and
>> observed that the subscribing Bria softphone did not update to 'Away'.  At
>> the same time, I executed the sqlite3 app and pasted each of the 3 SQL
>> select statements I saw in the FS console log, and pasted them below.  I'm
>> new to sqlite3.  Do you see what my issue is?
>>
>> sqlite>
>> sqlite> select
>> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
>> from sip_subscriptions left join sip_presence on
>> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
>> sip_subscriptions.sub_to_host=sip_presence.sip_host and
>> sip_subscriptions.profile_name=sip_presence.profile_name) where
>> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
>> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
>> like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'internal' or
>> sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
>> sip|5382|192.168.72.79|5401|192.168.72.79|presence|"5382 on 79" <
>> sip:5382 at 192.168.72.150:34672>|ZTQ2ZWQwZGRlZjRiNTdkYTJjNGM5NTgzOWIyNmIwZmU.|"5382
>> on 79" <sip:5382 at 192.168.72.79 <sip%3A5382 at 192.168.72.79>>;tag=68bb4eb6|SIP/2.0/UDP
>> 192.168.72.150:34672;branch=z9hG4bK-d8754z-eafc60166305eaef-1---d8754z-;rport=34672|1264546204|Teo
>> Softphone release 2.5.4 stamp
>> 55958||internal|Away|away|192.168.72.79|Away|away
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite> select
>> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
>> from sip_subscriptions left join sip_presence on
>> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
>> sip_subscriptions.sub_to_host=sip_presence.sip_host and
>> sip_subscriptions.profile_name=sip_presence.profile_name) where
>> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
>> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
>> like '%192.168.72.79%') and (sip_subscriptions.profile_name =
>> 'internal-ipv6' or sip_subscriptions.presence_hosts !=
>> sip_subscriptions.sub_to_host);
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite> select
>> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
>> from sip_subscriptions left join sip_presence on
>> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
>> sip_subscriptions.sub_to_host=sip_presence.sip_host and
>> sip_subscriptions.profile_name=sip_presence.profile_name) where
>> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
>> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
>> like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'external' or
>> sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
>> sqlite>
>>  Thanks and Best Regards,
>> Jerry
>>
>>  ------------------------------
>> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
>> *Sent:* Monday, January 25, 2010 11:05 AM
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>>
>>   the notify will be instant after the publish
>> the notify you see are not triggered by the publish or they would be
>> instant.
>>
>> Same drill, turn on presence debugging in sofia.conf.xml
>> and look at the sql stmts and see why
>>
>>
>> On Mon, Jan 25, 2010 at 12:30 PM, Jerry Richards <
>> jerry.richards at teotech.com> wrote:
>>
>>>  Okay, I notice that if I reduce the Presence SUBSCRIBE Expires duration
>>> (from 3600 seconds to 60 seconds), then the delay between PUBLISH's and
>>> NOTIFY's is reduced, but FS still waits nearly 45 seconds to send the
>>> NOTIFY's after it receives a PUBLISH.
>>>
>>> Can a change be made in FS so that NOTIFYs are sent as a direct result of
>>> receipt of the PUBLISH message?  And not tied to the SUBSCRIBE expiration?
>>> I really don't want to configure all my phones to re-subscribe every 30 or
>>> 15 seconds.
>>>
>>> Thanks and Best Regards,
>>> Jerry
>>>
>>>
>>>  ------------------------------
>>> *From:* RobertT [mailto:siniypin at gmail.com]
>>> *Sent:* Tuesday, December 29, 2009 12:02 AM
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>>>
>>>   You can try to reduce your registration time.
>>> I for one made my client apps send PUBLISH message every minute in
>>> addition to reduced registration time.
>>>
>>> Regards, Robert.
>>>
>>> 2009/12/28 Jerry Richards <jerry.richards at teotech.com>
>>>
>>>> Is there a setting to control how fast FS distributes presence changes
>>>> to
>>>> subscribers?  Currently, it appears to take several minutes before I see
>>>> presence changes.  I would like to see them almost instantaneously, if
>>>> possible.
>>>>
>>>> Thanks and Best Regards,
>>>> Jerry
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From codecomplete at free.fr  Thu Jan 28 02:24:07 2010
From: codecomplete at free.fr (Fred-145)
Date: Thu, 28 Jan 2010 11:24:07 +0100
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
References: <27338355.post@talk.nabble.com>
	<27338355.post-WJuSqJV8a7jJsTRiRinrng@public.gmane.org>
	<9BB5BC82B9F54466ACFA5BA610669FD7@cune.pri>
Message-ID: <mcp2m55d16dlaq51kt5a0k7m4iut33c1k1@4ax.com>

On Wed, 27 Jan 2010 06:47:28 -0600, "Russell Mosemann"
<Russell.Mosemann at cune.org> wrote:
>http://wiki.freeswitch.org/wiki/Bypass_Media

Thanks everyone. I'll experiment with Freeswitch in the LAN behind a
NAT router and check if performance is good enough to avoid using the
bypass-media option.



From codecomplete at free.fr  Thu Jan 28 03:04:46 2010
From: codecomplete at free.fr (Fred-145)
Date: Thu, 28 Jan 2010 12:04:46 +0100
Subject: [Freeswitch-users] Investigating one-way sound?
References: <27341219.post@talk.nabble.com>
	<27341219.post-WJuSqJV8a7jJsTRiRinrng@public.gmane.org>
	<22DD670E-B920-4328-9939-56447375D5C7@freeswitch.org>
Message-ID: <jfp2m55k5vedrabmubv801lgd404s5769j@4ax.com>

On Wed, 27 Jan 2010 09:30:41 -0600, Brian West
<brian at freeswitch.org> wrote:

> I'm going to guess right off the X-Lite is putting its public IP and
> since maybe your NAT router can't hair pin the media you get on way media.

Thanks Brian for the tip. I tried this, but it didn't work. Turns out
it's a bug in how XLite and the Siemens negotiate codecs, and it's now
working.



From codecomplete at free.fr  Thu Jan 28 03:54:16 2010
From: codecomplete at free.fr (Fred-145)
Date: Thu, 28 Jan 2010 12:54:16 +0100
Subject: [Freeswitch-users] [fs_cli] Only display SIP messages?
Message-ID: <mku2m5lrafrcvq4aqr0pclu1p9530hs73j@4ax.com>

Hello

Before I learn how to use Wireshark and filter stuff... can the fs_cli
console be configured so that only SIP messages are displayed? I'd
like to do this so I can learn more about what goes on when I play
with SIP clients.

Thank you.



From lakindia89 at gmail.com  Thu Jan 28 04:11:02 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 28 Jan 2010 17:41:02 +0530
Subject: [Freeswitch-users] Multiple DTMF on group_confirm_key
Message-ID: <7d79b3931001280411u6262f627xca723c64de5e118d@mail.gmail.com>

Hi all,

I've experimented with group confirm key and group confirm file. It works
great. However, I was unable to give multiple DTMF digits to get the
confirmation.

I've set group_confirm_key=1234, I thought it will ask the 4 digits from the
user. But it simply taken 1 and when the user presses 1, the call got
bridged.

Is there any way to specify multiple dtmf to be confirmed??
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From javieraristizabal at gmail.com  Thu Jan 28 05:57:14 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Thu, 28 Jan 2010 08:57:14 -0500
Subject: [Freeswitch-users] [fs_cli] Only display SIP messages?
In-Reply-To: <mku2m5lrafrcvq4aqr0pclu1p9530hs73j@4ax.com>
References: <mku2m5lrafrcvq4aqr0pclu1p9530hs73j@4ax.com>
Message-ID: <b6f11f301001280557w57114acbn39c4e749b43b2d0@mail.gmail.com>

Hello. Try:

sofia profile internal siptrace on

Replace internal for the profile that you need.


Javier

On Thu, Jan 28, 2010 at 6:54 AM, Fred-145 <codecomplete at free.fr> wrote:

> Hello
>
> Before I learn how to use Wireshark and filter stuff... can the fs_cli
> console be configured so that only SIP messages are displayed? I'd
> like to do this so I can learn more about what goes on when I play
> with SIP clients.
>
> Thank you.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From wiltingtree at gmail.com  Thu Jan 28 06:41:42 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Thu, 28 Jan 2010 09:41:42 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
Message-ID: <eb7797d71001280641j48a0d93bt7da6c27aa8a317ee@mail.gmail.com>

Man, I'm in the process of switching from Python to Lua because of bugs in
the Python support.
If Lua doesn't work with odbc, that's a big problem.
Does anybody have any other advice?


On Thu, Jan 28, 2010 at 2:43 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello,
>
> That works fine:
>
> box:~# lua testdb.lua
> box:~#
>
>
> David
>
> On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
> <mike at van.lammeren.net> wrote:
> > Have you tried running a Lua script that includes the library from
> outside
> > of FreeSWITCH? What does that do?
> >
> > On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com>
> wrote:
> >>
> >> I tried running ldconfig on the directory containing mysql.so, but it
> did
> >> not help.
> >> So it sounds like there could be a bug in the latter versions?
> >>
> >> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
> >> <david.villasmil.work at gmail.com> wrote:
> >>>
> >>> I got the same error, my script was working with no problems before an
> >>> update to trunk.
> >>>
> >>> David
> >>>
> >>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com>
> wrote:
> >>> > Hi, I followed the instructions in the Lua documentation for setting
> up
> >>> > luasql, but when I try to run my script I get:
> >>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module
> >>> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
> >>> >         /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore
> segment
> >>> > prot
> >>> > after reloc: Permission denied
> >>> > stack traceback:
> >>> >         [C]: ?
> >>> >         [C]: in function 'require'
> >>> >         /usr/local/freeswitch/scripts/l.lua:2: in main chunk
> >>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
> >>> > I changed the permissions for mysql.so and for my script to 777, so
> I'm
> >>> > not
> >>> > sure where the permission problem could be.
> >>> > I'd appreciate any suggestions.
> >>> > Thanks,
> >>> > Adam
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From codecomplete at free.fr  Thu Jan 28 07:09:55 2010
From: codecomplete at free.fr (Fred-145)
Date: Thu, 28 Jan 2010 16:09:55 +0100
Subject: [Freeswitch-users] [fs_cli] Only display SIP messages?
References: <mku2m5lrafrcvq4aqr0pclu1p9530hs73j@4ax.com>
	<mku2m5lrafrcvq4aqr0pclu1p9530hs73j-e09XROE/p8c@public.gmane.org>
	<b6f11f301001280557w57114acbn39c4e749b43b2d0@mail.gmail.com>
Message-ID: <p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi@4ax.com>

On Thu, 28 Jan 2010 08:57:14 -0500, Javier Aristiz?bal
<javieraristizabal at gmail.com> wrote:
>sofia profile internal siptrace on

Thanks for the tip, but even with this command, I get a lot more data
than just the SIP dialog:

==========
Dialplan: sofia/internal/1004 at 192.168.0.7 Regex (FAIL)
[global-intercept] destination_number(1001) =~ /^886$/ break=on-false
Dialplan: sofia/internal/1004 at 192.168.0.7 parsing
[default->group-intercept] continue=false
[...]
2010-01-28 16:07:34.149826 [DEBUG] switch_core_state_machine.c:122
(sofia/internal/1004 at 192.168.0.7) State Change CS_ROUTING ->
CS_EXECUTE
2010-01-28 16:07:34.149826 [DEBUG] switch_core_session.c:1013 Send
signal sofia/internal/1004 at 192.168.0.7 [BREAK]
[...]
send 1369 bytes to udp/[192.168.0.1]:19354 at 15:07:34.017297:

------------------------------------------------------------------------
   INVITE sip:1001 at 192.168.0.1:19354;rinstance=804198d1614a592d
SIP/2.0
   Via: SIP/2.0/UDP 192.168.0.7;rport;branch=z9hG4bKpDgjj09Qv54Xj
   Max-Forwards: 69
   From: "Extension 1004" <sip:1004 at 192.168.0.7>;tag=Nra5SSp7aat4S
   To: <sip:1001 at 192.168.0.1:19354;rinstance=804198d1614a592d>
   Call-ID: b455d480-86c1-122d-f48c-00242116bb98
   CSeq: 126227291 INVITE
   Contact: <sip:mod_sofia at 192.168.0.7:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16456
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 425
   X-FS-Support: update_display
   Remote-Party-ID: "Extension 1004"
<sip:1004 at 192.168.0.7>;party=calling;screen=yes;privacy=off
   
   v=0
   o=FreeSWITCH 1264662214 1264662215 IN IP4 192.168.0.7
   s=FreeSWITCH
   c=IN IP4 192.168.0.7
   t=0 0
   m=audio 29040 RTP/AVP 8 115 107 9 0 3 101 13
   a=rtpmap:8 PCMA/8000
==========

Can fs_cli be configured to filter everything but the SIP protocol
itself?



From john at acsol.net  Thu Jan 28 07:26:43 2010
From: john at acsol.net (John)
Date: Thu, 28 Jan 2010 08:26:43 -0700
Subject: [Freeswitch-users] Voicemail via web interface
Message-ID: <4B61ACB3.50903@acsol.net>

Hello,
Can you point me to any additional information about the voice mail via 
web interface? I have it up and running; however if you click the play 
button there is no playback, if you click download it will play in MS 
media player. Thanks John


From rupa at rupa.com  Thu Jan 28 07:44:07 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 28 Jan 2010 09:44:07 -0600
Subject: [Freeswitch-users] [fs_cli] Only display SIP messages?
In-Reply-To: <p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi@4ax.com>
References: <mku2m5lrafrcvq4aqr0pclu1p9530hs73j@4ax.com>
	<mku2m5lrafrcvq4aqr0pclu1p9530hs73j-e09XROE/p8c@public.gmane.org>
	<b6f11f301001280557w57114acbn39c4e749b43b2d0@mail.gmail.com>
	<p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi@4ax.com>
Message-ID: <b39273531001280744j29f596c9qad879db501cac509@mail.gmail.com>

I would suggest using ngrep or sipgrep to get a capture of the sip
dialog.  Or going full bore and using wireshark.  I often use ngrep or
sipgrep when I want to follow a sip dialog.

On Thu, Jan 28, 2010 at 9:09 AM, Fred-145 <codecomplete at free.fr> wrote:
> On Thu, 28 Jan 2010 08:57:14 -0500, Javier Aristiz?bal
> <javieraristizabal at gmail.com> wrote:
>>sofia profile internal siptrace on
>
> Thanks for the tip, but even with this command, I get a lot more data
> than just the SIP dialog:
>
> ==========
> Dialplan: sofia/internal/1004 at 192.168.0.7 Regex (FAIL)
> [global-intercept] destination_number(1001) =~ /^886$/ break=on-false
> Dialplan: sofia/internal/1004 at 192.168.0.7 parsing
> [default->group-intercept] continue=false
> [...]
> 2010-01-28 16:07:34.149826 [DEBUG] switch_core_state_machine.c:122
> (sofia/internal/1004 at 192.168.0.7) State Change CS_ROUTING ->
> CS_EXECUTE
> 2010-01-28 16:07:34.149826 [DEBUG] switch_core_session.c:1013 Send
> signal sofia/internal/1004 at 192.168.0.7 [BREAK]
> [...]
> send 1369 bytes to udp/[192.168.0.1]:19354 at 15:07:34.017297:
>
> ------------------------------------------------------------------------
> ? INVITE sip:1001 at 192.168.0.1:19354;rinstance=804198d1614a592d
> SIP/2.0
> ? Via: SIP/2.0/UDP 192.168.0.7;rport;branch=z9hG4bKpDgjj09Qv54Xj
> ? Max-Forwards: 69
> ? From: "Extension 1004" <sip:1004 at 192.168.0.7>;tag=Nra5SSp7aat4S
> ? To: <sip:1001 at 192.168.0.1:19354;rinstance=804198d1614a592d>
> ? Call-ID: b455d480-86c1-122d-f48c-00242116bb98
> ? CSeq: 126227291 INVITE
> ? Contact: <sip:mod_sofia at 192.168.0.7:5060>
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16456
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> ? Supported: timer, precondition, path, replaces
> ? Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> ? Content-Type: application/sdp
> ? Content-Disposition: session
> ? Content-Length: 425
> ? X-FS-Support: update_display
> ? Remote-Party-ID: "Extension 1004"
> <sip:1004 at 192.168.0.7>;party=calling;screen=yes;privacy=off
>
> ? v=0
> ? o=FreeSWITCH 1264662214 1264662215 IN IP4 192.168.0.7
> ? s=FreeSWITCH
> ? c=IN IP4 192.168.0.7
> ? t=0 0
> ? m=audio 29040 RTP/AVP 8 115 107 9 0 3 101 13
> ? a=rtpmap:8 PCMA/8000
> ==========
>
> Can fs_cli be configured to filter everything but the SIP protocol
> itself?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From codecomplete at free.fr  Thu Jan 28 08:22:14 2010
From: codecomplete at free.fr (Fred-145)
Date: Thu, 28 Jan 2010 17:22:14 +0100
Subject: [Freeswitch-users] [fs_cli] Only display SIP messages?
References: <mku2m5lrafrcvq4aqr0pclu1p9530hs73j@4ax.com>
	<mku2m5lrafrcvq4aqr0pclu1p9530hs73j-e09XROE/p8c@public.gmane.org>
	<b6f11f301001280557w57114acbn39c4e749b43b2d0@mail.gmail.com>
	<p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi@4ax.com>
	<p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi-e09XROE/p8c@public.gmane.org>
	<b39273531001280744j29f596c9qad879db501cac509@mail.gmail.com>
Message-ID: <uce3m59t58cs4o8r51cjvslbljmdv266pt@4ax.com>

On Thu, 28 Jan 2010 09:44:07 -0600, Rupa Schomaker
<rupa at rupa.com> wrote:
> I often use ngrep or >sipgrep when I want to follow a sip dialog.

Thanks for the tip.



From anthony.minessale at gmail.com  Thu Jan 28 08:28:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 Jan 2010 10:28:56 -0600
Subject: [Freeswitch-users] [fs_cli] Only display SIP messages?
In-Reply-To: <p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi@4ax.com>
References: <mku2m5lrafrcvq4aqr0pclu1p9530hs73j@4ax.com>
	<mku2m5lrafrcvq4aqr0pclu1p9530hs73j-e09XROE/p8c@public.gmane.org>
	<b6f11f301001280557w57114acbn39c4e749b43b2d0@mail.gmail.com>
	<p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi@4ax.com>
Message-ID: <191c3a031001280828m785a077fm5bccc7eb7abe2a6f@mail.gmail.com>

console loglevel 0
as well

On Thu, Jan 28, 2010 at 9:09 AM, Fred-145 <codecomplete at free.fr> wrote:

> On Thu, 28 Jan 2010 08:57:14 -0500, Javier Aristiz?bal
> <javieraristizabal at gmail.com> wrote:
> >sofia profile internal siptrace on
>
> Thanks for the tip, but even with this command, I get a lot more data
> than just the SIP dialog:
>
> ==========
> Dialplan: sofia/internal/1004 at 192.168.0.7 Regex (FAIL)
> [global-intercept] destination_number(1001) =~ /^886$/ break=on-false
> Dialplan: sofia/internal/1004 at 192.168.0.7 parsing
> [default->group-intercept] continue=false
> [...]
> 2010-01-28 16:07:34.149826 [DEBUG] switch_core_state_machine.c:122
> (sofia/internal/1004 at 192.168.0.7) State Change CS_ROUTING ->
> CS_EXECUTE
> 2010-01-28 16:07:34.149826 [DEBUG] switch_core_session.c:1013 Send
> signal sofia/internal/1004 at 192.168.0.7 [BREAK]
> [...]
> send 1369 bytes to udp/[192.168.0.1]:19354 at 15:07:34.017297:
>
> ------------------------------------------------------------------------
>   INVITE sip:1001 at 192.168.0.1:19354;rinstance=804198d1614a592d
> SIP/2.0
>   Via: SIP/2.0/UDP 192.168.0.7;rport;branch=z9hG4bKpDgjj09Qv54Xj
>   Max-Forwards: 69
>   From: "Extension 1004" <sip:1004 at 192.168.0.7 <sip%3A1004 at 192.168.0.7>
> >;tag=Nra5SSp7aat4S
>   To: <sip:1001 at 192.168.0.1:19354;rinstance=804198d1614a592d>
>   Call-ID: b455d480-86c1-122d-f48c-00242116bb98
>   CSeq: 126227291 INVITE
>   Contact: <sip:mod_sofia at 192.168.0.7:5060>
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16456
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>   Content-Type: application/sdp
>   Content-Disposition: session
>   Content-Length: 425
>   X-FS-Support: update_display
>   Remote-Party-ID: "Extension 1004"
> <sip:1004 at 192.168.0.7 <sip%3A1004 at 192.168.0.7>
> >;party=calling;screen=yes;privacy=off
>
>   v=0
>   o=FreeSWITCH 1264662214 1264662215 IN IP4 192.168.0.7
>   s=FreeSWITCH
>   c=IN IP4 192.168.0.7
>   t=0 0
>   m=audio 29040 RTP/AVP 8 115 107 9 0 3 101 13
>   a=rtpmap:8 PCMA/8000
> ==========
>
> Can fs_cli be configured to filter everything but the SIP protocol
> itself?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From codecomplete at free.fr  Thu Jan 28 08:37:31 2010
From: codecomplete at free.fr (Fred)
Date: Thu, 28 Jan 2010 17:37:31 +0100
Subject: [Freeswitch-users] [fs_cli] Only display SIP messages?
Message-ID: <7.0.1.0.2.20100128173441.0266b8e0@free.fr>

At 17:17 28/01/2010, Diego Viola <diego.viola at gmail.com> wrote:
>You could try this from fs_cli:
>
>fsctl loglevel 0
>sofia profile internal siptrace on

Works great. Thank you.  



From codecomplete at free.fr  Thu Jan 28 08:41:00 2010
From: codecomplete at free.fr (Fred-145)
Date: Thu, 28 Jan 2010 17:41:00 +0100
Subject: [Freeswitch-users] [fs_cli] Only display SIP messages?
References: <mku2m5lrafrcvq4aqr0pclu1p9530hs73j@4ax.com>
	<mku2m5lrafrcvq4aqr0pclu1p9530hs73j-e09XROE/p8c@public.gmane.org>
	<b6f11f301001280557w57114acbn39c4e749b43b2d0@mail.gmail.com>
	<p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi@4ax.com>
	<p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi-e09XROE/p8c@public.gmane.org>
	<191c3a031001280828m785a077fm5bccc7eb7abe2a6f@mail.gmail.com>
Message-ID: <1gf3m55r2oc346agcop9fp5m89tt13g1j8@4ax.com>

On Thu, 28 Jan 2010 10:28:56 -0600, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
>console loglevel 0
>as well

Thanks, just what I was looking for.



From troy at tlainvestments.com  Thu Jan 28 08:49:20 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 28 Jan 2010 09:49:20 -0700
Subject: [Freeswitch-users] Call Dropping with SIP 503 status
In-Reply-To: <191c3a031001271431re6cfad2w1389e96db4e6c1a0@mail.gmail.com>
References: <d021d3be1001271126y7a40e209t13e3d5b71af0d1a4@mail.gmail.com>
	<ADD964DC-4B88-4DFA-A330-D391BE786F78@freeswitch.org>
	<BD4CD56A-D488-4CCD-955D-0A838CB3CFD1@tlainvestments.com>
	<191c3a031001271431re6cfad2w1389e96db4e6c1a0@mail.gmail.com>
Message-ID: <910698FF-EDB7-43B1-B776-8CC7E69E8023@tlainvestments.com>

Of course the error didn't show up in the 4 hours I had the sip trace on...

I downgraded the firmware on the Polycom 301's to 3.3.1RevB in stead of 3.2.2 and don't seem to be having the problem any more.

If it comes back, we'll break out sip trance again to see what's up.

Thanks!
-Troy

On Jan 27, 2010, at 3:31 PM, Anthony Minessale wrote:

> try turning on sip trace as well to see the sip traffic
> 
> sofia profile internal siptrace on (from cli)
> probably its something that said it could do session timers but was lying
> 
> 
> On Wed, Jan 27, 2010 at 2:05 PM, Troy Anderson <troy at tlainvestments.com> wrote:
> We are experiencing an odd issue.  We have many calls that don't drop, but some do after being up a minute or two.
> 
> The reason code is NORMAL_TEMPORARY_FAILURE and the sip status that is triggering that is 503 (Service Unavailable).  With only one or two calls up at a time, I don't think it's a session limit issue (set to 1000).
> 
> Here is the console log from just before the 503 status - any help is greatly appreciated!
> 
> 2010-01-27 12:49:11.879251 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:49:11.899334 [INFO] sofia.c:597 Update Callee ID to "400" <400>
> 2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [ready][200]
> 2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4011 Duplicate SDP
> v=0
> o=- 1264621687 1264621687 IN IP4 192.168.0.46
> s=Polycom IP Phone
> c=IN IP4 192.168.0.46
> t=0 0
> a=sendrecv
> m=audio 2222 RTP/AVP 0 8 18 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
> 
> 2010-01-27 12:50:06.068999 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [ready][200]
> 2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4011 Duplicate SDP
> v=0
> o=- 1264621687 1264621687 IN IP4 192.168.0.46
> s=Polycom IP Phone
> c=IN IP4 192.168.0.46
> t=0 0
> a=sendrecv
> m=audio 2222 RTP/AVP 0 8 18 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
> 
> 2010-01-27 12:51:05.259614 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [ready][200]
> 2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4011 Duplicate SDP
> v=0
> o=- 1264621687 1264621687 IN IP4 192.168.0.46
> s=Polycom IP Phone
> c=IN IP4 192.168.0.46
> t=0 0
> a=sendrecv
> m=audio 2222 RTP/AVP 0 8 18 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
> 
> 2010-01-27 12:52:05.369138 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [ready][200]
> 2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4011 Duplicate SDP
> v=0
> o=- 1264621687 1264621687 IN IP4 192.168.0.46
> s=Polycom IP Phone
> c=IN IP4 192.168.0.46
> t=0 0
> a=sendrecv
> m=audio 2222 RTP/AVP 0 8 18 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
> 
> 2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [terminating][503]
> 2010-01-27 12:53:09.599069 [NOTICE] sofia.c:4647 Hangup sofia/internal/400 at 192.168.0.31 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
> 2010-01-27 12:53:09.599069 [DEBUG] switch_ivr_bridge.c:466 sofia/internal/400 at 192.168.0.31 ending bridge by request from write function
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Thu Jan 28 09:14:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 Jan 2010 11:14:33 -0600
Subject: [Freeswitch-users] Multiple DTMF on group_confirm_key
In-Reply-To: <7d79b3931001280411u6262f627xca723c64de5e118d@mail.gmail.com>
References: <7d79b3931001280411u6262f627xca723c64de5e118d@mail.gmail.com>
Message-ID: <191c3a031001280914l56a035s74fd6d4b00dd9b3d@mail.gmail.com>

you have to use a script (See the wiki for executing a script)
then you can read in as many digits as you want and do what you need.


On Thu, Jan 28, 2010 at 6:11 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi all,
>
> I've experimented with group confirm key and group confirm file. It works
> great. However, I was unable to give multiple DTMF digits to get the
> confirmation.
>
> I've set group_confirm_key=1234, I thought it will ask the 4 digits from
> the user. But it simply taken 1 and when the user presses 1, the call got
> bridged.
>
> Is there any way to specify multiple dtmf to be confirmed??
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From david.villasmil.work at gmail.com  Thu Jan 28 09:49:31 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 28 Jan 2010 18:49:31 +0100
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <eb7797d71001280641j48a0d93bt7da6c27aa8a317ee@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
	<eb7797d71001280641j48a0d93bt7da6c27aa8a317ee@mail.gmail.com>
Message-ID: <9853f4ff1001280949u41336092j6b3ed6f3d3b6545@mail.gmail.com>

It does support SQL. Not sure about ODBC specifically. I know it does
support PostgreSQL, MySQL and Oracle
(http://www.keplerproject.org/luasql/)

And for it was working, it is actually still working on 1 box.

David

On Thu, Jan 28, 2010 at 3:41 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
> Man, I'm in the process of switching from Python to Lua because of bugs in
> the Python support.
> If Lua doesn't work with odbc, that's a big problem.
> Does anybody have any other advice?
>
> On Thu, Jan 28, 2010 at 2:43 AM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> Hello,
>>
>> That works fine:
>>
>> box:~# lua testdb.lua
>> box:~#
>>
>>
>> David
>>
>> On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
>> <mike at van.lammeren.net> wrote:
>> > Have you tried running a Lua script that includes the library from
>> > outside
>> > of FreeSWITCH? What does that do?
>> >
>> > On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com>
>> > wrote:
>> >>
>> >> I tried running ldconfig on the directory containing mysql.so, but it
>> >> did
>> >> not help.
>> >> So it sounds like there could be a bug in the latter versions?
>> >>
>> >> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
>> >> <david.villasmil.work at gmail.com> wrote:
>> >>>
>> >>> I got the same error, my script was working with no problems before an
>> >>> update to trunk.
>> >>>
>> >>> David
>> >>>
>> >>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com>
>> >>> wrote:
>> >>> > Hi, I followed the instructions in the Lua documentation for setting
>> >>> > up
>> >>> > luasql, but when I try to run my script I get:
>> >>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading
>> >>> > module
>> >>> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
>> >>> > ?? ? ? ?/usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore
>> >>> > segment
>> >>> > prot
>> >>> > after reloc: Permission denied
>> >>> > stack traceback:
>> >>> > ?? ? ? ?[C]: ?
>> >>> > ?? ? ? ?[C]: in function 'require'
>> >>> > ?? ? ? ?/usr/local/freeswitch/scripts/l.lua:2: in main chunk
>> >>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
>> >>> > I changed the permissions for mysql.so and for my script to 777, so
>> >>> > I'm
>> >>> > not
>> >>> > sure where the permission problem could be.
>> >>> > I'd appreciate any suggestions.
>> >>> > Thanks,
>> >>> > Adam
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> >
>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From john at acsol.net  Thu Jan 28 10:43:16 2010
From: john at acsol.net (John)
Date: Thu, 28 Jan 2010 11:43:16 -0700
Subject: [Freeswitch-users] Voicemail in MP3
Message-ID: <4B61DAC4.2030301@acsol.net>

I have installed mod_shout and edited the modules.conf.xml to installed 
it as well. I have updated the user.xml to include <variable 
name="vm_message_ext" value="mp3"/>.... Messages are still being saved 
in WAV format. Missing Step? Thanks




From ranjtech at gmail.com  Thu Jan 28 10:46:56 2010
From: ranjtech at gmail.com (RR)
Date: Thu, 28 Jan 2010 13:46:56 -0500
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
Message-ID: <022701caa04a$44f60b80$cee22280$@com>

Hi brian,

 

Do you need the sip trace from within FS or just any packet capture like
from snoop/tcpdump etc will do? I don't know how to do a sip trace from
within FS

 

\R

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, January 28, 2010 2:50 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 409 Conflict When registering FS with
Softswitch

 

Can you provide a SIP Trace?

 

/b

 

On Jan 28, 2010, at 1:11 AM, RR wrote:





Gentlemen,

 

I have a probably a simple problem but I have no idea why it's occurring. I
am beyond novice/new to configuring FS and as my first try, tried to
configure it to register with our softswitch so I could have the basic
registration to happen between FS and it. However, I keep getting the "409
Conflict" message when that username is not actually registered with the
softswitch. The profile / config file under sip_profiles/external is
configured like so

 

<include>

  <gateway name="Test-Inbound">

  <param name="username" value="user1"/>-->

  <param name="password" value="user1pwd"/>-->

  <param name="realm" value="mysoftswitch.net.au"/>-->

  <param name="extension" value="1000"/>

  <param name="from-user" value="user1"/>-->

  <param name="expire-seconds" value="60"/>-->

  <param name="register" value="true"/>-->

  </gateway>-->

</include>

 

Any ideas why? And how to resolve it or what kind of traces I can send to
you guys to help me resolve it? Like I said I am so new to FS that I don't
even know what I can turn on in FS to see what's going on

 

Thanks a lot

\RR

 



__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4812 (20100128) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 



__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4815 (20100128) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From brian at freeswitch.org  Thu Jan 28 10:59:04 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 28 Jan 2010 12:59:04 -0600
Subject: [Freeswitch-users] Voicemail in MP3
In-Reply-To: <4B61DAC4.2030301@acsol.net>
References: <4B61DAC4.2030301@acsol.net>
Message-ID: <FB99FBD9-0E9B-4D8C-9B9E-C512EF23554B@freeswitch.org>

yes you're not loading mod_shout

/b

On Jan 28, 2010, at 12:43 PM, John wrote:

> I have installed mod_shout and edited the modules.conf.xml to installed 
> it as well. I have updated the user.xml to include <variable 
> name="vm_message_ext" value="mp3"/>.... Messages are still being saved 
> in WAV format. Missing Step? Thanks



From brian at freeswitch.org  Thu Jan 28 10:59:50 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 28 Jan 2010 12:59:50 -0600
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <022701caa04a$44f60b80$cee22280$@com>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
Message-ID: <697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>

Any sip trace would work.. text.. or what not.. 



/b

On Jan 28, 2010, at 12:46 PM, RR wrote:

> Hi brian,
>  
> Do you need the sip trace from within FS or just any packet capture like from snoop/tcpdump etc will do? I don?t know how to do a sip trace from within FS
>  
> \R

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From john at acsol.net  Thu Jan 28 11:11:11 2010
From: john at acsol.net (John)
Date: Thu, 28 Jan 2010 12:11:11 -0700
Subject: [Freeswitch-users] Voicemail in MP3
In-Reply-To: <FB99FBD9-0E9B-4D8C-9B9E-C512EF23554B@freeswitch.org>
References: <4B61DAC4.2030301@acsol.net>
	<FB99FBD9-0E9B-4D8C-9B9E-C512EF23554B@freeswitch.org>
Message-ID: <4B61E14F.4090801@acsol.net>

It shows that it's loaded without issues. I did uncomment the line in 
modules.conf.xml . Any way to know beyond that? Could anything else be 
wrong?
thanks




On 1/28/2010 11:59 AM, Brian West wrote:
> yes you're not loading mod_shout
>
> /b
>
> On Jan 28, 2010, at 12:43 PM, John wrote:
>
>    
>> I have installed mod_shout and edited the modules.conf.xml to installed
>> it as well. I have updated the user.xml to include<variable
>> name="vm_message_ext" value="mp3"/>.... Messages are still being saved
>> in WAV format. Missing Step? Thanks
>>      
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    



From jeff at jefflenk.com  Thu Jan 28 11:12:53 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 28 Jan 2010 13:12:53 -0600
Subject: [Freeswitch-users] Voicemail in MP3
In-Reply-To: <FB99FBD9-0E9B-4D8C-9B9E-C512EF23554B@freeswitch.org>
References: <4B61DAC4.2030301@acsol.net>,
	<FB99FBD9-0E9B-4D8C-9B9E-C512EF23554B@freeswitch.org>
Message-ID: <COL117-W304F063A2308794AE83AD3B95C0@phx.gbl>


There is a small problem with the logic using the callers setup rather than the callee. Been to busy to submit fix. If someone doesnt beat me to it I will submit fix later today.


 
> From: brian at freeswitch.org
> Date: Thu, 28 Jan 2010 12:59:04 -0600
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Voicemail in MP3
> 
> yes you're not loading mod_shout
> 
> /b
> 
> On Jan 28, 2010, at 12:43 PM, John wrote:
> 
> > I have installed mod_shout and edited the modules.conf.xml to installed 
> > it as well. I have updated the user.xml to include <variable 
> > name="vm_message_ext" value="mp3"/>.... Messages are still being saved 
> > in WAV format. Missing Step? Thanks
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Hotmail: Free, trusted and rich email service.
http://clk.atdmt.com/GBL/go/196390708/direct/01/
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From robert.hadley at teotech.com  Thu Jan 28 11:24:29 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 28 Jan 2010 11:24:29 -0800
Subject: [Freeswitch-users] Voicemail in MP3
In-Reply-To: <4B61DAC4.2030301@acsol.net>
References: <4B61DAC4.2030301@acsol.net>
Message-ID: <63E67C0D94CA4805822B2126EBA7D483@greyhawk.tonecommander.com>

A couple of things to check:

1. Make sure to enable mod_shout in the runtime installation folder:
   e.g. /usr/local/freeswitch/conf/autoload_conf/modules.conf.xml
2. I found the variable vm_message_ext to apply to sent emails, not
received.  For an internal extensions test set this variable for both ends.

-RobertH

-----Original Message-----
From: John [mailto:john at acsol.net] 
Sent: Thursday, January 28, 2010 10:43 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Voicemail in MP3

I have installed mod_shout and edited the modules.conf.xml to installed 
it as well. I have updated the user.xml to include <variable 
name="vm_message_ext" value="mp3"/>.... Messages are still being saved 
in WAV format. Missing Step? Thanks







From robert.hadley at teotech.com  Thu Jan 28 11:28:05 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 28 Jan 2010 11:28:05 -0800
Subject: [Freeswitch-users] Voicemail via web interface
In-Reply-To: <4B61ACB3.50903@acsol.net>
References: <4B61ACB3.50903@acsol.net>
Message-ID: <090DDB5F675E4FC48512CA3FC52FA096@greyhawk.tonecommander.com>


Using Firefox I was asked to install the latest Flash plugin and then I
could play the messages from the webpage directly.  IE8 never asked to add
the plugin that I noticed.
-RobertH


-----Original Message-----
From: John [mailto:john at acsol.net] 
Sent: Thursday, January 28, 2010 7:27 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Voicemail via web interface

Hello,
Can you point me to any additional information about the voice mail via 
web interface? I have it up and running; however if you click the play 
button there is no playback, if you click download it will play in MS 
media player. Thanks John





From troy at tlainvestments.com  Thu Jan 28 11:49:10 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 28 Jan 2010 12:49:10 -0700
Subject: [Freeswitch-users] ERR root tag missing
In-Reply-To: <090DDB5F675E4FC48512CA3FC52FA096@greyhawk.tonecommander.com>
References: <4B61ACB3.50903@acsol.net>
	<090DDB5F675E4FC48512CA3FC52FA096@greyhawk.tonecommander.com>
Message-ID: <F5D03273-03A1-4829-BDFA-A1173BA6CEEC@tlainvestments.com>

I'm seeing this error quite often on my systems:
	2010-01-28 12:35:46.703112 [ERR] switch_xml.c:1571 Error[[error near line 1]: root tag missing]

I've looked at freeswitch.xml.fsxml to see if I could find some kind of malformed XML, but with no luck.  Which Is line 1is it referring to?

Line 1 of freeswitch.xml.fsxml is a comment, with the first actual XML line like:
	<document type="freeswitch/xml">

This error always happens right after after a mod_dialplan_xml.c:408 log message, so I'm led to believe my dialplan XML is messed up, but I cannot see where.

In freeswitch.xml.fsxml near the dialplan section, this is what I have:

...
<section name="configuration" description="Various Configuration">
	<!-- CUT a bunch of stuff ... CUT -->
</section>
<section name="dialplan" description="Regex/XML Dialplan">
	<context name="my_office">
		<extension name="unloop">
			<condition field="${unroll_loops}" expression="^true$"/>
			<condition field="${sip_looped_call}" expression="^true$">
				<action application="deflect" data="${destination_number}"/>
			</condition>
		</extension>
		<extension name="set_domain" continue="true">
			<condition field="${domain_name}" expression="^$"/>
			<condition field="source" expression="mod_sofia"/>
			<condition field="${sip_auth_realm}" expression="^$">
				<action application="set" data="domain_name=10.0.0.120"/>
				<anti-action application="set" data="domain_name=${sip_auth_realm}"/>
			</condition>
		</extension>
...
</section>
...

Thanks for any ideas!

-Troy



From john at acsol.net  Thu Jan 28 11:52:04 2010
From: john at acsol.net (John)
Date: Thu, 28 Jan 2010 12:52:04 -0700
Subject: [Freeswitch-users] Voicemail via web interface
In-Reply-To: <090DDB5F675E4FC48512CA3FC52FA096@greyhawk.tonecommander.com>
References: <4B61ACB3.50903@acsol.net>
	<090DDB5F675E4FC48512CA3FC52FA096@greyhawk.tonecommander.com>
Message-ID: <4B61EAE4.2070607@acsol.net>

Thanks Robert. I believe the issue is probably because our files are in 
WAV format and not MP3.

On 1/28/2010 12:28 PM, Robert Hadley wrote:
> Using Firefox I was asked to install the latest Flash plugin and then I
> could play the messages from the webpage directly.  IE8 never asked to add
> the plugin that I noticed.
> -RobertH
>
>
> -----Original Message-----
> From: John [mailto:john at acsol.net]
> Sent: Thursday, January 28, 2010 7:27 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Voicemail via web interface
>
> Hello,
> Can you point me to any additional information about the voice mail via
> web interface? I have it up and running; however if you click the play
> button there is no playback, if you click download it will play in MS
> media player. Thanks John
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    



From john at acsol.net  Thu Jan 28 11:53:03 2010
From: john at acsol.net (John)
Date: Thu, 28 Jan 2010 12:53:03 -0700
Subject: [Freeswitch-users] Voicemail in MP3
In-Reply-To: <63E67C0D94CA4805822B2126EBA7D483@greyhawk.tonecommander.com>
References: <4B61DAC4.2030301@acsol.net>
	<63E67C0D94CA4805822B2126EBA7D483@greyhawk.tonecommander.com>
Message-ID: <4B61EB1F.2070105@acsol.net>

Would I setup the vm_message_ext in the dialplan then? Thanks

On 1/28/2010 12:24 PM, Robert Hadley wrote:
> A couple of things to check:
>
> 1. Make sure to enable mod_shout in the runtime installation folder:
>     e.g. /usr/local/freeswitch/conf/autoload_conf/modules.conf.xml
> 2. I found the variable vm_message_ext to apply to sent emails, not
> received.  For an internal extensions test set this variable for both ends.
>
> -RobertH
>
> -----Original Message-----
> From: John [mailto:john at acsol.net]
> Sent: Thursday, January 28, 2010 10:43 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Voicemail in MP3
>
> I have installed mod_shout and edited the modules.conf.xml to installed
> it as well. I have updated the user.xml to include<variable
> name="vm_message_ext" value="mp3"/>.... Messages are still being saved
> in WAV format. Missing Step? Thanks
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    



From Prometheus001 at gmx.net  Thu Jan 28 12:00:32 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 28 Jan 2010 21:00:32 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com>
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com>
	<4B6070A8.6050607@gmx.net>	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com>
	<4B607944.4040700@gmx.net> <4B608014.4030902@gmx.net>
	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>
Message-ID: <4B61ECE0.10409@gmx.net>

Hello Giovanni,

I did so but the same problem again.

Did you ever test in on Debian 5.0?

Best reards
Peter

Giovanni Maruzzelli schrieb:
> good, so you have only one sound device, the right one.
>
> Use the one with hw:0 in the window that skype gives you to set sound devices
>
> -gm
>
> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> I installed alsa-utile,
>>
>> now I get:
>>
>> skype:/var/cache/apt/archives# aplay -l
>> **** List of PLAYBACK Hardware Devices ****
>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>  Subdevices: 127/128
>>  Subdevice #0: subdevice #0
>>  Subdevice #1: subdevice #1
>>  Subdevice #2: subdevice #2
>>  Subdevice #3: subdevice #3
>>  Subdevice #4: subdevice #4
>>  Subdevice #5: subdevice #5
>>  Subdevice #6: subdevice #6
>>  Subdevice #7: subdevice #7
>>  Subdevice #8: subdevice #8
>>  Subdevice #9: subdevice #9
>>  Subdevice #10: subdevice #10
>>  Subdevice #11: subdevice #11
>>  Subdevice #12: subdevice #12
>>  Subdevice #13: subdevice #13
>>  Subdevice #14: subdevice #14
>>  Subdevice #15: subdevice #15
>>  Subdevice #16: subdevice #16
>>  Subdevice #17: subdevice #17
>>  Subdevice #18: subdevice #18
>>  Subdevice #19: subdevice #19
>>  Subdevice #20: subdevice #20
>>  Subdevice #21: subdevice #21
>>  Subdevice #22: subdevice #22
>>  Subdevice #23: subdevice #23
>>  Subdevice #24: subdevice #24
>>  Subdevice #25: subdevice #25
>>  Subdevice #26: subdevice #26
>>  Subdevice #27: subdevice #27
>>  Subdevice #28: subdevice #28
>>  Subdevice #29: subdevice #29
>>  Subdevice #30: subdevice #30
>>  Subdevice #31: subdevice #31
>>  Subdevice #32: subdevice #32
>>  Subdevice #33: subdevice #33
>>  Subdevice #34: subdevice #34
>>  Subdevice #35: subdevice #35
>>  Subdevice #36: subdevice #36
>>  Subdevice #37: subdevice #37
>>  Subdevice #38: subdevice #38
>>  Subdevice #39: subdevice #39
>>  Subdevice #40: subdevice #40
>>  Subdevice #41: subdevice #41
>>  Subdevice #42: subdevice #42
>>  Subdevice #43: subdevice #43
>>  Subdevice #44: subdevice #44
>>  Subdevice #45: subdevice #45
>>  Subdevice #46: subdevice #46
>>  Subdevice #47: subdevice #47
>>  Subdevice #48: subdevice #48
>>  Subdevice #49: subdevice #49
>>  Subdevice #50: subdevice #50
>>  Subdevice #51: subdevice #51
>>  Subdevice #52: subdevice #52
>>  Subdevice #53: subdevice #53
>>  Subdevice #54: subdevice #54
>>  Subdevice #55: subdevice #55
>>  Subdevice #56: subdevice #56
>>  Subdevice #57: subdevice #57
>>  Subdevice #58: subdevice #58
>>  Subdevice #59: subdevice #59
>>  Subdevice #60: subdevice #60
>>  Subdevice #61: subdevice #61
>>  Subdevice #62: subdevice #62
>>  Subdevice #63: subdevice #63
>>  Subdevice #64: subdevice #64
>>  Subdevice #65: subdevice #65
>>  Subdevice #66: subdevice #66
>>  Subdevice #67: subdevice #67
>>  Subdevice #68: subdevice #68
>>  Subdevice #69: subdevice #69
>>  Subdevice #70: subdevice #70
>>  Subdevice #71: subdevice #71
>>  Subdevice #72: subdevice #72
>>  Subdevice #73: subdevice #73
>>  Subdevice #74: subdevice #74
>>  Subdevice #75: subdevice #75
>>  Subdevice #76: subdevice #76
>>  Subdevice #77: subdevice #77
>>  Subdevice #78: subdevice #78
>>  Subdevice #79: subdevice #79
>>  Subdevice #80: subdevice #80
>>  Subdevice #81: subdevice #81
>>  Subdevice #82: subdevice #82
>>  Subdevice #83: subdevice #83
>>  Subdevice #84: subdevice #84
>>  Subdevice #85: subdevice #85
>>  Subdevice #86: subdevice #86
>>  Subdevice #87: subdevice #87
>>  Subdevice #88: subdevice #88
>>  Subdevice #89: subdevice #89
>>  Subdevice #90: subdevice #90
>>  Subdevice #91: subdevice #91
>>  Subdevice #92: subdevice #92
>>  Subdevice #93: subdevice #93
>>  Subdevice #94: subdevice #94
>>  Subdevice #95: subdevice #95
>>  Subdevice #96: subdevice #96
>>  Subdevice #97: subdevice #97
>>  Subdevice #98: subdevice #98
>>  Subdevice #99: subdevice #99
>>  Subdevice #100: subdevice #100
>>  Subdevice #101: subdevice #101
>>  Subdevice #102: subdevice #102
>>  Subdevice #103: subdevice #103
>>  Subdevice #104: subdevice #104
>>  Subdevice #105: subdevice #105
>>  Subdevice #106: subdevice #106
>>  Subdevice #107: subdevice #107
>>  Subdevice #108: subdevice #108
>>  Subdevice #109: subdevice #109
>>  Subdevice #110: subdevice #110
>>  Subdevice #111: subdevice #111
>>  Subdevice #112: subdevice #112
>>  Subdevice #113: subdevice #113
>>  Subdevice #114: subdevice #114
>>  Subdevice #115: subdevice #115
>>  Subdevice #116: subdevice #116
>>  Subdevice #117: subdevice #117
>>  Subdevice #118: subdevice #118
>>  Subdevice #119: subdevice #119
>>  Subdevice #120: subdevice #120
>>  Subdevice #121: subdevice #121
>>  Subdevice #122: subdevice #122
>>  Subdevice #123: subdevice #123
>>  Subdevice #124: subdevice #124
>>  Subdevice #125: subdevice #125
>>  Subdevice #126: subdevice #126
>>  Subdevice #127: subdevice #127
>>
>>
>> Peter P GMX schrieb:
>>     
>>> Her's the output:
>>>
>>> skype:~# aplay -l
>>> bash: aplay: command not found
>>>
>>> Giovanni Maruzzelli schrieb:
>>>
>>>       
>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>> what's the output of:
>>>>
>>>> aplay -l
>>>>
>>>> ?
>>>>
>>>> If instead you are referring to the choices that skype clients offers
>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>
>>>> Eg: not the "default", but the "hardware" one
>>>>
>>>>
>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>
>>>>
>>>>         
>>>>> Thanks Giovanni,
>>>>>
>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>> since I compiled alsa newly)
>>>>>
>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>>
>>>>> Giovanni Maruzzelli schrieb:
>>>>>
>>>>>
>>>>>           
>>>>>> This warning is harmless:
>>>>>>
>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>
>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>
>>>>>>
>>>>>>
>>>>>>             
>>>>>>> Ciao Peter
>>>>>>>
>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>> snd-dummy-customized.
>>>>>>>
>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>> 8.04).
>>>>>>>
>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>
>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>> do this with the modprobe command, as in:
>>>>>>>
>>>>>>> rmmod snd-dummy
>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>
>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>> soundcard.
>>>>>>>
>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>> just committed some improvements.
>>>>>>>
>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>> write the mailing list again,
>>>>>>>
>>>>>>> ciao for now,
>>>>>>>
>>>>>>> -giovanni
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>> to all contributors, excellent work!
>>>>>>>>
>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>> instructions in
>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>> ist Debian 5.0R3)
>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>> one-way-audio and my logs show
>>>>>>>>
>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>> suppressed
>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>> suppressed
>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>> with the current limitations).
>>>>>>>>
>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>
>>>>>>>> Second question:
>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>
>>>>>>>> Best regards
>>>>>>>> Peter
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>> --
>>>>>>> Sincerely,
>>>>>>>
>>>>>>> Giovanni Maruzzelli
>>>>>>> Cell : +39-347-2665618
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>             
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>           
>>>>
>>>>         
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   


From mike at van.lammeren.net  Thu Jan 28 12:05:11 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 28 Jan 2010 15:05:11 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
Message-ID: <5d2828f1001281205u2eccc999re6f25a55b07156c7@mail.gmail.com>

And you can make queries against your MySQL database, and get results, etc.?

On Thu, Jan 28, 2010 at 2:43 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello,
>
> That works fine:
>
> box:~# lua testdb.lua
> box:~#
>
>
> David
>
> On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
> <mike at van.lammeren.net> wrote:
> > Have you tried running a Lua script that includes the library from
> outside
> > of FreeSWITCH? What does that do?
> >
> > On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com>
> wrote:
> >>
> >> I tried running ldconfig on the directory containing mysql.so, but it
> did
> >> not help.
> >> So it sounds like there could be a bug in the latter versions?
> >>
> >> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
> >> <david.villasmil.work at gmail.com> wrote:
> >>>
> >>> I got the same error, my script was working with no problems before an
> >>> update to trunk.
> >>>
> >>> David
> >>>
> >>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com>
> wrote:
> >>> > Hi, I followed the instructions in the Lua documentation for setting
> up
> >>> > luasql, but when I try to run my script I get:
> >>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module
> >>> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
> >>> >         /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore
> segment
> >>> > prot
> >>> > after reloc: Permission denied
> >>> > stack traceback:
> >>> >         [C]: ?
> >>> >         [C]: in function 'require'
> >>> >         /usr/local/freeswitch/scripts/l.lua:2: in main chunk
> >>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
> >>> > I changed the permissions for mysql.so and for my script to 777, so
> I'm
> >>> > not
> >>> > sure where the permission problem could be.
> >>> > I'd appreciate any suggestions.
> >>> > Thanks,
> >>> > Adam
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at celliax.org  Thu Jan 28 12:10:51 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 28 Jan 2010 21:10:51 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B61ECE0.10409@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com> 
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com> 
	<4B6070A8.6050607@gmx.net>
	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com> 
	<4B607944.4040700@gmx.net> <4B608014.4030902@gmx.net>
	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com> 
	<4B61ECE0.10409@gmx.net>
Message-ID: <7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>

Ciao Peter,

Never tested on Debian 5.

When you write "same problem" you are referring to the audio going one
way only (btw, which way?) with the custom audio driver?

Have you tried with multiple instances of the regular Debian
snd-dummy, as I wrote in a mail before?

-gm



On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello Giovanni,
>
> I did so but the same problem again.
>
> Did you ever test in on Debian 5.0?
>
> Best reards
> Peter
>
> Giovanni Maruzzelli schrieb:
>> good, so you have only one sound device, the right one.
>>
>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>
>> -gm
>>
>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>
>>> I installed alsa-utile,
>>>
>>> now I get:
>>>
>>> skype:/var/cache/apt/archives# aplay -l
>>> **** List of PLAYBACK Hardware Devices ****
>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>> ?Subdevices: 127/128
>>> ?Subdevice #0: subdevice #0
>>> ?Subdevice #1: subdevice #1
>>> ?Subdevice #2: subdevice #2
>>> ?Subdevice #3: subdevice #3
>>> ?Subdevice #4: subdevice #4
>>> ?Subdevice #5: subdevice #5
>>> ?Subdevice #6: subdevice #6
>>> ?Subdevice #7: subdevice #7
>>> ?Subdevice #8: subdevice #8
>>> ?Subdevice #9: subdevice #9
>>> ?Subdevice #10: subdevice #10
>>> ?Subdevice #11: subdevice #11
>>> ?Subdevice #12: subdevice #12
>>> ?Subdevice #13: subdevice #13
>>> ?Subdevice #14: subdevice #14
>>> ?Subdevice #15: subdevice #15
>>> ?Subdevice #16: subdevice #16
>>> ?Subdevice #17: subdevice #17
>>> ?Subdevice #18: subdevice #18
>>> ?Subdevice #19: subdevice #19
>>> ?Subdevice #20: subdevice #20
>>> ?Subdevice #21: subdevice #21
>>> ?Subdevice #22: subdevice #22
>>> ?Subdevice #23: subdevice #23
>>> ?Subdevice #24: subdevice #24
>>> ?Subdevice #25: subdevice #25
>>> ?Subdevice #26: subdevice #26
>>> ?Subdevice #27: subdevice #27
>>> ?Subdevice #28: subdevice #28
>>> ?Subdevice #29: subdevice #29
>>> ?Subdevice #30: subdevice #30
>>> ?Subdevice #31: subdevice #31
>>> ?Subdevice #32: subdevice #32
>>> ?Subdevice #33: subdevice #33
>>> ?Subdevice #34: subdevice #34
>>> ?Subdevice #35: subdevice #35
>>> ?Subdevice #36: subdevice #36
>>> ?Subdevice #37: subdevice #37
>>> ?Subdevice #38: subdevice #38
>>> ?Subdevice #39: subdevice #39
>>> ?Subdevice #40: subdevice #40
>>> ?Subdevice #41: subdevice #41
>>> ?Subdevice #42: subdevice #42
>>> ?Subdevice #43: subdevice #43
>>> ?Subdevice #44: subdevice #44
>>> ?Subdevice #45: subdevice #45
>>> ?Subdevice #46: subdevice #46
>>> ?Subdevice #47: subdevice #47
>>> ?Subdevice #48: subdevice #48
>>> ?Subdevice #49: subdevice #49
>>> ?Subdevice #50: subdevice #50
>>> ?Subdevice #51: subdevice #51
>>> ?Subdevice #52: subdevice #52
>>> ?Subdevice #53: subdevice #53
>>> ?Subdevice #54: subdevice #54
>>> ?Subdevice #55: subdevice #55
>>> ?Subdevice #56: subdevice #56
>>> ?Subdevice #57: subdevice #57
>>> ?Subdevice #58: subdevice #58
>>> ?Subdevice #59: subdevice #59
>>> ?Subdevice #60: subdevice #60
>>> ?Subdevice #61: subdevice #61
>>> ?Subdevice #62: subdevice #62
>>> ?Subdevice #63: subdevice #63
>>> ?Subdevice #64: subdevice #64
>>> ?Subdevice #65: subdevice #65
>>> ?Subdevice #66: subdevice #66
>>> ?Subdevice #67: subdevice #67
>>> ?Subdevice #68: subdevice #68
>>> ?Subdevice #69: subdevice #69
>>> ?Subdevice #70: subdevice #70
>>> ?Subdevice #71: subdevice #71
>>> ?Subdevice #72: subdevice #72
>>> ?Subdevice #73: subdevice #73
>>> ?Subdevice #74: subdevice #74
>>> ?Subdevice #75: subdevice #75
>>> ?Subdevice #76: subdevice #76
>>> ?Subdevice #77: subdevice #77
>>> ?Subdevice #78: subdevice #78
>>> ?Subdevice #79: subdevice #79
>>> ?Subdevice #80: subdevice #80
>>> ?Subdevice #81: subdevice #81
>>> ?Subdevice #82: subdevice #82
>>> ?Subdevice #83: subdevice #83
>>> ?Subdevice #84: subdevice #84
>>> ?Subdevice #85: subdevice #85
>>> ?Subdevice #86: subdevice #86
>>> ?Subdevice #87: subdevice #87
>>> ?Subdevice #88: subdevice #88
>>> ?Subdevice #89: subdevice #89
>>> ?Subdevice #90: subdevice #90
>>> ?Subdevice #91: subdevice #91
>>> ?Subdevice #92: subdevice #92
>>> ?Subdevice #93: subdevice #93
>>> ?Subdevice #94: subdevice #94
>>> ?Subdevice #95: subdevice #95
>>> ?Subdevice #96: subdevice #96
>>> ?Subdevice #97: subdevice #97
>>> ?Subdevice #98: subdevice #98
>>> ?Subdevice #99: subdevice #99
>>> ?Subdevice #100: subdevice #100
>>> ?Subdevice #101: subdevice #101
>>> ?Subdevice #102: subdevice #102
>>> ?Subdevice #103: subdevice #103
>>> ?Subdevice #104: subdevice #104
>>> ?Subdevice #105: subdevice #105
>>> ?Subdevice #106: subdevice #106
>>> ?Subdevice #107: subdevice #107
>>> ?Subdevice #108: subdevice #108
>>> ?Subdevice #109: subdevice #109
>>> ?Subdevice #110: subdevice #110
>>> ?Subdevice #111: subdevice #111
>>> ?Subdevice #112: subdevice #112
>>> ?Subdevice #113: subdevice #113
>>> ?Subdevice #114: subdevice #114
>>> ?Subdevice #115: subdevice #115
>>> ?Subdevice #116: subdevice #116
>>> ?Subdevice #117: subdevice #117
>>> ?Subdevice #118: subdevice #118
>>> ?Subdevice #119: subdevice #119
>>> ?Subdevice #120: subdevice #120
>>> ?Subdevice #121: subdevice #121
>>> ?Subdevice #122: subdevice #122
>>> ?Subdevice #123: subdevice #123
>>> ?Subdevice #124: subdevice #124
>>> ?Subdevice #125: subdevice #125
>>> ?Subdevice #126: subdevice #126
>>> ?Subdevice #127: subdevice #127
>>>
>>>
>>> Peter P GMX schrieb:
>>>
>>>> Her's the output:
>>>>
>>>> skype:~# aplay -l
>>>> bash: aplay: command not found
>>>>
>>>> Giovanni Maruzzelli schrieb:
>>>>
>>>>
>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>> what's the output of:
>>>>>
>>>>> aplay -l
>>>>>
>>>>> ?
>>>>>
>>>>> If instead you are referring to the choices that skype clients offers
>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>
>>>>> Eg: not the "default", but the "hardware" one
>>>>>
>>>>>
>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>
>>>>>
>>>>>
>>>>>> Thanks Giovanni,
>>>>>>
>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>> since I compiled alsa newly)
>>>>>>
>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>
>>>>>> Best regards
>>>>>> Peter
>>>>>>
>>>>>>
>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>
>>>>>>
>>>>>>
>>>>>>> This warning is harmless:
>>>>>>>
>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>
>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> Ciao Peter
>>>>>>>>
>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>> snd-dummy-customized.
>>>>>>>>
>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>> 8.04).
>>>>>>>>
>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>
>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>
>>>>>>>> rmmod snd-dummy
>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>
>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>> soundcard.
>>>>>>>>
>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>> just committed some improvements.
>>>>>>>>
>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>> write the mailing list again,
>>>>>>>>
>>>>>>>> ciao for now,
>>>>>>>>
>>>>>>>> -giovanni
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>
>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>> instructions in
>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>
>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>> suppressed
>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>> suppressed
>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>> with the current limitations).
>>>>>>>>>
>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>
>>>>>>>>> Second question:
>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>
>>>>>>>>> Best regards
>>>>>>>>> Peter
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>> --
>>>>>>>> Sincerely,
>>>>>>>>
>>>>>>>> Giovanni Maruzzelli
>>>>>>>> Cell : +39-347-2665618
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From robert.hadley at teotech.com  Thu Jan 28 12:12:18 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 28 Jan 2010 12:12:18 -0800
Subject: [Freeswitch-users] Voicemail in MP3
In-Reply-To: <4B61EB1F.2070105@acsol.net>
References: <4B61DAC4.2030301@acsol.net><63E67C0D94CA4805822B2126EBA7D483@greyhawk.tonecommander.com>
	<4B61EB1F.2070105@acsol.net>
Message-ID: <9C598BC10A7142BF8791A37C7ABFFF38@greyhawk.tonecommander.com>


No, I was using the default dialplan.  If 1000 is calling 1007 and leaving
1007 a voicmail, add the <variable> name="vm_message_ext" value="mp3"/> to
the conf/directory/default/1000.xml.  All of the other email "params"
changes should be made to 1007.xml.  Another email suggested that the caller
vs. callee behavior may be fixed in future.

-Robert

-----Original Message-----
From: John [mailto:john at acsol.net] 
Sent: Thursday, January 28, 2010 11:53 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Voicemail in MP3

Would I setup the vm_message_ext in the dialplan then? Thanks

On 1/28/2010 12:24 PM, Robert Hadley wrote:
> A couple of things to check:
>
> 1. Make sure to enable mod_shout in the runtime installation folder:
>     e.g. /usr/local/freeswitch/conf/autoload_conf/modules.conf.xml
> 2. I found the variable vm_message_ext to apply to sent emails, not
> received.  For an internal extensions test set this variable for both
ends.
>
> -RobertH
>
> -----Original Message-----
> From: John [mailto:john at acsol.net]
> Sent: Thursday, January 28, 2010 10:43 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Voicemail in MP3
>
> I have installed mod_shout and edited the modules.conf.xml to installed
> it as well. I have updated the user.xml to include<variable
> name="vm_message_ext" value="mp3"/>.... Messages are still being saved
> in WAV format. Missing Step? Thanks
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    






From john at acsol.net  Thu Jan 28 12:16:59 2010
From: john at acsol.net (John)
Date: Thu, 28 Jan 2010 13:16:59 -0700
Subject: [Freeswitch-users] Voicemail in MP3
In-Reply-To: <63E67C0D94CA4805822B2126EBA7D483@greyhawk.tonecommander.com>
References: <4B61DAC4.2030301@acsol.net>
	<63E67C0D94CA4805822B2126EBA7D483@greyhawk.tonecommander.com>
Message-ID: <4B61F0BB.4070904@acsol.net>

Robert - You were correct, I added the vm_message_ext to the sending 
extension and now it works.
Thanks
On 1/28/2010 12:24 PM, Robert Hadley wrote:
> A couple of things to check:
>
> 1. Make sure to enable mod_shout in the runtime installation folder:
>     e.g. /usr/local/freeswitch/conf/autoload_conf/modules.conf.xml
> 2. I found the variable vm_message_ext to apply to sent emails, not
> received.  For an internal extensions test set this variable for both ends.
>
> -RobertH
>
> -----Original Message-----
> From: John [mailto:john at acsol.net]
> Sent: Thursday, January 28, 2010 10:43 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Voicemail in MP3
>
> I have installed mod_shout and edited the modules.conf.xml to installed
> it as well. I have updated the user.xml to include<variable
> name="vm_message_ext" value="mp3"/>.... Messages are still being saved
> in WAV format. Missing Step? Thanks
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    



From Prometheus001 at gmx.net  Thu Jan 28 13:07:55 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 28 Jan 2010 22:07:55 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com>
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com>
	<4B6070A8.6050607@gmx.net>	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com>
	<4B607944.4040700@gmx.net>
	<4B608014.4030902@gmx.net>	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>
	<4B61ECE0.10409@gmx.net>
	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>
Message-ID: <4B61FCAB.5040707@gmx.net>

I crated 3 instances of snd-dummy, this worked. I assigned then Instance
#2 to the Skype accounts. Still no sound.
On the frist call there is one way audio, on the following calls there
is no audio at all.
This is weird.

Best regards
Peter

Giovanni Maruzzelli schrieb:
> Ciao Peter,
>
> Never tested on Debian 5.
>
> When you write "same problem" you are referring to the audio going one
> way only (btw, which way?) with the custom audio driver?
>
> Have you tried with multiple instances of the regular Debian
> snd-dummy, as I wrote in a mail before?
>
> -gm
>
>
>
> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> Hello Giovanni,
>>
>> I did so but the same problem again.
>>
>> Did you ever test in on Debian 5.0?
>>
>> Best reards
>> Peter
>>
>> Giovanni Maruzzelli schrieb:
>>     
>>> good, so you have only one sound device, the right one.
>>>
>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>
>>> -gm
>>>
>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>       
>>>> I installed alsa-utile,
>>>>
>>>> now I get:
>>>>
>>>> skype:/var/cache/apt/archives# aplay -l
>>>> **** List of PLAYBACK Hardware Devices ****
>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>  Subdevices: 127/128
>>>>  Subdevice #0: subdevice #0
>>>>  Subdevice #1: subdevice #1
>>>>  Subdevice #2: subdevice #2
>>>>  Subdevice #3: subdevice #3
>>>>  Subdevice #4: subdevice #4
>>>>  Subdevice #5: subdevice #5
>>>>  Subdevice #6: subdevice #6
>>>>  Subdevice #7: subdevice #7
>>>>  Subdevice #8: subdevice #8
>>>>  Subdevice #9: subdevice #9
>>>>  Subdevice #10: subdevice #10
>>>>  Subdevice #11: subdevice #11
>>>>  Subdevice #12: subdevice #12
>>>>  Subdevice #13: subdevice #13
>>>>  Subdevice #14: subdevice #14
>>>>  Subdevice #15: subdevice #15
>>>>  Subdevice #16: subdevice #16
>>>>  Subdevice #17: subdevice #17
>>>>  Subdevice #18: subdevice #18
>>>>  Subdevice #19: subdevice #19
>>>>  Subdevice #20: subdevice #20
>>>>  Subdevice #21: subdevice #21
>>>>  Subdevice #22: subdevice #22
>>>>  Subdevice #23: subdevice #23
>>>>  Subdevice #24: subdevice #24
>>>>  Subdevice #25: subdevice #25
>>>>  Subdevice #26: subdevice #26
>>>>  Subdevice #27: subdevice #27
>>>>  Subdevice #28: subdevice #28
>>>>  Subdevice #29: subdevice #29
>>>>  Subdevice #30: subdevice #30
>>>>  Subdevice #31: subdevice #31
>>>>  Subdevice #32: subdevice #32
>>>>  Subdevice #33: subdevice #33
>>>>  Subdevice #34: subdevice #34
>>>>  Subdevice #35: subdevice #35
>>>>  Subdevice #36: subdevice #36
>>>>  Subdevice #37: subdevice #37
>>>>  Subdevice #38: subdevice #38
>>>>  Subdevice #39: subdevice #39
>>>>  Subdevice #40: subdevice #40
>>>>  Subdevice #41: subdevice #41
>>>>  Subdevice #42: subdevice #42
>>>>  Subdevice #43: subdevice #43
>>>>  Subdevice #44: subdevice #44
>>>>  Subdevice #45: subdevice #45
>>>>  Subdevice #46: subdevice #46
>>>>  Subdevice #47: subdevice #47
>>>>  Subdevice #48: subdevice #48
>>>>  Subdevice #49: subdevice #49
>>>>  Subdevice #50: subdevice #50
>>>>  Subdevice #51: subdevice #51
>>>>  Subdevice #52: subdevice #52
>>>>  Subdevice #53: subdevice #53
>>>>  Subdevice #54: subdevice #54
>>>>  Subdevice #55: subdevice #55
>>>>  Subdevice #56: subdevice #56
>>>>  Subdevice #57: subdevice #57
>>>>  Subdevice #58: subdevice #58
>>>>  Subdevice #59: subdevice #59
>>>>  Subdevice #60: subdevice #60
>>>>  Subdevice #61: subdevice #61
>>>>  Subdevice #62: subdevice #62
>>>>  Subdevice #63: subdevice #63
>>>>  Subdevice #64: subdevice #64
>>>>  Subdevice #65: subdevice #65
>>>>  Subdevice #66: subdevice #66
>>>>  Subdevice #67: subdevice #67
>>>>  Subdevice #68: subdevice #68
>>>>  Subdevice #69: subdevice #69
>>>>  Subdevice #70: subdevice #70
>>>>  Subdevice #71: subdevice #71
>>>>  Subdevice #72: subdevice #72
>>>>  Subdevice #73: subdevice #73
>>>>  Subdevice #74: subdevice #74
>>>>  Subdevice #75: subdevice #75
>>>>  Subdevice #76: subdevice #76
>>>>  Subdevice #77: subdevice #77
>>>>  Subdevice #78: subdevice #78
>>>>  Subdevice #79: subdevice #79
>>>>  Subdevice #80: subdevice #80
>>>>  Subdevice #81: subdevice #81
>>>>  Subdevice #82: subdevice #82
>>>>  Subdevice #83: subdevice #83
>>>>  Subdevice #84: subdevice #84
>>>>  Subdevice #85: subdevice #85
>>>>  Subdevice #86: subdevice #86
>>>>  Subdevice #87: subdevice #87
>>>>  Subdevice #88: subdevice #88
>>>>  Subdevice #89: subdevice #89
>>>>  Subdevice #90: subdevice #90
>>>>  Subdevice #91: subdevice #91
>>>>  Subdevice #92: subdevice #92
>>>>  Subdevice #93: subdevice #93
>>>>  Subdevice #94: subdevice #94
>>>>  Subdevice #95: subdevice #95
>>>>  Subdevice #96: subdevice #96
>>>>  Subdevice #97: subdevice #97
>>>>  Subdevice #98: subdevice #98
>>>>  Subdevice #99: subdevice #99
>>>>  Subdevice #100: subdevice #100
>>>>  Subdevice #101: subdevice #101
>>>>  Subdevice #102: subdevice #102
>>>>  Subdevice #103: subdevice #103
>>>>  Subdevice #104: subdevice #104
>>>>  Subdevice #105: subdevice #105
>>>>  Subdevice #106: subdevice #106
>>>>  Subdevice #107: subdevice #107
>>>>  Subdevice #108: subdevice #108
>>>>  Subdevice #109: subdevice #109
>>>>  Subdevice #110: subdevice #110
>>>>  Subdevice #111: subdevice #111
>>>>  Subdevice #112: subdevice #112
>>>>  Subdevice #113: subdevice #113
>>>>  Subdevice #114: subdevice #114
>>>>  Subdevice #115: subdevice #115
>>>>  Subdevice #116: subdevice #116
>>>>  Subdevice #117: subdevice #117
>>>>  Subdevice #118: subdevice #118
>>>>  Subdevice #119: subdevice #119
>>>>  Subdevice #120: subdevice #120
>>>>  Subdevice #121: subdevice #121
>>>>  Subdevice #122: subdevice #122
>>>>  Subdevice #123: subdevice #123
>>>>  Subdevice #124: subdevice #124
>>>>  Subdevice #125: subdevice #125
>>>>  Subdevice #126: subdevice #126
>>>>  Subdevice #127: subdevice #127
>>>>
>>>>
>>>> Peter P GMX schrieb:
>>>>
>>>>         
>>>>> Her's the output:
>>>>>
>>>>> skype:~# aplay -l
>>>>> bash: aplay: command not found
>>>>>
>>>>> Giovanni Maruzzelli schrieb:
>>>>>
>>>>>
>>>>>           
>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>> what's the output of:
>>>>>>
>>>>>> aplay -l
>>>>>>
>>>>>> ?
>>>>>>
>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>
>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>
>>>>>>
>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>
>>>>>>
>>>>>>
>>>>>>             
>>>>>>> Thanks Giovanni,
>>>>>>>
>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>> since I compiled alsa newly)
>>>>>>>
>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>
>>>>>>> Best regards
>>>>>>> Peter
>>>>>>>
>>>>>>>
>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>>> This warning is harmless:
>>>>>>>>
>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>
>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>>>> Ciao Peter
>>>>>>>>>
>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>> snd-dummy-customized.
>>>>>>>>>
>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>> 8.04).
>>>>>>>>>
>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>
>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>
>>>>>>>>> rmmod snd-dummy
>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>
>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>> soundcard.
>>>>>>>>>
>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>> just committed some improvements.
>>>>>>>>>
>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>> write the mailing list again,
>>>>>>>>>
>>>>>>>>> ciao for now,
>>>>>>>>>
>>>>>>>>> -giovanni
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>
>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>> instructions in
>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>
>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>> suppressed
>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>> suppressed
>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>> with the current limitations).
>>>>>>>>>>
>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>
>>>>>>>>>> Second question:
>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>
>>>>>>>>>> Best regards
>>>>>>>>>> Peter
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>                     
>>>>>>>>> --
>>>>>>>>> Sincerely,
>>>>>>>>>
>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>             
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   


From fvillarroel at yahoo.com  Thu Jan 28 13:14:37 2010
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Thu, 28 Jan 2010 13:14:37 -0800 (PST)
Subject: [Freeswitch-users] prefix on exten
Message-ID: <968062.52038.qm@web34307.mail.mud.yahoo.com>

Dear.

If i receive a call from a customer with some prefx like 1234

How i can do in order to forward this call with out prefix like Asterisk

{ENTEN:4}

Regards.


      


From gmaruzz at celliax.org  Thu Jan 28 13:41:16 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 28 Jan 2010 22:41:16 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B61FCAB.5040707@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com> 
	<4B6070A8.6050607@gmx.net>
	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com> 
	<4B607944.4040700@gmx.net> <4B608014.4030902@gmx.net>
	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com> 
	<4B61ECE0.10409@gmx.net>
	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com> 
	<4B61FCAB.5040707@gmx.net>
Message-ID: <7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>

with three instances you will assign the hw:0 device to skype client
0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
Must work. Pay attention to assign the same device name to all devices
needed by a skype instance (sound devices window): playback, capture
AND ring.

Or maybe is a bug of ALSA on Debian...

-giovanni

On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
> #2 to the Skype accounts. Still no sound.
> On the frist call there is one way audio, on the following calls there
> is no audio at all.
> This is weird.
>
> Best regards
> Peter
>
> Giovanni Maruzzelli schrieb:
>> Ciao Peter,
>>
>> Never tested on Debian 5.
>>
>> When you write "same problem" you are referring to the audio going one
>> way only (btw, which way?) with the custom audio driver?
>>
>> Have you tried with multiple instances of the regular Debian
>> snd-dummy, as I wrote in a mail before?
>>
>> -gm
>>
>>
>>
>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>
>>> Hello Giovanni,
>>>
>>> I did so but the same problem again.
>>>
>>> Did you ever test in on Debian 5.0?
>>>
>>> Best reards
>>> Peter
>>>
>>> Giovanni Maruzzelli schrieb:
>>>
>>>> good, so you have only one sound device, the right one.
>>>>
>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>
>>>> -gm
>>>>
>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>
>>>>
>>>>> I installed alsa-utile,
>>>>>
>>>>> now I get:
>>>>>
>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>> ?Subdevices: 127/128
>>>>> ?Subdevice #0: subdevice #0
>>>>> ?Subdevice #1: subdevice #1
>>>>> ?Subdevice #2: subdevice #2
>>>>> ?Subdevice #3: subdevice #3
>>>>> ?Subdevice #4: subdevice #4
>>>>> ?Subdevice #5: subdevice #5
>>>>> ?Subdevice #6: subdevice #6
>>>>> ?Subdevice #7: subdevice #7
>>>>> ?Subdevice #8: subdevice #8
>>>>> ?Subdevice #9: subdevice #9
>>>>> ?Subdevice #10: subdevice #10
>>>>> ?Subdevice #11: subdevice #11
>>>>> ?Subdevice #12: subdevice #12
>>>>> ?Subdevice #13: subdevice #13
>>>>> ?Subdevice #14: subdevice #14
>>>>> ?Subdevice #15: subdevice #15
>>>>> ?Subdevice #16: subdevice #16
>>>>> ?Subdevice #17: subdevice #17
>>>>> ?Subdevice #18: subdevice #18
>>>>> ?Subdevice #19: subdevice #19
>>>>> ?Subdevice #20: subdevice #20
>>>>> ?Subdevice #21: subdevice #21
>>>>> ?Subdevice #22: subdevice #22
>>>>> ?Subdevice #23: subdevice #23
>>>>> ?Subdevice #24: subdevice #24
>>>>> ?Subdevice #25: subdevice #25
>>>>> ?Subdevice #26: subdevice #26
>>>>> ?Subdevice #27: subdevice #27
>>>>> ?Subdevice #28: subdevice #28
>>>>> ?Subdevice #29: subdevice #29
>>>>> ?Subdevice #30: subdevice #30
>>>>> ?Subdevice #31: subdevice #31
>>>>> ?Subdevice #32: subdevice #32
>>>>> ?Subdevice #33: subdevice #33
>>>>> ?Subdevice #34: subdevice #34
>>>>> ?Subdevice #35: subdevice #35
>>>>> ?Subdevice #36: subdevice #36
>>>>> ?Subdevice #37: subdevice #37
>>>>> ?Subdevice #38: subdevice #38
>>>>> ?Subdevice #39: subdevice #39
>>>>> ?Subdevice #40: subdevice #40
>>>>> ?Subdevice #41: subdevice #41
>>>>> ?Subdevice #42: subdevice #42
>>>>> ?Subdevice #43: subdevice #43
>>>>> ?Subdevice #44: subdevice #44
>>>>> ?Subdevice #45: subdevice #45
>>>>> ?Subdevice #46: subdevice #46
>>>>> ?Subdevice #47: subdevice #47
>>>>> ?Subdevice #48: subdevice #48
>>>>> ?Subdevice #49: subdevice #49
>>>>> ?Subdevice #50: subdevice #50
>>>>> ?Subdevice #51: subdevice #51
>>>>> ?Subdevice #52: subdevice #52
>>>>> ?Subdevice #53: subdevice #53
>>>>> ?Subdevice #54: subdevice #54
>>>>> ?Subdevice #55: subdevice #55
>>>>> ?Subdevice #56: subdevice #56
>>>>> ?Subdevice #57: subdevice #57
>>>>> ?Subdevice #58: subdevice #58
>>>>> ?Subdevice #59: subdevice #59
>>>>> ?Subdevice #60: subdevice #60
>>>>> ?Subdevice #61: subdevice #61
>>>>> ?Subdevice #62: subdevice #62
>>>>> ?Subdevice #63: subdevice #63
>>>>> ?Subdevice #64: subdevice #64
>>>>> ?Subdevice #65: subdevice #65
>>>>> ?Subdevice #66: subdevice #66
>>>>> ?Subdevice #67: subdevice #67
>>>>> ?Subdevice #68: subdevice #68
>>>>> ?Subdevice #69: subdevice #69
>>>>> ?Subdevice #70: subdevice #70
>>>>> ?Subdevice #71: subdevice #71
>>>>> ?Subdevice #72: subdevice #72
>>>>> ?Subdevice #73: subdevice #73
>>>>> ?Subdevice #74: subdevice #74
>>>>> ?Subdevice #75: subdevice #75
>>>>> ?Subdevice #76: subdevice #76
>>>>> ?Subdevice #77: subdevice #77
>>>>> ?Subdevice #78: subdevice #78
>>>>> ?Subdevice #79: subdevice #79
>>>>> ?Subdevice #80: subdevice #80
>>>>> ?Subdevice #81: subdevice #81
>>>>> ?Subdevice #82: subdevice #82
>>>>> ?Subdevice #83: subdevice #83
>>>>> ?Subdevice #84: subdevice #84
>>>>> ?Subdevice #85: subdevice #85
>>>>> ?Subdevice #86: subdevice #86
>>>>> ?Subdevice #87: subdevice #87
>>>>> ?Subdevice #88: subdevice #88
>>>>> ?Subdevice #89: subdevice #89
>>>>> ?Subdevice #90: subdevice #90
>>>>> ?Subdevice #91: subdevice #91
>>>>> ?Subdevice #92: subdevice #92
>>>>> ?Subdevice #93: subdevice #93
>>>>> ?Subdevice #94: subdevice #94
>>>>> ?Subdevice #95: subdevice #95
>>>>> ?Subdevice #96: subdevice #96
>>>>> ?Subdevice #97: subdevice #97
>>>>> ?Subdevice #98: subdevice #98
>>>>> ?Subdevice #99: subdevice #99
>>>>> ?Subdevice #100: subdevice #100
>>>>> ?Subdevice #101: subdevice #101
>>>>> ?Subdevice #102: subdevice #102
>>>>> ?Subdevice #103: subdevice #103
>>>>> ?Subdevice #104: subdevice #104
>>>>> ?Subdevice #105: subdevice #105
>>>>> ?Subdevice #106: subdevice #106
>>>>> ?Subdevice #107: subdevice #107
>>>>> ?Subdevice #108: subdevice #108
>>>>> ?Subdevice #109: subdevice #109
>>>>> ?Subdevice #110: subdevice #110
>>>>> ?Subdevice #111: subdevice #111
>>>>> ?Subdevice #112: subdevice #112
>>>>> ?Subdevice #113: subdevice #113
>>>>> ?Subdevice #114: subdevice #114
>>>>> ?Subdevice #115: subdevice #115
>>>>> ?Subdevice #116: subdevice #116
>>>>> ?Subdevice #117: subdevice #117
>>>>> ?Subdevice #118: subdevice #118
>>>>> ?Subdevice #119: subdevice #119
>>>>> ?Subdevice #120: subdevice #120
>>>>> ?Subdevice #121: subdevice #121
>>>>> ?Subdevice #122: subdevice #122
>>>>> ?Subdevice #123: subdevice #123
>>>>> ?Subdevice #124: subdevice #124
>>>>> ?Subdevice #125: subdevice #125
>>>>> ?Subdevice #126: subdevice #126
>>>>> ?Subdevice #127: subdevice #127
>>>>>
>>>>>
>>>>> Peter P GMX schrieb:
>>>>>
>>>>>
>>>>>> Her's the output:
>>>>>>
>>>>>> skype:~# aplay -l
>>>>>> bash: aplay: command not found
>>>>>>
>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>
>>>>>>
>>>>>>
>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>> what's the output of:
>>>>>>>
>>>>>>> aplay -l
>>>>>>>
>>>>>>> ?
>>>>>>>
>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>
>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> Thanks Giovanni,
>>>>>>>>
>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>> since I compiled alsa newly)
>>>>>>>>
>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>
>>>>>>>> Best regards
>>>>>>>> Peter
>>>>>>>>
>>>>>>>>
>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>> This warning is harmless:
>>>>>>>>>
>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>
>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>> Ciao Peter
>>>>>>>>>>
>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>
>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>> 8.04).
>>>>>>>>>>
>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>
>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>
>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>
>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>> soundcard.
>>>>>>>>>>
>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>> just committed some improvements.
>>>>>>>>>>
>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>> write the mailing list again,
>>>>>>>>>>
>>>>>>>>>> ciao for now,
>>>>>>>>>>
>>>>>>>>>> -giovanni
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>
>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>> instructions in
>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>
>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>> suppressed
>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>> suppressed
>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>
>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>
>>>>>>>>>>> Second question:
>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>
>>>>>>>>>>> Best regards
>>>>>>>>>>> Peter
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>> Sincerely,
>>>>>>>>>>
>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From testeador01 at gmail.com  Thu Jan 28 13:51:31 2010
From: testeador01 at gmail.com (Milena)
Date: Thu, 28 Jan 2010 16:51:31 -0500
Subject: [Freeswitch-users] ERR root tag missing
Message-ID: <c29b930b1001281351r3bebc2f2ta0279a9bbd4b6c0e@mail.gmail.com>

Hello :)
First, do not hijack threads, if you want to post about a different problem,
do not click reply and then change the subject, please create a NEW MESSAGE.

About your question, are you using xml_curl? or any other dialplan seekers?

-Milena

2010/1/28 Troy Anderson <troy at tlainvestments.com>

> I'm seeing this error quite often on my systems:
>        2010-01-28 12:35:46.703112 [ERR] switch_xml.c:1571 Error[[error near
> line 1]: root tag missing]
>
> I've looked at freeswitch.xml.fsxml to see if I could find some kind of
> malformed XML, but with no luck.  Which Is line 1is it referring to?
>
> Line 1 of freeswitch.xml.fsxml is a comment, with the first actual XML line
> like:
>        <document type="freeswitch/xml">
>
> This error always happens right after after a mod_dialplan_xml.c:408 log
> message, so I'm led to believe my dialplan XML is messed up, but I cannot
> see where.
>
> In freeswitch.xml.fsxml near the dialplan section, this is what I have:
>
> ...
> <section name="configuration" description="Various Configuration">
>        <!-- CUT a bunch of stuff ... CUT -->
> </section>
> <section name="dialplan" description="Regex/XML Dialplan">
>        <context name="my_office">
>                <extension name="unloop">
>                        <condition field="${unroll_loops}"
> expression="^true$"/>
>                        <condition field="${sip_looped_call}"
> expression="^true$">
>                                <action application="deflect"
> data="${destination_number}"/>
>                        </condition>
>                </extension>
>                <extension name="set_domain" continue="true">
>                        <condition field="${domain_name}" expression="^$"/>
>                        <condition field="source" expression="mod_sofia"/>
>                        <condition field="${sip_auth_realm}"
> expression="^$">
>                                <action application="set"
> data="domain_name=10.0.0.120"/>
>                                <anti-action application="set"
> data="domain_name=${sip_auth_realm}"/>
>                        </condition>
>                </extension>
> ...
> </section>
> ...
>
> Thanks for any ideas!
>
> -Troy
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Prometheus001 at gmx.net  Thu Jan 28 13:54:19 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 28 Jan 2010 22:54:19 +0100
Subject: [Freeswitch-users] Set an gateway parameter for external gateway
	(skype)
Message-ID: <4B62078B.5060406@gmx.net>

Hello,

I have a main freeswitch server and a separate Freeswitch/Skype server
with mod_skypiax.

I want main freeswitch server to tell the Freeswitch/Skype server to use
a dedicated Skype interface(interface1, interface2 etc). What is the
best way to pass this variable?
Some ideas from my side

    * set caller_id_number or caller_id_number_name, as these are
      overwritten by Skype anyway. But this is not a clean solution
    * use another UDP port for each gateway, but this is a huge effort
      if the number of gateways becomes larger
    * set a dedicated sip header for this. Reading sip-header is easy
      (documented) but how to set my own sip header entry in FS?
    * any other idea?


Best regards
Peter



From anthony.minessale at gmail.com  Thu Jan 28 14:42:20 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 Jan 2010 16:42:20 -0600
Subject: [Freeswitch-users] Voicemail via web interface
In-Reply-To: <4B61EAE4.2070607@acsol.net>
References: <4B61ACB3.50903@acsol.net>
	<090DDB5F675E4FC48512CA3FC52FA096@greyhawk.tonecommander.com>
	<4B61EAE4.2070607@acsol.net>
Message-ID: <191c3a031001281442m2b74ebb5xffa19113f54f0948@mail.gmail.com>

yes sadly mp3 up sampled to 11khz is the only thing that works with that
flash player.


On Thu, Jan 28, 2010 at 1:52 PM, John <john at acsol.net> wrote:

> Thanks Robert. I believe the issue is probably because our files are in
> WAV format and not MP3.
>
> On 1/28/2010 12:28 PM, Robert Hadley wrote:
> > Using Firefox I was asked to install the latest Flash plugin and then I
> > could play the messages from the webpage directly.  IE8 never asked to
> add
> > the plugin that I noticed.
> > -RobertH
> >
> >
> > -----Original Message-----
> > From: John [mailto:john at acsol.net]
> > Sent: Thursday, January 28, 2010 7:27 AM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: [Freeswitch-users] Voicemail via web interface
> >
> > Hello,
> > Can you point me to any additional information about the voice mail via
> > web interface? I have it up and running; however if you click the play
> > button there is no playback, if you click download it will play in MS
> > media player. Thanks John
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Thu Jan 28 14:43:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 Jan 2010 16:43:50 -0600
Subject: [Freeswitch-users] prefix on exten
In-Reply-To: <968062.52038.qm@web34307.mail.mud.yahoo.com>
References: <968062.52038.qm@web34307.mail.mud.yahoo.com>
Message-ID: <191c3a031001281443y3e03ca6n5c8880b159ae57fc@mail.gmail.com>

1) capture it in the regex and put the () around the part without the
prefix.
2) do the same thing as asterisk with ${destination_number:4}



On Thu, Jan 28, 2010 at 3:14 PM, FERNANDO VILLARROEL
<fvillarroel at yahoo.com>wrote:

> Dear.
>
> If i receive a call from a customer with some prefx like 1234
>
> How i can do in order to forward this call with out prefix like Asterisk
>
> {ENTEN:4}
>
> Regards.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mustafa.pk at gmail.com  Thu Jan 28 15:21:33 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Fri, 29 Jan 2010 04:21:33 +0500
Subject: [Freeswitch-users] Set an gateway parameter for external
	gateway (skype)
In-Reply-To: <4B62078B.5060406@gmx.net>
References: <4B62078B.5060406@gmx.net>
Message-ID: <8213d6071001281521u27347956nf2c89c3cd2e72745@mail.gmail.com>

i don't know if it's a wise solution, but you can make a db insert
before bridge on freeswitch server, kypiax server can query db after
answer to get an idea!

On Fri, Jan 29, 2010 at 2:54 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello,
>
> I have a main freeswitch server and a separate Freeswitch/Skype server
> with mod_skypiax.
>
> I want main freeswitch server to tell the Freeswitch/Skype server to use
> a dedicated Skype interface(interface1, interface2 etc). What is the
> best way to pass this variable?
> Some ideas from my side
>
> ? ?* set caller_id_number or caller_id_number_name, as these are
> ? ? ?overwritten by Skype anyway. But this is not a clean solution
> ? ?* use another UDP port for each gateway, but this is a huge effort
> ? ? ?if the number of gateways becomes larger
> ? ?* set a dedicated sip header for this. Reading sip-header is easy
> ? ? ?(documented) but how to set my own sip header entry in FS?
> ? ?* any other idea?
>
>
> Best regards
> Peter
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com


From wasim at convergence.pk  Thu Jan 28 15:29:54 2010
From: wasim at convergence.pk (Wasim Baig)
Date: Fri, 29 Jan 2010 04:29:54 +0500
Subject: [Freeswitch-users] Set an gateway parameter for external
	gateway (skype)
In-Reply-To: <8213d6071001281521u27347956nf2c89c3cd2e72745@mail.gmail.com>
References: <4B62078B.5060406@gmx.net>
	<8213d6071001281521u27347956nf2c89c3cd2e72745@mail.gmail.com>
Message-ID: <b8ad2a5b1001281529n4f5ac2acy266ad324b0da9318@mail.gmail.com>

Setting a sip header is a more elegant way ...

http://wiki.freeswitch.org/wiki/Sofia#Adding_Request_Headers

-wasim

On Fri, Jan 29, 2010 at 4:21 AM, Ghulam Mustafa <mustafa.pk at gmail.com>wrote:

> i don't know if it's a wise solution, but you can make a db insert
> before bridge on freeswitch server, kypiax server can query db after
> answer to get an idea!
>
> On Fri, Jan 29, 2010 at 2:54 AM, Peter P GMX <Prometheus001 at gmx.net>
> wrote:
> > Hello,
> >
> > I have a main freeswitch server and a separate Freeswitch/Skype server
> > with mod_skypiax.
> >
> > I want main freeswitch server to tell the Freeswitch/Skype server to use
> > a dedicated Skype interface(interface1, interface2 etc). What is the
> > best way to pass this variable?
> > Some ideas from my side
> >
> >    * set caller_id_number or caller_id_number_name, as these are
> >      overwritten by Skype anyway. But this is not a clean solution
> >    * use another UDP port for each gateway, but this is a huge effort
> >      if the number of gateways becomes larger
> >    * set a dedicated sip header for this. Reading sip-header is easy
> >      (documented) but how to set my own sip header entry in FS?
> >    * any other idea?
> >
> >
> > Best regards
> > Peter
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
peace be upon you ...
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From emptysands at gmail.com  Wed Jan 27 21:54:03 2010
From: emptysands at gmail.com (Nicholas Lee)
Date: Thu, 28 Jan 2010 18:54:03 +1300
Subject: [Freeswitch-users] Hybrid Encryption?
Message-ID: <2b6116b31001272154l3c0bbe80y8bf3db94961e8e1d@mail.gmail.com>

The TLS wiki page talks about [1] Freeswitch being able to act as a past
though proxy for a SIP phone. Turning an unencrypted SIP+RTP session into a
SIPS+SRTP session.

Is there howto guide for the above?

I'm also wondering if Freeswitch could do a drop-in encryption.  ie. Phone
<-> Asterisk replaced with Phone <-> FS  <==> FS <-> Asterisk, where <==>
is encrypted.


Nicholas

[1] http://wiki.freeswitch.org/wiki/Tls#Hybrid_Encryption
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From bobc at panztel.biz  Thu Jan 28 14:28:30 2010
From: bobc at panztel.biz (Bob Coleman)
Date: Fri, 29 Jan 2010 11:28:30 +1300
Subject: [Freeswitch-users] prefix on exten
In-Reply-To: <968062.52038.qm@web34307.mail.mud.yahoo.com>
References: <968062.52038.qm@web34307.mail.mud.yahoo.com>
Message-ID: <8543d2b11001281428m737fb7bev23c7bfd802d2dfee@mail.gmail.com>

In your dialplan do something like this:

<include>
  <extension name="remotefs">
    <condition field="destination_number" expression="^1234(\d+)$">
     <action application="bridge" data="
sofia/internal/$1 at 192.168.50.195:5080"/<sofia/internal/$1 at 192.168.50.195:5080%22/>
>
    </condition>
  </extension>
</include>

The $1 variable has the number without the prefix of 1234.

I am sending the call to another FS box in this example

Bob

On Fri, Jan 29, 2010 at 10:14 AM, FERNANDO VILLARROEL <fvillarroel at yahoo.com
> wrote:

> Dear.
>
> If i receive a call from a customer with some prefx like 1234
>
> How i can do in order to forward this call with out prefix like Asterisk
>
> {ENTEN:4}
>
> Regards.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Thu Jan 28 16:19:14 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 28 Jan 2010 18:19:14 -0600
Subject: [Freeswitch-users] Hybrid Encryption?
In-Reply-To: <2b6116b31001272154l3c0bbe80y8bf3db94961e8e1d@mail.gmail.com>
References: <2b6116b31001272154l3c0bbe80y8bf3db94961e8e1d@mail.gmail.com>
Message-ID: <8A9EDC4E-C49B-488D-9DBF-169A185462AB@freeswitch.org>

Or you could do Phone <-> FS <=====> FS <-> Phone...

;)  Less complex.

/b

On Jan 27, 2010, at 11:54 PM, Nicholas Lee wrote:

> The TLS wiki page talks about [1] Freeswitch being able to act as a past though proxy for a SIP phone. Turning an unencrypted SIP+RTP session into a SIPS+SRTP session.
> 
> Is there howto guide for the above?
> 
> I'm also wondering if Freeswitch could do a drop-in encryption.  ie. Phone <-> Asterisk replaced with Phone <-> FS  <==> FS <-> Asterisk, where <==> is encrypted.
> 
> 
> Nicholas



From emptysands at gmail.com  Thu Jan 28 16:31:02 2010
From: emptysands at gmail.com (Nicholas Lee)
Date: Fri, 29 Jan 2010 08:31:02 +0800
Subject: [Freeswitch-users] Hybrid Encryption?
In-Reply-To: <8A9EDC4E-C49B-488D-9DBF-169A185462AB@freeswitch.org>
References: <2b6116b31001272154l3c0bbe80y8bf3db94961e8e1d@mail.gmail.com>
	<8A9EDC4E-C49B-488D-9DBF-169A185462AB@freeswitch.org>
Message-ID: <2b6116b31001281631u7ada7876wc419bb7afadd7ef7@mail.gmail.com>

Unfortunately it's not going to cover every situation.


Nicholas

On Fri, Jan 29, 2010 at 8:19 AM, Brian West <brian at freeswitch.org> wrote:

> Or you could do Phone <-> FS <=====> FS <-> Phone...
>
> ;)  Less complex.
>
> /b
>
> On Jan 27, 2010, at 11:54 PM, Nicholas Lee wrote:
>
> > The TLS wiki page talks about [1] Freeswitch being able to act as a past
> though proxy for a SIP phone. Turning an unencrypted SIP+RTP session into a
> SIPS+SRTP session.
> >
> > Is there howto guide for the above?
> >
> > I'm also wondering if Freeswitch could do a drop-in encryption.  ie.
> Phone <-> Asterisk replaced with Phone <-> FS  <==> FS <-> Asterisk, where
> <==> is encrypted.
> >
> >
> > Nicholas
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Thu Jan 28 16:52:39 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 28 Jan 2010 18:52:39 -0600
Subject: [Freeswitch-users] Hybrid Encryption?
In-Reply-To: <2b6116b31001281631u7ada7876wc419bb7afadd7ef7@mail.gmail.com>
References: <2b6116b31001272154l3c0bbe80y8bf3db94961e8e1d@mail.gmail.com>
	<8A9EDC4E-C49B-488D-9DBF-169A185462AB@freeswitch.org>
	<2b6116b31001281631u7ada7876wc419bb7afadd7ef7@mail.gmail.com>
Message-ID: <46C06209-9515-4B1B-B449-F55A51FF548B@freeswitch.org>

Then yes you could use FreeSWITCH to augment your Asterisk install and enable encryption from site to site.

/b


> Unfortunately it's not going to cover every situation.
> 
> 
> Nicholas



From dujinfang at gmail.com  Thu Jan 28 17:29:36 2010
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 29 Jan 2010 09:29:36 +0800
Subject: [Freeswitch-users] Set an gateway parameter for external
	gateway (skype)
In-Reply-To: <b8ad2a5b1001281529n4f5ac2acy266ad324b0da9318@mail.gmail.com>
References: <4B62078B.5060406@gmx.net>
	<8213d6071001281521u27347956nf2c89c3cd2e72745@mail.gmail.com>
	<b8ad2a5b1001281529n4f5ac2acy266ad324b0da9318@mail.gmail.com>
Message-ID: <23f91031001281729s403a9954nc7400d73fa69f6ea@mail.gmail.com>

why not just use dialplan matching?

assume FS1 has gateway named skype,

originate sofia/gateway/skype/<your_dedicated_skype_name>+<other_skype_name_you_want_to_dial


on your FS/skype server, set dialplan:


$destination_number to match    (.*)\+(.*), then you can

bridge skypiax/$1/$2

2010/1/29 Wasim Baig <wasim at convergence.pk>:
> Setting a sip header is a more elegant way ...
>
> http://wiki.freeswitch.org/wiki/Sofia#Adding_Request_Headers
>
> -wasim
>
> On Fri, Jan 29, 2010 at 4:21 AM, Ghulam Mustafa <mustafa.pk at gmail.com>
> wrote:
>>
>> i don't know if it's a wise solution, but you can make a db insert
>> before bridge on freeswitch server, kypiax server can query db after
>> answer to get an idea!
>>
>> On Fri, Jan 29, 2010 at 2:54 AM, Peter P GMX <Prometheus001 at gmx.net>
>> wrote:
>> > Hello,
>> >
>> > I have a main freeswitch server and a separate Freeswitch/Skype server
>> > with mod_skypiax.
>> >
>> > I want main freeswitch server to tell the Freeswitch/Skype server to use
>> > a dedicated Skype interface(interface1, interface2 etc). What is the
>> > best way to pass this variable?
>> > Some ideas from my side
>> >
>> > ? ?* set caller_id_number or caller_id_number_name, as these are
>> > ? ? ?overwritten by Skype anyway. But this is not a clean solution
>> > ? ?* use another UDP port for each gateway, but this is a huge effort
>> > ? ? ?if the number of gateways becomes larger
>> > ? ?* set a dedicated sip header for this. Reading sip-header is easy
>> > ? ? ?(documented) but how to set my own sip header entry in FS?
>> > ? ?* any other idea?
>> >
>> >
>> > Best regards
>> > Peter
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Ghulam Mustafa
>> cell: +92 333.611.7681
>> sip: cyrenity at ekiga.net
>> mail: mustafa.pk at gmail.com
>> web: cyrenity.wordpress.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
> peace be upon you ...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From emptysands at gmail.com  Thu Jan 28 18:08:29 2010
From: emptysands at gmail.com (Nicholas Lee)
Date: Fri, 29 Jan 2010 15:08:29 +1300
Subject: [Freeswitch-users] Hybrid Encryption?
In-Reply-To: <46C06209-9515-4B1B-B449-F55A51FF548B@freeswitch.org>
References: <2b6116b31001272154l3c0bbe80y8bf3db94961e8e1d@mail.gmail.com>
	<8A9EDC4E-C49B-488D-9DBF-169A185462AB@freeswitch.org>
	<2b6116b31001281631u7ada7876wc419bb7afadd7ef7@mail.gmail.com>
	<46C06209-9515-4B1B-B449-F55A51FF548B@freeswitch.org>
Message-ID: <2b6116b31001281808x1a004cd2ne7e8dcb9f16fec3e@mail.gmail.com>

Is there a way to do it transparently?  The FS proxies will past though the
extension creds.

On Fri, Jan 29, 2010 at 1:52 PM, Brian West <brian at freeswitch.org> wrote:

> Then yes you could use FreeSWITCH to augment your Asterisk install and
> enable encryption from site to site.
>
> /b
>
>
> > Unfortunately it's not going to cover every situation.
> >
> >
> > Nicholas
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From troy at tlainvestments.com  Thu Jan 28 19:18:03 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 28 Jan 2010 20:18:03 -0700
Subject: [Freeswitch-users] ERR root tag missing
In-Reply-To: <c29b930b1001281351r3bebc2f2ta0279a9bbd4b6c0e@mail.gmail.com>
References: <c29b930b1001281351r3bebc2f2ta0279a9bbd4b6c0e@mail.gmail.com>
Message-ID: <5B9401A1-DF37-4ED1-827A-2B95DAE7AEF2@tlainvestments.com>

Sorry about hijacking this thread!  I now know better.

I am using xml_curl, and I think that's the heads up I needed. Lemme check into what I'm returning from that...

Thanks!

On Jan 28, 2010, at 2:51 PM, Milena wrote:

> Hello :)
> First, do not hijack threads, if you want to post about a different problem, do not click reply and then change the subject, please create a NEW MESSAGE.
> 
> About your question, are you using xml_curl? or any other dialplan seekers?
> 
> -Milena
> 
> 2010/1/28 Troy Anderson <troy at tlainvestments.com>
> I'm seeing this error quite often on my systems:
>        2010-01-28 12:35:46.703112 [ERR] switch_xml.c:1571 Error[[error near line 1]: root tag missing]
> 
> I've looked at freeswitch.xml.fsxml to see if I could find some kind of malformed XML, but with no luck.  Which Is line 1is it referring to?
> 
> Line 1 of freeswitch.xml.fsxml is a comment, with the first actual XML line like:
>        <document type="freeswitch/xml">
> 
> This error always happens right after after a mod_dialplan_xml.c:408 log message, so I'm led to believe my dialplan XML is messed up, but I cannot see where.
> 
> In freeswitch.xml.fsxml near the dialplan section, this is what I have:
> 
> ...
> <section name="configuration" description="Various Configuration">
>        <!-- CUT a bunch of stuff ... CUT -->
> </section>
> <section name="dialplan" description="Regex/XML Dialplan">
>        <context name="my_office">
>                <extension name="unloop">
>                        <condition field="${unroll_loops}" expression="^true$"/>
>                        <condition field="${sip_looped_call}" expression="^true$">
>                                <action application="deflect" data="${destination_number}"/>
>                        </condition>
>                </extension>
>                <extension name="set_domain" continue="true">
>                        <condition field="${domain_name}" expression="^$"/>
>                        <condition field="source" expression="mod_sofia"/>
>                        <condition field="${sip_auth_realm}" expression="^$">
>                                <action application="set" data="domain_name=10.0.0.120"/>
>                                <anti-action application="set" data="domain_name=${sip_auth_realm}"/>
>                        </condition>
>                </extension>
> ...
> </section>
> ...
> 
> Thanks for any ideas!
> 
> -Troy
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mcampbellsmith at gmail.com  Thu Jan 28 19:32:26 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 29 Jan 2010 14:32:26 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <191c3a031001272125t2b19d134lbb65dfebbb7a52b9@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
	<33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
	<191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>
	<33c87fa31001271939i668e9676y1f687dc182b4ca89@mail.gmail.com>
	<CF0D5382-EE44-48CB-BF14-27A443014336@freeswitch.org>
	<33c87fa31001272035t7c46e054t7ce2dd598fa40ebb@mail.gmail.com>
	<191c3a031001272125t2b19d134lbb65dfebbb7a52b9@mail.gmail.com>
Message-ID: <33c87fa31001281932h73466f87yf4715058f4632558@mail.gmail.com>

Hi !

I confirmed yesterday that if the SPA is not NAT'd, then the event is
sent.  I just removed NAT from the extension that I was having
problems with.

Looking at the db tables, it appears there are two - the
sofia_reg_internal.db and sofia_reg_internal_nat.db

Could it be that the sendevent command is only looking in the
sofia_reg_internal.db database and not sofia_reg_internal_nat.db ?



On Thu, Jan 28, 2010 at 4:25 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> You have to look in the sql db and compare the specified vals with the ones
> looked up from the event again the user and host need to match the db
>
> On Jan 27, 2010 10:41 PM, "Mark Campbell-Smith" <mcampbellsmith at gmail.com>
> wrote:
>
> Hi Brian,
>
> ?I've previously enabled siptrace for internal profile, but I see
> nothing sent and nothing received.
>
> On Thu, Jan 28, 2010 at 2:54 PM, Brian West <brian at freeswitch.org> wrote:
>> I'm suspecting the code...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From ranjtech at gmail.com  Thu Jan 28 21:57:03 2010
From: ranjtech at gmail.com (RR)
Date: Fri, 29 Jan 2010 00:57:03 -0500
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
Message-ID: <025701caa0a7$e1ca6200$a55f2600$@com>

Hi Brian,

 

Ok here's the sip trace captured at the softswitch. 

 

BTW, I noticed that during startup, I see FS printing out this message:

2010-01-29 01:30:05.634220 [NOTICE] sofia_reg.c:2267 Added gateway
'Test-Inbound' to profile 'external'

2010-01-29 01:30:05.634547 [ERR] sofia.c:1663 ERROR: username param is
REQUIRED!

2010-01-29 01:30:05.634657 [ERR] sofia.c:1663 ERROR: username param is
REQUIRED!

 

Which is weird, because as you can see from the config, the username is
infact present. Weird! Anyway, here's the trace

 

REGISTER sip:myswitch.net.au SIP/2.0

Via: SIP/2.0/UDP 173.xxx.xxx.xxx:5080;rport;branch=z9hG4bK1U32gpr9vj5eF

Max-Forwards: 70

From: <sip:user1 at myswitch.net.au;transport=udp>;tag=11Qey4tcrUH9g

To: <sip:user1 at myswitch.net.au;transport=udp>

Call-ID: cd589b73-031e-445f-a4d8-2fe334d81bbc

CSeq: 126255215 REGISTER

Contact:
<sip:gw+Test-Inbound at 173.xxx.xxx.xxx:5080;transport=udp;gw=Test-Inbound>

Expires: 60

User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15980

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY

Supported: timer, precondition, path, replaces

Content-Length: 0

SIP/2.0 409 Conflict

Via: SIP/2.0/UDP 173.xxx.xxx.xxx:5080;rport;branch=z9hG4bK1U32gpr9vj5eF

From: <sip:user1 at 10.1.3.20;transport=udp>;tag=11Qey4tcrUH9g

To: <sip:user1 at 10.1.3.20;transport=udp>

Call-ID: cd589b73-031e-445f-a4d8-2fe334d81bbc

CSeq: 126255215 REGISTER

Content-Length: 0

 

.and then this message just repeats again and again with every REGISTER
request.

 

Thanks for your help

\RR

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, January 28, 2010 2:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 409 Conflict When registering FS with
Softswitch

 

Any sip trace would work.. text.. or what not.. 

 

 

 

/b

 

On Jan 28, 2010, at 12:46 PM, RR wrote:





Hi brian,

 

Do you need the sip trace from within FS or just any packet capture like
from snoop/tcpdump etc will do? I don't know how to do a sip trace from
within FS

 

\R

 



__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4815 (20100128) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From magesh.freeswitch at gmail.com  Thu Jan 28 22:32:08 2010
From: magesh.freeswitch at gmail.com (Magesh R)
Date: Fri, 29 Jan 2010 01:32:08 -0500
Subject: [Freeswitch-users] min_dtmf_duration has not changed
Message-ID: <369c72d81001282232r6f7ef2f2m745e71fcdc6b73e2@mail.gmail.com>

Dear All,
   I am trying to change the min_dtmf_duration value by using fsctl. But it
didn't changed.

freeswitch at debian> fsctl min_dtmf_duration 800

+OK min dtmf duration: 400

     Is there any thing need to be set before changing this?

Thanks,
Mag.
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From christian.loeschenkohl at xpirio.com  Fri Jan 29 00:32:05 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Fri, 29 Jan 2010 09:32:05 +0100
Subject: [Freeswitch-users] wiki password recovery - no mail is send
Message-ID: <4B629D05.1060908@xpirio.com>

hello

the password recovery for the fs wiki doesn't seem to work.
no e-mail is send when entering the username and press "e-mail new password".

may i assist here, we do maintain a few wikis for ourself.

br


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From Prometheus001 at gmx.net  Fri Jan 29 03:32:05 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 29 Jan 2010 12:32:05 +0100
Subject: [Freeswitch-users] Set an gateway parameter for
 external	gateway (skype)
In-Reply-To: <23f91031001281729s403a9954nc7400d73fa69f6ea@mail.gmail.com>
References: <4B62078B.5060406@gmx.net>	<8213d6071001281521u27347956nf2c89c3cd2e72745@mail.gmail.com>	<b8ad2a5b1001281529n4f5ac2acy266ad324b0da9318@mail.gmail.com>
	<23f91031001281729s403a9954nc7400d73fa69f6ea@mail.gmail.com>
Message-ID: <4B62C735.4010309@gmx.net>

Hello Sven

I think that's a good idea. I just tried: "+" is not valid in a Skype
username. So I may use it.

I will try that and give some feedback.

Best regards
Peter


Seven Du schrieb:
> why not just use dialplan matching?
>
> assume FS1 has gateway named skype,
>
> originate sofia/gateway/skype/<your_dedicated_skype_name>+<other_skype_name_you_want_to_dial
>
>
> on your FS/skype server, set dialplan:
>
>
> $destination_number to match    (.*)\+(.*), then you can
>
> bridge skypiax/$1/$2
>
> 2010/1/29 Wasim Baig <wasim at convergence.pk>:
>   
>> Setting a sip header is a more elegant way ...
>>
>> http://wiki.freeswitch.org/wiki/Sofia#Adding_Request_Headers
>>
>> -wasim
>>
>> On Fri, Jan 29, 2010 at 4:21 AM, Ghulam Mustafa <mustafa.pk at gmail.com>
>> wrote:
>>     
>>> i don't know if it's a wise solution, but you can make a db insert
>>> before bridge on freeswitch server, kypiax server can query db after
>>> answer to get an idea!
>>>
>>> On Fri, Jan 29, 2010 at 2:54 AM, Peter P GMX <Prometheus001 at gmx.net>
>>> wrote:
>>>       
>>>> Hello,
>>>>
>>>> I have a main freeswitch server and a separate Freeswitch/Skype server
>>>> with mod_skypiax.
>>>>
>>>> I want main freeswitch server to tell the Freeswitch/Skype server to use
>>>> a dedicated Skype interface(interface1, interface2 etc). What is the
>>>> best way to pass this variable?
>>>> Some ideas from my side
>>>>
>>>>    * set caller_id_number or caller_id_number_name, as these are
>>>>      overwritten by Skype anyway. But this is not a clean solution
>>>>    * use another UDP port for each gateway, but this is a huge effort
>>>>      if the number of gateways becomes larger
>>>>    * set a dedicated sip header for this. Reading sip-header is easy
>>>>      (documented) but how to set my own sip header entry in FS?
>>>>    * any other idea?
>>>>
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>         
>>>
>>> --
>>> Ghulam Mustafa
>>> cell: +92 333.611.7681
>>> sip: cyrenity at ekiga.net
>>> mail: mustafa.pk at gmail.com
>>> web: cyrenity.wordpress.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>       
>>
>> --
>> wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
>> peace be upon you ...
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From Prometheus001 at gmx.net  Fri Jan 29 03:41:22 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 29 Jan 2010 12:41:22 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com>
	<4B6070A8.6050607@gmx.net>	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com>
	<4B607944.4040700@gmx.net>
	<4B608014.4030902@gmx.net>	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>
	<4B61ECE0.10409@gmx.net>	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>
	<4B61FCAB.5040707@gmx.net>
	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>
Message-ID: <4B62C962.7000601@gmx.net>

I now reinstalled the original sound drivers
Unfortunaltely the sound problems remain, not that worse but they are there:
Audio is still (almost) one way. Almost means:

    * SIP -> Skype ok
    * Skype=> SIP I hear only some scratching on very loud audio

Could it be a volume problem? But snd-dummy should have no volume
properties, right?

Best regards
Peter


Giovanni Maruzzelli schrieb:
> with three instances you will assign the hw:0 device to skype client
> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
> Must work. Pay attention to assign the same device name to all devices
> needed by a skype instance (sound devices window): playback, capture
> AND ring.
>
> Or maybe is a bug of ALSA on Debian...
>
> -giovanni
>
> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>> #2 to the Skype accounts. Still no sound.
>> On the frist call there is one way audio, on the following calls there
>> is no audio at all.
>> This is weird.
>>
>> Best regards
>> Peter
>>
>> Giovanni Maruzzelli schrieb:
>>     
>>> Ciao Peter,
>>>
>>> Never tested on Debian 5.
>>>
>>> When you write "same problem" you are referring to the audio going one
>>> way only (btw, which way?) with the custom audio driver?
>>>
>>> Have you tried with multiple instances of the regular Debian
>>> snd-dummy, as I wrote in a mail before?
>>>
>>> -gm
>>>
>>>
>>>
>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>       
>>>> Hello Giovanni,
>>>>
>>>> I did so but the same problem again.
>>>>
>>>> Did you ever test in on Debian 5.0?
>>>>
>>>> Best reards
>>>> Peter
>>>>
>>>> Giovanni Maruzzelli schrieb:
>>>>
>>>>         
>>>>> good, so you have only one sound device, the right one.
>>>>>
>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>
>>>>> -gm
>>>>>
>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>
>>>>>
>>>>>           
>>>>>> I installed alsa-utile,
>>>>>>
>>>>>> now I get:
>>>>>>
>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>  Subdevices: 127/128
>>>>>>  Subdevice #0: subdevice #0
>>>>>>  Subdevice #1: subdevice #1
>>>>>>  Subdevice #2: subdevice #2
>>>>>>  Subdevice #3: subdevice #3
>>>>>>  Subdevice #4: subdevice #4
>>>>>>  Subdevice #5: subdevice #5
>>>>>>  Subdevice #6: subdevice #6
>>>>>>  Subdevice #7: subdevice #7
>>>>>>  Subdevice #8: subdevice #8
>>>>>>  Subdevice #9: subdevice #9
>>>>>>  Subdevice #10: subdevice #10
>>>>>>  Subdevice #11: subdevice #11
>>>>>>  Subdevice #12: subdevice #12
>>>>>>  Subdevice #13: subdevice #13
>>>>>>  Subdevice #14: subdevice #14
>>>>>>  Subdevice #15: subdevice #15
>>>>>>  Subdevice #16: subdevice #16
>>>>>>  Subdevice #17: subdevice #17
>>>>>>  Subdevice #18: subdevice #18
>>>>>>  Subdevice #19: subdevice #19
>>>>>>  Subdevice #20: subdevice #20
>>>>>>  Subdevice #21: subdevice #21
>>>>>>  Subdevice #22: subdevice #22
>>>>>>  Subdevice #23: subdevice #23
>>>>>>  Subdevice #24: subdevice #24
>>>>>>  Subdevice #25: subdevice #25
>>>>>>  Subdevice #26: subdevice #26
>>>>>>  Subdevice #27: subdevice #27
>>>>>>  Subdevice #28: subdevice #28
>>>>>>  Subdevice #29: subdevice #29
>>>>>>  Subdevice #30: subdevice #30
>>>>>>  Subdevice #31: subdevice #31
>>>>>>  Subdevice #32: subdevice #32
>>>>>>  Subdevice #33: subdevice #33
>>>>>>  Subdevice #34: subdevice #34
>>>>>>  Subdevice #35: subdevice #35
>>>>>>  Subdevice #36: subdevice #36
>>>>>>  Subdevice #37: subdevice #37
>>>>>>  Subdevice #38: subdevice #38
>>>>>>  Subdevice #39: subdevice #39
>>>>>>  Subdevice #40: subdevice #40
>>>>>>  Subdevice #41: subdevice #41
>>>>>>  Subdevice #42: subdevice #42
>>>>>>  Subdevice #43: subdevice #43
>>>>>>  Subdevice #44: subdevice #44
>>>>>>  Subdevice #45: subdevice #45
>>>>>>  Subdevice #46: subdevice #46
>>>>>>  Subdevice #47: subdevice #47
>>>>>>  Subdevice #48: subdevice #48
>>>>>>  Subdevice #49: subdevice #49
>>>>>>  Subdevice #50: subdevice #50
>>>>>>  Subdevice #51: subdevice #51
>>>>>>  Subdevice #52: subdevice #52
>>>>>>  Subdevice #53: subdevice #53
>>>>>>  Subdevice #54: subdevice #54
>>>>>>  Subdevice #55: subdevice #55
>>>>>>  Subdevice #56: subdevice #56
>>>>>>  Subdevice #57: subdevice #57
>>>>>>  Subdevice #58: subdevice #58
>>>>>>  Subdevice #59: subdevice #59
>>>>>>  Subdevice #60: subdevice #60
>>>>>>  Subdevice #61: subdevice #61
>>>>>>  Subdevice #62: subdevice #62
>>>>>>  Subdevice #63: subdevice #63
>>>>>>  Subdevice #64: subdevice #64
>>>>>>  Subdevice #65: subdevice #65
>>>>>>  Subdevice #66: subdevice #66
>>>>>>  Subdevice #67: subdevice #67
>>>>>>  Subdevice #68: subdevice #68
>>>>>>  Subdevice #69: subdevice #69
>>>>>>  Subdevice #70: subdevice #70
>>>>>>  Subdevice #71: subdevice #71
>>>>>>  Subdevice #72: subdevice #72
>>>>>>  Subdevice #73: subdevice #73
>>>>>>  Subdevice #74: subdevice #74
>>>>>>  Subdevice #75: subdevice #75
>>>>>>  Subdevice #76: subdevice #76
>>>>>>  Subdevice #77: subdevice #77
>>>>>>  Subdevice #78: subdevice #78
>>>>>>  Subdevice #79: subdevice #79
>>>>>>  Subdevice #80: subdevice #80
>>>>>>  Subdevice #81: subdevice #81
>>>>>>  Subdevice #82: subdevice #82
>>>>>>  Subdevice #83: subdevice #83
>>>>>>  Subdevice #84: subdevice #84
>>>>>>  Subdevice #85: subdevice #85
>>>>>>  Subdevice #86: subdevice #86
>>>>>>  Subdevice #87: subdevice #87
>>>>>>  Subdevice #88: subdevice #88
>>>>>>  Subdevice #89: subdevice #89
>>>>>>  Subdevice #90: subdevice #90
>>>>>>  Subdevice #91: subdevice #91
>>>>>>  Subdevice #92: subdevice #92
>>>>>>  Subdevice #93: subdevice #93
>>>>>>  Subdevice #94: subdevice #94
>>>>>>  Subdevice #95: subdevice #95
>>>>>>  Subdevice #96: subdevice #96
>>>>>>  Subdevice #97: subdevice #97
>>>>>>  Subdevice #98: subdevice #98
>>>>>>  Subdevice #99: subdevice #99
>>>>>>  Subdevice #100: subdevice #100
>>>>>>  Subdevice #101: subdevice #101
>>>>>>  Subdevice #102: subdevice #102
>>>>>>  Subdevice #103: subdevice #103
>>>>>>  Subdevice #104: subdevice #104
>>>>>>  Subdevice #105: subdevice #105
>>>>>>  Subdevice #106: subdevice #106
>>>>>>  Subdevice #107: subdevice #107
>>>>>>  Subdevice #108: subdevice #108
>>>>>>  Subdevice #109: subdevice #109
>>>>>>  Subdevice #110: subdevice #110
>>>>>>  Subdevice #111: subdevice #111
>>>>>>  Subdevice #112: subdevice #112
>>>>>>  Subdevice #113: subdevice #113
>>>>>>  Subdevice #114: subdevice #114
>>>>>>  Subdevice #115: subdevice #115
>>>>>>  Subdevice #116: subdevice #116
>>>>>>  Subdevice #117: subdevice #117
>>>>>>  Subdevice #118: subdevice #118
>>>>>>  Subdevice #119: subdevice #119
>>>>>>  Subdevice #120: subdevice #120
>>>>>>  Subdevice #121: subdevice #121
>>>>>>  Subdevice #122: subdevice #122
>>>>>>  Subdevice #123: subdevice #123
>>>>>>  Subdevice #124: subdevice #124
>>>>>>  Subdevice #125: subdevice #125
>>>>>>  Subdevice #126: subdevice #126
>>>>>>  Subdevice #127: subdevice #127
>>>>>>
>>>>>>
>>>>>> Peter P GMX schrieb:
>>>>>>
>>>>>>
>>>>>>             
>>>>>>> Her's the output:
>>>>>>>
>>>>>>> skype:~# aplay -l
>>>>>>> bash: aplay: command not found
>>>>>>>
>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>> what's the output of:
>>>>>>>>
>>>>>>>> aplay -l
>>>>>>>>
>>>>>>>> ?
>>>>>>>>
>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>
>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>
>>>>>>>>
>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>>>> Thanks Giovanni,
>>>>>>>>>
>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>
>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>
>>>>>>>>> Best regards
>>>>>>>>> Peter
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>>> This warning is harmless:
>>>>>>>>>>
>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>
>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>                     
>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>
>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>
>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>> 8.04).
>>>>>>>>>>>
>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>
>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>
>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>
>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>> soundcard.
>>>>>>>>>>>
>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>
>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>
>>>>>>>>>>> ciao for now,
>>>>>>>>>>>
>>>>>>>>>>> -giovanni
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>
>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>> instructions in
>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>
>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>> suppressed
>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>> suppressed
>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>
>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>
>>>>>>>>>>>> Second question:
>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>
>>>>>>>>>>>> Best regards
>>>>>>>>>>>> Peter
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>                         
>>>>>>>>>>> --
>>>>>>>>>>> Sincerely,
>>>>>>>>>>>
>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>>             
>>>>>
>>>>>           
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   


From codecomplete at free.fr  Fri Jan 29 03:52:26 2010
From: codecomplete at free.fr (Fred-145)
Date: Fri, 29 Jan 2010 12:52:26 +0100
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
References: <27338355.post@talk.nabble.com>
	<27338355.post-WJuSqJV8a7jJsTRiRinrng@public.gmane.org>
	<201001270750.16772.sos@sokhapkin.dyndns.org>
Message-ID: <eui5m5hg6tqemri4r5u250hu72mjddmjnt@4ax.com>

On Wed, 27 Jan 2010 07:50:16 -0500, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
>set bypass_media=true

Thanks everyone for the feedback.

Are there drawbacks to having RTP pakets flow directly between the SIP
end-points?



From mustafa.pk at gmail.com  Fri Jan 29 04:03:39 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Fri, 29 Jan 2010 17:03:39 +0500
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B62C962.7000601@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com>
	<4B607944.4040700@gmx.net> <4B608014.4030902@gmx.net>
	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>
	<4B61ECE0.10409@gmx.net>
	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>
	<4B61FCAB.5040707@gmx.net>
	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>
	<4B62C962.7000601@gmx.net>
Message-ID: <8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>

did you enable debug mode while compiling custom snd-dummy? if  yes
try re-compiling with debug mode disabled.

-m

On Fri, Jan 29, 2010 at 4:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> I now reinstalled the original sound drivers
> Unfortunaltely the sound problems remain, not that worse but they are there:
> Audio is still (almost) one way. Almost means:
>
> ? ?* SIP -> Skype ok
> ? ?* Skype=> SIP I hear only some scratching on very loud audio
>
> Could it be a volume problem? But snd-dummy should have no volume
> properties, right?
>
> Best regards
> Peter
>
>
> Giovanni Maruzzelli schrieb:
>> with three instances you will assign the hw:0 device to skype client
>> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
>> Must work. Pay attention to assign the same device name to all devices
>> needed by a skype instance (sound devices window): playback, capture
>> AND ring.
>>
>> Or maybe is a bug of ALSA on Debian...
>>
>> -giovanni
>>
>> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>
>>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>>> #2 to the Skype accounts. Still no sound.
>>> On the frist call there is one way audio, on the following calls there
>>> is no audio at all.
>>> This is weird.
>>>
>>> Best regards
>>> Peter
>>>
>>> Giovanni Maruzzelli schrieb:
>>>
>>>> Ciao Peter,
>>>>
>>>> Never tested on Debian 5.
>>>>
>>>> When you write "same problem" you are referring to the audio going one
>>>> way only (btw, which way?) with the custom audio driver?
>>>>
>>>> Have you tried with multiple instances of the regular Debian
>>>> snd-dummy, as I wrote in a mail before?
>>>>
>>>> -gm
>>>>
>>>>
>>>>
>>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>
>>>>
>>>>> Hello Giovanni,
>>>>>
>>>>> I did so but the same problem again.
>>>>>
>>>>> Did you ever test in on Debian 5.0?
>>>>>
>>>>> Best reards
>>>>> Peter
>>>>>
>>>>> Giovanni Maruzzelli schrieb:
>>>>>
>>>>>
>>>>>> good, so you have only one sound device, the right one.
>>>>>>
>>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>>
>>>>>> -gm
>>>>>>
>>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>
>>>>>>
>>>>>>
>>>>>>> I installed alsa-utile,
>>>>>>>
>>>>>>> now I get:
>>>>>>>
>>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>> ?Subdevices: 127/128
>>>>>>> ?Subdevice #0: subdevice #0
>>>>>>> ?Subdevice #1: subdevice #1
>>>>>>> ?Subdevice #2: subdevice #2
>>>>>>> ?Subdevice #3: subdevice #3
>>>>>>> ?Subdevice #4: subdevice #4
>>>>>>> ?Subdevice #5: subdevice #5
>>>>>>> ?Subdevice #6: subdevice #6
>>>>>>> ?Subdevice #7: subdevice #7
>>>>>>> ?Subdevice #8: subdevice #8
>>>>>>> ?Subdevice #9: subdevice #9
>>>>>>> ?Subdevice #10: subdevice #10
>>>>>>> ?Subdevice #11: subdevice #11
>>>>>>> ?Subdevice #12: subdevice #12
>>>>>>> ?Subdevice #13: subdevice #13
>>>>>>> ?Subdevice #14: subdevice #14
>>>>>>> ?Subdevice #15: subdevice #15
>>>>>>> ?Subdevice #16: subdevice #16
>>>>>>> ?Subdevice #17: subdevice #17
>>>>>>> ?Subdevice #18: subdevice #18
>>>>>>> ?Subdevice #19: subdevice #19
>>>>>>> ?Subdevice #20: subdevice #20
>>>>>>> ?Subdevice #21: subdevice #21
>>>>>>> ?Subdevice #22: subdevice #22
>>>>>>> ?Subdevice #23: subdevice #23
>>>>>>> ?Subdevice #24: subdevice #24
>>>>>>> ?Subdevice #25: subdevice #25
>>>>>>> ?Subdevice #26: subdevice #26
>>>>>>> ?Subdevice #27: subdevice #27
>>>>>>> ?Subdevice #28: subdevice #28
>>>>>>> ?Subdevice #29: subdevice #29
>>>>>>> ?Subdevice #30: subdevice #30
>>>>>>> ?Subdevice #31: subdevice #31
>>>>>>> ?Subdevice #32: subdevice #32
>>>>>>> ?Subdevice #33: subdevice #33
>>>>>>> ?Subdevice #34: subdevice #34
>>>>>>> ?Subdevice #35: subdevice #35
>>>>>>> ?Subdevice #36: subdevice #36
>>>>>>> ?Subdevice #37: subdevice #37
>>>>>>> ?Subdevice #38: subdevice #38
>>>>>>> ?Subdevice #39: subdevice #39
>>>>>>> ?Subdevice #40: subdevice #40
>>>>>>> ?Subdevice #41: subdevice #41
>>>>>>> ?Subdevice #42: subdevice #42
>>>>>>> ?Subdevice #43: subdevice #43
>>>>>>> ?Subdevice #44: subdevice #44
>>>>>>> ?Subdevice #45: subdevice #45
>>>>>>> ?Subdevice #46: subdevice #46
>>>>>>> ?Subdevice #47: subdevice #47
>>>>>>> ?Subdevice #48: subdevice #48
>>>>>>> ?Subdevice #49: subdevice #49
>>>>>>> ?Subdevice #50: subdevice #50
>>>>>>> ?Subdevice #51: subdevice #51
>>>>>>> ?Subdevice #52: subdevice #52
>>>>>>> ?Subdevice #53: subdevice #53
>>>>>>> ?Subdevice #54: subdevice #54
>>>>>>> ?Subdevice #55: subdevice #55
>>>>>>> ?Subdevice #56: subdevice #56
>>>>>>> ?Subdevice #57: subdevice #57
>>>>>>> ?Subdevice #58: subdevice #58
>>>>>>> ?Subdevice #59: subdevice #59
>>>>>>> ?Subdevice #60: subdevice #60
>>>>>>> ?Subdevice #61: subdevice #61
>>>>>>> ?Subdevice #62: subdevice #62
>>>>>>> ?Subdevice #63: subdevice #63
>>>>>>> ?Subdevice #64: subdevice #64
>>>>>>> ?Subdevice #65: subdevice #65
>>>>>>> ?Subdevice #66: subdevice #66
>>>>>>> ?Subdevice #67: subdevice #67
>>>>>>> ?Subdevice #68: subdevice #68
>>>>>>> ?Subdevice #69: subdevice #69
>>>>>>> ?Subdevice #70: subdevice #70
>>>>>>> ?Subdevice #71: subdevice #71
>>>>>>> ?Subdevice #72: subdevice #72
>>>>>>> ?Subdevice #73: subdevice #73
>>>>>>> ?Subdevice #74: subdevice #74
>>>>>>> ?Subdevice #75: subdevice #75
>>>>>>> ?Subdevice #76: subdevice #76
>>>>>>> ?Subdevice #77: subdevice #77
>>>>>>> ?Subdevice #78: subdevice #78
>>>>>>> ?Subdevice #79: subdevice #79
>>>>>>> ?Subdevice #80: subdevice #80
>>>>>>> ?Subdevice #81: subdevice #81
>>>>>>> ?Subdevice #82: subdevice #82
>>>>>>> ?Subdevice #83: subdevice #83
>>>>>>> ?Subdevice #84: subdevice #84
>>>>>>> ?Subdevice #85: subdevice #85
>>>>>>> ?Subdevice #86: subdevice #86
>>>>>>> ?Subdevice #87: subdevice #87
>>>>>>> ?Subdevice #88: subdevice #88
>>>>>>> ?Subdevice #89: subdevice #89
>>>>>>> ?Subdevice #90: subdevice #90
>>>>>>> ?Subdevice #91: subdevice #91
>>>>>>> ?Subdevice #92: subdevice #92
>>>>>>> ?Subdevice #93: subdevice #93
>>>>>>> ?Subdevice #94: subdevice #94
>>>>>>> ?Subdevice #95: subdevice #95
>>>>>>> ?Subdevice #96: subdevice #96
>>>>>>> ?Subdevice #97: subdevice #97
>>>>>>> ?Subdevice #98: subdevice #98
>>>>>>> ?Subdevice #99: subdevice #99
>>>>>>> ?Subdevice #100: subdevice #100
>>>>>>> ?Subdevice #101: subdevice #101
>>>>>>> ?Subdevice #102: subdevice #102
>>>>>>> ?Subdevice #103: subdevice #103
>>>>>>> ?Subdevice #104: subdevice #104
>>>>>>> ?Subdevice #105: subdevice #105
>>>>>>> ?Subdevice #106: subdevice #106
>>>>>>> ?Subdevice #107: subdevice #107
>>>>>>> ?Subdevice #108: subdevice #108
>>>>>>> ?Subdevice #109: subdevice #109
>>>>>>> ?Subdevice #110: subdevice #110
>>>>>>> ?Subdevice #111: subdevice #111
>>>>>>> ?Subdevice #112: subdevice #112
>>>>>>> ?Subdevice #113: subdevice #113
>>>>>>> ?Subdevice #114: subdevice #114
>>>>>>> ?Subdevice #115: subdevice #115
>>>>>>> ?Subdevice #116: subdevice #116
>>>>>>> ?Subdevice #117: subdevice #117
>>>>>>> ?Subdevice #118: subdevice #118
>>>>>>> ?Subdevice #119: subdevice #119
>>>>>>> ?Subdevice #120: subdevice #120
>>>>>>> ?Subdevice #121: subdevice #121
>>>>>>> ?Subdevice #122: subdevice #122
>>>>>>> ?Subdevice #123: subdevice #123
>>>>>>> ?Subdevice #124: subdevice #124
>>>>>>> ?Subdevice #125: subdevice #125
>>>>>>> ?Subdevice #126: subdevice #126
>>>>>>> ?Subdevice #127: subdevice #127
>>>>>>>
>>>>>>>
>>>>>>> Peter P GMX schrieb:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> Her's the output:
>>>>>>>>
>>>>>>>> skype:~# aplay -l
>>>>>>>> bash: aplay: command not found
>>>>>>>>
>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>>> what's the output of:
>>>>>>>>>
>>>>>>>>> aplay -l
>>>>>>>>>
>>>>>>>>> ?
>>>>>>>>>
>>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>>
>>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>> Thanks Giovanni,
>>>>>>>>>>
>>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>>
>>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>>
>>>>>>>>>> Best regards
>>>>>>>>>> Peter
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> This warning is harmless:
>>>>>>>>>>>
>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>
>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>>
>>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>>
>>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>>> 8.04).
>>>>>>>>>>>>
>>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>>
>>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>>
>>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>>
>>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>>> soundcard.
>>>>>>>>>>>>
>>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>>
>>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>>
>>>>>>>>>>>> ciao for now,
>>>>>>>>>>>>
>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>>
>>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>>> instructions in
>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>>
>>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>>
>>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>>
>>>>>>>>>>>>> Second question:
>>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>>
>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>> --
>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>
>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com


From lakindia89 at gmail.com  Fri Jan 29 04:36:12 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 29 Jan 2010 18:06:12 +0530
Subject: [Freeswitch-users] Multiple DTMF on group_confirm_key
In-Reply-To: <191c3a031001280914l56a035s74fd6d4b00dd9b3d@mail.gmail.com>
References: <7d79b3931001280411u6262f627xca723c64de5e118d@mail.gmail.com>
	<191c3a031001280914l56a035s74fd6d4b00dd9b3d@mail.gmail.com>
Message-ID: <7d79b3931001290436j2061a8dcg5cf7c5144103eb00@mail.gmail.com>

I tested by executing a script. It works great. But a small doubt.
Assume that I made a parallel dial using bridge application.
Normally, when one party answer the call, other party end will be hanged up.

But if I use group_confirm_key=exec and group_confirm_file=perl script.pl,
both the end can answer, and call bridged with the person who finished the
script first.

What I've to do if I need to execute the script only for the person who
answer's the call first?


On Thu, Jan 28, 2010 at 10:44 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you have to use a script (See the wiki for executing a script)
> then you can read in as many digits as you want and do what you need.
>
>
> On Thu, Jan 28, 2010 at 6:11 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Hi all,
>>
>> I've experimented with group confirm key and group confirm file. It works
>> great. However, I was unable to give multiple DTMF digits to get the
>> confirmation.
>>
>> I've set group_confirm_key=1234, I thought it will ask the 4 digits from
>> the user. But it simply taken 1 and when the user presses 1, the call got
>> bridged.
>>
>> Is there any way to specify multiple dtmf to be confirmed??
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at celliax.org  Fri Jan 29 05:13:00 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 29 Jan 2010 14:13:00 +0100
Subject: [Freeswitch-users] Set an gateway parameter for external
	gateway (skype)
In-Reply-To: <4B62C735.4010309@gmx.net>
References: <4B62078B.5060406@gmx.net>
	<8213d6071001281521u27347956nf2c89c3cd2e72745@mail.gmail.com> 
	<b8ad2a5b1001281529n4f5ac2acy266ad324b0da9318@mail.gmail.com> 
	<23f91031001281729s403a9954nc7400d73fa69f6ea@mail.gmail.com> 
	<4B62C735.4010309@gmx.net>
Message-ID: <7b197bef1001290513i14d3530avfb7298f5f97582f7@mail.gmail.com>

I would use SIP for intra-FS communications (as anthm rightly told me
when I was implementing a remote-protocol-kludge to have the skype
clients running in a machine different from the primary FS server -
that's probably what you want to obtain - not having skype clients
running in the primary FS server)

so FS1 <-> SIP <-> FS2 <-> skype

you can make good use of an URI like "skype/skypeusername" as detailed
in the wiki page
http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#Dialplan

So, you can have something like FS1 calling sip:skype/skypeusername at
FS2 , not sure about syntax tough.

-gm

On Fri, Jan 29, 2010 at 12:32 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello Sven
>
> I think that's a good idea. I just tried: "+" is not valid in a Skype
> username. So I may use it.
>
> I will try that and give some feedback.
>
> Best regards
> Peter
>
>
> Seven Du schrieb:
>> why not just use dialplan matching?
>>
>> assume FS1 has gateway named skype,
>>
>> originate sofia/gateway/skype/<your_dedicated_skype_name>+<other_skype_name_you_want_to_dial
>>
>>
>> on your FS/skype server, set dialplan:
>>
>>
>> $destination_number to match ? ?(.*)\+(.*), then you can
>>
>> bridge skypiax/$1/$2
>>
>> 2010/1/29 Wasim Baig <wasim at convergence.pk>:
>>
>>> Setting a sip header is a more elegant way ...
>>>
>>> http://wiki.freeswitch.org/wiki/Sofia#Adding_Request_Headers
>>>
>>> -wasim
>>>
>>> On Fri, Jan 29, 2010 at 4:21 AM, Ghulam Mustafa <mustafa.pk at gmail.com>
>>> wrote:
>>>
>>>> i don't know if it's a wise solution, but you can make a db insert
>>>> before bridge on freeswitch server, kypiax server can query db after
>>>> answer to get an idea!
>>>>
>>>> On Fri, Jan 29, 2010 at 2:54 AM, Peter P GMX <Prometheus001 at gmx.net>
>>>> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> I have a main freeswitch server and a separate Freeswitch/Skype server
>>>>> with mod_skypiax.
>>>>>
>>>>> I want main freeswitch server to tell the Freeswitch/Skype server to use
>>>>> a dedicated Skype interface(interface1, interface2 etc). What is the
>>>>> best way to pass this variable?
>>>>> Some ideas from my side
>>>>>
>>>>> ? ?* set caller_id_number or caller_id_number_name, as these are
>>>>> ? ? ?overwritten by Skype anyway. But this is not a clean solution
>>>>> ? ?* use another UDP port for each gateway, but this is a huge effort
>>>>> ? ? ?if the number of gateways becomes larger
>>>>> ? ?* set a dedicated sip header for this. Reading sip-header is easy
>>>>> ? ? ?(documented) but how to set my own sip header entry in FS?
>>>>> ? ?* any other idea?
>>>>>
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> --
>>>> Ghulam Mustafa
>>>> cell: +92 333.611.7681
>>>> sip: cyrenity at ekiga.net
>>>> mail: mustafa.pk at gmail.com
>>>> web: cyrenity.wordpress.com
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> --
>>> wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
>>> peace be upon you ...
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Fri Jan 29 05:20:38 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 29 Jan 2010 14:20:38 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>
References: <4B60555B.2020004@gmx.net> <4B607944.4040700@gmx.net> 
	<4B608014.4030902@gmx.net>
	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com> 
	<4B61ECE0.10409@gmx.net>
	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com> 
	<4B61FCAB.5040707@gmx.net>
	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com> 
	<4B62C962.7000601@gmx.net>
	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>
Message-ID: <7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>

Peter,

Can you connect on IRC (irc.freenode.net #freeswitch)? I'm gmaruzz there.

Can you restate your problems? I've lost connection :)

with snd-dummy custom you can create *one only* snd-dummy instance, so
*one only* fake soundcard. If you create more, will not work. But with
that one fake soundcard you can use 64 skype client instances, all
with the same soundcard hardware device (hw:n).

with original snd-dummy you can create a max of 8 instances, so 8 fake
soundcards, and with each fake soundcard you can  use a max of 8 skype
client instances.

use the hardware devices, not the default devices (use the "hw:n")

-giovanni

On Fri, Jan 29, 2010 at 1:03 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:
> did you enable debug mode while compiling custom snd-dummy? if ?yes
> try re-compiling with debug mode disabled.
>
> -m
>
> On Fri, Jan 29, 2010 at 4:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>> I now reinstalled the original sound drivers
>> Unfortunaltely the sound problems remain, not that worse but they are there:
>> Audio is still (almost) one way. Almost means:
>>
>> ? ?* SIP -> Skype ok
>> ? ?* Skype=> SIP I hear only some scratching on very loud audio
>>
>> Could it be a volume problem? But snd-dummy should have no volume
>> properties, right?
>>
>> Best regards
>> Peter
>>
>>
>> Giovanni Maruzzelli schrieb:
>>> with three instances you will assign the hw:0 device to skype client
>>> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
>>> Must work. Pay attention to assign the same device name to all devices
>>> needed by a skype instance (sound devices window): playback, capture
>>> AND ring.
>>>
>>> Or maybe is a bug of ALSA on Debian...
>>>
>>> -giovanni
>>>
>>> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>>>> #2 to the Skype accounts. Still no sound.
>>>> On the frist call there is one way audio, on the following calls there
>>>> is no audio at all.
>>>> This is weird.
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>> Giovanni Maruzzelli schrieb:
>>>>
>>>>> Ciao Peter,
>>>>>
>>>>> Never tested on Debian 5.
>>>>>
>>>>> When you write "same problem" you are referring to the audio going one
>>>>> way only (btw, which way?) with the custom audio driver?
>>>>>
>>>>> Have you tried with multiple instances of the regular Debian
>>>>> snd-dummy, as I wrote in a mail before?
>>>>>
>>>>> -gm
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>
>>>>>
>>>>>> Hello Giovanni,
>>>>>>
>>>>>> I did so but the same problem again.
>>>>>>
>>>>>> Did you ever test in on Debian 5.0?
>>>>>>
>>>>>> Best reards
>>>>>> Peter
>>>>>>
>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>
>>>>>>
>>>>>>> good, so you have only one sound device, the right one.
>>>>>>>
>>>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>>>
>>>>>>> -gm
>>>>>>>
>>>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> I installed alsa-utile,
>>>>>>>>
>>>>>>>> now I get:
>>>>>>>>
>>>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>>> ?Subdevices: 127/128
>>>>>>>> ?Subdevice #0: subdevice #0
>>>>>>>> ?Subdevice #1: subdevice #1
>>>>>>>> ?Subdevice #2: subdevice #2
>>>>>>>> ?Subdevice #3: subdevice #3
>>>>>>>> ?Subdevice #4: subdevice #4
>>>>>>>> ?Subdevice #5: subdevice #5
>>>>>>>> ?Subdevice #6: subdevice #6
>>>>>>>> ?Subdevice #7: subdevice #7
>>>>>>>> ?Subdevice #8: subdevice #8
>>>>>>>> ?Subdevice #9: subdevice #9
>>>>>>>> ?Subdevice #10: subdevice #10
>>>>>>>> ?Subdevice #11: subdevice #11
>>>>>>>> ?Subdevice #12: subdevice #12
>>>>>>>> ?Subdevice #13: subdevice #13
>>>>>>>> ?Subdevice #14: subdevice #14
>>>>>>>> ?Subdevice #15: subdevice #15
>>>>>>>> ?Subdevice #16: subdevice #16
>>>>>>>> ?Subdevice #17: subdevice #17
>>>>>>>> ?Subdevice #18: subdevice #18
>>>>>>>> ?Subdevice #19: subdevice #19
>>>>>>>> ?Subdevice #20: subdevice #20
>>>>>>>> ?Subdevice #21: subdevice #21
>>>>>>>> ?Subdevice #22: subdevice #22
>>>>>>>> ?Subdevice #23: subdevice #23
>>>>>>>> ?Subdevice #24: subdevice #24
>>>>>>>> ?Subdevice #25: subdevice #25
>>>>>>>> ?Subdevice #26: subdevice #26
>>>>>>>> ?Subdevice #27: subdevice #27
>>>>>>>> ?Subdevice #28: subdevice #28
>>>>>>>> ?Subdevice #29: subdevice #29
>>>>>>>> ?Subdevice #30: subdevice #30
>>>>>>>> ?Subdevice #31: subdevice #31
>>>>>>>> ?Subdevice #32: subdevice #32
>>>>>>>> ?Subdevice #33: subdevice #33
>>>>>>>> ?Subdevice #34: subdevice #34
>>>>>>>> ?Subdevice #35: subdevice #35
>>>>>>>> ?Subdevice #36: subdevice #36
>>>>>>>> ?Subdevice #37: subdevice #37
>>>>>>>> ?Subdevice #38: subdevice #38
>>>>>>>> ?Subdevice #39: subdevice #39
>>>>>>>> ?Subdevice #40: subdevice #40
>>>>>>>> ?Subdevice #41: subdevice #41
>>>>>>>> ?Subdevice #42: subdevice #42
>>>>>>>> ?Subdevice #43: subdevice #43
>>>>>>>> ?Subdevice #44: subdevice #44
>>>>>>>> ?Subdevice #45: subdevice #45
>>>>>>>> ?Subdevice #46: subdevice #46
>>>>>>>> ?Subdevice #47: subdevice #47
>>>>>>>> ?Subdevice #48: subdevice #48
>>>>>>>> ?Subdevice #49: subdevice #49
>>>>>>>> ?Subdevice #50: subdevice #50
>>>>>>>> ?Subdevice #51: subdevice #51
>>>>>>>> ?Subdevice #52: subdevice #52
>>>>>>>> ?Subdevice #53: subdevice #53
>>>>>>>> ?Subdevice #54: subdevice #54
>>>>>>>> ?Subdevice #55: subdevice #55
>>>>>>>> ?Subdevice #56: subdevice #56
>>>>>>>> ?Subdevice #57: subdevice #57
>>>>>>>> ?Subdevice #58: subdevice #58
>>>>>>>> ?Subdevice #59: subdevice #59
>>>>>>>> ?Subdevice #60: subdevice #60
>>>>>>>> ?Subdevice #61: subdevice #61
>>>>>>>> ?Subdevice #62: subdevice #62
>>>>>>>> ?Subdevice #63: subdevice #63
>>>>>>>> ?Subdevice #64: subdevice #64
>>>>>>>> ?Subdevice #65: subdevice #65
>>>>>>>> ?Subdevice #66: subdevice #66
>>>>>>>> ?Subdevice #67: subdevice #67
>>>>>>>> ?Subdevice #68: subdevice #68
>>>>>>>> ?Subdevice #69: subdevice #69
>>>>>>>> ?Subdevice #70: subdevice #70
>>>>>>>> ?Subdevice #71: subdevice #71
>>>>>>>> ?Subdevice #72: subdevice #72
>>>>>>>> ?Subdevice #73: subdevice #73
>>>>>>>> ?Subdevice #74: subdevice #74
>>>>>>>> ?Subdevice #75: subdevice #75
>>>>>>>> ?Subdevice #76: subdevice #76
>>>>>>>> ?Subdevice #77: subdevice #77
>>>>>>>> ?Subdevice #78: subdevice #78
>>>>>>>> ?Subdevice #79: subdevice #79
>>>>>>>> ?Subdevice #80: subdevice #80
>>>>>>>> ?Subdevice #81: subdevice #81
>>>>>>>> ?Subdevice #82: subdevice #82
>>>>>>>> ?Subdevice #83: subdevice #83
>>>>>>>> ?Subdevice #84: subdevice #84
>>>>>>>> ?Subdevice #85: subdevice #85
>>>>>>>> ?Subdevice #86: subdevice #86
>>>>>>>> ?Subdevice #87: subdevice #87
>>>>>>>> ?Subdevice #88: subdevice #88
>>>>>>>> ?Subdevice #89: subdevice #89
>>>>>>>> ?Subdevice #90: subdevice #90
>>>>>>>> ?Subdevice #91: subdevice #91
>>>>>>>> ?Subdevice #92: subdevice #92
>>>>>>>> ?Subdevice #93: subdevice #93
>>>>>>>> ?Subdevice #94: subdevice #94
>>>>>>>> ?Subdevice #95: subdevice #95
>>>>>>>> ?Subdevice #96: subdevice #96
>>>>>>>> ?Subdevice #97: subdevice #97
>>>>>>>> ?Subdevice #98: subdevice #98
>>>>>>>> ?Subdevice #99: subdevice #99
>>>>>>>> ?Subdevice #100: subdevice #100
>>>>>>>> ?Subdevice #101: subdevice #101
>>>>>>>> ?Subdevice #102: subdevice #102
>>>>>>>> ?Subdevice #103: subdevice #103
>>>>>>>> ?Subdevice #104: subdevice #104
>>>>>>>> ?Subdevice #105: subdevice #105
>>>>>>>> ?Subdevice #106: subdevice #106
>>>>>>>> ?Subdevice #107: subdevice #107
>>>>>>>> ?Subdevice #108: subdevice #108
>>>>>>>> ?Subdevice #109: subdevice #109
>>>>>>>> ?Subdevice #110: subdevice #110
>>>>>>>> ?Subdevice #111: subdevice #111
>>>>>>>> ?Subdevice #112: subdevice #112
>>>>>>>> ?Subdevice #113: subdevice #113
>>>>>>>> ?Subdevice #114: subdevice #114
>>>>>>>> ?Subdevice #115: subdevice #115
>>>>>>>> ?Subdevice #116: subdevice #116
>>>>>>>> ?Subdevice #117: subdevice #117
>>>>>>>> ?Subdevice #118: subdevice #118
>>>>>>>> ?Subdevice #119: subdevice #119
>>>>>>>> ?Subdevice #120: subdevice #120
>>>>>>>> ?Subdevice #121: subdevice #121
>>>>>>>> ?Subdevice #122: subdevice #122
>>>>>>>> ?Subdevice #123: subdevice #123
>>>>>>>> ?Subdevice #124: subdevice #124
>>>>>>>> ?Subdevice #125: subdevice #125
>>>>>>>> ?Subdevice #126: subdevice #126
>>>>>>>> ?Subdevice #127: subdevice #127
>>>>>>>>
>>>>>>>>
>>>>>>>> Peter P GMX schrieb:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>> Her's the output:
>>>>>>>>>
>>>>>>>>> skype:~# aplay -l
>>>>>>>>> bash: aplay: command not found
>>>>>>>>>
>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>>>> what's the output of:
>>>>>>>>>>
>>>>>>>>>> aplay -l
>>>>>>>>>>
>>>>>>>>>> ?
>>>>>>>>>>
>>>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>>>
>>>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> Thanks Giovanni,
>>>>>>>>>>>
>>>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>>>
>>>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>>>
>>>>>>>>>>> Best regards
>>>>>>>>>>> Peter
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>> This warning is harmless:
>>>>>>>>>>>>
>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>
>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>>>
>>>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>>>
>>>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>>>> 8.04).
>>>>>>>>>>>>>
>>>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>>>
>>>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>>>
>>>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>>>
>>>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>>>> soundcard.
>>>>>>>>>>>>>
>>>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>>>
>>>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>>>
>>>>>>>>>>>>> ciao for now,
>>>>>>>>>>>>>
>>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>>>> instructions in
>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Second question:
>>>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>> --
>>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>>
>>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From q.edward at gmail.com  Fri Jan 29 06:04:23 2010
From: q.edward at gmail.com (Edward Q.)
Date: Fri, 29 Jan 2010 09:04:23 -0500
Subject: [Freeswitch-users] Blind dialed number
Message-ID: <89313a91001290604q500c114cs31ce39f5a4d65922@mail.gmail.com>

Hi guys.
Sorry for been a newbie on all this. And thank you for your help in
advanced.
Here is the scenario I am trying to accomplish.
I would like to have the phone number inside a database not in the
extension.xml file.
Here is what I am trying to do.
Create a let's say 1001.xml file ( if needed )  but that file instead of
having the phone number to be dialed,  go to MySQL and look for the phone
number there. Or when freeswitch is told to call that extension instead of
looking for the 1001.xml file go directly to mysql to look for the phone
number associated to that extension to dial it out.
That way the phone numbers are always kept private at all times.
What we are doing is.
>From the webserver customer clicks on a link -> link dials the phone number
in the xml file automatically but I would like to keep the phone number on
the DB server not in the xml file.
Can anyone point me please in the right direction?
Really appreciated.
Ed
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From brian at freeswitch.org  Fri Jan 29 06:26:53 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 29 Jan 2010 08:26:53 -0600
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
In-Reply-To: <eui5m5hg6tqemri4r5u250hu72mjddmjnt@4ax.com>
References: <27338355.post@talk.nabble.com>
	<27338355.post-WJuSqJV8a7jJsTRiRinrng@public.gmane.org>
	<201001270750.16772.sos@sokhapkin.dyndns.org>
	<eui5m5hg6tqemri4r5u250hu72mjddmjnt@4ax.com>
Message-ID: <57E0B3FB-166A-4CF2-82E8-267D2FADB9DC@freeswitch.org>

Well if nat is involved you might have issues.

/b

On Jan 29, 2010, at 5:52 AM, Fred-145 wrote:

> Are there drawbacks to having RTP pakets flow directly between the SIP
> end-points?



From jmesquita at freeswitch.org  Fri Jan 29 06:33:54 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 29 Jan 2010 12:33:54 -0200
Subject: [Freeswitch-users] Blind dialed number
In-Reply-To: <89313a91001290604q500c114cs31ce39f5a4d65922@mail.gmail.com>
References: <89313a91001290604q500c114cs31ce39f5a4d65922@mail.gmail.com>
Message-ID: <ebbc2d551001290633t57fae340v80a9a82153f28c64@mail.gmail.com>

A couple of options there.

ESL, mod_xml_curl or even mod_python, mod_spidermonkey, lua, etc...

Pick your flavor.

Jo?o Mesquita


On Fri, Jan 29, 2010 at 12:04 PM, Edward Q. <q.edward at gmail.com> wrote:

> Hi guys.
> Sorry for been a newbie on all this. And thank you for your help in
> advanced.
> Here is the scenario I am trying to accomplish.
> I would like to have the phone number inside a database not in the
> extension.xml file.
> Here is what I am trying to do.
> Create a let's say 1001.xml file ( if needed )  but that file instead of
> having the phone number to be dialed,  go to MySQL and look for the phone
> number there. Or when freeswitch is told to call that extension instead of
> looking for the 1001.xml file go directly to mysql to look for the phone
> number associated to that extension to dial it out.
> That way the phone numbers are always kept private at all times.
> What we are doing is.
> From the webserver customer clicks on a link -> link dials the phone number
> in the xml file automatically but I would like to keep the phone number on
> the DB server not in the xml file.
> Can anyone point me please in the right direction?
> Really appreciated.
> Ed
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rob4manhere at gmail.com  Fri Jan 29 06:48:00 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 29 Jan 2010 08:48:00 -0600
Subject: [Freeswitch-users] Blind dialed number
In-Reply-To: <ebbc2d551001290633t57fae340v80a9a82153f28c64@mail.gmail.com>
References: <89313a91001290604q500c114cs31ce39f5a4d65922@mail.gmail.com>
	<ebbc2d551001290633t57fae340v80a9a82153f28c64@mail.gmail.com>
Message-ID: <81B863DC-CF42-4ADE-8F82-EC62B51B03F3@gmail.com>

Agreed- you have lots of options.

A side note though- if your requirement is that the number is kept  
private from freeswitch, you're going to run into issues.  Freeswitch  
will eventually have to know the number in order to call it, and it  
will log its activity... so if there are people who aren't supposed to  
see the number but can see 1001.xml, they could also see the  
freeswitch logs.  They could also open the fs_cli and see activity  
there.  I guess you could lock everything like that down but just  
things to think through if that kind of compartmentalization is a big  
requirement.

Rob

On Jan 29, 2010, at 8:33 AM, Jo?o Mesquita wrote:

> A couple of options there.
>
> ESL, mod_xml_curl or even mod_python, mod_spidermonkey, lua, etc...
>
> Pick your flavor.
>
> Jo?o Mesquita
>
>
> On Fri, Jan 29, 2010 at 12:04 PM, Edward Q. <q.edward at gmail.com>  
> wrote:
> Hi guys.
> Sorry for been a newbie on all this. And thank you for your help in  
> advanced.
> Here is the scenario I am trying to accomplish.
> I would like to have the phone number inside a database not in the  
> extension.xml file.
> Here is what I am trying to do.
> Create a let's say 1001.xml file ( if needed )  but that file  
> instead of having the phone number to be dialed,  go to MySQL and  
> look for the phone number there. Or when freeswitch is told to call  
> that extension instead of looking for the 1001.xml file go directly  
> to mysql to look for the phone number associated to that extension  
> to dial it out.
> That way the phone numbers are always kept private at all times.
> What we are doing is.
> From the webserver customer clicks on a link -> link dials the phone  
> number in the xml file automatically but I would like to keep the  
> phone number on the DB server not in the xml file.
> Can anyone point me please in the right direction?
> Really appreciated.
> Ed
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Fri Jan 29 06:53:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jan 2010 06:53:50 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Agenda
Message-ID: <87f2f3b91001290653x417398aaj8900a4da9b1aa83b@mail.gmail.com>

Greetings,

This week's conference call agenda is posted here:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_29

Please add your items as the agenda is very light this week. We do have a
few things to discuss, though, so please hop on and bring a friend!

Talk to you all soon,
Michael
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From tim at communicatefreely.net  Fri Jan 29 06:59:53 2010
From: tim at communicatefreely.net (Tim St. Pierre)
Date: Fri, 29 Jan 2010 09:59:53 -0500
Subject: [Freeswitch-users] Blind dialed number
In-Reply-To: <89313a91001290604q500c114cs31ce39f5a4d65922@mail.gmail.com>
References: <89313a91001290604q500c114cs31ce39f5a4d65922@mail.gmail.com>
Message-ID: <4B62F7E9.6000202@communicatefreely.net>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Edward,

I am building a dialplan that runs almost everything out of a database, and what seemed easiest was
to use mod_xml_curl and a php script (substitute your favorite language here).

When a call comes in, FS makes a query to the web server.  A PHP script looks at the posted values,
and if it matches certain criteria (I usually match on context first, before I do anything.), a
small piece of XML dialplan is returned.

Usually what I do is return an extension with a pattern match that will always match what is dialed,
along with the actions I want FS to take.  In this way, the PHP script is handling all the decision
making - FS only posts the call data, and the exact action to take is returned.  I'm not using the
XML files for dialplan at all - it is rendered on the fly by PHP.

The other nice thing about this is that it scales very well.  The web server, database, and FS
components can all be on separate boxes if it's a very high volume environment.  mod_xml_curl can be
set up with multiple failover options.

If you want to mix and match text vs. curl XML dialplan, just have some logic in your script that
returns only the root tags if the call doesn't match one of the database driven extensions.  FS will
include the returned result in with the XML dialplan, so the call can match either.

Good luck!

- -Tim

Edward Q. wrote:
> Hi guys.
> Sorry for been a newbie on all this. And thank you for your help in
> advanced.
> Here is the scenario I am trying to accomplish.
> I would like to have the phone number inside a database not in the
> extension.xml file.
> Here is what I am trying to do.
> Create a let's say 1001.xml file ( if needed )  but that file instead of
> having the phone number to be dialed,  go to MySQL and look for the
> phone number there. Or when freeswitch is told to call that extension
> instead of looking for the 1001.xml file go directly to mysql to look
> for the phone number associated to that extension to dial it out.
> That way the phone numbers are always kept private at all times.
> What we are doing is.
> From the webserver customer clicks on a link -> link dials the phone
> number in the xml file automatically but I would like to keep the phone
> number on the DB server not in the xml file.
> Can anyone point me please in the right direction?
> Really appreciated.
> Ed
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

- --
Tim St. Pierre
IP Voice technician
Communicate Freely
1-877-291-8647 x5101
sip:5101 at communicatefreely.net
tim at communicatefreely.net
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From msc at freeswitch.org  Fri Jan 29 07:11:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jan 2010 07:11:07 -0800
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E90BF91C@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E90BD7F4@prato.activenetwerx.local>
	<20100127210836.5ED5D2E3B2D@cuneorg-email.cune.pri>
	<CA5A491E9DEFBE4CB777DE97E21575E90BF91C@prato.activenetwerx.local>
Message-ID: <87f2f3b91001290711t3ee71079wa0674941d8dbc741@mail.gmail.com>

Are you really dialing that phone number or is that just redacted? Try
dialing 919-386-9900 for testing.
-MC

On Wed, Jan 27, 2010 at 6:09 PM, Joseph L. Casale <jcasale at activenetwerx.com
> wrote:

> >Debian 5.0.3
>
> Well, given the time I had tonight, I tried on my CentOS 5.3 box.
> The incoming log is the first block, and an outgoing log is the
> second block at http://pastebin.freeswitch.org/11965
>
> When I call in, I can hear it get answered, as I play the wav file
> I hear the tone go very low, but no sound.
>
> When I try to call out, nothing happens?
>
> Is there anything in the log that might standout from your perspective?
>
> Thanks everyone!
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.villasmil.work at gmail.com  Fri Jan 29 07:22:37 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 29 Jan 2010 16:22:37 +0100
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <5d2828f1001281205u2eccc999re6f25a55b07156c7@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
	<5d2828f1001281205u2eccc999re6f25a55b07156c7@mail.gmail.com>
Message-ID: <9853f4ff1001290722m4007e620sf6db3c61c1495a4b@mail.gmail.com>

Oh yes, here's an example:

function dbConnect()
     -- connect to db
     require "luasql.mysql"
     env = assert(luasql.mysql())
     conn = assert(env:connect("freeswitch","user","userpass","localhost"))
end

function getpin()
     session:streamFile(card_greeting_audio_file)

     card_pin = session:getDigits(4, "#", 3000);

     if card_pin > "" then
          freeswitch.consoleLog("info", "CARD INFO: PIN...........:
".. card_pin .."\n");
          cur = assert(
               conn:execute( "select * from cards_table where pin ="..
card_pin ..";" )
               )

          -- print all rows, the rows will be indexed by field names
          row = cur:fetch ({}, "a")
          fsLog("ROWS: ".. cur:numrows() )
          if cur:numrows() > 0 then pinok=true end
          while row do

               fsLog("CARD INFO: Batch.........: ".. row.batch        )
               fsLog("CARD INFO: Card Name.....: ".. row.card_name    )
               fsLog("CARD INFO: Ratetable.....: ".. row.ratetable    )
               fsLog("CARD INFO: Initial Bal...: ".. row.init_bal     )
               fsLog("CARD INFO: Curr Balance..: ".. row.balance      )

               batch, ratetable, init_bal, balance = row.batch,
row.ratetable, row.init_bal, row.balance

               SetVar("card_pin",card_pin)
               SetVar("card_batch", batch)
               SetVar("card_ratetable", ratetable)
               SetVar("card_init_bal", init_bal)
               SetVar("card_balance", balance)

            -- reusing the table of results
            row = cur:fetch (row, "a")
          end
     else
          pinok=false
          session:streamFile(card_invalid_pin_audio_file)
     end
end


On Thu, Jan 28, 2010 at 9:05 PM, Mike van Lammeren
<mike at van.lammeren.net> wrote:
> And you can make queries against your MySQL database, and get results, etc.?
>
> On Thu, Jan 28, 2010 at 2:43 AM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> Hello,
>>
>> That works fine:
>>
>> box:~# lua testdb.lua
>> box:~#
>>
>>
>> David
>>
>> On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
>> <mike at van.lammeren.net> wrote:
>> > Have you tried running a Lua script that includes the library from
>> > outside
>> > of FreeSWITCH? What does that do?
>> >
>> > On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com>
>> > wrote:
>> >>
>> >> I tried running ldconfig on the directory containing mysql.so, but it
>> >> did
>> >> not help.
>> >> So it sounds like there could be a bug in the latter versions?
>> >>
>> >> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
>> >> <david.villasmil.work at gmail.com> wrote:
>> >>>
>> >>> I got the same error, my script was working with no problems before an
>> >>> update to trunk.
>> >>>
>> >>> David
>> >>>
>> >>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com>
>> >>> wrote:
>> >>> > Hi, I followed the instructions in the Lua documentation for setting
>> >>> > up
>> >>> > luasql, but when I try to run my script I get:
>> >>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading
>> >>> > module
>> >>> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
>> >>> > ?? ? ? ?/usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore
>> >>> > segment
>> >>> > prot
>> >>> > after reloc: Permission denied
>> >>> > stack traceback:
>> >>> > ?? ? ? ?[C]: ?
>> >>> > ?? ? ? ?[C]: in function 'require'
>> >>> > ?? ? ? ?/usr/local/freeswitch/scripts/l.lua:2: in main chunk
>> >>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
>> >>> > I changed the permissions for mysql.so and for my script to 777, so
>> >>> > I'm
>> >>> > not
>> >>> > sure where the permission problem could be.
>> >>> > I'd appreciate any suggestions.
>> >>> > Thanks,
>> >>> > Adam
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> >
>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Fri Jan 29 07:25:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jan 2010 07:25:53 -0800
Subject: [Freeswitch-users] wiki password recovery - no mail is send
In-Reply-To: <4B629D05.1060908@xpirio.com>
References: <4B629D05.1060908@xpirio.com>
Message-ID: <87f2f3b91001290725o13dde05fq9426d8ef45e46035@mail.gmail.com>

Email me off list and I'll help you with this. I've never been able to
reproduce this symptom but some others have. In any case I will assist you
with getting your wiki account updated.
-MC

2010/1/29 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> hello
>
> the password recovery for the fs wiki doesn't seem to work.
> no e-mail is send when entering the username and press "e-mail new
> password".
>
> may i assist here, we do maintain a few wikis for ourself.
>
> br
>
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Jan 29 07:30:55 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jan 2010 07:30:55 -0800
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <9853f4ff1001290722m4007e620sf6db3c61c1495a4b@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
	<5d2828f1001281205u2eccc999re6f25a55b07156c7@mail.gmail.com>
	<9853f4ff1001290722m4007e620sf6db3c61c1495a4b@mail.gmail.com>
Message-ID: <87f2f3b91001290730y652086far4008036c47beed89@mail.gmail.com>

David,

Are you using Lua and lusql for some exotic call handling scenarios? If so,
would you mind posting some examples to the wiki and then linking here?
Also, if you can join the community conference call today that would be
great!

Thanks,
MC

On Fri, Jan 29, 2010 at 7:22 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Oh yes, here's an example:
>
> function dbConnect()
>     -- connect to db
>     require "luasql.mysql"
>     env = assert(luasql.mysql())
>     conn = assert(env:connect("freeswitch","user","userpass","localhost"))
> end
>
> function getpin()
>     session:streamFile(card_greeting_audio_file)
>
>     card_pin = session:getDigits(4, "#", 3000);
>
>     if card_pin > "" then
>          freeswitch.consoleLog("info", "CARD INFO: PIN...........:
> ".. card_pin .."\n");
>          cur = assert(
>               conn:execute( "select * from cards_table where pin ="..
> card_pin ..";" )
>               )
>
>          -- print all rows, the rows will be indexed by field names
>          row = cur:fetch ({}, "a")
>          fsLog("ROWS: ".. cur:numrows() )
>          if cur:numrows() > 0 then pinok=true end
>          while row do
>
>               fsLog("CARD INFO: Batch.........: ".. row.batch        )
>               fsLog("CARD INFO: Card Name.....: ".. row.card_name    )
>               fsLog("CARD INFO: Ratetable.....: ".. row.ratetable    )
>               fsLog("CARD INFO: Initial Bal...: ".. row.init_bal     )
>               fsLog("CARD INFO: Curr Balance..: ".. row.balance      )
>
>               batch, ratetable, init_bal, balance = row.batch,
> row.ratetable, row.init_bal, row.balance
>
>               SetVar("card_pin",card_pin)
>               SetVar("card_batch", batch)
>               SetVar("card_ratetable", ratetable)
>               SetVar("card_init_bal", init_bal)
>               SetVar("card_balance", balance)
>
>            -- reusing the table of results
>            row = cur:fetch (row, "a")
>          end
>     else
>          pinok=false
>          session:streamFile(card_invalid_pin_audio_file)
>     end
> end
>
>
> On Thu, Jan 28, 2010 at 9:05 PM, Mike van Lammeren
> <mike at van.lammeren.net> wrote:
> > And you can make queries against your MySQL database, and get results,
> etc.?
> >
> > On Thu, Jan 28, 2010 at 2:43 AM, David Villasmil
> > <david.villasmil.work at gmail.com> wrote:
> >>
> >> Hello,
> >>
> >> That works fine:
> >>
> >> box:~# lua testdb.lua
> >> box:~#
> >>
> >>
> >> David
> >>
> >> On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
> >> <mike at van.lammeren.net> wrote:
> >> > Have you tried running a Lua script that includes the library from
> >> > outside
> >> > of FreeSWITCH? What does that do?
> >> >
> >> > On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com>
> >> > wrote:
> >> >>
> >> >> I tried running ldconfig on the directory containing mysql.so, but it
> >> >> did
> >> >> not help.
> >> >> So it sounds like there could be a bug in the latter versions?
> >> >>
> >> >> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
> >> >> <david.villasmil.work at gmail.com> wrote:
> >> >>>
> >> >>> I got the same error, my script was working with no problems before
> an
> >> >>> update to trunk.
> >> >>>
> >> >>> David
> >> >>>
> >> >>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com>
> >> >>> wrote:
> >> >>> > Hi, I followed the instructions in the Lua documentation for
> setting
> >> >>> > up
> >> >>> > luasql, but when I try to run my script I get:
> >> >>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading
> >> >>> > module
> >> >>> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
> >> >>> >         /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore
> >> >>> > segment
> >> >>> > prot
> >> >>> > after reloc: Permission denied
> >> >>> > stack traceback:
> >> >>> >         [C]: ?
> >> >>> >         [C]: in function 'require'
> >> >>> >         /usr/local/freeswitch/scripts/l.lua:2: in main chunk
> >> >>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
> >> >>> > I changed the permissions for mysql.so and for my script to 777,
> so
> >> >>> > I'm
> >> >>> > not
> >> >>> > sure where the permission problem could be.
> >> >>> > I'd appreciate any suggestions.
> >> >>> > Thanks,
> >> >>> > Adam
> >> >>> >
> >> >>> > _______________________________________________
> >> >>> > FreeSWITCH-users mailing list
> >> >>> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >
> >> >>> >
> >> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> > http://www.freeswitch.org
> >> >>> >
> >> >>> >
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.villasmil.work at gmail.com  Fri Jan 29 07:51:36 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 29 Jan 2010 16:51:36 +0100
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <87f2f3b91001290730y652086far4008036c47beed89@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
	<5d2828f1001281205u2eccc999re6f25a55b07156c7@mail.gmail.com>
	<9853f4ff1001290722m4007e620sf6db3c61c1495a4b@mail.gmail.com>
	<87f2f3b91001290730y652086far4008036c47beed89@mail.gmail.com>
Message-ID: <9853f4ff1001290751q29c12e58oc550bd37fbf225c9@mail.gmail.com>

Today what time/timezone?

On Fri, Jan 29, 2010 at 4:30 PM, Michael Collins <msc at freeswitch.org> wrote:
> David,
>
> Are you using Lua and lusql for some exotic call handling scenarios? If so,
> would you mind posting some examples to the wiki and then linking here?
> Also, if you can join the community conference call today that would be
> great!
>
> Thanks,
> MC
>
> On Fri, Jan 29, 2010 at 7:22 AM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> Oh yes, here's an example:
>>
>> function dbConnect()
>> ? ? -- connect to db
>> ? ? require "luasql.mysql"
>> ? ? env = assert(luasql.mysql())
>> ? ? conn = assert(env:connect("freeswitch","user","userpass","localhost"))
>> end
>>
>> function getpin()
>> ? ? session:streamFile(card_greeting_audio_file)
>>
>> ? ? card_pin = session:getDigits(4, "#", 3000);
>>
>> ? ? if card_pin > "" then
>> ? ? ? ? ?freeswitch.consoleLog("info", "CARD INFO: PIN...........:
>> ".. card_pin .."\n");
>> ? ? ? ? ?cur = assert(
>> ? ? ? ? ? ? ? conn:execute( "select * from cards_table where pin ="..
>> card_pin ..";" )
>> ? ? ? ? ? ? ? )
>>
>> ? ? ? ? ?-- print all rows, the rows will be indexed by field names
>> ? ? ? ? ?row = cur:fetch ({}, "a")
>> ? ? ? ? ?fsLog("ROWS: ".. cur:numrows() )
>> ? ? ? ? ?if cur:numrows() > 0 then pinok=true end
>> ? ? ? ? ?while row do
>>
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Batch.........: ".. row.batch ? ? ? ?)
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Card Name.....: ".. row.card_name ? ?)
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Ratetable.....: ".. row.ratetable ? ?)
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Initial Bal...: ".. row.init_bal ? ? )
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Curr Balance..: ".. row.balance ? ? ?)
>>
>> ? ? ? ? ? ? ? batch, ratetable, init_bal, balance = row.batch,
>> row.ratetable, row.init_bal, row.balance
>>
>> ? ? ? ? ? ? ? SetVar("card_pin",card_pin)
>> ? ? ? ? ? ? ? SetVar("card_batch", batch)
>> ? ? ? ? ? ? ? SetVar("card_ratetable", ratetable)
>> ? ? ? ? ? ? ? SetVar("card_init_bal", init_bal)
>> ? ? ? ? ? ? ? SetVar("card_balance", balance)
>>
>> ? ? ? ? ? ?-- reusing the table of results
>> ? ? ? ? ? ?row = cur:fetch (row, "a")
>> ? ? ? ? ?end
>> ? ? else
>> ? ? ? ? ?pinok=false
>> ? ? ? ? ?session:streamFile(card_invalid_pin_audio_file)
>> ? ? end
>> end
>>
>>
>> On Thu, Jan 28, 2010 at 9:05 PM, Mike van Lammeren
>> <mike at van.lammeren.net> wrote:
>> > And you can make queries against your MySQL database, and get results,
>> > etc.?
>> >
>> > On Thu, Jan 28, 2010 at 2:43 AM, David Villasmil
>> > <david.villasmil.work at gmail.com> wrote:
>> >>
>> >> Hello,
>> >>
>> >> That works fine:
>> >>
>> >> box:~# lua testdb.lua
>> >> box:~#
>> >>
>> >>
>> >> David
>> >>
>> >> On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
>> >> <mike at van.lammeren.net> wrote:
>> >> > Have you tried running a Lua script that includes the library from
>> >> > outside
>> >> > of FreeSWITCH? What does that do?
>> >> >
>> >> > On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> I tried running ldconfig on the directory containing mysql.so, but
>> >> >> it
>> >> >> did
>> >> >> not help.
>> >> >> So it sounds like there could be a bug in the latter versions?
>> >> >>
>> >> >> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
>> >> >> <david.villasmil.work at gmail.com> wrote:
>> >> >>>
>> >> >>> I got the same error, my script was working with no problems before
>> >> >>> an
>> >> >>> update to trunk.
>> >> >>>
>> >> >>> David
>> >> >>>
>> >> >>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com>
>> >> >>> wrote:
>> >> >>> > Hi, I followed the instructions in the Lua documentation for
>> >> >>> > setting
>> >> >>> > up
>> >> >>> > luasql, but when I try to run my script I get:
>> >> >>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading
>> >> >>> > module
>> >> >>> > 'luasql.mysql' from file
>> >> >>> > '/usr/local/lib/lua/5.1/luasql/mysql.so':
>> >> >>> > ?? ? ? ?/usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore
>> >> >>> > segment
>> >> >>> > prot
>> >> >>> > after reloc: Permission denied
>> >> >>> > stack traceback:
>> >> >>> > ?? ? ? ?[C]: ?
>> >> >>> > ?? ? ? ?[C]: in function 'require'
>> >> >>> > ?? ? ? ?/usr/local/freeswitch/scripts/l.lua:2: in main chunk
>> >> >>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
>> >> >>> > I changed the permissions for mysql.so and for my script to 777,
>> >> >>> > so
>> >> >>> > I'm
>> >> >>> > not
>> >> >>> > sure where the permission problem could be.
>> >> >>> > I'd appreciate any suggestions.
>> >> >>> > Thanks,
>> >> >>> > Adam
>> >> >>> >
>> >> >>> > _______________________________________________
>> >> >>> > FreeSWITCH-users mailing list
>> >> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>> >
>> >> >>> >
>> >> >>> >
>> >> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> > http://www.freeswitch.org
>> >> >>> >
>> >> >>> >
>> >> >>>
>> >> >>> _______________________________________________
>> >> >>> FreeSWITCH-users mailing list
>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>
>> >> >>>
>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> http://www.freeswitch.org
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From freeswitch-list at puzzled.xs4all.nl  Fri Jan 29 07:56:45 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick)
Date: Fri, 29 Jan 2010 16:56:45 +0100
Subject: [Freeswitch-users] wiki password recovery - no mail is send
In-Reply-To: <87f2f3b91001290725o13dde05fq9426d8ef45e46035@mail.gmail.com>
References: <4B629D05.1060908@xpirio.com>
	<87f2f3b91001290725o13dde05fq9426d8ef45e46035@mail.gmail.com>
Message-ID: <4B63053D.20805@puzzled.xs4all.nl>

On 01/29/2010 04:25 PM, Michael Collins wrote:
> Email me off list and I'll help you with this. I've never been able to
> reproduce this symptom but some others have. In any case I will assist
> you with getting your wiki account updated.
> -MC

I have seen this happening if the receiving mailserver uses greylisting
and the email is sent directly by the Wiki app. If all the Wiki does is
fire email and forget then it will not try to resend after being
greylisted. The solution would be to have the Wiki send the email to a
local MTA which takes care of delivering it.

Regards,
Patrick


From rob4manhere at gmail.com  Fri Jan 29 07:59:08 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 29 Jan 2010 09:59:08 -0600
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <9853f4ff1001290751q29c12e58oc550bd37fbf225c9@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
	<5d2828f1001281205u2eccc999re6f25a55b07156c7@mail.gmail.com>
	<9853f4ff1001290722m4007e620sf6db3c61c1495a4b@mail.gmail.com>
	<87f2f3b91001290730y652086far4008036c47beed89@mail.gmail.com>
	<9853f4ff1001290751q29c12e58oc550bd37fbf225c9@mail.gmail.com>
Message-ID: <EDB39B7D-87D2-40F4-B18F-EDBFA77ADFBB@gmail.com>

http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_29


Calling Instructions

Friday January 29 at 1700 UTC (1100 CST)


sip:888 at conference.freeswitch.org or via the good old PSTN at  
+1-919-386-9900
Or click on this link
Or call Skype the skype user "skypiax5", then press "1" on the Skype  
dialpad (max 20 concurrent users).
Codecs: PCMU/PCMA, G.722, CELT, Speex, Skype, among others




On Jan 29, 2010, at 9:51 AM, David Villasmil wrote:

> Today what time/timezone?
>
> On Fri, Jan 29, 2010 at 4:30 PM, Michael Collins  
> <msc at freeswitch.org> wrote:
>> David,
>>
>> Are you using Lua and lusql for some exotic call handling  
>> scenarios? If so,
>> would you mind posting some examples to the wiki and then linking  
>> here?
>> Also, if you can join the community conference call today that  
>> would be
>> great!
>>
>> Thanks,
>> MC
>>
>> On Fri, Jan 29, 2010 at 7:22 AM, David Villasmil
>> <david.villasmil.work at gmail.com> wrote:
>>>
>>> Oh yes, here's an example:
>>>
>>> function dbConnect()
>>>     -- connect to db
>>>     require "luasql.mysql"
>>>     env = assert(luasql.mysql())
>>>     conn =  
>>> assert(env:connect("freeswitch","user","userpass","localhost"))
>>> end
>>>
>>> function getpin()
>>>     session:streamFile(card_greeting_audio_file)
>>>
>>>     card_pin = session:getDigits(4, "#", 3000);
>>>
>>>     if card_pin > "" then
>>>          freeswitch.consoleLog("info", "CARD INFO: PIN...........:
>>> ".. card_pin .."\n");
>>>          cur = assert(
>>>               conn:execute( "select * from cards_table where pin  
>>> ="..
>>> card_pin ..";" )
>>>               )
>>>
>>>          -- print all rows, the rows will be indexed by field names
>>>          row = cur:fetch ({}, "a")
>>>          fsLog("ROWS: ".. cur:numrows() )
>>>          if cur:numrows() > 0 then pinok=true end
>>>          while row do
>>>
>>>               fsLog("CARD INFO: Batch.........: "..  
>>> row.batch        )
>>>               fsLog("CARD INFO: Card Name.....: "..  
>>> row.card_name    )
>>>               fsLog("CARD INFO: Ratetable.....: "..  
>>> row.ratetable    )
>>>               fsLog("CARD INFO: Initial Bal...: "..  
>>> row.init_bal     )
>>>               fsLog("CARD INFO: Curr Balance..: "..  
>>> row.balance      )
>>>
>>>               batch, ratetable, init_bal, balance = row.batch,
>>> row.ratetable, row.init_bal, row.balance
>>>
>>>               SetVar("card_pin",card_pin)
>>>               SetVar("card_batch", batch)
>>>               SetVar("card_ratetable", ratetable)
>>>               SetVar("card_init_bal", init_bal)
>>>               SetVar("card_balance", balance)
>>>
>>>            -- reusing the table of results
>>>            row = cur:fetch (row, "a")
>>>          end
>>>     else
>>>          pinok=false
>>>          session:streamFile(card_invalid_pin_audio_file)
>>>     end
>>> end
>>>
>>>
>>> On Thu, Jan 28, 2010 at 9:05 PM, Mike van Lammeren
>>> <mike at van.lammeren.net> wrote:
>>>> And you can make queries against your MySQL database, and get  
>>>> results,
>>>> etc.?
>>>>
>>>> On Thu, Jan 28, 2010 at 2:43 AM, David Villasmil
>>>> <david.villasmil.work at gmail.com> wrote:
>>>>>
>>>>> Hello,
>>>>>
>>>>> That works fine:
>>>>>
>>>>> box:~# lua testdb.lua
>>>>> box:~#
>>>>>
>>>>>
>>>>> David
>>>>>
>>>>> On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
>>>>> <mike at van.lammeren.net> wrote:
>>>>>> Have you tried running a Lua script that includes the library  
>>>>>> from
>>>>>> outside
>>>>>> of FreeSWITCH? What does that do?
>>>>>>
>>>>>> On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com 
>>>>>> >
>>>>>> wrote:
>>>>>>>
>>>>>>> I tried running ldconfig on the directory containing mysql.so,  
>>>>>>> but
>>>>>>> it
>>>>>>> did
>>>>>>> not help.
>>>>>>> So it sounds like there could be a bug in the latter versions?
>>>>>>>
>>>>>>> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
>>>>>>> <david.villasmil.work at gmail.com> wrote:
>>>>>>>>
>>>>>>>> I got the same error, my script was working with no problems  
>>>>>>>> before
>>>>>>>> an
>>>>>>>> update to trunk.
>>>>>>>>
>>>>>>>> David
>>>>>>>>
>>>>>>>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com 
>>>>>>>> >
>>>>>>>> wrote:
>>>>>>>>> Hi, I followed the instructions in the Lua documentation for
>>>>>>>>> setting
>>>>>>>>> up
>>>>>>>>> luasql, but when I try to run my script I get:
>>>>>>>>> 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading
>>>>>>>>> module
>>>>>>>>> 'luasql.mysql' from file
>>>>>>>>> '/usr/local/lib/lua/5.1/luasql/mysql.so':
>>>>>>>>>         /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore
>>>>>>>>> segment
>>>>>>>>> prot
>>>>>>>>> after reloc: Permission denied
>>>>>>>>> stack traceback:
>>>>>>>>>         [C]: ?
>>>>>>>>>         [C]: in function 'require'
>>>>>>>>>         /usr/local/freeswitch/scripts/l.lua:2: in main chunk
>>>>>>>>> I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
>>>>>>>>> I changed the permissions for mysql.so and for my script to  
>>>>>>>>> 777,
>>>>>>>>> so
>>>>>>>>> I'm
>>>>>>>>> not
>>>>>>>>> sure where the permission problem could be.
>>>>>>>>> I'd appreciate any suggestions.
>>>>>>>>> Thanks,
>>>>>>>>> Adam
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>
>>>>>>>>
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>>
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From william.suffill at gmail.com  Fri Jan 29 08:02:06 2010
From: william.suffill at gmail.com (William Suffill)
Date: Fri, 29 Jan 2010 11:02:06 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <9853f4ff1001290751q29c12e58oc550bd37fbf225c9@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com> 
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com> 
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com> 
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com> 
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com> 
	<5d2828f1001281205u2eccc999re6f25a55b07156c7@mail.gmail.com> 
	<9853f4ff1001290722m4007e620sf6db3c61c1495a4b@mail.gmail.com> 
	<87f2f3b91001290730y652086far4008036c47beed89@mail.gmail.com> 
	<9853f4ff1001290751q29c12e58oc550bd37fbf225c9@mail.gmail.com>
Message-ID: <6b65470d1001290802q4f92b1e8wc60e8b61615be812@mail.gmail.com>

Friday January 29 at 1700 UTC (1100 CST)
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org> or
via the good old PSTN at +1-919-386-9900
Or click on this link<http://www.phonefromhere.com/freeswitch/freeswitch.xsql>
Or call Skype the skype user "skypiax5", then press "1" on the Skype dialpad
(max 20 concurrent users).
Codecs: PCMU/PCMA, G.722, CELT, Speex, Skype, among others


More info here http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_29
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From jcasale at activenetwerx.com  Fri Jan 29 08:21:34 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 29 Jan 2010 16:21:34 +0000
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
In-Reply-To: <87f2f3b91001290711t3ee71079wa0674941d8dbc741@mail.gmail.com>
References: <CA5A491E9DEFBE4CB777DE97E21575E90BD7F4@prato.activenetwerx.local>
	<20100127210836.5ED5D2E3B2D@cuneorg-email.cune.pri>
	<CA5A491E9DEFBE4CB777DE97E21575E90BF91C@prato.activenetwerx.local>
	<87f2f3b91001290711t3ee71079wa0674941d8dbc741@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90C2546@prato.activenetwerx.local>

>Are you really dialing that phone number or is that just redacted? Try dialing 919-386-9900 for testing.

Hi Michael,
It was redacted. I can get back in at night to continue testing, but
given incoming doesn't work I doubt I will have any luck. The dialplan
to bypass the sip provider requires a 9, then the 10 digit #.

On the incoming call, I see the two errors:
[ERR] zap_io.c:1599 I/O backend does not support command 2[4|5]!

Any idea what that pertains to? From there on, everything looks
positive as it find the dialplan and plays the wave, I just can't
hear anything at all.

Even though this card doesn't have a hw ec, w/ mg2 or hpec the quality
was flawless. I'd love to have it running again.

For the record, using the 4630 rev of zaptel as per
http://wiki.freeswitch.org/wiki/Zaptel_Tutorial
I was able to get some success, but it was very intermittent.

Thanks for any insight,
jlc


From codecomplete at free.fr  Fri Jan 29 08:21:52 2010
From: codecomplete at free.fr (Fred-145)
Date: Fri, 29 Jan 2010 17:21:52 +0100
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
References: <27338355.post@talk.nabble.com>
	<27338355.post-WJuSqJV8a7jJsTRiRinrng@public.gmane.org>
	<201001270750.16772.sos@sokhapkin.dyndns.org>
	<eui5m5hg6tqemri4r5u250hu72mjddmjnt@4ax.com>
	<eui5m5hg6tqemri4r5u250hu72mjddmjnt-e09XROE/p8c@public.gmane.org>
	<57E0B3FB-166A-4CF2-82E8-267D2FADB9DC@freeswitch.org>
Message-ID: <gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b@4ax.com>

On Fri, 29 Jan 2010 08:26:53 -0600, Brian West
<brian at freeswitch.org> wrote:
>> Are there drawbacks to having RTP pakets flow directly between the SIP
>> end-points?
>
>Well if nat is involved you might have issues.

Since Freeswitch has handled the initial connection and the NAT box
has opened the range of UDP ports for RTP/RTCP, in what case would the
two end-points have a problem with NAT? Is it possible for end-points
to send IP/port information in the midst of a conversation?



From david.villasmil.work at gmail.com  Fri Jan 29 08:23:45 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 29 Jan 2010 17:23:45 +0100
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <87f2f3b91001290730y652086far4008036c47beed89@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
	<5d2828f1001281205u2eccc999re6f25a55b07156c7@mail.gmail.com>
	<9853f4ff1001290722m4007e620sf6db3c61c1495a4b@mail.gmail.com>
	<87f2f3b91001290730y652086far4008036c47beed89@mail.gmail.com>
Message-ID: <9853f4ff1001290823l5ef053e2j2feaf1190a23d834@mail.gmail.com>

Where in the wiki?

On Fri, Jan 29, 2010 at 4:30 PM, Michael Collins <msc at freeswitch.org> wrote:
> David,
>
> Are you using Lua and lusql for some exotic call handling scenarios? If so,
> would you mind posting some examples to the wiki and then linking here?
> Also, if you can join the community conference call today that would be
> great!
>
> Thanks,
> MC
>
> On Fri, Jan 29, 2010 at 7:22 AM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> Oh yes, here's an example:
>>
>> function dbConnect()
>> ? ? -- connect to db
>> ? ? require "luasql.mysql"
>> ? ? env = assert(luasql.mysql())
>> ? ? conn = assert(env:connect("freeswitch","user","userpass","localhost"))
>> end
>>
>> function getpin()
>> ? ? session:streamFile(card_greeting_audio_file)
>>
>> ? ? card_pin = session:getDigits(4, "#", 3000);
>>
>> ? ? if card_pin > "" then
>> ? ? ? ? ?freeswitch.consoleLog("info", "CARD INFO: PIN...........:
>> ".. card_pin .."\n");
>> ? ? ? ? ?cur = assert(
>> ? ? ? ? ? ? ? conn:execute( "select * from cards_table where pin ="..
>> card_pin ..";" )
>> ? ? ? ? ? ? ? )
>>
>> ? ? ? ? ?-- print all rows, the rows will be indexed by field names
>> ? ? ? ? ?row = cur:fetch ({}, "a")
>> ? ? ? ? ?fsLog("ROWS: ".. cur:numrows() )
>> ? ? ? ? ?if cur:numrows() > 0 then pinok=true end
>> ? ? ? ? ?while row do
>>
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Batch.........: ".. row.batch ? ? ? ?)
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Card Name.....: ".. row.card_name ? ?)
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Ratetable.....: ".. row.ratetable ? ?)
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Initial Bal...: ".. row.init_bal ? ? )
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Curr Balance..: ".. row.balance ? ? ?)
>>
>> ? ? ? ? ? ? ? batch, ratetable, init_bal, balance = row.batch,
>> row.ratetable, row.init_bal, row.balance
>>
>> ? ? ? ? ? ? ? SetVar("card_pin",card_pin)
>> ? ? ? ? ? ? ? SetVar("card_batch", batch)
>> ? ? ? ? ? ? ? SetVar("card_ratetable", ratetable)
>> ? ? ? ? ? ? ? SetVar("card_init_bal", init_bal)
>> ? ? ? ? ? ? ? SetVar("card_balance", balance)
>>
>> ? ? ? ? ? ?-- reusing the table of results
>> ? ? ? ? ? ?row = cur:fetch (row, "a")
>> ? ? ? ? ?end
>> ? ? else
>> ? ? ? ? ?pinok=false
>> ? ? ? ? ?session:streamFile(card_invalid_pin_audio_file)
>> ? ? end
>> end
>>
>>
>> On Thu, Jan 28, 2010 at 9:05 PM, Mike van Lammeren
>> <mike at van.lammeren.net> wrote:
>> > And you can make queries against your MySQL database, and get results,
>> > etc.?
>> >
>> > On Thu, Jan 28, 2010 at 2:43 AM, David Villasmil
>> > <david.villasmil.work at gmail.com> wrote:
>> >>
>> >> Hello,
>> >>
>> >> That works fine:
>> >>
>> >> box:~# lua testdb.lua
>> >> box:~#
>> >>
>> >>
>> >> David
>> >>
>> >> On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
>> >> <mike at van.lammeren.net> wrote:
>> >> > Have you tried running a Lua script that includes the library from
>> >> > outside
>> >> > of FreeSWITCH? What does that do?
>> >> >
>> >> > On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> I tried running ldconfig on the directory containing mysql.so, but
>> >> >> it
>> >> >> did
>> >> >> not help.
>> >> >> So it sounds like there could be a bug in the latter versions?
>> >> >>
>> >> >> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
>> >> >> <david.villasmil.work at gmail.com> wrote:
>> >> >>>
>> >> >>> I got the same error, my script was working with no problems before
>> >> >>> an
>> >> >>> update to trunk.
>> >> >>>
>> >> >>> David
>> >> >>>
>> >> >>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com>
>> >> >>> wrote:
>> >> >>> > Hi, I followed the instructions in the Lua documentation for
>> >> >>> > setting
>> >> >>> > up
>> >> >>> > luasql, but when I try to run my script I get:
>> >> >>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading
>> >> >>> > module
>> >> >>> > 'luasql.mysql' from file
>> >> >>> > '/usr/local/lib/lua/5.1/luasql/mysql.so':
>> >> >>> > ?? ? ? ?/usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore
>> >> >>> > segment
>> >> >>> > prot
>> >> >>> > after reloc: Permission denied
>> >> >>> > stack traceback:
>> >> >>> > ?? ? ? ?[C]: ?
>> >> >>> > ?? ? ? ?[C]: in function 'require'
>> >> >>> > ?? ? ? ?/usr/local/freeswitch/scripts/l.lua:2: in main chunk
>> >> >>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
>> >> >>> > I changed the permissions for mysql.so and for my script to 777,
>> >> >>> > so
>> >> >>> > I'm
>> >> >>> > not
>> >> >>> > sure where the permission problem could be.
>> >> >>> > I'd appreciate any suggestions.
>> >> >>> > Thanks,
>> >> >>> > Adam
>> >> >>> >
>> >> >>> > _______________________________________________
>> >> >>> > FreeSWITCH-users mailing list
>> >> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>> >
>> >> >>> >
>> >> >>> >
>> >> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> > http://www.freeswitch.org
>> >> >>> >
>> >> >>> >
>> >> >>>
>> >> >>> _______________________________________________
>> >> >>> FreeSWITCH-users mailing list
>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>
>> >> >>>
>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> http://www.freeswitch.org
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Fri Jan 29 08:38:01 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jan 2010 08:38:01 -0800
Subject: [Freeswitch-users] wiki password recovery - no mail is send
In-Reply-To: <4B63053D.20805@puzzled.xs4all.nl>
References: <4B629D05.1060908@xpirio.com>
	<87f2f3b91001290725o13dde05fq9426d8ef45e46035@mail.gmail.com>
	<4B63053D.20805@puzzled.xs4all.nl>
Message-ID: <87f2f3b91001290838m337ea44dwe7e3ea7e9c3f1ac2@mail.gmail.com>

On Fri, Jan 29, 2010 at 7:56 AM, Patrick
<freeswitch-list at puzzled.xs4all.nl>wrote:

> On 01/29/2010 04:25 PM, Michael Collins wrote:
> > Email me off list and I'll help you with this. I've never been able to
> > reproduce this symptom but some others have. In any case I will assist
> > you with getting your wiki account updated.
> > -MC
>
> I have seen this happening if the receiving mailserver uses greylisting
> and the email is sent directly by the Wiki app. If all the Wiki does is
> fire email and forget then it will not try to resend after being
> greylisted. The solution would be to have the Wiki send the email to a
> local MTA which takes care of delivering it.
>
> Sounds like a job for ... Super Raymond! :D
-MC
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From Russell.Mosemann at cune.org  Fri Jan 29 08:43:59 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 29 Jan 2010 16:43:59 -0000
Subject: [Freeswitch-users] Equivalent to
	Asterisk's	"directrtpsetup=yes"?
In-Reply-To: <gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b@4ax.com>
Message-ID: <20100129164359.9B3C22B10D@cuneorg-email.cune.pri>

Fred-145 <codecomplete at free.fr> said:

> Since Freeswitch has handled the initial connection and the NAT box
> has opened the range of UDP ports for RTP/RTCP, in what case would the
> two end-points have a problem with NAT? 

The ports are open between the endpoint and Freeswitch. The ports are not
open between the two endpoints themselves. If each endpoint is behind its
own NAT, neither endpoint will be able to contact the other endpoint
unless some kind of forwarding is set up on the firewall to map the
external IP address and port to an internal IP address and port.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From brian at freeswitch.org  Fri Jan 29 09:04:07 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 29 Jan 2010 11:04:07 -0600
Subject: [Freeswitch-users] wiki password recovery - no mail is send
In-Reply-To: <87f2f3b91001290838m337ea44dwe7e3ea7e9c3f1ac2@mail.gmail.com>
References: <4B629D05.1060908@xpirio.com>
	<87f2f3b91001290725o13dde05fq9426d8ef45e46035@mail.gmail.com>
	<4B63053D.20805@puzzled.xs4all.nl>
	<87f2f3b91001290838m337ea44dwe7e3ea7e9c3f1ac2@mail.gmail.com>
Message-ID: <9F5F7247-3ED1-497E-8DDB-0ED43C91158B@freeswitch.org>

aka Mud Puddle!

/b

On Jan 29, 2010, at 10:38 AM, Michael Collins wrote:

> Sounds like a job for ... Super Raymond! :D
> -MC



From ranjtech at gmail.com  Fri Jan 29 09:57:46 2010
From: ranjtech at gmail.com (RR)
Date: Fri, 29 Jan 2010 12:57:46 -0500
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <025701caa0a7$e1ca6200$a55f2600$@com>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
Message-ID: <02b001caa10c$913b9100$b3b2b300$@com>

Any ideas? Anyone? All I can tell is that the Contact Header looks a bit
dodgey but how that's causing a 409 is what I don't get. 

 

Help please?

\RR

 

From: RR [mailto:ranjtech at gmail.com] 
Sent: Friday, January 29, 2010 12:57 AM
To: freeswitch-users at lists.freeswitch.org
Subject: RE: [Freeswitch-users] 409 Conflict When registering FS with
Softswitch

 

Hi Brian,

 

Ok here's the sip trace captured at the softswitch. 

 

BTW, I noticed that during startup, I see FS printing out this message:

2010-01-29 01:30:05.634220 [NOTICE] sofia_reg.c:2267 Added gateway
'Test-Inbound' to profile 'external'

2010-01-29 01:30:05.634547 [ERR] sofia.c:1663 ERROR: username param is
REQUIRED!

2010-01-29 01:30:05.634657 [ERR] sofia.c:1663 ERROR: username param is
REQUIRED!

 

Which is weird, because as you can see from the config, the username is
infact present. Weird! Anyway, here's the trace

 

REGISTER sip:myswitch.net.au SIP/2.0

Via: SIP/2.0/UDP 173.xxx.xxx.xxx:5080;rport;branch=z9hG4bK1U32gpr9vj5eF

Max-Forwards: 70

From: <sip:user1 at myswitch.net.au;transport=udp>;tag=11Qey4tcrUH9g

To: <sip:user1 at myswitch.net.au;transport=udp>

Call-ID: cd589b73-031e-445f-a4d8-2fe334d81bbc

CSeq: 126255215 REGISTER

Contact:
<sip:gw+Test-Inbound at 173.xxx.xxx.xxx:5080;transport=udp;gw=Test-Inbound>

Expires: 60

User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15980

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY

Supported: timer, precondition, path, replaces

Content-Length: 0

 

SIP/2.0 409 Conflict

Via: SIP/2.0/UDP 173.xxx.xxx.xxx:5080;rport;branch=z9hG4bK1U32gpr9vj5eF

From: <sip:user1 at 10.1.3.20;transport=udp>;tag=11Qey4tcrUH9g

To: <sip:user1 at 10.1.3.20;transport=udp>

Call-ID: cd589b73-031e-445f-a4d8-2fe334d81bbc

CSeq: 126255215 REGISTER

Content-Length: 0

 

.and then this message just repeats again and again with every REGISTER
request.

 

Thanks for your help

\RR

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, January 28, 2010 2:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 409 Conflict When registering FS with
Softswitch

 

Any sip trace would work.. text.. or what not.. 

 

 

 

/b

 

On Jan 28, 2010, at 12:46 PM, RR wrote:

 

Hi brian,

 

Do you need the sip trace from within FS or just any packet capture like
from snoop/tcpdump etc will do? I don't know how to do a sip trace from
within FS

 

\R

 



__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4815 (20100128) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com



__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4816 (20100128) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4818 (20100129) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From brian at freeswitch.org  Fri Jan 29 10:04:53 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 29 Jan 2010 12:04:53 -0600
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <02b001caa10c$913b9100$b3b2b300$@com>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
Message-ID: <F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>

contact header is fine... but if your switch requires the username in the contact then you can set extension-in-contact on the gateway to force that.

/b

On Jan 29, 2010, at 11:57 AM, RR wrote:

> Any ideas? Anyone? All I can tell is that the Contact Header looks a bit dodgey but how that?s causing a 409 is what I don?t get.
>  
> Help please?
> \RR
>  

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From ranjtech at gmail.com  Fri Jan 29 10:44:17 2010
From: ranjtech at gmail.com (RR)
Date: Fri, 29 Jan 2010 13:44:17 -0500
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
	<F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
Message-ID: <02c501caa113$105532b0$30ff9810$@com>

Thanks mate! Specifying the extension same as the username and then using
extension-in-contact fixed the problem. It now registers successfully with
the switch. 

 

I do though still get the message:

2010-01-29 01:30:05.634220 [NOTICE] sofia_reg.c:2267 Added gateway
'Test-Inbound' to profile 'external'

2010-01-29 01:30:05.634547 [ERR] sofia.c:1663 ERROR: username param is
REQUIRED!

2010-01-29 01:30:05.634657 [ERR] sofia.c:1663 ERROR: username param is
REQUIRED!

 

What is that about? I have the username param stated in the gateway
profile!!

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Friday, January 29, 2010 1:05 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 409 Conflict When registering FS with
Softswitch

 

contact header is fine... but if your switch requires the username in the
contact then you can set extension-in-contact on the gateway to force that.

 

/b

 

On Jan 29, 2010, at 11:57 AM, RR wrote:





Any ideas? Anyone? All I can tell is that the Contact Header looks a bit
dodgey but how that's causing a 409 is what I don't get.

 

Help please?

\RR

 

 



__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4818 (20100129) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From jerry.richards at teotech.com  Fri Jan 29 11:04:09 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 29 Jan 2010 11:04:09 -0800
Subject: [Freeswitch-users] Posting a Bounty
In-Reply-To: <mailman.2031.1251316570.2787.freeswitch-users@lists.freeswitch.org>
References: <mailman.2031.1251316570.2787.freeswitch-users@lists.freeswitch.org>
Message-ID: <F5D23A7C72D34626AC8BA8EFC1331AF1@greyhawk.tonecommander.com>

 
How do I post a new Bounty request?

Thanks,
Jerry



From brian at freeswitch.org  Fri Jan 29 11:08:17 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 29 Jan 2010 13:08:17 -0600
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <02c501caa113$105532b0$30ff9810$@com>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
	<F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
	<02c501caa113$105532b0$30ff9810$@com>
Message-ID: <E91AAF1B-2BCB-4D9E-9A49-221480CD8BF5@freeswitch.org>

You'll require a username param .. are you on svn trunk?

/b

On Jan 29, 2010, at 12:44 PM, RR wrote:

> Thanks mate! Specifying the extension same as the username and then using extension-in-contact fixed the problem. It now registers successfully with the switch.
>  
> I do though still get the message:
> 2010-01-29 01:30:05.634220 [NOTICE] sofia_reg.c:2267 Added gateway 'Test-Inbound' to profile 'external'
> 2010-01-29 01:30:05.634547 [ERR] sofia.c:1663 ERROR: username param is REQUIRED!
> 2010-01-29 01:30:05.634657 [ERR] sofia.c:1663 ERROR: username param is REQUIRED!

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From msc at freeswitch.org  Fri Jan 29 11:10:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jan 2010 11:10:49 -0800
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <02c501caa113$105532b0$30ff9810$@com>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
	<F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
	<02c501caa113$105532b0$30ff9810$@com>
Message-ID: <87f2f3b91001291110l226ce6d8u6e806d0a1b782c5f@mail.gmail.com>

On Fri, Jan 29, 2010 at 10:44 AM, RR <ranjtech at gmail.com> wrote:

>  Thanks mate! Specifying the extension same as the username and then using
> extension-in-contact fixed the problem. It now registers successfully with
> the switch.
>
>
>
> I do though still get the message:
>
> *2010-01-29 01:30:05.634220 [NOTICE] sofia_reg.c:2267 Added gateway
> 'Test-Inbound' to profile 'external'*
>
> *2010-01-29 01:30:05.634547 [ERR] sofia.c:1663 ERROR: username param is
> REQUIRED!*
>
> *2010-01-29 01:30:05.634657 [ERR] sofia.c:1663 ERROR: username param is
> REQUIRED!*
>
>
>
> What is that about? I have the username param stated in the gateway
> profile!!
>

Do you possibly have some other XML files floating around that don't have a
username param? It's curious that it said this error twice. It makes me
think that possibly a different file or files is causing that...

-MC
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From msc at freeswitch.org  Fri Jan 29 11:14:56 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jan 2010 11:14:56 -0800
Subject: [Freeswitch-users] Posting a Bounty
In-Reply-To: <F5D23A7C72D34626AC8BA8EFC1331AF1@greyhawk.tonecommander.com>
References: <mailman.2031.1251316570.2787.freeswitch-users@lists.freeswitch.org>
	<F5D23A7C72D34626AC8BA8EFC1331AF1@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91001291114s91dc125n324f215a71a80946@mail.gmail.com>

On Fri, Jan 29, 2010 at 11:04 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

>
> How do I post a new Bounty request?
>
> In JIRA you can do this:
http://jira.freeswitch.org/browse/BOUNTY

Once you post the bounty you can link to it here in the mailing list to let
everyone know about it.
-MC
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From ranjtech at gmail.com  Fri Jan 29 13:08:50 2010
From: ranjtech at gmail.com (RR)
Date: Fri, 29 Jan 2010 16:08:50 -0500
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <87f2f3b91001291110l226ce6d8u6e806d0a1b782c5f@mail.gmail.com>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
	<F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
	<02c501caa113$105532b0$30ff9810$@com>
	<87f2f3b91001291110l226ce6d8u6e806d0a1b782c5f@mail.gmail.com>
Message-ID: <02dd01caa127$428aa760$c79ff620$@com>

I don't think so..this is the first time I've started to configure FS, and
this was the first xml file I have touched. 

 

Anyway, I saved this gateway xml file, rm -rf'ed the entire conf directory,
did a make current, did a make samples, restored the gateway xml config and
now I don't see the error about username param.

 

Thanks for the help. Now let's see what I can do next in FS, I know it's a
lot to learn. Wish I'd jumped right into it 2 years ago when it was being
developed!

 

I guess the first thing to learn is Dialplan because now I want to make an
inbound call from another endpoint registered to my softswitch to be able to
call an extension registered to FS by calling user1

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Friday, January 29, 2010 2:11 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 409 Conflict When registering FS with
Softswitch

 

 

On Fri, Jan 29, 2010 at 10:44 AM, RR <ranjtech at gmail.com> wrote:

Thanks mate! Specifying the extension same as the username and then using
extension-in-contact fixed the problem. It now registers successfully with
the switch. 

 

I do though still get the message:

2010-01-29 01:30:05.634220 [NOTICE] sofia_reg.c:2267 Added gateway
'Test-Inbound' to profile 'external'

2010-01-29 01:30:05.634547 [ERR] sofia.c:1663 ERROR: username param is
REQUIRED!

2010-01-29 01:30:05.634657 [ERR] sofia.c:1663 ERROR: username param is
REQUIRED!

 

What is that about? I have the username param stated in the gateway
profile!!


Do you possibly have some other XML files floating around that don't have a
username param? It's curious that it said this error twice. It makes me
think that possibly a different file or files is causing that...

-MC

 

 

__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4818 (20100129) __________

 

The message was checked by ESET NOD32 Antivirus.

 

http://www.eset.com

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From jerry.richards at teotech.com  Fri Jan 29 13:14:45 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 29 Jan 2010 13:14:45 -0800
Subject: [Freeswitch-users] PRI Simulator
In-Reply-To: <mailman.5734.1264787871.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.5734.1264787871.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <AF84ED7F907F493DAC0A29C8FBC54E44@greyhawk.tonecommander.com>


Can anyone recommend a good PRI simulator?  Sorry this is off topic a bit.

Thanks,
Jerry



From mouncifbb at gmail.com  Fri Jan 29 14:43:38 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Fri, 29 Jan 2010 17:43:38 -0500
Subject: [Freeswitch-users] mod_lcr problem
Message-ID: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>

i can't make use of mod_lcr using Intra/Interstate rating, I am using
svn: FreeSWITCH Version 1.0.trunk (16517)


lcr mysql table structure:

CREATE TABLE `lcr` (
  `id` INT(11) NOT NULL AUTO_INCREMENT,
  `digits` VARCHAR(15) DEFAULT NULL,
  `rate` FLOAT(11,5) DEFAULT NULL,
  `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
  `intralata_rate` FLOAT(11,5) DEFAULT NULL,
  `carrier_id` INT(11) NOT NULL,
  `lead_strip` INT(11) NOT NULL,
  `trail_strip` INT(11) NOT NULL,
  `prefix` VARCHAR(16) NOT NULL,
  `suffix` VARCHAR(16) NOT NULL,
  `lcr_profile` VARCHAR(32) DEFAULT NULL,
  `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
  `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
  `quality` FLOAT(10,6) NOT NULL,
  `reliability` FLOAT(10,6) NOT NULL,
  `cid` VARCHAR(32) NOT NULL DEFAULT '',
  `enabled` TINYINT(1) NOT NULL DEFAULT '1',
  PRIMARY KEY  (`id`),
  KEY `carrier_id` (`carrier_id`),
  KEY `digits` (`digits`),
  KEY `lcr_profile` (`lcr_profile`),
  KEY `digits_profile_cid_rate` USING BTREE (`digits`),
  CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES `carriers`
(`id`) ON DELETE CASCADE ON UPDATE CASCADE
) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1



*lcr_admin show profiles*

Name:           default
 custom sql:    SELECT l.digits, c.carrier_name, l.${lcr_rate_field},
cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip, l.trail_strip,
l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
(${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN date_start AND
date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality DESC,
 reliability DESC, rand();
 has %:         false
 has vars:      true
 has intrastate:        true
 has intralata: true
 has npanxx:    true
 Reorder rate:  enabled
 Info in headers:       disabled
 Quote IN() List:       disabled




*lc**r 617642  default*  returns rate from the rate field table and not
intra/inter state fields rates.


Any ideas? thanks!
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From freeswitch at aastral.net  Fri Jan 29 15:52:21 2010
From: freeswitch at aastral.net (Bill W)
Date: Fri, 29 Jan 2010 18:52:21 -0500
Subject: [Freeswitch-users] Posting a Bounty
In-Reply-To: <F5D23A7C72D34626AC8BA8EFC1331AF1@greyhawk.tonecommander.com>
References: <mailman.2031.1251316570.2787.freeswitch-users@lists.freeswitch.org>
	<F5D23A7C72D34626AC8BA8EFC1331AF1@greyhawk.tonecommander.com>
Message-ID: <4B6374B5.1050500@aastral.net>

Hey Jerry,

Just go to jira.freeswitch.org, log in, create new issue with project=bounty

Hope this helps,
Bill



Jerry Richards wrote:
>  
> How do I post a new Bounty request?
> 
> Thanks,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Fri Jan 29 16:21:39 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 29 Jan 2010 16:21:39 -0800
Subject: [Freeswitch-users] PRI Simulator
In-Reply-To: <AF84ED7F907F493DAC0A29C8FBC54E44@greyhawk.tonecommander.com>
References: <mailman.5734.1264787871.2495.freeswitch-users@lists.freeswitch.org>
	<AF84ED7F907F493DAC0A29C8FBC54E44@greyhawk.tonecommander.com>
Message-ID: <1AF7532D-2B17-4D6E-97A0-421C595FF674@freeswitch.org>

What's your budget?

Sent from my iPhone

On Jan 29, 2010, at 1:14 PM, "Jerry Richards" <jerry.richards at teotech.com 
 > wrote:

>
> Can anyone recommend a good PRI simulator?  Sorry this is off topic  
> a bit.
>
> Thanks,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From rupa at rupa.com  Fri Jan 29 16:37:07 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 29 Jan 2010 18:37:07 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
Message-ID: <b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>

turn console logging up to debug and redo the lcr lookup.  The sql
statements along with status info will show up.  This should give
enough information to debug what is happening.

I'm assuming the npanxx table is actually populated and not just existing?

When doing the lookup from the cli you have to tell lcr what CID to
use (remember, it is relative to the src/dest number).  I'm pretty
sure you get something on the console log when you don't specify a CID
when using the commandline.  Anyway:

lcr 617642 ?default 6176421212

should give you intralata.

Note that the definition of intralata doesn't mean "local" for some
providers.  Some providers define local to "same ratecenter" which is
even more restrictive.

On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <mouncifbb at gmail.com> wrote:
> i can't make use of mod_lcr using Intra/Interstate rating, I am using
> svn:?FreeSWITCH Version 1.0.trunk (16517)
>
> lcr mysql table structure:
> CREATE TABLE `lcr` (
> ??`id` INT(11) NOT NULL AUTO_INCREMENT,
> ??`digits` VARCHAR(15) DEFAULT NULL,
> ??`rate` FLOAT(11,5) DEFAULT NULL,
> ??`intrastate_rate` FLOAT(11,5) DEFAULT NULL,
> ??`intralata_rate` FLOAT(11,5) DEFAULT NULL,
> ??`carrier_id` INT(11) NOT NULL,
> ??`lead_strip` INT(11) NOT NULL,
> ??`trail_strip` INT(11) NOT NULL,
> ??`prefix` VARCHAR(16) NOT NULL,
> ??`suffix` VARCHAR(16) NOT NULL,
> ??`lcr_profile` VARCHAR(32) DEFAULT NULL,
> ??`date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
> ??`date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
> ??`quality` FLOAT(10,6) NOT NULL,
> ??`reliability` FLOAT(10,6) NOT NULL,
> ??`cid` VARCHAR(32) NOT NULL DEFAULT '',
> ??`enabled` TINYINT(1) NOT NULL DEFAULT '1',
> ??PRIMARY KEY ?(`id`),
> ??KEY `carrier_id` (`carrier_id`),
> ??KEY `digits` (`digits`),
> ??KEY `lcr_profile` (`lcr_profile`),
> ??KEY `digits_profile_cid_rate` USING BTREE (`digits`),
> ??CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES `carriers`
> (`id`) ON DELETE CASCADE ON UPDATE CASCADE
> ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>
>
> lcr_admin show profiles
> Name: ? ? ? ? ? default
> ?custom sql: ? ?SELECT l.digits, c.carrier_name, l.${lcr_rate_field},
> cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip, l.trail_strip,
> l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
> l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
> c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
> (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN date_start AND
> date_end ORDER BY digits DESC, ?${lcr_rate_field}, ?quality DESC,
> ?reliability DESC, rand();
> ?has %: ? ? ? ? false
> ?has vars: ? ? ?true
> ?has intrastate: ? ? ? ?true
> ?has intralata: true
> ?has npanxx: ? ?true
> ?Reorder rate: ?enabled
> ?Info in headers: ? ? ? disabled
> ?Quote IN() List: ? ? ? disabled
>
>
>
> lcr 617642 ?default ?returns rate from the rate field table and not
> intra/inter state fields rates.
>
> Any ideas? thanks!
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From mouncifbb at gmail.com  Fri Jan 29 20:30:36 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Fri, 29 Jan 2010 23:30:36 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
Message-ID: <ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>

Tried it and it's not giving me intralata  instead I get interstate, does
the npa_nxx_company_ocn table needs to be used in this case?, also do I have
to have the rate field in lcr table?


lcr 617642  default 6176421212

 | Digit Match | Carrier  | Rate     | Codec | CID Regexp | Dialstring
                                                     |
 | 617642      | carrier1 | 0.00500  |       |            |
[lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |


2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr is
[617642  default 6176421212]
2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
[6176421212]
2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing [state:0
lata:0] so rate field is [rate]
2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
c.id=cg.carrier_id
WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
(617642, 61764, 6176, 617, 61, 6) AND CURRENT_TIMESTAMP BETWEEN date_start
AND date_end ORDER BY digits DESC,  rate, rand();
2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
[lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to head of
list
2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
[lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642


Thank you Rupa!

On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> turn console logging up to debug and redo the lcr lookup.  The sql
> statements along with status info will show up.  This should give
> enough information to debug what is happening.
>
> I'm assuming the npanxx table is actually populated and not just existing?
>
> When doing the lookup from the cli you have to tell lcr what CID to
> use (remember, it is relative to the src/dest number).  I'm pretty
> sure you get something on the console log when you don't specify a CID
> when using the commandline.  Anyway:
>
> lcr 617642  default 6176421212
>
> should give you intralata.
>
> Note that the definition of intralata doesn't mean "local" for some
> providers.  Some providers define local to "same ratecenter" which is
> even more restrictive.
>
> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <mouncifbb at gmail.com>
> wrote:
> > i can't make use of mod_lcr using Intra/Interstate rating, I am using
> > svn: FreeSWITCH Version 1.0.trunk (16517)
> >
> > lcr mysql table structure:
> > CREATE TABLE `lcr` (
> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
> >   `digits` VARCHAR(15) DEFAULT NULL,
> >   `rate` FLOAT(11,5) DEFAULT NULL,
> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
> >   `carrier_id` INT(11) NOT NULL,
> >   `lead_strip` INT(11) NOT NULL,
> >   `trail_strip` INT(11) NOT NULL,
> >   `prefix` VARCHAR(16) NOT NULL,
> >   `suffix` VARCHAR(16) NOT NULL,
> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
> >   `quality` FLOAT(10,6) NOT NULL,
> >   `reliability` FLOAT(10,6) NOT NULL,
> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
> >   PRIMARY KEY  (`id`),
> >   KEY `carrier_id` (`carrier_id`),
> >   KEY `digits` (`digits`),
> >   KEY `lcr_profile` (`lcr_profile`),
> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
> `carriers`
> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
> >
> >
> > lcr_admin show profiles
> > Name:           default
> >  custom sql:    SELECT l.digits, c.carrier_name, l.${lcr_rate_field},
> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
> l.trail_strip,
> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN date_start
> AND
> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality DESC,
> >  reliability DESC, rand();
> >  has %:         false
> >  has vars:      true
> >  has intrastate:        true
> >  has intralata: true
> >  has npanxx:    true
> >  Reorder rate:  enabled
> >  Info in headers:       disabled
> >  Quote IN() List:       disabled
> >
> >
> >
> > lcr 617642  default  returns rate from the rate field table and not
> > intra/inter state fields rates.
> >
> > Any ideas? thanks!
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mouncifbb at gmail.com  Fri Jan 29 20:42:04 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Fri, 29 Jan 2010 23:42:04 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
Message-ID: <ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>

Also the Provider has presented the rates in this format?

NPANXXLATA OCN INTER INTRA


On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> Tried it and it's not giving me intralata  instead I get interstate, does
> the npa_nxx_company_ocn table needs to be used in this case?, also do I have
> to have the rate field in lcr table?
>
>
> lcr 617642  default 6176421212
>
>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp | Dialstring
>                                                        |
>  | 617642      | carrier1 | 0.00500  |       |            |
> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>
>
> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr is
> [617642  default 6176421212]
> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
> [6176421212]
> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing [state:0
> lata:0] so rate field is [rate]
> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
> (617642, 61764, 6176, 617, 61, 6) AND CURRENT_TIMESTAMP BETWEEN date_start
> AND date_end ORDER BY digits DESC,  rate, rand();
> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to head of
> list
> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>
>
> Thank you Rupa!
>
>
> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> turn console logging up to debug and redo the lcr lookup.  The sql
>> statements along with status info will show up.  This should give
>> enough information to debug what is happening.
>>
>> I'm assuming the npanxx table is actually populated and not just existing?
>>
>> When doing the lookup from the cli you have to tell lcr what CID to
>> use (remember, it is relative to the src/dest number).  I'm pretty
>> sure you get something on the console log when you don't specify a CID
>> when using the commandline.  Anyway:
>>
>> lcr 617642  default 6176421212
>>
>> should give you intralata.
>>
>> Note that the definition of intralata doesn't mean "local" for some
>> providers.  Some providers define local to "same ratecenter" which is
>> even more restrictive.
>>
>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <mouncifbb at gmail.com>
>> wrote:
>> > i can't make use of mod_lcr using Intra/Interstate rating, I am using
>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>> >
>> > lcr mysql table structure:
>> > CREATE TABLE `lcr` (
>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>> >   `digits` VARCHAR(15) DEFAULT NULL,
>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>> >   `carrier_id` INT(11) NOT NULL,
>> >   `lead_strip` INT(11) NOT NULL,
>> >   `trail_strip` INT(11) NOT NULL,
>> >   `prefix` VARCHAR(16) NOT NULL,
>> >   `suffix` VARCHAR(16) NOT NULL,
>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>> >   `quality` FLOAT(10,6) NOT NULL,
>> >   `reliability` FLOAT(10,6) NOT NULL,
>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>> >   PRIMARY KEY  (`id`),
>> >   KEY `carrier_id` (`carrier_id`),
>> >   KEY `digits` (`digits`),
>> >   KEY `lcr_profile` (`lcr_profile`),
>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>> `carriers`
>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>> >
>> >
>> > lcr_admin show profiles
>> > Name:           default
>> >  custom sql:    SELECT l.digits, c.carrier_name, l.${lcr_rate_field},
>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>> l.trail_strip,
>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN date_start
>> AND
>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality DESC,
>> >  reliability DESC, rand();
>> >  has %:         false
>> >  has vars:      true
>> >  has intrastate:        true
>> >  has intralata: true
>> >  has npanxx:    true
>> >  Reorder rate:  enabled
>> >  Info in headers:       disabled
>> >  Quote IN() List:       disabled
>> >
>> >
>> >
>> > lcr 617642  default  returns rate from the rate field table and not
>> > intra/inter state fields rates.
>> >
>> > Any ideas? thanks!
>> >
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From marketing at cluecon.com  Fri Jan 29 22:47:45 2010
From: marketing at cluecon.com (Michael Collins)
Date: Fri, 29 Jan 2010 22:47:45 -0800
Subject: [Freeswitch-users] ClueCon MMX - Save the Date!
Message-ID: <87f2f3b91001292247u5f058054yca8590bb9c39ae65@mail.gmail.com>

ClueCon MMX (2010) will be here before you know it! Please mark your
calendars: August 3-5, 2010. Start talking up ClueCon with your peers,
coworkers, business owners, CEOs, potential sponsors, and anyone else you
can think of. It's coming fast, so start getting ready now. Lots more
information will be coming soon.

Looking forward to seeing everyone this August!
-ClueCon Team
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From robin at swip.net  Fri Jan 29 07:31:29 2010
From: robin at swip.net (Robin Vleij)
Date: Fri, 29 Jan 2010 16:31:29 +0100
Subject: [Freeswitch-users] CPS per mod_sofia profile limited?
Message-ID: <4B62FF51.8070608@swip.net>

Hi guys,

Doing a bit of testing / benchmarking with FS 1.04 (and 1.05 SVN). Found
out some interesting things, I think.

The setup is like this:

SIPP Client -> FS -> SIPP Server

The dialplan is as simple as it gets:

<extension name="test-out">
  <condition field="network_addr" expression="^130\.244\.190\.150$"/>
  <condition field="destination_number" expression="^(\d+)$">
    <action application="set" data="proxy_media=true"/>
    <action application="set" data="hangup_after_bridge=true"/>
    <action application="bridge"
data="{sip_contact_user=transit}sofia/gateway/${distributor(gwg1)}/$1|sofia/gateway/${distributor(gwg2)}/$1"/>
  </condition>
</extension>

For the rest it's running CSV cdr's, commented out all modules I'm not
using, etc etc all that I could find on the wiki and the Interwebs.
Hardware / OS: from the shelve quadcore Xeon, debian 64-bit, 12GB
memory. SIPP is running a 500ms RTP pcap and the other side echos back.

I had a few test setups then:

1: FS SVN, 1 sofia profile where the gateways were configured and the
server_IP:5060 was used.

2. FS SVN, 2 sofia profiles where the gateways where in an seperate
profile (server_IP:5070) and the "customer facing" side was the original
profile.

3. FS 1.04, same as above

4. FS 1.04, 4 sofia profiles, distributor to spread load over 2 incoming
and 2 outgoing profiles.

Now the interesting thing was that under 1 I could go up, almost without
any CPU load, to 50cps. As soon as I went over this, calls where handled
 slower and "ongoing" calls would pile up untill it became really slow.
CPU load went to 100% on the FS process (both user and system time).
Lots of interupts and context switches. No throughput anymore untill I
lower and wait till the "buffer" is empty and FS is keeping up again.

Under 2, I was able to increase the CPS to about 100 with the same effect.

3 then went much better, I was able to increase CPS to about 200 cps and
response times in SIPP went up slighty untill it just hits some kind of
limit and calls are handled slower.

4 is pretty cool. Here I can run 2 sipp clients both doing 150cps to the
gateways that are spread on the distribution module, so I spread traffic
over 2 profiles. With 300 cps in total, FS is keeping up and I have 30%
idle CPU. However, increasing to over 300cps gives problems again, even
though I have idle CPU left!

All in all, I have a feeling that a single sip profile can't run more
than a certain limit untill it gets into some problem. Depending on if
I'm running SVN or 1.04 that limit seems to be 50cps or a bit higher.
After that limit it starts piling up "ongoing" calls, by taking time to
handle them and when that limit gets too high it's too late. All in all
really fine, I just set the system wide limit to a little under that
"threshold". But when I'm running just UNDER the threshold it's not CPU
that's a problem. Theoretically I should be able to run (based on the
CPU usage at 300cps) about 400cps.

When running at 300 I get SOME failed calls and I see

"switch_core_state_machine.c:525 a9a60636-0cea-11df-85a1-09c991f2afc5
Timeout waiting for next instruction in CS_NEW!"

in the console.

I didn't find much on how people do high cps setups and it feels a bit
like a "friday afternoon solution" to run multiple sofia profiles on the
same machine in order to max out the system.

Maybe I'm missing something and I know it's not an exact science this,
but I'm not sure "all is OK" because I'm not slowly getting to a 100%
cpu (or disk / network) usage, I hit some kind of limit after which
stuff goes wrong.

Anyone any input!

/Robin


From paul.gore.j at gmail.com  Fri Jan 29 20:20:46 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Fri, 29 Jan 2010 23:20:46 -0500
Subject: [Freeswitch-users] Logging question
Message-ID: <da4ab26e1001292020o280f330ft62a278b93c7fb1fd@mail.gmail.com>

Hi there,
I am running FS 1.0.trunk (14501) (I know it's old but we serve a small
community and don't have time to upgrade/test the latest/greatest). I am
having troubles understanding how to switch SIP trace in log files, I tried

fsctl loglevel debug
sofia tracelevel debug

but it seem to have no effect, I only get sofia debug messages but no
detailed SIP info.
What also puzzling me is if I do

console loglevel 0

I still get debug information on console.
What am I doing wrong?
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From mike at jerris.com  Fri Jan 29 23:50:45 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 Jan 2010 02:50:45 -0500
Subject: [Freeswitch-users] Hybrid Encryption?
In-Reply-To: <2b6116b31001281808x1a004cd2ne7e8dcb9f16fec3e@mail.gmail.com>
References: <2b6116b31001272154l3c0bbe80y8bf3db94961e8e1d@mail.gmail.com>
	<8A9EDC4E-C49B-488D-9DBF-169A185462AB@freeswitch.org>
	<2b6116b31001281631u7ada7876wc419bb7afadd7ef7@mail.gmail.com>
	<46C06209-9515-4B1B-B449-F55A51FF548B@freeswitch.org>
	<2b6116b31001281808x1a004cd2ne7e8dcb9f16fec3e@mail.gmail.com>
Message-ID: <B2BE2748-ADA4-4D1C-8004-99AD72FB7806@jerris.com>

Freeswitch isn't a proxy, and no, we don't provide support for passthrough auth like this.  A proxy would, but not sure of any proxy based solution that would do the srtp work for you.

Mike

On Jan 28, 2010, at 9:08 PM, Nicholas Lee wrote:

> Is there a way to do it transparently?  The FS proxies will past though the extension creds.
> 
> On Fri, Jan 29, 2010 at 1:52 PM, Brian West <brian at freeswitch.org> wrote:
> Then yes you could use FreeSWITCH to augment your Asterisk install and enable encryption from site to site.
> 
> /b
> 
> 
> > Unfortunately it's not going to cover every situation.
> >
> >
> > Nicholas
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Fri Jan 29 23:56:30 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 Jan 2010 02:56:30 -0500
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001281932h73466f87yf4715058f4632558@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
	<33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
	<191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>
	<33c87fa31001271939i668e9676y1f687dc182b4ca89@mail.gmail.com>
	<CF0D5382-EE44-48CB-BF14-27A443014336@freeswitch.org>
	<33c87fa31001272035t7c46e054t7ce2dd598fa40ebb@mail.gmail.com>
	<191c3a031001272125t2b19d134lbb65dfebbb7a52b9@mail.gmail.com>
	<33c87fa31001281932h73466f87yf4715058f4632558@mail.gmail.com>
Message-ID: <69FCAB67-B24A-4B10-B4E9-00A0FC55324E@jerris.com>

So they are in 2 different profiles then.  If you are doing them exactly the same as you said, the issue is your telling it profile internal, when it is really nat, so it does not find it.

Mike

On Jan 28, 2010, at 10:32 PM, Mark Campbell-Smith wrote:

> Hi !
> 
> I confirmed yesterday that if the SPA is not NAT'd, then the event is
> sent.  I just removed NAT from the extension that I was having
> problems with.
> 
> Looking at the db tables, it appears there are two - the
> sofia_reg_internal.db and sofia_reg_internal_nat.db
> 
> Could it be that the sendevent command is only looking in the
> sofia_reg_internal.db database and not sofia_reg_internal_nat.db ?
> 



From oseslija at gmail.com  Sat Jan 30 01:14:19 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sat, 30 Jan 2010 10:14:19 +0100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001281932h73466f87yf4715058f4632558@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
	<33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
	<191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>
	<33c87fa31001271939i668e9676y1f687dc182b4ca89@mail.gmail.com>
	<CF0D5382-EE44-48CB-BF14-27A443014336@freeswitch.org>
	<33c87fa31001272035t7c46e054t7ce2dd598fa40ebb@mail.gmail.com>
	<191c3a031001272125t2b19d134lbb65dfebbb7a52b9@mail.gmail.com>
	<33c87fa31001281932h73466f87yf4715058f4632558@mail.gmail.com>
Message-ID: <4468a6771001300114m68a18e4fk74627314ad7182fd@mail.gmail.com>

I have reboot working from fs_cli with the NATed SPA.

Regards,
Ognjen

On Fri, Jan 29, 2010 at 4:32 AM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Hi !
>
> I confirmed yesterday that if the SPA is not NAT'd, then the event is
> sent.  I just removed NAT from the extension that I was having
> problems with.
>
> Looking at the db tables, it appears there are two - the
> sofia_reg_internal.db and sofia_reg_internal_nat.db
>
> Could it be that the sendevent command is only looking in the
> sofia_reg_internal.db database and not sofia_reg_internal_nat.db ?
>
>
>
> On Thu, Jan 28, 2010 at 4:25 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > You have to look in the sql db and compare the specified vals with the
> ones
> > looked up from the event again the user and host need to match the db
> >
> > On Jan 27, 2010 10:41 PM, "Mark Campbell-Smith" <
> mcampbellsmith at gmail.com>
> > wrote:
> >
> > Hi Brian,
> >
> >  I've previously enabled siptrace for internal profile, but I see
> > nothing sent and nothing received.
> >
> > On Thu, Jan 28, 2010 at 2:54 PM, Brian West <brian at freeswitch.org>
> wrote:
> >> I'm suspecting the code...
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From errotan at gmail.com  Sat Jan 30 03:33:21 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Sat, 30 Jan 2010 12:33:21 +0100
Subject: [Freeswitch-users] CPS per mod_sofia profile limited?
In-Reply-To: <4B62FF51.8070608@swip.net>
References: <4B62FF51.8070608@swip.net>
Message-ID: <201001301233.21516.errotan@gmail.com>

2010. janu?r 29. 16.31.29 Robin Vleij d?tummal ezt ?rta:
> Hi guys,
> 
> Doing a bit of testing / benchmarking with FS 1.04 (and 1.05 SVN). Found
> out some interesting things, I think.
> 
> The setup is like this:
> 
> SIPP Client -> FS -> SIPP Server
> 
> The dialplan is as simple as it gets:
> 
> <extension name="test-out">
>   <condition field="network_addr" expression="^130\.244\.190\.150$"/>
>   <condition field="destination_number" expression="^(\d+)$">
>     <action application="set" data="proxy_media=true"/>
>     <action application="set" data="hangup_after_bridge=true"/>
>     <action application="bridge"
> data="{sip_contact_user=transit}sofia/gateway/${distributor(gwg1)}/$1|sofia
> /gateway/${distributor(gwg2)}/$1"/> </condition>
> </extension>
> 
> For the rest it's running CSV cdr's, commented out all modules I'm not
> using, etc etc all that I could find on the wiki and the Interwebs.
> Hardware / OS: from the shelve quadcore Xeon, debian 64-bit, 12GB
> memory. SIPP is running a 500ms RTP pcap and the other side echos back.
> 
> I had a few test setups then:
> 
> 1: FS SVN, 1 sofia profile where the gateways were configured and the
> server_IP:5060 was used.
> 
> 2. FS SVN, 2 sofia profiles where the gateways where in an seperate
> profile (server_IP:5070) and the "customer facing" side was the original
> profile.
> 
> 3. FS 1.04, same as above
> 
> 4. FS 1.04, 4 sofia profiles, distributor to spread load over 2 incoming
> and 2 outgoing profiles.
> 
> Now the interesting thing was that under 1 I could go up, almost without
> any CPU load, to 50cps. As soon as I went over this, calls where handled
>  slower and "ongoing" calls would pile up untill it became really slow.
> CPU load went to 100% on the FS process (both user and system time).
> Lots of interupts and context switches. No throughput anymore untill I
> lower and wait till the "buffer" is empty and FS is keeping up again.
> 
> Under 2, I was able to increase the CPS to about 100 with the same effect.
> 
> 3 then went much better, I was able to increase CPS to about 200 cps and
> response times in SIPP went up slighty untill it just hits some kind of
> limit and calls are handled slower.
> 
> 4 is pretty cool. Here I can run 2 sipp clients both doing 150cps to the
> gateways that are spread on the distribution module, so I spread traffic
> over 2 profiles. With 300 cps in total, FS is keeping up and I have 30%
> idle CPU. However, increasing to over 300cps gives problems again, even
> though I have idle CPU left!
> 
> All in all, I have a feeling that a single sip profile can't run more
> than a certain limit untill it gets into some problem. Depending on if
> I'm running SVN or 1.04 that limit seems to be 50cps or a bit higher.
> After that limit it starts piling up "ongoing" calls, by taking time to
> handle them and when that limit gets too high it's too late. All in all
> really fine, I just set the system wide limit to a little under that
> "threshold". But when I'm running just UNDER the threshold it's not CPU
> that's a problem. Theoretically I should be able to run (based on the
> CPU usage at 300cps) about 400cps.
> 
> When running at 300 I get SOME failed calls and I see
> 
> "switch_core_state_machine.c:525 a9a60636-0cea-11df-85a1-09c991f2afc5
> Timeout waiting for next instruction in CS_NEW!"
> 
> in the console.
> 
> I didn't find much on how people do high cps setups and it feels a bit
> like a "friday afternoon solution" to run multiple sofia profiles on the
> same machine in order to max out the system.
> 
> Maybe I'm missing something and I know it's not an exact science this,
> but I'm not sure "all is OK" because I'm not slowly getting to a 100%
> cpu (or disk / network) usage, I hit some kind of limit after which
> stuff goes wrong.
> 
> Anyone any input!
> 
> /Robin
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

CPU usage is not the only thing that limit your calls. Have you set the 
recommended ulimit settings and / or started fs with the -waste option ?

http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations


From codecomplete at free.fr  Sat Jan 30 05:53:25 2010
From: codecomplete at free.fr (Fred-145)
Date: Sat, 30 Jan 2010 14:53:25 +0100
Subject: [Freeswitch-users] Equivalent to
	Asterisk's	"directrtpsetup=yes"?
References: <gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b@4ax.com>
	<gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b-e09XROE/p8c@public.gmane.org>
	<20100129164359.9B3C22B10D@cuneorg-email.cune.pri>
Message-ID: <j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8@4ax.com>

On Fri, 29 Jan 2010 16:43:59 -0000,
<Russell.Mosemann at cune.org> wrote:
>The ports are open between the endpoint and Freeswitch. The ports are not
>open between the two endpoints themselves. If each endpoint is behind its
>own NAT, neither endpoint will be able to contact the other endpoint
>unless some kind of forwarding is set up on the firewall to map the
>external IP address and port to an internal IP address and port.

Thanks but the context I was refering to is...
1. Freeswitch is configured in BypassMedia mode
2. The firewall and the local end-points are configured so that a
series of UDP ranges are mapped to their respective end-point (eg.
UDP100-1003 for extension #1, 1004-1007 for #2, etc.)
... so that RTP packets flow directly between the two end-points

Brian says above that there might be cases where NAT could be a
problem. When could this happen?

I'd like to get to the bottom of this so that in case a server is a
bit short on CPU/network power, I know that there's the alternative of
RTP packets by-passing the server... but I also need to know what
issues this setup can cause.

Thank you.



From yehavi.bourvine at gmail.com  Sat Jan 30 06:01:22 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 30 Jan 2010 16:01:22 +0200
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001251206x21242589he3014081a7e110d1@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<D56D65EB-B150-4A59-8B66-68BF4C58108B@jerris.com>
	<d361ca7b1001241004kcb73f11hc400bdb37c9d372b@mail.gmail.com>
	<3A27F063-E0C0-4178-A3AF-068956B55846@jerris.com>
	<d361ca7b1001241117q7f1379bewe1bc85279ef67c83@mail.gmail.com>
	<alpine.GSO.2.00.1001250716120.17161@h1.pp.uhwv.np.vy>
	<d361ca7b1001242150t29b36ebcy6c2e40ca3d360c00@mail.gmail.com>
	<d361ca7b1001251143p7f396dd8h11d372731a26f4d7@mail.gmail.com>
	<224C684A-B357-42E4-98AA-0EE238A27A49@jerris.com>
	<d361ca7b1001251206x21242589he3014081a7e110d1@mail.gmail.com>
Message-ID: <d361ca7b1001300601r29530dcfm65c67e9121d3ddde@mail.gmail.com>

It works ok now (fixed on r16534).

                Thanks! __Yehavi:

2010/1/25 Yehavi Bourvine <yehavi.bourvine at gmail.com>

> OK. The sources are under /home/freeswitch. Note that it has Hebrew modules
> which we are now developing, but the problem is seen with the vanilla
> version as well.
>
> The execs and all other stuff is at /freeswitch.
>
>                       regards, __Yehavi:
>
> 2010/1/25 Michael Jerris <mike at jerris.com>
>
> I have not had a chance to actually try it yet.  I will let you know.
>>
>> Mike
>>
>> On Jan 25, 2010, at 2:43 PM, Yehavi Bourvine wrote:
>>
>> > Hello Mike,
>> >
>> >   I see you are logged-in into our machine. Before I go to sleep: Do you
>> need any help?
>> >
>> >                     Thanks, __yehavi:
>> >
>>
>>
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From dftoro at yahoo.com  Sat Jan 30 06:27:44 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Sat, 30 Jan 2010 06:27:44 -0800 (PST)
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <191c3a031001261426h7fd87e1fpf95824788d639557@mail.gmail.com>
Message-ID: <406990.22243.qm@web33501.mail.mud.yahoo.com>

Hi, 

With 4 slashes (\\\\) works fine

Thank you

Diego Toro
http://lacarretade.blogspot.com/


--- On Tue, 1/26/10, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] mutiple playback files (unescape_char) Windows
> To: freeswitch-users at lists.freeswitch.org
> Date: Tuesday, January 26, 2010, 5:26 PM
> please update again and try 4 slashes
> 
> you need 4 because the expand vars on the data=""
> will eat the 4 down to 2
> then the splitter on ! will turn \\s into \s
> 
> 
> 
> 
> 
> 
> On Tue, Jan 26, 2010 at 3:02 PM, Diego Toro <dftoro at yahoo.com>
> wrote:
> 
> Hi, sorry, I explain better. Using \\\\ is
> also changed when path matches a character such as
> \s,\n... My alternative on Windows is to use
> '/' like path separator.
> 
> 
> 
> 
> 
> Diego Toro
> 
> http://lacarretade.blogspot.com/
> 
> 
> 
> 
> 
> --- On Tue, 1/26/10, Anthony
> Minessale <anthony.minessale at gmail.com>
> wrote:
> 
> 
> 
> > From: Anthony Minessale <anthony.minessale at gmail.com>
> 
> > Subject: Re: [Freeswitch-users] mutiple playback files
> (unescape_char) Windows
> 
> > To: freeswitch-users at lists.freeswitch.org
> 
> > Date: Tuesday, January 26, 2010, 11:27 AM
> 
> > I didn't
> understand that
> 
> >
> 
> > On Tue, Jan 26, 2010 at 9:58 AM,
> 
> > Diego Toro <dftoro at yahoo.com>
> 
> > wrote:
> 
> >
> 
> > Hi, using \\\\ the is changed also
> when
> 
> > there is a match with an escape character
> 
> > (\s,\n...)
> 
> >
> 
> >
> 
> >
> 
> > Thank you
> 
> >
> 
> >
> 
> >
> 
> > Diego Toro
> 
> >
> 
> > http://lacarretade.blogspot.com/
> 
> >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> > --- On Mon, 1/25/10, Anthony Minessale <anthony.minessale at gmail.com>
> 
> > wrote:
> 
> >
> 
> >
> 
> >
> 
> > > From: Anthony Minessale <anthony.minessale at gmail.com>
> 
> >
> 
> > > Subject: Re: [Freeswitch-users]
> 
> > mutiple playback files (unescape_char) Windows
> 
> >
> 
> > > To: freeswitch-users at lists.freeswitch.org
> 
> >
> 
> > > Date: Monday, January 25, 2010, 12:20 PM
> 
> >
> 
> > > its possible your
> 
> > string hits the parser
> 
> >
> 
> > > more than once.
> 
> >
> 
> > > try using 4 \
> 
> >
> 
> > >
> 
> >
> 
> > > \\\\sound
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > > On Sun, Jan 24, 2010 at 4:03 AM,
> 
> >
> 
> > > Michael Jerris <mike at jerris.com>
> 
> >
> 
> > > wrote:
> 
> >
> 
> > >
> 
> >
> 
> > > As noted on that bug, you should be
> 
> >
> 
> > > able to either use \\ or / for the path
> 
> > separator
> 
> >
> 
> > > there and it should work.
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > > Mike
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > > On Jan 22, 2010, at 9:18 AM, Diego Toro wrote:
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > > > Hi, with svn r16440 the problem persists, I
> 
> > creted a
> 
> >
> 
> > > jira report http://jira.freeswitch.org/browse/LBSNDF-8
> 
> >
> 
> > > this is a minor issue, but activing playback
> delimiter
> 
> > no
> 
> >
> 
> > > audio file can be played. On FS the audio files
> are
> 
> > placed
> 
> >
> 
> > > in the \sound\ directory, building the
> path
> 
> > on
> 
> >
> 
> > > Windows would be \sound '\s'
> which is
> 
> >
> 
> > > replaced by 'ound'.
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > > >
> 
> >
> 
> > >
> 
> >
> 
> > > > Thank you
> 
> >
> 
> > >
> 
> >
> 
> > > >
> 
> >
> 
> > >
> 
> >
> 
> > > > Diego Toro
> 
> >
> 
> > >
> 
> >
> 
> > > > http://lacarretade.blogspot.com/
> 
> >
> 
> > >
> 
> >
> 
> > > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > > _______________________________________________
> 
> >
> 
> > >
> 
> >
> 
> > > FreeSWITCH-users mailing list
> 
> >
> 
> > >
> 
> >
> 
> > > FreeSWITCH-users at lists.freeswitch.org
> 
> >
> 
> > >
> 
> >
> 
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> >
> 
> > >
> 
> >
> 
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> >
> 
> > >
> 
> >
> 
> > > http://www.freeswitch.org
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > > --
> 
> >
> 
> > > Anthony Minessale II
> 
> >
> 
> > >
> 
> >
> 
> > > FreeSWITCH http://www.freeswitch.org/
> 
> >
> 
> > > ClueCon http://www.cluecon.com/
> 
> >
> 
> > >
> 
> >
> 
> > > Twitter: http://twitter.com/FreeSWITCH_wire
> 
> >
> 
> > >
> 
> >
> 
> > > AIM: anthm
> 
> >
> 
> > > MSN:anthony_minessale at hotmail.com
> 
> >
> 
> > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> 
> >
> 
> > >
> 
> >
> 
> > > IRC: irc.freenode.net
> 
> >
> 
> > > #freeswitch
> 
> >
> 
> > >
> 
> >
> 
> > > FreeSWITCH Developer Conference
> 
> >
> 
> > > sip:888 at conference.freeswitch.org
> 
> >
> 
> > > iax:guest at conference.freeswitch.org/888
> 
> >
> 
> > >
> 
> >
> 
> > > googletalk:conf+888 at conference.freeswitch.org
> 
> >
> 
> > > pstn:+19193869900
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > > -----Inline Attachment Follows-----
> 
> >
> 
> > >
> 
> >
> 
> > > _______________________________________________
> 
> >
> 
> > > FreeSWITCH-users mailing list
> 
> >
> 
> > > FreeSWITCH-users at lists.freeswitch.org
> 
> >
> 
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> >
> 
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> >
> 
> > > http://www.freeswitch.org
> 
> >
> 
> > >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> > _______________________________________________
> 
> >
> 
> > FreeSWITCH-users mailing list
> 
> >
> 
> > FreeSWITCH-users at lists.freeswitch.org
> 
> >
> 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> >
> 
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> >
> 
> > http://www.freeswitch.org
> 
> >
> 
> >
> 
> >
> 
> >
> 
> > --
> 
> > Anthony Minessale II
> 
> >
> 
> > FreeSWITCH http://www.freeswitch.org/
> 
> > ClueCon http://www.cluecon.com/
> 
> >
> 
> > Twitter: http://twitter.com/FreeSWITCH_wire
> 
> >
> 
> > AIM: anthm
> 
> > MSN:anthony_minessale at hotmail.com
> 
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> 
> >
> 
> > IRC: irc.freenode.net
> 
> > #freeswitch
> 
> >
> 
> > FreeSWITCH Developer Conference
> 
> > sip:888 at conference.freeswitch.org
> 
> > iax:guest at conference.freeswitch.org/888
> 
> >
> 
> > googletalk:conf+888 at conference.freeswitch.org
> 
> > pstn:+19193869900
> 
> >
> 
> >
> 
> > -----Inline Attachment Follows-----
> 
> >
> 
> > _______________________________________________
> 
> > FreeSWITCH-users mailing list
> 
> > FreeSWITCH-users at lists.freeswitch.org
> 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> > http://www.freeswitch.org
> 
> >
> 
> 
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> 
> FreeSWITCH-users mailing list
> 
> FreeSWITCH-users at lists.freeswitch.org
> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> 
> IRC: irc.freenode.net
> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> 
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From sharad at coraltele.com  Sat Jan 30 06:43:10 2010
From: sharad at coraltele.com (Sharad)
Date: Sat, 30 Jan 2010 06:43:10 -0800 (PST)
Subject: [Freeswitch-users] freeswitch with T.38
Message-ID: <1264862590045-4485495.post@n2.nabble.com>


Is paid freeswitch available with T.38 support. ?

regards
Sharad
-- 
View this message in context: http://n2.nabble.com/freeswitch-with-T-38-tp4485495p4485495.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lakindia89 at gmail.com  Sat Jan 30 06:48:53 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Sat, 30 Jan 2010 20:18:53 +0530
Subject: [Freeswitch-users] nixevent behavior
Message-ID: <7d79b3931001300648j6aa55258yfc496d9cea5c4b8b@mail.gmail.com>

Dear all

I've done the following sample script to experiment the nixevent. I found
some difference in behavior because of nixevent. Let me explain my question
down the script.

require ESL;
use IO::Socket::INET;

my $ip = "192.168.1.222";
my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
Proto => 'tcp',  Listen => 1,  Reuse => 1 );
die "Could not create socket: $!\n" unless $sock;
for(;;) {
    my $new_sock = $sock->accept();
    next if (not defined ($new_sock));
    my $pid = fork();
    if ($pid) {
        close($new_sock);
        next;
    }
    print "CHILD PID: $$\n";
    my $host = $new_sock->sockhost();
    my $fd = fileno($new_sock);

    my $con = new ESL::ESLconnection($fd);
    my $info = $con->getInfo();

    my $uuid = $info->getHeader("unique-id");

    printf "Connected call %s, from %s\n", $uuid,
$info->getHeader("caller-caller-id-number");
    my $r=$con->execute("answer");
    $con->events("plain","all");
##########################
    $con->send("nixevent DTMF");
    my $val=$con->api("create_uuid");
    $val = $val->getBody();            # LINE 1
    chomp($val);
    print "UUID is $val\n";
    my $e = $con->recvEvent();
    $val = $e->getBody();               # LINE 2
    chomp($val);
    print "UUID is $val\n";
    close($new_sock);
}

# If the line ($con->send("nixevent DTMF");) is commented, then the result
of create_uuid is obtained in LINE 1.
# else, the result isn't obtained in the LINE 1 and it has nothing. The
result is obtained only when I do a recvEvent,
# followed by a getBody (LINE 2)

Just want to know why the behavior differs when nixevent is present???
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From rupa at rupa.com  Sat Jan 30 07:02:31 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 30 Jan 2010 09:02:31 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
Message-ID: <b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>

Something is still missing from the logs. Note the query of the npanxx
table, the flags being set, and the rate field being chosen.  Umm..
oh, what version of fs are you running?

Yes, the npa_nxx_ocn table needs to be loaded up as described in:
http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
(there is a link to that from mod_lcr's wiki page).

An example from my own setup:

2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
is [12148267711 default 12148267712]
2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to [12148267712]
2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR (npa=214
AND nxx=826)
2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count: 1
2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
[state:1 lata:1] so rate field is [intralata_rate]
2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT l.digits
AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
l.intralata_rate as lcr_rate_field,        cg.prefix AS lcr_gw_prefix,
cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
 WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
date_start AND date_end   ORDER BY digits DESC, intralata_rate,
random();
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
[lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
head of list
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
[lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of list
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
[lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of list
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
[lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to end of list
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
[lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
[...]

On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <mouncifbb at gmail.com> wrote:
> Also the Provider has presented the rates in this format?
> NPANXXLATA OCN INTER INTRA
>
>
> On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <mouncifbb at gmail.com>
> wrote:
>>
>> Tried it and it's not giving me?intralata??instead I get interstate, does
>> the?npa_nxx_company_ocn table needs to be used in this case?, also do I have
>> to have the rate field in lcr table?
>>
>> lcr 617642 ?default 6176421212
>> ?| Digit Match | Carrier ?| Rate ? ? | Codec | CID Regexp | Dialstring
>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?|
>> ?| 617642 ? ? ?| carrier1 | 0.00500 ?| ? ? ? | ? ? ? ? ? ?|
>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>> [617642 ?default 6176421212]
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>> [6176421212]
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing [state:0
>> lata:0] so rate field is [rate]
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
>> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled
>> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND CURRENT_TIMESTAMP
>> BETWEEN date_start AND date_end ORDER BY digits DESC, ?rate, rand();
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>> of list
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>
>> Thank you Rupa!
>>
>> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>> turn console logging up to debug and redo the lcr lookup. ?The sql
>>> statements along with status info will show up. ?This should give
>>> enough information to debug what is happening.
>>>
>>> I'm assuming the npanxx table is actually populated and not just
>>> existing?
>>>
>>> When doing the lookup from the cli you have to tell lcr what CID to
>>> use (remember, it is relative to the src/dest number). ?I'm pretty
>>> sure you get something on the console log when you don't specify a CID
>>> when using the commandline. ?Anyway:
>>>
>>> lcr 617642 ?default 6176421212
>>>
>>> should give you intralata.
>>>
>>> Note that the definition of intralata doesn't mean "local" for some
>>> providers. ?Some providers define local to "same ratecenter" which is
>>> even more restrictive.
>>>
>>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <mouncifbb at gmail.com>
>>> wrote:
>>> > i can't make use of mod_lcr using Intra/Interstate rating, I am using
>>> > svn:?FreeSWITCH Version 1.0.trunk (16517)
>>> >
>>> > lcr mysql table structure:
>>> > CREATE TABLE `lcr` (
>>> > ??`id` INT(11) NOT NULL AUTO_INCREMENT,
>>> > ??`digits` VARCHAR(15) DEFAULT NULL,
>>> > ??`rate` FLOAT(11,5) DEFAULT NULL,
>>> > ??`intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>> > ??`intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>> > ??`carrier_id` INT(11) NOT NULL,
>>> > ??`lead_strip` INT(11) NOT NULL,
>>> > ??`trail_strip` INT(11) NOT NULL,
>>> > ??`prefix` VARCHAR(16) NOT NULL,
>>> > ??`suffix` VARCHAR(16) NOT NULL,
>>> > ??`lcr_profile` VARCHAR(32) DEFAULT NULL,
>>> > ??`date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>>> > ??`date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>> > ??`quality` FLOAT(10,6) NOT NULL,
>>> > ??`reliability` FLOAT(10,6) NOT NULL,
>>> > ??`cid` VARCHAR(32) NOT NULL DEFAULT '',
>>> > ??`enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>> > ??PRIMARY KEY ?(`id`),
>>> > ??KEY `carrier_id` (`carrier_id`),
>>> > ??KEY `digits` (`digits`),
>>> > ??KEY `lcr_profile` (`lcr_profile`),
>>> > ??KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>> > ??CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>>> > `carriers`
>>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>> >
>>> >
>>> > lcr_admin show profiles
>>> > Name: ? ? ? ? ? default
>>> > ?custom sql: ? ?SELECT l.digits, c.carrier_name, l.${lcr_rate_field},
>>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>> > l.trail_strip,
>>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
>>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
>>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN date_start
>>> > AND
>>> > date_end ORDER BY digits DESC, ?${lcr_rate_field}, ?quality DESC,
>>> > ?reliability DESC, rand();
>>> > ?has %: ? ? ? ? false
>>> > ?has vars: ? ? ?true
>>> > ?has intrastate: ? ? ? ?true
>>> > ?has intralata: true
>>> > ?has npanxx: ? ?true
>>> > ?Reorder rate: ?enabled
>>> > ?Info in headers: ? ? ? disabled
>>> > ?Quote IN() List: ? ? ? disabled
>>> >
>>> >
>>> >
>>> > lcr 617642 ?default ?returns rate from the rate field table and not
>>> > intra/inter state fields rates.
>>> >
>>> > Any ideas? thanks!
>>> >
>>> >
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From rupa at rupa.com  Sat Jan 30 07:03:48 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 30 Jan 2010 09:03:48 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
Message-ID: <b39273531001300703oe2d8289j6279eec3bdd3dc0d@mail.gmail.com>

Can you give me the first few lines of their rate table?

Is it:

NPANXXLATA = prefix
OCN = rate for same ocn
INTER = rate for interlata
INTRA = rate for intralata

or something else?

On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <mouncifbb at gmail.com> wrote:
> Also the Provider has presented the rates in this format?
> NPANXXLATA OCN INTER INTRA
>
>
> On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <mouncifbb at gmail.com>
> wrote:
>>
>> Tried it and it's not giving me?intralata??instead I get interstate, does
>> the?npa_nxx_company_ocn table needs to be used in this case?, also do I have
>> to have the rate field in lcr table?
>>
>> lcr 617642 ?default 6176421212
>> ?| Digit Match | Carrier ?| Rate ? ? | Codec | CID Regexp | Dialstring
>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?|
>> ?| 617642 ? ? ?| carrier1 | 0.00500 ?| ? ? ? | ? ? ? ? ? ?|
>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>> [617642 ?default 6176421212]
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>> [6176421212]
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing [state:0
>> lata:0] so rate field is [rate]
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
>> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled
>> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND CURRENT_TIMESTAMP
>> BETWEEN date_start AND date_end ORDER BY digits DESC, ?rate, rand();
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>> of list
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>
>> Thank you Rupa!
>>
>> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>> turn console logging up to debug and redo the lcr lookup. ?The sql
>>> statements along with status info will show up. ?This should give
>>> enough information to debug what is happening.
>>>
>>> I'm assuming the npanxx table is actually populated and not just
>>> existing?
>>>
>>> When doing the lookup from the cli you have to tell lcr what CID to
>>> use (remember, it is relative to the src/dest number). ?I'm pretty
>>> sure you get something on the console log when you don't specify a CID
>>> when using the commandline. ?Anyway:
>>>
>>> lcr 617642 ?default 6176421212
>>>
>>> should give you intralata.
>>>
>>> Note that the definition of intralata doesn't mean "local" for some
>>> providers. ?Some providers define local to "same ratecenter" which is
>>> even more restrictive.
>>>
>>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <mouncifbb at gmail.com>
>>> wrote:
>>> > i can't make use of mod_lcr using Intra/Interstate rating, I am using
>>> > svn:?FreeSWITCH Version 1.0.trunk (16517)
>>> >
>>> > lcr mysql table structure:
>>> > CREATE TABLE `lcr` (
>>> > ??`id` INT(11) NOT NULL AUTO_INCREMENT,
>>> > ??`digits` VARCHAR(15) DEFAULT NULL,
>>> > ??`rate` FLOAT(11,5) DEFAULT NULL,
>>> > ??`intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>> > ??`intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>> > ??`carrier_id` INT(11) NOT NULL,
>>> > ??`lead_strip` INT(11) NOT NULL,
>>> > ??`trail_strip` INT(11) NOT NULL,
>>> > ??`prefix` VARCHAR(16) NOT NULL,
>>> > ??`suffix` VARCHAR(16) NOT NULL,
>>> > ??`lcr_profile` VARCHAR(32) DEFAULT NULL,
>>> > ??`date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>>> > ??`date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>> > ??`quality` FLOAT(10,6) NOT NULL,
>>> > ??`reliability` FLOAT(10,6) NOT NULL,
>>> > ??`cid` VARCHAR(32) NOT NULL DEFAULT '',
>>> > ??`enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>> > ??PRIMARY KEY ?(`id`),
>>> > ??KEY `carrier_id` (`carrier_id`),
>>> > ??KEY `digits` (`digits`),
>>> > ??KEY `lcr_profile` (`lcr_profile`),
>>> > ??KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>> > ??CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>>> > `carriers`
>>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>> >
>>> >
>>> > lcr_admin show profiles
>>> > Name: ? ? ? ? ? default
>>> > ?custom sql: ? ?SELECT l.digits, c.carrier_name, l.${lcr_rate_field},
>>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>> > l.trail_strip,
>>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
>>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
>>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN date_start
>>> > AND
>>> > date_end ORDER BY digits DESC, ?${lcr_rate_field}, ?quality DESC,
>>> > ?reliability DESC, rand();
>>> > ?has %: ? ? ? ? false
>>> > ?has vars: ? ? ?true
>>> > ?has intrastate: ? ? ? ?true
>>> > ?has intralata: true
>>> > ?has npanxx: ? ?true
>>> > ?Reorder rate: ?enabled
>>> > ?Info in headers: ? ? ? disabled
>>> > ?Quote IN() List: ? ? ? disabled
>>> >
>>> >
>>> >
>>> > lcr 617642 ?default ?returns rate from the rate field table and not
>>> > intra/inter state fields rates.
>>> >
>>> > Any ideas? thanks!
>>> >
>>> >
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From abid_freeswitch at live.com  Fri Jan 29 23:48:38 2010
From: abid_freeswitch at live.com (Abid Saleem)
Date: Sat, 30 Jan 2010 12:48:38 +0500
Subject: [Freeswitch-users] SS7 & MGCP support
Message-ID: <SNT106-W1023694FA10CC0F668ADCDF45A0@phx.gbl>


Hi,

 

I am new to FreeSwitch. Please help me answer my following
questions I could not find on wiki documentation.

 

?        
Since it is a softswitch also, does it support
SS7, MGCP and Megaco protocols to control media gateways?

?        
Does it support call shops business model?

?        
How to add new SIP user accounts into it that
can be used to register to it. I know one way is to copy and paste          1000.xml
file and edit it in the conf/directory folder. What is the optimal way to do
this task?o   Is there any GUI available. If yes how can I make it work and private label it.? 	

 

Thanks for your great help.
Regards-----------Abid SaleemProduct ManagerComcerto Bahrain W.L.L 		 	   		  
_________________________________________________________________
Hotmail: Powerful Free email with security by Microsoft.
https://signup.live.com/signup.aspx?id=60969
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From mike at jerris.com  Sat Jan 30 08:12:35 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 Jan 2010 11:12:35 -0500
Subject: [Freeswitch-users] little hangup problem - prepaid application
In-Reply-To: <7abab2411001252034kaf90e64g97b5d5c8f7d65e20@mail.gmail.com>
References: <4B5E0608.3070001@xpirio.com>
	<201001251620.35308.sos@sokhapkin.dyndns.org>
	<7abab2411001251713p31e542b2xf6204ff13c556d03@mail.gmail.com>
	<201001252234.08162.sos@sokhapkin.dyndns.org>
	<7abab2411001252034kaf90e64g97b5d5c8f7d65e20@mail.gmail.com>
Message-ID: <11212980-94E0-4C75-BACC-D2069644D663@jerris.com>

http://wiki.freeswitch.org/wiki/Lua is a good place to start.

On Jan 25, 2010, at 11:34 PM, Frank Church wrote:

> That is new to me, does that mean that all the languages linked in
> with Freeswitch have access to the events and variables in FS at all
> times?
> 
> Can you link me to the documenation that describes this part in more
> detail and some examples?
> 
> 2010/1/26 Sergey Okhapkin <sos at sokhapkin.dyndns.org>:
>> No record is sent to the script, but the script has access to all channel
>> variables.
>> 
>> On Monday 25 January 2010, Frank Church wrote:
>>> I am new to Freeswitch and I am interested in how it works. When the
>>> record is sent to the lua program what format is it sent in?
>>> 
> Frank Church

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From mike at jerris.com  Sat Jan 30 08:19:59 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 Jan 2010 11:19:59 -0500
Subject: [Freeswitch-users] External Profile Problem
In-Reply-To: <BAY143-W267C2F235ED9FCFA619FA1A15D0@phx.gbl>
References: <BAY143-W20A3102CC2FF36EF1C170FA15D0@phx.gbl>,
	<443888.41110.qm@web33505.mail.mud.yahoo.com>
	<BAY143-W267C2F235ED9FCFA619FA1A15D0@phx.gbl>
Message-ID: <74639FD5-11D5-4828-9D14-1E659CC30F52@jerris.com>

Contexts are sets of dialplan rules.  This allows you to have different rules for different "contexts"  such as, people dialing from the outside world are in one context, your internal users are in another.

Mike

On Jan 27, 2010, at 10:48 AM, juan camilo ospina quintero wrote:

> hi thanks
> 
> sorry but i dont really understand what a context is.
> 
> so, when i put <action application="transfer" data="$0 XML default"/>
> what does it really does, what it means that transfer to new context,
> 
> 
> bye
> 
> > Date: Wed, 27 Jan 2010 05:27:38 -0800
> > From: dftoro at yahoo.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] External Profile Problem
> > 
> > Hi, 
> > 
> > <action application="transfer" data="$0 XML default"/>
> > 
> > You must take into account that transfer application not "transfer" a call to destination, only transfer it to a new context
> > 
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer
> > 
> > 
> > Diego Toro
> > http://lacarretade.blogspot.com/
> > 
> > 
> > --- On Tue, 1/26/10, juan camilo ospina quintero <camilin2212 at hotmail.com> wrote:
> > 
> > > From: juan camilo ospina quintero <camilin2212 at hotmail.com>
> > > Subject: Re: [Freeswitch-users] External Profile Problem
> > > To: "freeswitch" <freeswitch-users at lists.freeswitch.org>
> > > Date: Tuesday, January 26, 2010, 11:06 PM
> > > 
> > > 
> > > 
> > > 
> > > 
> > > Hi
> > > 
> > > This works fine
> > > 
> > > <extension name="redirect">
> > >  <condition field="network_addr"
> > > expression="^192\.168\.2\.9$"/> 
> > > 
> > >  <condition field="destination_number"
> > > expression="^1(\d+)$">
> > >  <action application="bridge"
> > > data="sofia/external/$0 at 127.0.0.1:5070"/>
> > > </extension>
> > > 
> > > this redirects from freeswitch to sailfin (127.0.0.1:5070),
> > > and is in default.xml, in the dialplan.
> > > 
> > > the problem is this
> > > 
> > > <extension name="receive">
> > > <condition field="network_addr"
> > > expression="^127\.0\.0\.1$"/>
> > > <condition field="destination_number"
> > > expression="^1(\d+)$">
> > > <action application="transfer"
> > > data="$0 XML default"/>
> > > </condition>
> > > </extension>
> > > this doesnt work, this configuration can be found in
> > > public.xml in the dialplan, the idea of
> > > this is that when a sip invite comes from sailfin
> > > (127.0.0.1) transfer the invite to the destination number
> > > 
> > > the both configurations above are the only configuration i
> > > have change from  the default instalation of 
> > > freeswitch.
> > > 
> > > i would like to have some hep with this thanks
> > > 
> > > here is the trace log again
> > > 
> > > 2010-01-26 20:14:29.512927 [NOTICE]
> > > switch_channel.c:602 New Channel sofia/external/1000
> > > at 192.168.2.9 [5177e93a-0ae1-11df-afc9-db39c681a2f1]
> > > 2010-01-26 20:14:29.512927 [NOTICE] sofia.c:3527 Hangup
> > > sofia/external/1000
> > > at 192.168.2.9 [CS_NEW] [MANDATORY_IE_MISSING]
> > > send 632 bytes to udp/[192.168.2.9]:5070 at
> > > 01:14:29.517927:
> > > 
> > > ------------------------------------------------------------------------
> > > SIP/2.0 480 Temporarily Unavailable
> > > Via: SIP/2.0/UDP
> > > 192.168.153.1:5070;branch=z9hG4bKdaacdd64d693615c451ab9db43f9c71c2626;received=192.168.2.9
> > > From: <sip:1000
> > > at 192.168.2.9>;tag=g4xfbi12-3
> > > To: <sip:1001
> > > at 192.168.2.9:5080>;tag=4r91165pvcycB
> > > Call-Id: 192.168.153.1_3_3990383226484831353
> > > Cseq: 1 INVITE
> > > User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
> > > Accept: application/sdp
> > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
> > > MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
> > > Supported: timer, precondition, path, replaces
> > > Allow-Events: talk, refer
> > > Reason:
> > > Q.850;cause=96;text="MANDATORY_IE_MISSING"
> > > Content-Length: 0
> > > 
> > > 
> > > ------------------------------------------------------------------------
> > > 2010-01-26 20:14:29.525646 [NOTICE]
> > > switch_core_session.c:1086 Session 9 (sofia/external/1000
> > > at 192.168.2.9) Ended
> > > 2010-01-26 20:14:29.525646 [NOTICE]
> > > switch_core_session.c:1088 Close Channel sofia/external/1000
> > > at 192.168.2.9 [CS_DESTROY]
> > > 
> > > 
> > > 
> > > > Date: Tue, 26 Jan 2010 22:09:29 -0500
> > > > From: frank at carmickle.com
> > > > To: freeswitch-users at lists.freeswitch.org
> > > > Subject: Re: [Freeswitch-users] External Profile
> > > Problem
> > > > 
> > > > On Tue, Jan 26, juan camilo ospina quintero wrote:
> > > > > 
> > > > > Hi, 
> > > > > 
> > > > > im trying to establish a simple conference using
> > > freeswitch and sailfin, sailfin is
> > > > > and application server that works with
> > > SipSevlets.
> > > > > the all thing works as follow.
> > > > > 
> > > > > two softphone register with freeswitch, extension
> > > 1000 and 1001
> > > > > 1000 sends and invite to 1001, this invite goes
> > > to sailfin, i use this
> > > > > 
> > > > > <extension name="redirect">
> > > > > <condition
> > > field="network_addr"
> > > expression="^192\.168\.2\.9$"/> 
> > > 
> > > > > <condition
> > > field="destination_number"
> > > expression="^1(\d+)$">
> > > > > <action application="bridge"
> > > data="sofia/external/$0 at 127.0.0.1:5070"/>
> > > > 
> > > > And what is the external profile listening on? 
> > > Probably not the loopback address. Set up another profile
> > > listening on 127.0.0.1 and bridge to that.
> > > > 
> > > > I could be off base here because you haven't given
> > > us very much info about your freeswitch configurations.
> > > > 
> > > > --FC
> > > > 
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> > > 
> > > Windows Live: Friends
> > > get your Flickr, Yelp, and Digg updates when they e-mail
> > > you. 
> > > 
> > > -----Inline Attachment Follows-----
> > > 
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > > 
> > 
> > 
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> Hotmail: Trusted email with powerful SPAM protection. Sign up now. _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Sat Jan 30 08:48:18 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 30 Jan 2010 10:48:18 -0600
Subject: [Freeswitch-users] nixevent behavior
In-Reply-To: <7d79b3931001300648j6aa55258yfc496d9cea5c4b8b@mail.gmail.com>
References: <7d79b3931001300648j6aa55258yfc496d9cea5c4b8b@mail.gmail.com>
Message-ID: <191c3a031001300848h65d65c9cg9b355cd07e922@mail.gmail.com>

use $e = $con->sendRecv("command");

every time
for each send you do you must do a recv so this does both.


On Sat, Jan 30, 2010 at 8:48 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Dear all
>
> I've done the following sample script to experiment the nixevent. I found
> some difference in behavior because of nixevent. Let me explain my question
> down the script.
>
> require ESL;
> use IO::Socket::INET;
>
> my $ip = "192.168.1.222";
> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
> Proto => 'tcp',  Listen => 1,  Reuse => 1 );
> die "Could not create socket: $!\n" unless $sock;
> for(;;) {
>     my $new_sock = $sock->accept();
>     next if (not defined ($new_sock));
>     my $pid = fork();
>     if ($pid) {
>         close($new_sock);
>         next;
>     }
>     print "CHILD PID: $$\n";
>     my $host = $new_sock->sockhost();
>     my $fd = fileno($new_sock);
>
>     my $con = new ESL::ESLconnection($fd);
>     my $info = $con->getInfo();
>
>     my $uuid = $info->getHeader("unique-id");
>
>     printf "Connected call %s, from %s\n", $uuid,
> $info->getHeader("caller-caller-id-number");
>     my $r=$con->execute("answer");
>     $con->events("plain","all");
> ##########################
>     $con->send("nixevent DTMF");
>     my $val=$con->api("create_uuid");
>     $val = $val->getBody();            # LINE 1
>     chomp($val);
>     print "UUID is $val\n";
>     my $e = $con->recvEvent();
>     $val = $e->getBody();               # LINE 2
>     chomp($val);
>     print "UUID is $val\n";
>     close($new_sock);
> }
>
> # If the line ($con->send("nixevent DTMF");) is commented, then the result
> of create_uuid is obtained in LINE 1.
> # else, the result isn't obtained in the LINE 1 and it has nothing. The
> result is obtained only when I do a recvEvent,
> # followed by a getBody (LINE 2)
>
> Just want to know why the behavior differs when nixevent is present???
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Sat Jan 30 08:54:02 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 30 Jan 2010 10:54:02 -0600
Subject: [Freeswitch-users] freeswitch with T.38
In-Reply-To: <1264862590045-4485495.post@n2.nabble.com>
References: <1264862590045-4485495.post@n2.nabble.com>
Message-ID: <191c3a031001300854r2dbc1cbeje06393183271b629@mail.gmail.com>

You can pay for commercial support contract to get priority bug fixes.
As for t38 you will have to wait for that feature a bit longer.


On Sat, Jan 30, 2010 at 8:43 AM, Sharad <sharad at coraltele.com> wrote:

>
> Is paid freeswitch available with T.38 support. ?
>
> regards
> Sharad
> --
> View this message in context:
> http://n2.nabble.com/freeswitch-with-T-38-tp4485495p4485495.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Sat Jan 30 08:57:02 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 30 Jan 2010 10:57:02 -0600
Subject: [Freeswitch-users] CPS per mod_sofia profile limited?
In-Reply-To: <201001301233.21516.errotan@gmail.com>
References: <4B62FF51.8070608@swip.net> <201001301233.21516.errotan@gmail.com>
Message-ID: <191c3a031001300857w36920b94r183f2e861db6baae@mail.gmail.com>

Also keep in mind that the industry standard is 50 which is the capacity to
take over for the real standard of 25 in a fail-over scenario.  So you
should be happy you even get 300cps for free.

The sofia stack can be improved but we are not the creators of this sip
stack.  There is little to no work being done on that project right now and
we are happy with what we have until we can get the lead dev to work on
improving it with us when he has the time.



On Sat, Jan 30, 2010 at 5:33 AM, Pusk?s Zsolt <errotan at gmail.com> wrote:

> 2010. janu?r 29. 16.31.29 Robin Vleij d?tummal ezt ?rta:
> > Hi guys,
> >
> > Doing a bit of testing / benchmarking with FS 1.04 (and 1.05 SVN). Found
> > out some interesting things, I think.
> >
> > The setup is like this:
> >
> > SIPP Client -> FS -> SIPP Server
> >
> > The dialplan is as simple as it gets:
> >
> > <extension name="test-out">
> >   <condition field="network_addr" expression="^130\.244\.190\.150$"/>
> >   <condition field="destination_number" expression="^(\d+)$">
> >     <action application="set" data="proxy_media=true"/>
> >     <action application="set" data="hangup_after_bridge=true"/>
> >     <action application="bridge"
> >
> data="{sip_contact_user=transit}sofia/gateway/${distributor(gwg1)}/$1|sofia
> > /gateway/${distributor(gwg2)}/$1"/> </condition>
> > </extension>
> >
> > For the rest it's running CSV cdr's, commented out all modules I'm not
> > using, etc etc all that I could find on the wiki and the Interwebs.
> > Hardware / OS: from the shelve quadcore Xeon, debian 64-bit, 12GB
> > memory. SIPP is running a 500ms RTP pcap and the other side echos back.
> >
> > I had a few test setups then:
> >
> > 1: FS SVN, 1 sofia profile where the gateways were configured and the
> > server_IP:5060 was used.
> >
> > 2. FS SVN, 2 sofia profiles where the gateways where in an seperate
> > profile (server_IP:5070) and the "customer facing" side was the original
> > profile.
> >
> > 3. FS 1.04, same as above
> >
> > 4. FS 1.04, 4 sofia profiles, distributor to spread load over 2 incoming
> > and 2 outgoing profiles.
> >
> > Now the interesting thing was that under 1 I could go up, almost without
> > any CPU load, to 50cps. As soon as I went over this, calls where handled
> >  slower and "ongoing" calls would pile up untill it became really slow.
> > CPU load went to 100% on the FS process (both user and system time).
> > Lots of interupts and context switches. No throughput anymore untill I
> > lower and wait till the "buffer" is empty and FS is keeping up again.
> >
> > Under 2, I was able to increase the CPS to about 100 with the same
> effect.
> >
> > 3 then went much better, I was able to increase CPS to about 200 cps and
> > response times in SIPP went up slighty untill it just hits some kind of
> > limit and calls are handled slower.
> >
> > 4 is pretty cool. Here I can run 2 sipp clients both doing 150cps to the
> > gateways that are spread on the distribution module, so I spread traffic
> > over 2 profiles. With 300 cps in total, FS is keeping up and I have 30%
> > idle CPU. However, increasing to over 300cps gives problems again, even
> > though I have idle CPU left!
> >
> > All in all, I have a feeling that a single sip profile can't run more
> > than a certain limit untill it gets into some problem. Depending on if
> > I'm running SVN or 1.04 that limit seems to be 50cps or a bit higher.
> > After that limit it starts piling up "ongoing" calls, by taking time to
> > handle them and when that limit gets too high it's too late. All in all
> > really fine, I just set the system wide limit to a little under that
> > "threshold". But when I'm running just UNDER the threshold it's not CPU
> > that's a problem. Theoretically I should be able to run (based on the
> > CPU usage at 300cps) about 400cps.
> >
> > When running at 300 I get SOME failed calls and I see
> >
> > "switch_core_state_machine.c:525 a9a60636-0cea-11df-85a1-09c991f2afc5
> > Timeout waiting for next instruction in CS_NEW!"
> >
> > in the console.
> >
> > I didn't find much on how people do high cps setups and it feels a bit
> > like a "friday afternoon solution" to run multiple sofia profiles on the
> > same machine in order to max out the system.
> >
> > Maybe I'm missing something and I know it's not an exact science this,
> > but I'm not sure "all is OK" because I'm not slowly getting to a 100%
> > cpu (or disk / network) usage, I hit some kind of limit after which
> > stuff goes wrong.
> >
> > Anyone any input!
> >
> > /Robin
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> CPU usage is not the only thing that limit your calls. Have you set the
> recommended ulimit settings and / or started fs with the -waste option ?
>
> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From mouncifbb at gmail.com  Sat Jan 30 13:38:22 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Sat, 30 Jan 2010 16:38:22 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
Message-ID: <ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>

  NPANXX,"LATA","OCN","NTER","INTRA"  201007,"224","7229","0.0059","0.0127"
201040,"224","9206","0.0036","0.0036"
FreeSWITCH Version 1.0.trunk (16540)


Also I noticed the *npa_nxx_ocn* table never get consulted.

I also see this now when making a real call instead of running thorugh CLI

EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890 NANPA_STD)
2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate channel var is
[undef]*
2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based on
interstate rates
2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on 16179470893
using profile NANPA_STD
2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing [state:0
lata:0] so rate field is [rate]

called number 6179470890  caller ID: 6179472456

any ideas??




On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> Something is still missing from the logs. Note the query of the npanxx
> table, the flags being set, and the rate field being chosen.  Umm..
> oh, what version of fs are you running?
>
> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>
> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
> (there is a link to that from mod_lcr's wiki page).
>
> An example from my own setup:
>
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
> is [12148267711 default 12148267712]
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
> [12148267712]
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR (npa=214
> AND nxx=826)
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count: 1
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
> [state:1 lata:1] so rate field is [intralata_rate]
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT l.digits
> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
> l.intralata_rate as lcr_rate_field,        cg.prefix AS lcr_gw_prefix,
> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
> random();
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>
> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
> head of list
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of
> list
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of
> list
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to end of
> list
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>
> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
> [...]
>
> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <mouncifbb at gmail.com>
> wrote:
> > Also the Provider has presented the rates in this format?
> > NPANXXLATA OCN INTER INTRA
> >
> >
> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <mouncifbb at gmail.com>
> > wrote:
> >>
> >> Tried it and it's not giving me intralata  instead I get interstate,
> does
> >> the npa_nxx_company_ocn table needs to be used in this case?, also do I
> have
> >> to have the rate field in lcr table?
> >>
> >> lcr 617642  default 6176421212
> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp | Dialstring
> >>                                                        |
> >>  | 617642      | carrier1 | 0.00500  |       |            |
> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
> >>
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr is
> >> [617642  default 6176421212]
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
> >> [6176421212]
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing [state:0
> >> lata:0] so rate field is [rate]
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM
> lcr
> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
> l.enabled
> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
> CURRENT_TIMESTAMP
> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate, rand();
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
> >> of list
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
> >>
> >> Thank you Rupa!
> >>
> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> >>>
> >>> turn console logging up to debug and redo the lcr lookup.  The sql
> >>> statements along with status info will show up.  This should give
> >>> enough information to debug what is happening.
> >>>
> >>> I'm assuming the npanxx table is actually populated and not just
> >>> existing?
> >>>
> >>> When doing the lookup from the cli you have to tell lcr what CID to
> >>> use (remember, it is relative to the src/dest number).  I'm pretty
> >>> sure you get something on the console log when you don't specify a CID
> >>> when using the commandline.  Anyway:
> >>>
> >>> lcr 617642  default 6176421212
> >>>
> >>> should give you intralata.
> >>>
> >>> Note that the definition of intralata doesn't mean "local" for some
> >>> providers.  Some providers define local to "same ratecenter" which is
> >>> even more restrictive.
> >>>
> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <mouncifbb at gmail.com
> >
> >>> wrote:
> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I am using
> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
> >>> >
> >>> > lcr mysql table structure:
> >>> > CREATE TABLE `lcr` (
> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
> >>> >   `carrier_id` INT(11) NOT NULL,
> >>> >   `lead_strip` INT(11) NOT NULL,
> >>> >   `trail_strip` INT(11) NOT NULL,
> >>> >   `prefix` VARCHAR(16) NOT NULL,
> >>> >   `suffix` VARCHAR(16) NOT NULL,
> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
> >>> >   `quality` FLOAT(10,6) NOT NULL,
> >>> >   `reliability` FLOAT(10,6) NOT NULL,
> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
> >>> >   PRIMARY KEY  (`id`),
> >>> >   KEY `carrier_id` (`carrier_id`),
> >>> >   KEY `digits` (`digits`),
> >>> >   KEY `lcr_profile` (`lcr_profile`),
> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
> >>> > `carriers`
> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
> >>> >
> >>> >
> >>> > lcr_admin show profiles
> >>> > Name:           default
> >>> >  custom sql:    SELECT l.digits, c.carrier_name, l.${lcr_rate_field},
> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
> >>> > l.trail_strip,
> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
> WHERE
> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits
> IN
> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
> date_start
> >>> > AND
> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality DESC,
> >>> >  reliability DESC, rand();
> >>> >  has %:         false
> >>> >  has vars:      true
> >>> >  has intrastate:        true
> >>> >  has intralata: true
> >>> >  has npanxx:    true
> >>> >  Reorder rate:  enabled
> >>> >  Info in headers:       disabled
> >>> >  Quote IN() List:       disabled
> >>> >
> >>> >
> >>> >
> >>> > lcr 617642  default  returns rate from the rate field table and not
> >>> > intra/inter state fields rates.
> >>> >
> >>> > Any ideas? thanks!
> >>> >
> >>> >
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>>
> >>>
> >>> --
> >>> -Rupa
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rupa at rupa.com  Sat Jan 30 15:59:27 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 30 Jan 2010 17:59:27 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
Message-ID: <b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>

Stuff inline.

On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

>  NPANXX,"LATA","OCN","NTER","INTRA"  201007,"224","7229","0.0059","0.0127"
> 201040,"224","9206","0.0036","0.0036"
>

Looks like they give you the LATA and OCN values with the prefix.  We
(should) look that up ourselves.


> FreeSWITCH Version 1.0.trunk (16540)
>
>
> Also I noticed the *npa_nxx_ocn* table never get consulted.
>
> I also see this now when making a real call instead of running thorugh CLI
>
> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890 NANPA_STD)
> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate channel var
> is [undef]*


This is fine.  it is a leftover from when you would tell mod_lcr via a
channel var that it should do intrastate.  I later had mod_lcr do the lookup
itself, but we still honor the old var.  There are no channel vars
associated with the cli, so you wouldn't see that msg.


>
> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based on
> interstate rates
> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on 16179470893
> using profile NANPA_STD
> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing [state:0
> lata:0] so rate field is [rate]
>
> called number 6179470890  caller ID: 6179472456
>
> any ideas??
>
>
Only thing that jumps out at me.

The output from lcr_admin show profiles showed only the default one.  On the
dialplan you use the NANPA_STD profile.  Can you check lcr_admin list and
see if that profile is defined and if so if it says it is using the npanxx
table?




>
>
>
>
> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> Something is still missing from the logs. Note the query of the npanxx
>> table, the flags being set, and the rate field being chosen.  Umm..
>> oh, what version of fs are you running?
>>
>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>
>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>> (there is a link to that from mod_lcr's wiki page).
>>
>> An example from my own setup:
>>
>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>> is [12148267711 default 12148267712]
>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>> [12148267712]
>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR (npa=214
>> AND nxx=826)
>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count: 1
>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>> [state:1 lata:1] so rate field is [intralata_rate]
>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT l.digits
>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>> l.intralata_rate as lcr_rate_field,        cg.prefix AS lcr_gw_prefix,
>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>> random();
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>
>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
>> head of list
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of
>> list
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of
>> list
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to end of
>> list
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>
>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>> [...]
>>
>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <mouncifbb at gmail.com>
>> wrote:
>> > Also the Provider has presented the rates in this format?
>> > NPANXXLATA OCN INTER INTRA
>> >
>> >
>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <mouncifbb at gmail.com
>> >
>> > wrote:
>> >>
>> >> Tried it and it's not giving me intralata  instead I get interstate,
>> does
>> >> the npa_nxx_company_ocn table needs to be used in this case?, also do I
>> have
>> >> to have the rate field in lcr table?
>> >>
>> >> lcr 617642  default 6176421212
>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp | Dialstring
>> >>                                                        |
>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>> >>
>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>> >> [617642  default 6176421212]
>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>> >> [6176421212]
>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing [state:0
>> >> lata:0] so rate field is [rate]
>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM
>> lcr
>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
>> l.enabled
>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>> CURRENT_TIMESTAMP
>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate, rand();
>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to
>> head
>> >> of list
>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>> >>
>> >> Thank you Rupa!
>> >>
>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>> >>>
>> >>> turn console logging up to debug and redo the lcr lookup.  The sql
>> >>> statements along with status info will show up.  This should give
>> >>> enough information to debug what is happening.
>> >>>
>> >>> I'm assuming the npanxx table is actually populated and not just
>> >>> existing?
>> >>>
>> >>> When doing the lookup from the cli you have to tell lcr what CID to
>> >>> use (remember, it is relative to the src/dest number).  I'm pretty
>> >>> sure you get something on the console log when you don't specify a CID
>> >>> when using the commandline.  Anyway:
>> >>>
>> >>> lcr 617642  default 6176421212
>> >>>
>> >>> should give you intralata.
>> >>>
>> >>> Note that the definition of intralata doesn't mean "local" for some
>> >>> providers.  Some providers define local to "same ratecenter" which is
>> >>> even more restrictive.
>> >>>
>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <
>> mouncifbb at gmail.com>
>> >>> wrote:
>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I am
>> using
>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>> >>> >
>> >>> > lcr mysql table structure:
>> >>> > CREATE TABLE `lcr` (
>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>> >>> >   `carrier_id` INT(11) NOT NULL,
>> >>> >   `lead_strip` INT(11) NOT NULL,
>> >>> >   `trail_strip` INT(11) NOT NULL,
>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>> >>> >   PRIMARY KEY  (`id`),
>> >>> >   KEY `carrier_id` (`carrier_id`),
>> >>> >   KEY `digits` (`digits`),
>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>> >>> > `carriers`
>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>> >>> >
>> >>> >
>> >>> > lcr_admin show profiles
>> >>> > Name:           default
>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>> l.${lcr_rate_field},
>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>> >>> > l.trail_strip,
>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>> WHERE
>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits
>> IN
>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>> date_start
>> >>> > AND
>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality DESC,
>> >>> >  reliability DESC, rand();
>> >>> >  has %:         false
>> >>> >  has vars:      true
>> >>> >  has intrastate:        true
>> >>> >  has intralata: true
>> >>> >  has npanxx:    true
>> >>> >  Reorder rate:  enabled
>> >>> >  Info in headers:       disabled
>> >>> >  Quote IN() List:       disabled
>> >>> >
>> >>> >
>> >>> >
>> >>> > lcr 617642  default  returns rate from the rate field table and not
>> >>> > intra/inter state fields rates.
>> >>> >
>> >>> > Any ideas? thanks!
>> >>> >
>> >>> >
>> >>> >
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> -Rupa
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From mouncifbb at gmail.com  Sat Jan 30 16:23:33 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Sat, 30 Jan 2010 19:23:33 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
Message-ID: <ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>

yes I use NANPA_STD profile instead of default cause I thought the custom
profile was causing issues, but looks like it's returning same results.

There is this line in thw wiki:
intra lata/state selection is done manually by setting the channel variables
*intrastate* or *intralata* to the value *true*.

do I have to set these ? if yes how?

Thanks

On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Stuff inline.
>
> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>
>>  NPANXX,"LATA","OCN","NTER","INTRA"
>> 201007,"224","7229","0.0059","0.0127"
>> 201040,"224","9206","0.0036","0.0036"
>>
>
> Looks like they give you the LATA and OCN values with the prefix.  We
> (should) look that up ourselves.
>
>
>> FreeSWITCH Version 1.0.trunk (16540)
>>
>>
>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>
>> I also see this now when making a real call instead of running thorugh CLI
>>
>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890 NANPA_STD)
>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate channel var
>> is [undef]*
>
>
> This is fine.  it is a leftover from when you would tell mod_lcr via a
> channel var that it should do intrastate.  I later had mod_lcr do the lookup
> itself, but we still honor the old var.  There are no channel vars
> associated with the cli, so you wouldn't see that msg.
>
>
>>
>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based on
>> interstate rates
>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>> 16179470893 using profile NANPA_STD
>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing [state:0
>> lata:0] so rate field is [rate]
>>
>> called number 6179470890  caller ID: 6179472456
>>
>> any ideas??
>>
>>
> Only thing that jumps out at me.
>
> The output from lcr_admin show profiles showed only the default one.  On
> the dialplan you use the NANPA_STD profile.  Can you check lcr_admin list
> and see if that profile is defined and if so if it says it is using the
> npanxx table?
>
>
>
>
>>
>>
>>
>>
>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>> Something is still missing from the logs. Note the query of the npanxx
>>> table, the flags being set, and the rate field being chosen.  Umm..
>>> oh, what version of fs are you running?
>>>
>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>> (there is a link to that from mod_lcr's wiki page).
>>>
>>> An example from my own setup:
>>>
>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>>> is [12148267711 default 12148267712]
>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>> [12148267712]
>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR (npa=214
>>> AND nxx=826)
>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count: 1
>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>> [state:1 lata:1] so rate field is [intralata_rate]
>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT l.digits
>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS lcr_gw_prefix,
>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>> random();
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>
>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
>>> head of list
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of
>>> list
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of
>>> list
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to end
>>> of list
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>
>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>> [...]
>>>
>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <mouncifbb at gmail.com>
>>> wrote:
>>> > Also the Provider has presented the rates in this format?
>>> > NPANXXLATA OCN INTER INTRA
>>> >
>>> >
>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <
>>> mouncifbb at gmail.com>
>>> > wrote:
>>> >>
>>> >> Tried it and it's not giving me intralata  instead I get interstate,
>>> does
>>> >> the npa_nxx_company_ocn table needs to be used in this case?, also do
>>> I have
>>> >> to have the rate field in lcr table?
>>> >>
>>> >> lcr 617642  default 6176421212
>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp | Dialstring
>>> >>                                                        |
>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>> |
>>> >>
>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr
>>> is
>>> >> [617642  default 6176421212]
>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>> >> [6176421212]
>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>> [state:0
>>> >> lata:0] so rate field is [rate]
>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>> gw_suffix,
>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid
>>> FROM lcr
>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
>>> l.enabled
>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>> CURRENT_TIMESTAMP
>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate, rand();
>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to
>>> head
>>> >> of list
>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>> >>
>>> >> Thank you Rupa!
>>> >>
>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com>
>>> wrote:
>>> >>>
>>> >>> turn console logging up to debug and redo the lcr lookup.  The sql
>>> >>> statements along with status info will show up.  This should give
>>> >>> enough information to debug what is happening.
>>> >>>
>>> >>> I'm assuming the npanxx table is actually populated and not just
>>> >>> existing?
>>> >>>
>>> >>> When doing the lookup from the cli you have to tell lcr what CID to
>>> >>> use (remember, it is relative to the src/dest number).  I'm pretty
>>> >>> sure you get something on the console log when you don't specify a
>>> CID
>>> >>> when using the commandline.  Anyway:
>>> >>>
>>> >>> lcr 617642  default 6176421212
>>> >>>
>>> >>> should give you intralata.
>>> >>>
>>> >>> Note that the definition of intralata doesn't mean "local" for some
>>> >>> providers.  Some providers define local to "same ratecenter" which is
>>> >>> even more restrictive.
>>> >>>
>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <
>>> mouncifbb at gmail.com>
>>> >>> wrote:
>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I am
>>> using
>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>> >>> >
>>> >>> > lcr mysql table structure:
>>> >>> > CREATE TABLE `lcr` (
>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>> >>> >   PRIMARY KEY  (`id`),
>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>> >>> >   KEY `digits` (`digits`),
>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>>> >>> > `carriers`
>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>> >>> >
>>> >>> >
>>> >>> > lcr_admin show profiles
>>> >>> > Name:           default
>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>> l.${lcr_rate_field},
>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>> >>> > l.trail_strip,
>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>> WHERE
>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits
>>> IN
>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>> date_start
>>> >>> > AND
>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality DESC,
>>> >>> >  reliability DESC, rand();
>>> >>> >  has %:         false
>>> >>> >  has vars:      true
>>> >>> >  has intrastate:        true
>>> >>> >  has intralata: true
>>> >>> >  has npanxx:    true
>>> >>> >  Reorder rate:  enabled
>>> >>> >  Info in headers:       disabled
>>> >>> >  Quote IN() List:       disabled
>>> >>> >
>>> >>> >
>>> >>> >
>>> >>> > lcr 617642  default  returns rate from the rate field table and not
>>> >>> > intra/inter state fields rates.
>>> >>> >
>>> >>> > Any ideas? thanks!
>>> >>> >
>>> >>> >
>>> >>> >
>>> >>> >
>>> >>> > _______________________________________________
>>> >>> > FreeSWITCH-users mailing list
>>> >>> > FreeSWITCH-users at lists.freeswitch.org
>>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >
>>> >>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> > http://www.freeswitch.org
>>> >>> >
>>> >>> >
>>> >>>
>>> >>>
>>> >>>
>>> >>> --
>>> >>> -Rupa
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rupa at rupa.com  Sat Jan 30 16:45:35 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 30 Jan 2010 18:45:35 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
Message-ID: <b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>

turn up logging to debug again, and then reload mod_lcr.  It'll spit out a
bunch of crap when it tests out each profile you have defined.  Give me the
full log (here or in pastebin.freeswitch.org).  That may show more useful
info as to why things are mucked up?

On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> yes I use NANPA_STD profile instead of default cause I thought the custom
> profile was causing issues, but looks like it's returning same results.
>
> There is this line in thw wiki:
> intra lata/state selection is done manually by setting the channel
> variables *intrastate* or *intralata* to the value *true*.
>
> do I have to set these ? if yes how?
>
> Thanks
>
>
> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> Stuff inline.
>>
>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>>
>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>> 201007,"224","7229","0.0059","0.0127"
>>> 201040,"224","9206","0.0036","0.0036"
>>>
>>
>> Looks like they give you the LATA and OCN values with the prefix.  We
>> (should) look that up ourselves.
>>
>>
>>> FreeSWITCH Version 1.0.trunk (16540)
>>>
>>>
>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>
>>> I also see this now when making a real call instead of running thorugh
>>> CLI
>>>
>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890 NANPA_STD)
>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate channel
>>> var is [undef]*
>>
>>
>> This is fine.  it is a leftover from when you would tell mod_lcr via a
>> channel var that it should do intrastate.  I later had mod_lcr do the lookup
>> itself, but we still honor the old var.  There are no channel vars
>> associated with the cli, so you wouldn't see that msg.
>>
>>
>>>
>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based on
>>> interstate rates
>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>> 16179470893 using profile NANPA_STD
>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>> lata:0] so rate field is [rate]
>>>
>>> called number 6179470890  caller ID: 6179472456
>>>
>>> any ideas??
>>>
>>>
>> Only thing that jumps out at me.
>>
>> The output from lcr_admin show profiles showed only the default one.  On
>> the dialplan you use the NANPA_STD profile.  Can you check lcr_admin list
>> and see if that profile is defined and if so if it says it is using the
>> npanxx table?
>>
>>
>>
>>
>>>
>>>
>>>
>>>
>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>>> Something is still missing from the logs. Note the query of the npanxx
>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>> oh, what version of fs are you running?
>>>>
>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>
>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>> (there is a link to that from mod_lcr's wiki page).
>>>>
>>>> An example from my own setup:
>>>>
>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>>>> is [12148267711 default 12148267712]
>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>> [12148267712]
>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR (npa=214
>>>> AND nxx=826)
>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count: 1
>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT l.digits
>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS lcr_gw_prefix,
>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>> random();
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>
>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
>>>> head of list
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>
>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of
>>>> list
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of
>>>> list
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>
>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to end
>>>> of list
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>
>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>> [...]
>>>>
>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <mouncifbb at gmail.com>
>>>> wrote:
>>>> > Also the Provider has presented the rates in this format?
>>>> > NPANXXLATA OCN INTER INTRA
>>>> >
>>>> >
>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <
>>>> mouncifbb at gmail.com>
>>>> > wrote:
>>>> >>
>>>> >> Tried it and it's not giving me intralata  instead I get interstate,
>>>> does
>>>> >> the npa_nxx_company_ocn table needs to be used in this case?, also do
>>>> I have
>>>> >> to have the rate field in lcr table?
>>>> >>
>>>> >> lcr 617642  default 6176421212
>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>> Dialstring
>>>> >>                                                        |
>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>> |
>>>> >>
>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr
>>>> is
>>>> >> [617642  default 6176421212]
>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>> >> [6176421212]
>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>> [state:0
>>>> >> lata:0] so rate field is [rate]
>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>> l.digits,
>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>> gw_suffix,
>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid
>>>> FROM lcr
>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
>>>> l.enabled
>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>> CURRENT_TIMESTAMP
>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate, rand();
>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to
>>>> head
>>>> >> of list
>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>> >>
>>>> >> Thank you Rupa!
>>>> >>
>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com>
>>>> wrote:
>>>> >>>
>>>> >>> turn console logging up to debug and redo the lcr lookup.  The sql
>>>> >>> statements along with status info will show up.  This should give
>>>> >>> enough information to debug what is happening.
>>>> >>>
>>>> >>> I'm assuming the npanxx table is actually populated and not just
>>>> >>> existing?
>>>> >>>
>>>> >>> When doing the lookup from the cli you have to tell lcr what CID to
>>>> >>> use (remember, it is relative to the src/dest number).  I'm pretty
>>>> >>> sure you get something on the console log when you don't specify a
>>>> CID
>>>> >>> when using the commandline.  Anyway:
>>>> >>>
>>>> >>> lcr 617642  default 6176421212
>>>> >>>
>>>> >>> should give you intralata.
>>>> >>>
>>>> >>> Note that the definition of intralata doesn't mean "local" for some
>>>> >>> providers.  Some providers define local to "same ratecenter" which
>>>> is
>>>> >>> even more restrictive.
>>>> >>>
>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <
>>>> mouncifbb at gmail.com>
>>>> >>> wrote:
>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I am
>>>> using
>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>> >>> >
>>>> >>> > lcr mysql table structure:
>>>> >>> > CREATE TABLE `lcr` (
>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>> >>> >   PRIMARY KEY  (`id`),
>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>> >>> >   KEY `digits` (`digits`),
>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>>>> >>> > `carriers`
>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>> >>> >
>>>> >>> >
>>>> >>> > lcr_admin show profiles
>>>> >>> > Name:           default
>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>> l.${lcr_rate_field},
>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>> >>> > l.trail_strip,
>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c
>>>> ON
>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>> WHERE
>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>> digits IN
>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>> date_start
>>>> >>> > AND
>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality DESC,
>>>> >>> >  reliability DESC, rand();
>>>> >>> >  has %:         false
>>>> >>> >  has vars:      true
>>>> >>> >  has intrastate:        true
>>>> >>> >  has intralata: true
>>>> >>> >  has npanxx:    true
>>>> >>> >  Reorder rate:  enabled
>>>> >>> >  Info in headers:       disabled
>>>> >>> >  Quote IN() List:       disabled
>>>> >>> >
>>>> >>> >
>>>> >>> >
>>>> >>> > lcr 617642  default  returns rate from the rate field table and
>>>> not
>>>> >>> > intra/inter state fields rates.
>>>> >>> >
>>>> >>> > Any ideas? thanks!
>>>> >>> >
>>>> >>> >
>>>> >>> >
>>>> >>> >
>>>> >>> > _______________________________________________
>>>> >>> > FreeSWITCH-users mailing list
>>>> >>> > FreeSWITCH-users at lists.freeswitch.org
>>>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>> >
>>>> >>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>> > http://www.freeswitch.org
>>>> >>> >
>>>> >>> >
>>>> >>>
>>>> >>>
>>>> >>>
>>>> >>> --
>>>> >>> -Rupa
>>>> >>>
>>>> >>> _______________________________________________
>>>> >>> FreeSWITCH-users mailing list
>>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>> http://www.freeswitch.org
>>>> >>
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From jwssr at charter.net  Sat Jan 30 10:11:42 2010
From: jwssr at charter.net (jwssr)
Date: Sat, 30 Jan 2010 10:11:42 -0800 (PST)
Subject: [Freeswitch-users]  problem with hard phone
Message-ID: <27386032.post@talk.nabble.com>


Iam having probles with my hard phone...van access ip0020...which is not
completing the 'B' leg on incoming calls to it
and 
on outgoing calls....immedialtely issuing a 488...
and 
fs log states ...
switch_core_codec.c:537 Codec G722 Exists but not at the desired
implementation. 16000hz 20ms
2010-01-30 11:49:07.736157 [ERR] sofia_glue.c:2126 Can't load codec?
...
I have multiple soft phones (sflphones  extensions 1000-1019), a dlink wifi
dph-540 (extension 4001 ip=192.168.1.190), and this van access ip0020 hard
phone (extension 2001 ip=192.168.1.180).....fs running on 192.168.1.102.

all phones sans '2001"  work perfectly..to/fro vitelity and local on lan.

Im almost certain that the problem lies in the port number assigned to the
uri.  wireshark shows that call was cancel because port could not be reached 
and 
port number is missing on ua shown by fs cli.

Call-ID:        af4cfd49b52bf8b3b3ec8db3e8a309e5 at 192.168.1.190
User:           4001 at 192.168.1.102
Contact:        4001 <sip:4001 at 192.168.1.190:5060>
...
...

Call-ID:        4c1ed9ae64405367 at 192.168.1.180
User:           2001 at 192.168.1.102
Contact:        "user" <sip:2001 at 192.168.1.180>
...                                   notice missing port no. here

Call-ID:        9be86b04-720f-4f7c-af52-b4f67dccf76b
User:           1000 at 192.168.1.102
Contact:        "jon" <sip:1000 at 192.168.1.101:5060>  
....

wireshark.....
292 14.903431   192.168.1.102         192.168.1.180         ICMP    
Destination unreachable (Port unreachable)
295 14.927845   192.168.1.102         192.168.1.180         SIP     
Request: BYE sip:2001 at 192.168.1.180

I have output from siptrace...but did not want to clutter up forum with too
much detail but can provide.

I also have screenshots of html config of phone...

I would appreciate some help please

thanks
-- 
View this message in context: http://old.nabble.com/problem-with-hard-phone-tp27386032p27386032.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From mike at jerris.com  Sat Jan 30 17:57:38 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 Jan 2010 20:57:38 -0500
Subject: [Freeswitch-users] CPS per mod_sofia profile limited?
In-Reply-To: <191c3a031001300857w36920b94r183f2e861db6baae@mail.gmail.com>
References: <4B62FF51.8070608@swip.net> <201001301233.21516.errotan@gmail.com>
	<191c3a031001300857w36920b94r183f2e861db6baae@mail.gmail.com>
Message-ID: <A1448B87-D14D-4181-892A-9ECCD5EB41F0@jerris.com>

For clarification, is it correct that your getting worse numbers for sustainable cps on SVN then on 1.0.4?  I would be interested in the numbers you would get with bypass_media=true instead of proxy_media=true and with neither setting set as well.  Also, make sure your logging level is low and try putting the db dir on a ram disk.  Thanks for the info.  As tony said, there are clearly some bottlenecks in the sofia library, but if you really need to pass media, your test is not very accurate for real life, and the results are likely not very useful to you.  You should use a more realistic length of call.  In bypass media I would suspect that length of call matters very little.  In proxy media or normal mode, the performance of the box is much more of a calculation on number of calls than cps as a result of the context switching from having to move the media and you will see that this plays a much more significant role on realistic call lengths (unless you really have lots of 1/2 second calls).  Some other tips.  While this extension may be trivial, what else is there in your dialplan context?  Anything above that extension could cause a significant impact.  Do you have any of the presence features enabled?  These do significantly impact call handling performance even if your sipp scenarios do not send any of those packets.

Mike


On Jan 30, 2010, at 11:57 AM, Anthony Minessale wrote:

> Also keep in mind that the industry standard is 50 which is the capacity to take over for the real standard of 25 in a fail-over scenario.  So you should be happy you even get 300cps for free.
> 
> The sofia stack can be improved but we are not the creators of this sip stack.  There is little to no work being done on that project right now and we are happy with what we have until we can get the lead dev to work on improving it with us when he has the time.
> 
> 
> 
> On Sat, Jan 30, 2010 at 5:33 AM, Pusk?s Zsolt <errotan at gmail.com> wrote:
> 2010. janu?r 29. 16.31.29 Robin Vleij d?tummal ezt ?rta:
> > Hi guys,
> >
> > Doing a bit of testing / benchmarking with FS 1.04 (and 1.05 SVN). Found
> > out some interesting things, I think.
> >
> > The setup is like this:
> >
> > SIPP Client -> FS -> SIPP Server
> >
> > The dialplan is as simple as it gets:
> >
> > <extension name="test-out">
> >   <condition field="network_addr" expression="^130\.244\.190\.150$"/>
> >   <condition field="destination_number" expression="^(\d+)$">
> >     <action application="set" data="proxy_media=true"/>
> >     <action application="set" data="hangup_after_bridge=true"/>
> >     <action application="bridge"
> > data="{sip_contact_user=transit}sofia/gateway/${distributor(gwg1)}/$1|sofia
> > /gateway/${distributor(gwg2)}/$1"/> </condition>
> > </extension>
> >
> > For the rest it's running CSV cdr's, commented out all modules I'm not
> > using, etc etc all that I could find on the wiki and the Interwebs.
> > Hardware / OS: from the shelve quadcore Xeon, debian 64-bit, 12GB
> > memory. SIPP is running a 500ms RTP pcap and the other side echos back.
> >
> > I had a few test setups then:
> >
> > 1: FS SVN, 1 sofia profile where the gateways were configured and the
> > server_IP:5060 was used.
> >
> > 2. FS SVN, 2 sofia profiles where the gateways where in an seperate
> > profile (server_IP:5070) and the "customer facing" side was the original
> > profile.
> >
> > 3. FS 1.04, same as above
> >
> > 4. FS 1.04, 4 sofia profiles, distributor to spread load over 2 incoming
> > and 2 outgoing profiles.
> >
> > Now the interesting thing was that under 1 I could go up, almost without
> > any CPU load, to 50cps. As soon as I went over this, calls where handled
> >  slower and "ongoing" calls would pile up untill it became really slow.
> > CPU load went to 100% on the FS process (both user and system time).
> > Lots of interupts and context switches. No throughput anymore untill I
> > lower and wait till the "buffer" is empty and FS is keeping up again.
> >
> > Under 2, I was able to increase the CPS to about 100 with the same effect.
> >
> > 3 then went much better, I was able to increase CPS to about 200 cps and
> > response times in SIPP went up slighty untill it just hits some kind of
> > limit and calls are handled slower.
> >
> > 4 is pretty cool. Here I can run 2 sipp clients both doing 150cps to the
> > gateways that are spread on the distribution module, so I spread traffic
> > over 2 profiles. With 300 cps in total, FS is keeping up and I have 30%
> > idle CPU. However, increasing to over 300cps gives problems again, even
> > though I have idle CPU left!
> >
> > All in all, I have a feeling that a single sip profile can't run more
> > than a certain limit untill it gets into some problem. Depending on if
> > I'm running SVN or 1.04 that limit seems to be 50cps or a bit higher.
> > After that limit it starts piling up "ongoing" calls, by taking time to
> > handle them and when that limit gets too high it's too late. All in all
> > really fine, I just set the system wide limit to a little under that
> > "threshold". But when I'm running just UNDER the threshold it's not CPU
> > that's a problem. Theoretically I should be able to run (based on the
> > CPU usage at 300cps) about 400cps.
> >
> > When running at 300 I get SOME failed calls and I see
> >
> > "switch_core_state_machine.c:525 a9a60636-0cea-11df-85a1-09c991f2afc5
> > Timeout waiting for next instruction in CS_NEW!"
> >
> > in the console.
> >
> > I didn't find much on how people do high cps setups and it feels a bit
> > like a "friday afternoon solution" to run multiple sofia profiles on the
> > same machine in order to max out the system.
> >
> > Maybe I'm missing something and I know it's not an exact science this,
> > but I'm not sure "all is OK" because I'm not slowly getting to a 100%
> > cpu (or disk / network) usage, I hit some kind of limit after which
> > stuff goes wrong.
> >
> > Anyone any input!
> >
> > /Robin
> >
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From mike at jerris.com  Sat Jan 30 18:01:47 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 Jan 2010 21:01:47 -0500
Subject: [Freeswitch-users] problem with hard phone
In-Reply-To: <27386032.post@talk.nabble.com>
References: <27386032.post@talk.nabble.com>
Message-ID: <FA59609C-96AB-41CB-8922-EA444A124221@jerris.com>

Your log below seems to indicate this is a codec negotiation issue, I doubt it has anything to do with the port thing you mention below.  The message below looks like your phone is requesting g722 at 16000hz.  This would seem logical as its a wideband codec.  There is a long ugly story behind this, but the rfc requires everyone to lie and say 8000hz on g722 instead.  Your phone seems to not be following this rule and I think that is the cause of your issues.  You should update your phone to some firmware that fixes this bug, correct it with some configuration setting on the phone, or disable g722 on the phone altogether.

Mike


On Jan 30, 2010, at 1:11 PM, jwssr wrote:

> 
> Iam having probles with my hard phone...van access ip0020...which is not
> completing the 'B' leg on incoming calls to it
> and 
> on outgoing calls....immedialtely issuing a 488...
> and 
> fs log states ...
> switch_core_codec.c:537 Codec G722 Exists but not at the desired
> implementation. 16000hz 20ms
> 2010-01-30 11:49:07.736157 [ERR] sofia_glue.c:2126 Can't load codec?
> ...
> I have multiple soft phones (sflphones  extensions 1000-1019), a dlink wifi
> dph-540 (extension 4001 ip=192.168.1.190), and this van access ip0020 hard
> phone (extension 2001 ip=192.168.1.180).....fs running on 192.168.1.102.
> 
> all phones sans '2001"  work perfectly..to/fro vitelity and local on lan.
> 
> Im almost certain that the problem lies in the port number assigned to the
> uri.  wireshark shows that call was cancel because port could not be reached 
> and 
> port number is missing on ua shown by fs cli.
> 
> Call-ID:        af4cfd49b52bf8b3b3ec8db3e8a309e5 at 192.168.1.190
> User:           4001 at 192.168.1.102
> Contact:        4001 <sip:4001 at 192.168.1.190:5060>
> ...
> ...
> 
> Call-ID:        4c1ed9ae64405367 at 192.168.1.180
> User:           2001 at 192.168.1.102
> Contact:        "user" <sip:2001 at 192.168.1.180>
> ...                                   notice missing port no. here
> 
> Call-ID:        9be86b04-720f-4f7c-af52-b4f67dccf76b
> User:           1000 at 192.168.1.102
> Contact:        "jon" <sip:1000 at 192.168.1.101:5060>  
> ....
> 
> wireshark.....
> 292 14.903431   192.168.1.102         192.168.1.180         ICMP    
> Destination unreachable (Port unreachable)
> 295 14.927845   192.168.1.102         192.168.1.180         SIP     
> Request: BYE sip:2001 at 192.168.1.180
> 
> I have output from siptrace...but did not want to clutter up forum with too
> much detail but can provide.
> 
> I also have screenshots of html config of phone...
> 
> I would appreciate some help please
> 
> thanks
> -- 
> View this message in context: http://old.nabble.com/problem-with-hard-phone-tp27386032p27386032.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Sat Jan 30 18:35:14 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 Jan 2010 21:35:14 -0500
Subject: [Freeswitch-users] Strategies for reliably detecting nat on
	B-leg?
In-Reply-To: <4B608987.9090606@aastral.net>
References: <4B60785F.6030505@aastral.net>	<FB139E6B-F685-491C-A672-16844878200E@freeswitch.org>
	<191c3a031001271005t62631e77sf6d9ca406054ba00@mail.gmail.com>
	<4B608987.9090606@aastral.net>
Message-ID: <2ADEAAFC-502E-474B-92BD-D6CBACCF885F@jerris.com>

There is no even remotely reliable way to tell.  The only thing you can tell is for some devices you can know they ARE behind nat, you can't ever know reliably that they are not.  That being said, I am not sure that is really the measure you are looking for.  It may be enough to know that the devices reliably tell you their external ip and port in their sdp.  For the a leg, you certainly know this before hand.  For the b leg you would not at dialplan time.  If you are looking to save bandwidth, the only real way of dealing with this would be to perform these actions at some point after the initial call is set up.  This will allow freeswitch to handle any initial audio indications that may be necessary, and for you to definitively know that it has 2 remote endpoints capable of at least handling nat for rtp.  You could do something from a monitoring application such as using uuid_media to trigger the freeswitch box out of the media path after the call has set up and you have confirmed the endpoints are well behaved.  The one thing you should be careful of in this case would be if freeswitch ever has to do an auto-adjust when initially setting up the media, this means the other end lied about either its ip or port.  This could be an indication of both software that can not report its remote IP in sdp or of a firewall that is changing the port.  The firewall changing port scenario you can not use bypass media unless the other device uses an auto-adjust method like freeswitch.  If it just lies about its IP, you could conceivably re-write the sdp with the correct ip and port, but freeswitch does not support this and I don't see it being likely that this would be added.  For you to accomplish this, you likely wold need to add some events to better monitor auto-adjust in the rtp and have some application (either as a freeswitch module, script, or some app attached to event socket) monitor these events and take action.

Mike

On Jan 27, 2010, at 1:44 PM, Bill W wrote:

> Thanks for the reply!
> 
> Just to make sure we're on the same page,  my FreeSWITCH sesrver has a 
> public IP, and I'm trying to bypass media whenever possible to reduce my 
> bandwidth usage.   My concern is trying to bypass media when one of the 
> remote endpoints (b-leg) is behind NAT (since I can reliably detect nat 
> with aggressive-nat on the A-leg).
> 
> Isn't local-network-acl and autonat:x.x.x.x for FS behind nat?
> 
> Does this new information change your responses?
> 
> Thanks again!
> Bill
> 
> 
> 
> Anthony Minessale wrote:
>> also you can set
>> sip_sticky_contact=true
>> channel var which will make that session turn on nat lock in the b leg 
>> so they can't change the contact to a nat addr
>> 
>> add it in {} to your dial string like
>> 
>> {sip_sticky_contact=true}sofia/internal/foo at bar.com <mailto:foo at bar.com>
>> 
>> 
>> 
>> 
>> On Wed, Jan 27, 2010 at 11:50 AM, Brian West <brian at freeswitch.org 
>> <mailto:brian at freeswitch.org>> wrote:
>> 
>>    update to trunk.  and don't use agressive-nat, set
>>    local-network-acl, set the ext-rtp-ip and ext-sip-ip to
>>    autonat:x.x.x.x or if you're behind a natpmp or upnp router set it
>>    to auto-nat.
>> 
>>    It should just work.  Again you have no real way to know if the far
>>    end client never lies to you.  Which it should never do anyway.
>>     Endpoints should know how to traverse their own nat and not leave
>>    it up to the registrar to figure it out.
>> 
>>    /b
>> 
>>    On Jan 27, 2010, at 11:31 AM, Bill W wrote:
>> 
>>> Thoughts?  Suggestions?
>> 
>> 
>>    _______________________________________________
>>    FreeSWITCH-users mailing list
>>    FreeSWITCH-users at lists.freeswitch.org
>>    <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>    http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com 
>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>> 
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org 
>> <mailto:sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888 
>> <http://iax:guest at conference.freeswitch.org/888>
>> googletalk:conf+888 at conference.freeswitch.org 
>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>> 
>> 
>> ------------------------------------------------------------------------
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Sat Jan 30 18:44:55 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 Jan 2010 21:44:55 -0500
Subject: [Freeswitch-users] Multiple DTMF on group_confirm_key
In-Reply-To: <7d79b3931001290436j2061a8dcg5cf7c5144103eb00@mail.gmail.com>
References: <7d79b3931001280411u6262f627xca723c64de5e118d@mail.gmail.com>
	<191c3a031001280914l56a035s74fd6d4b00dd9b3d@mail.gmail.com>
	<7d79b3931001290436j2061a8dcg5cf7c5144103eb00@mail.gmail.com>
Message-ID: <A8C99DF5-D9ED-481C-84E7-3DA5100C2576@jerris.com>

This does not make any sense.  If the one person answers the call but fails to enter the digits, what are you going to do with the caller just hang up on them?  


On Jan 29, 2010, at 7:36 AM, lakshmanan ganapathy wrote:

> I tested by executing a script. It works great. But a small doubt. 
> Assume that I made a parallel dial using bridge application.
> Normally, when one party answer the call, other party end will be hanged up.
> 
> But if I use group_confirm_key=exec and group_confirm_file=perl script.pl, both the end can answer, and call bridged with the person who finished the script first.
> 
> What I've to do if I need to execute the script only for the person who answer's the call first?
> 
> 
> On Thu, Jan 28, 2010 at 10:44 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> you have to use a script (See the wiki for executing a script)
> then you can read in as many digits as you want and do what you need.
> 

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From anthony.minessale at gmail.com  Sat Jan 30 20:19:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 30 Jan 2010 22:19:07 -0600
Subject: [Freeswitch-users] Multiple DTMF on group_confirm_key
In-Reply-To: <191c3a031001302017r7dee2c09vb4ef934335bc2f87@mail.gmail.com>
References: <7d79b3931001280411u6262f627xca723c64de5e118d@mail.gmail.com>
	<191c3a031001280914l56a035s74fd6d4b00dd9b3d@mail.gmail.com>
	<7d79b3931001290436j2061a8dcg5cf7c5144103eb00@mail.gmail.com>
	<191c3a031001302017r7dee2c09vb4ef934335bc2f87@mail.gmail.com>
Message-ID: <191c3a031001302019keac00f7j2deef44daa810304@mail.gmail.com>

If the both answer you can hangup on the one who entered the wrong or no
info.  The winner is whichever one exits the script first and is not hungup.

On Jan 29, 2010 6:43 AM, "lakshmanan ganapathy" <lakindia89 at gmail.com>
wrote:

I tested by executing a script. It works great. But a small doubt.
Assume that I made a parallel dial using bridge application.
Normally, when one party answer the call, other party end will be hanged up.

But if I use group_confirm_key=exec and group_confirm_file=perl script.pl,
both the end can answer, and call bridged with the person who finished the
script first.

What I've to do if I need to execute the script only for the person who
answer's the call first?




On Thu, Jan 28, 2010 at 10:44 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:
>
> you ...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From mouncifbb at gmail.com  Sat Jan 30 20:57:08 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Sat, 30 Jan 2010 23:57:08 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
Message-ID: <ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>

OK going back to use default profile to keep things simple below 2 results

Using:

lcr 16179470890  default  19785223241  ( this one consult
npa_nxx_company_ocn)

lcr 6179470890  default  9785223241 ( this one don't!! )



freeswitch> lcr 16179470890  default  19785223241
2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
[16179470890  default  19785223241]
2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
[19785223241]
2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
lata:1] so rate field is [intralata_rate]
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway
cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
l.enabled = '1' AND digits IN (16179470890, 1617947089, 161794708, 16179470,
1617947, 161794, 16179, 1617, 161, 16, 1) AND CURRENT_TIMESTAMP BETWEEN
date_start AND date_end ORDER BY digits DESC,  intralata_rate, rand();
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
[lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to head of
list
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
[lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
06179470890 at proxy.carrier2.net:5060
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end of
list after carrier1

 | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
Dialstring
|
 | 1           | carrier1 | 0.00000  |       |            |
[lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
|
 | 1           | carrier2 | 0.00000  |       |            |
[lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
06179470890 at proxy.carrier2.net:5060 |

2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
[lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
[lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
06179470890 at proxy.carrier2.net:5060





freeswitch> lcr 6179470890  default  9785223241
2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
[6179470890  default  9785223241]
2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
[9785223241]
2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
lata:0] so rate field is [rate]
2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
c.id=cg.carrier_id
WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
(6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
rate, rand();
2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
[lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to head of
list
2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
[lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890

 | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
Dialstring                                                                |
 | 617947      | carrier1 | 0.09000  |       |            |
[lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |










On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> turn up logging to debug again, and then reload mod_lcr.  It'll spit out a
> bunch of crap when it tests out each profile you have defined.  Give me the
> full log (here or in pastebin.freeswitch.org).  That may show more useful
> info as to why things are mucked up?
>
>
> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>
>> yes I use NANPA_STD profile instead of default cause I thought the custom
>> profile was causing issues, but looks like it's returning same results.
>>
>> There is this line in thw wiki:
>> intra lata/state selection is done manually by setting the channel
>> variables *intrastate* or *intralata* to the value *true*.
>>
>> do I have to set these ? if yes how?
>>
>> Thanks
>>
>>
>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>> Stuff inline.
>>>
>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>>>
>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>> 201007,"224","7229","0.0059","0.0127"
>>>> 201040,"224","9206","0.0036","0.0036"
>>>>
>>>
>>> Looks like they give you the LATA and OCN values with the prefix.  We
>>> (should) look that up ourselves.
>>>
>>>
>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>
>>>>
>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>
>>>> I also see this now when making a real call instead of running thorugh
>>>> CLI
>>>>
>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890 NANPA_STD)
>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate channel
>>>> var is [undef]*
>>>
>>>
>>> This is fine.  it is a leftover from when you would tell mod_lcr via a
>>> channel var that it should do intrastate.  I later had mod_lcr do the lookup
>>> itself, but we still honor the old var.  There are no channel vars
>>> associated with the cli, so you wouldn't see that msg.
>>>
>>>
>>>>
>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based on
>>>> interstate rates
>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>> 16179470893 using profile NANPA_STD
>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>>> lata:0] so rate field is [rate]
>>>>
>>>> called number 6179470890  caller ID: 6179472456
>>>>
>>>> any ideas??
>>>>
>>>>
>>> Only thing that jumps out at me.
>>>
>>> The output from lcr_admin show profiles showed only the default one.  On
>>> the dialplan you use the NANPA_STD profile.  Can you check lcr_admin list
>>> and see if that profile is defined and if so if it says it is using the
>>> npanxx table?
>>>
>>>
>>>
>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>>
>>>>> Something is still missing from the logs. Note the query of the npanxx
>>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>>> oh, what version of fs are you running?
>>>>>
>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>
>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>
>>>>> An example from my own setup:
>>>>>
>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>>>>> is [12148267711 default 12148267712]
>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>> [12148267712]
>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR (npa=214
>>>>> AND nxx=826)
>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count: 1
>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT l.digits
>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS lcr_gw_prefix,
>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>> random();
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>
>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
>>>>> head of list
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>
>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end
>>>>> of list
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end
>>>>> of list
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>
>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to end
>>>>> of list
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>
>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>> [...]
>>>>>
>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <
>>>>> mouncifbb at gmail.com> wrote:
>>>>> > Also the Provider has presented the rates in this format?
>>>>> > NPANXXLATA OCN INTER INTRA
>>>>> >
>>>>> >
>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <
>>>>> mouncifbb at gmail.com>
>>>>> > wrote:
>>>>> >>
>>>>> >> Tried it and it's not giving me intralata  instead I get interstate,
>>>>> does
>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?, also
>>>>> do I have
>>>>> >> to have the rate field in lcr table?
>>>>> >>
>>>>> >> lcr 617642  default 6176421212
>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>> Dialstring
>>>>> >>                                                        |
>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>> >>
>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>> >>
>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr
>>>>> is
>>>>> >> [617642  default 6176421212]
>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>>> >> [6176421212]
>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>> [state:0
>>>>> >> lata:0] so rate field is [rate]
>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>> l.digits,
>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>> gw_suffix,
>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid
>>>>> FROM lcr
>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
>>>>> l.enabled
>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>> CURRENT_TIMESTAMP
>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate, rand();
>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>> Dialstring
>>>>> >>
>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to
>>>>> head
>>>>> >> of list
>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>> Dialstring
>>>>> >>
>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>> >>
>>>>> >> Thank you Rupa!
>>>>> >>
>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com>
>>>>> wrote:
>>>>> >>>
>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The sql
>>>>> >>> statements along with status info will show up.  This should give
>>>>> >>> enough information to debug what is happening.
>>>>> >>>
>>>>> >>> I'm assuming the npanxx table is actually populated and not just
>>>>> >>> existing?
>>>>> >>>
>>>>> >>> When doing the lookup from the cli you have to tell lcr what CID to
>>>>> >>> use (remember, it is relative to the src/dest number).  I'm pretty
>>>>> >>> sure you get something on the console log when you don't specify a
>>>>> CID
>>>>> >>> when using the commandline.  Anyway:
>>>>> >>>
>>>>> >>> lcr 617642  default 6176421212
>>>>> >>>
>>>>> >>> should give you intralata.
>>>>> >>>
>>>>> >>> Note that the definition of intralata doesn't mean "local" for some
>>>>> >>> providers.  Some providers define local to "same ratecenter" which
>>>>> is
>>>>> >>> even more restrictive.
>>>>> >>>
>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <
>>>>> mouncifbb at gmail.com>
>>>>> >>> wrote:
>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I am
>>>>> using
>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>> >>> >
>>>>> >>> > lcr mysql table structure:
>>>>> >>> > CREATE TABLE `lcr` (
>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>> >>> >   KEY `digits` (`digits`),
>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>>>>> >>> > `carriers`
>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>> >>> >
>>>>> >>> >
>>>>> >>> > lcr_admin show profiles
>>>>> >>> > Name:           default
>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>> l.${lcr_rate_field},
>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>>> >>> > l.trail_strip,
>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c
>>>>> ON
>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>> WHERE
>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>> digits IN
>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>> date_start
>>>>> >>> > AND
>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality DESC,
>>>>> >>> >  reliability DESC, rand();
>>>>> >>> >  has %:         false
>>>>> >>> >  has vars:      true
>>>>> >>> >  has intrastate:        true
>>>>> >>> >  has intralata: true
>>>>> >>> >  has npanxx:    true
>>>>> >>> >  Reorder rate:  enabled
>>>>> >>> >  Info in headers:       disabled
>>>>> >>> >  Quote IN() List:       disabled
>>>>> >>> >
>>>>> >>> >
>>>>> >>> >
>>>>> >>> > lcr 617642  default  returns rate from the rate field table and
>>>>> not
>>>>> >>> > intra/inter state fields rates.
>>>>> >>> >
>>>>> >>> > Any ideas? thanks!
>>>>> >>> >
>>>>> >>> >
>>>>> >>> >
>>>>> >>> >
>>>>> >>> > _______________________________________________
>>>>> >>> > FreeSWITCH-users mailing list
>>>>> >>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>> >
>>>>> >>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >>> > http://www.freeswitch.org
>>>>> >>> >
>>>>> >>> >
>>>>> >>>
>>>>> >>>
>>>>> >>>
>>>>> >>> --
>>>>> >>> -Rupa
>>>>> >>>
>>>>> >>> _______________________________________________
>>>>> >>> FreeSWITCH-users mailing list
>>>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >>> http://www.freeswitch.org
>>>>> >>
>>>>> >
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sat Jan 30 21:38:07 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 31 Jan 2010 00:38:07 -0500
Subject: [Freeswitch-users] Inbound sip invite from external gateway
In-Reply-To: <BAY143-W25E4C7D5F62FD42D7BA37BA15E0@phx.gbl>
References: <BAY143-W25E4C7D5F62FD42D7BA37BA15E0@phx.gbl>
Message-ID: <AB6382BE-DE8E-4239-8325-54A5A3538297@jerris.com>

If you look at the debug you can see all the condition matching and  
variable expansion.  Is it not matching or is $0 not expanding?  Maybe  
try to use $1 instead.  Regardless, the debug of the dialplan should  
point to exactly what is not matching.

On Jan 26, 2010, at 4:10 PM, juan camilo ospina quintero <camilin2212 at hotmail.com 
 > wrote:

> hi to all
>
> im already do the integration with. Freeswitch sends invite messages  
> to sailfin, in sailfin there is a sip
> servlet that acts as a proxy, this means it receives the invite from  
> extension1000 and send the invite back
> to freeswitch at extension 1001, but i get the freeswitch messages  
> go to sailfin, but i dont get freeswitch
> to understand sailfin messages.
>
> there is my configuration for sending messages and for receiving  
> messages
>
> In /freeswitch/conf/dialplan/default.xml
>
>     <extension name="redirect">
>       <condition field="network_addr" expression="^192\.168\.2\.9$"/>
>       <condition field="destination_number" expression="^1(\d+)$">
>           <action application="bridge" data="sofia/external/ 
> $0 at 127.0.0.1:5070"/>
>       </condition>
>     </extension>
>
> this works fine, it redirects the messages to sailfin in 127.0.0.1
>
>
> In /freeswitch/conf/dialplan/public.xml
>
>     <extension name="receive">
>       <condition field="network_addr" expression="^127\.0\.0\.1$"/>
>       <condition field="destination_number" expression="^1(\d+)$">
>           <action application="bridge" data="sofia/internal/ 
> $0 at 192.168.2.9:5060"/>
>       </condition>
>     </extension>
>
> this doesnt work, i also use <action application="transfer"  
> data="$1001 XML default/> instead <action application="bridge"  
> data="sofia/internal/$0 at 192.168.2.9:5060"/>
>
> but still doesnt work, the invite that sailfin sends appears in the  
> freeswitch console, but the 1001 extension doesnt get it
>
> Keep your friends updated? even when you?re not signed in.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From mike at jerris.com  Sat Jan 30 21:46:39 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 31 Jan 2010 00:46:39 -0500
Subject: [Freeswitch-users] NAT keep alive
In-Reply-To: <8657153B18DB4AB3B17DB7A4BAAFF862@Terminal>
References: <8657153B18DB4AB3B17DB7A4BAAFF862@Terminal>
Message-ID: <28B8210E-BF86-4478-8140-21F0C9A135E8@jerris.com>

http://wiki.freeswitch.org/wiki/Sofia.conf.xml#nat-options-ping

It is a Sofia profile param

On Jan 26, 2010, at 3:57 AM, "Airsignal" <airsignal at wavecable.com>  
wrote:

> Good Evening:
>
> I am trying to get my switch to send keep alives to the ata's in the  
> field.
>
> <param name="nat-options-ping" value="true"/>   seems to be meant  
> for this.
>
> where should it go?  I can little documentation discussing this...
>
>
> Thanks!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From mike at jerris.com  Sat Jan 30 21:58:07 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 31 Jan 2010 00:58:07 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
Message-ID: <536A30FA-3494-40C2-9B9D-D8F63CA6BCC6@jerris.com>

http://www.google.com/search?q=cannot+restore+segment+prot+after+reloc:+Permission+denied

Google says this is selinux configured to enforcing without setting it  
up properly to allow what your trying to do.  Try disabling selinux,  
and if that works and you want selinux enabled, you will need to come  
up with the propper config.

Mike

On Jan 27, 2010, at 11:47 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> I tried running ldconfig on the directory containing mysql.so, but  
> it did not help.
> So it sounds like there could be a bug in the latter versions?
>
>
> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil <david.villasmil.work at gmail.com 
> > wrote:
> I got the same error, my script was working with no problems before an
> update to trunk.
>
> David
>
> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com>  
> wrote:
> > Hi, I followed the instructions in the Lua documentation for  
> setting up
> > luasql, but when I try to run my script I get:
> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading  
> module
> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
> >         /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore  
> segment prot
> > after reloc: Permission denied
> > stack traceback:
> >         [C]: ?
> >         [C]: in function 'require'
> >         /usr/local/freeswitch/scripts/l.lua:2: in main chunk
> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
> > I changed the permissions for mysql.so and for my script to 777,  
> so I'm not
> > sure where the permission problem could be.
> > I'd appreciate any suggestions.
> > Thanks,
> > Adam
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From rm at callrica.co.za  Sat Jan 30 22:05:39 2010
From: rm at callrica.co.za (Roly Maz)
Date: Sun, 31 Jan 2010 08:05:39 +0200
Subject: [Freeswitch-users] How to populate Voice recording metadata with
	custom CRM data?
Message-ID: <003c01caa23b$83ed0800$8bc71800$@co.za>

Hi All,  

 

I am using Freeswitch 1.0.4 (Latest Windows Installer version) on Windows 7
Ultimate for testing

 

Assuming an outbound call, what would be the most sensible approach to pass
custom CRM data into the voice-recording? 

 

I would like the voice recording of the call to include the customers social
security number in the file title, or even metadata.

 

In other words, is there a way to pass custom info, at time of call, to the
dialplan for use in the creating the voice recording. 

 

Any pointers would be much appreciated

 

Rgds

 

Roly

 

 

 

 

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From oseslija at gmail.com  Sun Jan 31 02:47:42 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sun, 31 Jan 2010 11:47:42 +0100
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
In-Reply-To: <j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8@4ax.com>
References: <gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b@4ax.com>
	<gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b-e09XROE/p8c@public.gmane.org>
	<20100129164359.9B3C22B10D@cuneorg-email.cune.pri>
	<j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8@4ax.com>
Message-ID: <4468a6771001310247n4bf5d1a4rf4aea9f2d6c35a1f@mail.gmail.com>

If FreeSWITCH is configured in bypass-media mode, and the endpoint behing
NAT cannot use any of the NAT avoiding techiques to send public IP in the
SDP (STUN etc.) then you'll have issues.

You can do what do I do, which is to make different sofia profiles for NATed
and non-NATED endpoints (FS has many server-side nat traversal mechanisms).

Regards,
Ognjen

On Sat, Jan 30, 2010 at 2:53 PM, Fred-145 <codecomplete at free.fr> wrote:

> On Fri, 29 Jan 2010 16:43:59 -0000,
> <Russell.Mosemann at cune.org> wrote:
> >The ports are open between the endpoint and Freeswitch. The ports are not
> >open between the two endpoints themselves. If each endpoint is behind its
> >own NAT, neither endpoint will be able to contact the other endpoint
> >unless some kind of forwarding is set up on the firewall to map the
> >external IP address and port to an internal IP address and port.
>
> Thanks but the context I was refering to is...
> 1. Freeswitch is configured in BypassMedia mode
> 2. The firewall and the local end-points are configured so that a
> series of UDP ranges are mapped to their respective end-point (eg.
> UDP100-1003 for extension #1, 1004-1007 for #2, etc.)
> ... so that RTP packets flow directly between the two end-points
>
> Brian says above that there might be cases where NAT could be a
> problem. When could this happen?
>
> I'd like to get to the bottom of this so that in case a server is a
> bit short on CPU/network power, I know that there's the alternative of
> RTP packets by-passing the server... but I also need to know what
> issues this setup can cause.
>
> Thank you.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch-list at puzzled.xs4all.nl  Sun Jan 31 05:04:28 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick)
Date: Sun, 31 Jan 2010 14:04:28 +0100
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <536A30FA-3494-40C2-9B9D-D8F63CA6BCC6@jerris.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<536A30FA-3494-40C2-9B9D-D8F63CA6BCC6@jerris.com>
Message-ID: <4B657FDC.5080109@puzzled.xs4all.nl>

On 01/31/2010 06:58 AM, Michael Jerris wrote:
> http://www.google.com/search?q=cannot+restore+segment+prot+after+reloc:+Permission+denied
> 
> Google says this is selinux configured to enforcing without setting it
> up properly to allow what your trying to do.  Try disabling selinux, and
> if that works and you want selinux enabled, you will need to come up
> with the propper config.

To fix a similar error message this is what I had in an old spec file:
/sbin/restorecon -v /usr/lib64/somelib.so

Iirc this is not the proper way to fix this and one should use the chcon
command (chcon -t ...) or create an selinux policy. man chcon and google
has more info.

Regards,
Patrick





From rupa at rupa.com  Sun Jan 31 05:32:04 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 31 Jan 2010 07:32:04 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
Message-ID: <b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>

On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> OK going back to use default profile to keep things simple below 2 results
>
> Using:
>
> lcr 16179470890  default  19785223241  ( this one consult
> npa_nxx_company_ocn)
>
> lcr 6179470890  default  9785223241 ( this one don't!! )
>
>
>
Oh, right!  mod_lcr really expects you to normalize your prefix to e164
format.  I thought there was discussion about this in the wiki, but maybe
not.  For simple prefix matching it doesn't matter, but for things that make
decisions based on the # (like the lata/state stuff) it does.

npanxx lookup only makes sense for NANPA numbers.  NANPA numbers have a
country code of "1" and a total length of 11 (including the 1).

This is the only rational way to do it when you have a rate table with both
domestic (NANPA) and international prefixes.


> freeswitch> lcr 16179470890  default  19785223241
> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
> [16179470890  default  19785223241]
> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
> [19785223241]
> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
> OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
> npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
> lata:1] so rate field is [intralata_rate]
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
> c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
> gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
> l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway
> cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
> l.enabled = '1' AND digits IN (16179470890, 1617947089, 161794708, 16179470,
> 1617947, 161794, 16179, 1617, 161, 16, 1) AND CURRENT_TIMESTAMP BETWEEN
> date_start AND date_end ORDER BY digits DESC,  intralata_rate, rand();
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to head of
> list
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
> 06179470890 at proxy.carrier2.net:5060
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end of
> list after carrier1
>
>
>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
> Dialstring
> |
>  | 1           | carrier1 | 0.00000  |       |            |
> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
> |
>  | 1           | carrier2 | 0.00000  |       |            |
> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
> 06179470890 at proxy.carrier2.net:5060 |
>
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
> 06179470890 at proxy.carrier2.net:5060
>
>
>
>
>
> freeswitch> lcr 6179470890  default  9785223241
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
> [6179470890  default  9785223241]
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
> [9785223241]
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
> lata:0] so rate field is [rate]
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
> (6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
> AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
> rate, rand();
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to head of
> list
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>
>
>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
> Dialstring                                                                |
>  | 617947      | carrier1 | 0.09000  |       |            |
> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |
>
>
>
>
>
>
>
>
>
>
>
> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> turn up logging to debug again, and then reload mod_lcr.  It'll spit out a
>> bunch of crap when it tests out each profile you have defined.  Give me the
>> full log (here or in pastebin.freeswitch.org).  That may show more useful
>> info as to why things are mucked up?
>>
>>
>> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>>
>>> yes I use NANPA_STD profile instead of default cause I thought the custom
>>> profile was causing issues, but looks like it's returning same results.
>>>
>>> There is this line in thw wiki:
>>> intra lata/state selection is done manually by setting the channel
>>> variables *intrastate* or *intralata* to the value *true*.
>>>
>>> do I have to set these ? if yes how?
>>>
>>> Thanks
>>>
>>>
>>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>>> Stuff inline.
>>>>
>>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>>>>
>>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>>> 201007,"224","7229","0.0059","0.0127"
>>>>> 201040,"224","9206","0.0036","0.0036"
>>>>>
>>>>
>>>> Looks like they give you the LATA and OCN values with the prefix.  We
>>>> (should) look that up ourselves.
>>>>
>>>>
>>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>>
>>>>>
>>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>>
>>>>> I also see this now when making a real call instead of running thorugh
>>>>> CLI
>>>>>
>>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890 NANPA_STD)
>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate channel
>>>>> var is [undef]*
>>>>
>>>>
>>>> This is fine.  it is a leftover from when you would tell mod_lcr via a
>>>> channel var that it should do intrastate.  I later had mod_lcr do the lookup
>>>> itself, but we still honor the old var.  There are no channel vars
>>>> associated with the cli, so you wouldn't see that msg.
>>>>
>>>>
>>>>>
>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based
>>>>> on interstate rates
>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>>> 16179470893 using profile NANPA_STD
>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>>>> lata:0] so rate field is [rate]
>>>>>
>>>>> called number 6179470890  caller ID: 6179472456
>>>>>
>>>>> any ideas??
>>>>>
>>>>>
>>>> Only thing that jumps out at me.
>>>>
>>>> The output from lcr_admin show profiles showed only the default one.  On
>>>> the dialplan you use the NANPA_STD profile.  Can you check lcr_admin list
>>>> and see if that profile is defined and if so if it says it is using the
>>>> npanxx table?
>>>>
>>>>
>>>>
>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>
>>>>>> Something is still missing from the logs. Note the query of the npanxx
>>>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>>>> oh, what version of fs are you running?
>>>>>>
>>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>>
>>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>>
>>>>>> An example from my own setup:
>>>>>>
>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>>>>>> is [12148267711 default 12148267712]
>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>>> [12148267712]
>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
>>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
>>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR (npa=214
>>>>>> AND nxx=826)
>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count: 1
>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT l.digits
>>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS lcr_gw_prefix,
>>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
>>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
>>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>>> random();
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>
>>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
>>>>>> head of list
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>
>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end
>>>>>> of list
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end
>>>>>> of list
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>
>>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to
>>>>>> end of list
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>
>>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>>> [...]
>>>>>>
>>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <
>>>>>> mouncifbb at gmail.com> wrote:
>>>>>> > Also the Provider has presented the rates in this format?
>>>>>> > NPANXXLATA OCN INTER INTRA
>>>>>> >
>>>>>> >
>>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <
>>>>>> mouncifbb at gmail.com>
>>>>>> > wrote:
>>>>>> >>
>>>>>> >> Tried it and it's not giving me intralata  instead I get
>>>>>> interstate, does
>>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?, also
>>>>>> do I have
>>>>>> >> to have the rate field in lcr table?
>>>>>> >>
>>>>>> >> lcr 617642  default 6176421212
>>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>>> Dialstring
>>>>>> >>                                                        |
>>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>>> >>
>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>>> >>
>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to
>>>>>> lcr is
>>>>>> >> [617642  default 6176421212]
>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>>>> >> [6176421212]
>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>>> [state:0
>>>>>> >> lata:0] so rate field is [rate]
>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>>> l.digits,
>>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>>> gw_suffix,
>>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid
>>>>>> FROM lcr
>>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
>>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
>>>>>> l.enabled
>>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>>> CURRENT_TIMESTAMP
>>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate,
>>>>>> rand();
>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>> Dialstring
>>>>>> >>
>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to
>>>>>> head
>>>>>> >> of list
>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>> Dialstring
>>>>>> >>
>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>> >>
>>>>>> >> Thank you Rupa!
>>>>>> >>
>>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com>
>>>>>> wrote:
>>>>>> >>>
>>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The sql
>>>>>> >>> statements along with status info will show up.  This should give
>>>>>> >>> enough information to debug what is happening.
>>>>>> >>>
>>>>>> >>> I'm assuming the npanxx table is actually populated and not just
>>>>>> >>> existing?
>>>>>> >>>
>>>>>> >>> When doing the lookup from the cli you have to tell lcr what CID
>>>>>> to
>>>>>> >>> use (remember, it is relative to the src/dest number).  I'm pretty
>>>>>> >>> sure you get something on the console log when you don't specify a
>>>>>> CID
>>>>>> >>> when using the commandline.  Anyway:
>>>>>> >>>
>>>>>> >>> lcr 617642  default 6176421212
>>>>>> >>>
>>>>>> >>> should give you intralata.
>>>>>> >>>
>>>>>> >>> Note that the definition of intralata doesn't mean "local" for
>>>>>> some
>>>>>> >>> providers.  Some providers define local to "same ratecenter" which
>>>>>> is
>>>>>> >>> even more restrictive.
>>>>>> >>>
>>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <
>>>>>> mouncifbb at gmail.com>
>>>>>> >>> wrote:
>>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I am
>>>>>> using
>>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>>> >>> >
>>>>>> >>> > lcr mysql table structure:
>>>>>> >>> > CREATE TABLE `lcr` (
>>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>>> >>> >   KEY `digits` (`digits`),
>>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>>>>>> >>> > `carriers`
>>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>>> >>> >
>>>>>> >>> >
>>>>>> >>> > lcr_admin show profiles
>>>>>> >>> > Name:           default
>>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>>> l.${lcr_rate_field},
>>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>>>> >>> > l.trail_strip,
>>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c
>>>>>> ON
>>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>> WHERE
>>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>>> digits IN
>>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>>> date_start
>>>>>> >>> > AND
>>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality
>>>>>> DESC,
>>>>>> >>> >  reliability DESC, rand();
>>>>>> >>> >  has %:         false
>>>>>> >>> >  has vars:      true
>>>>>> >>> >  has intrastate:        true
>>>>>> >>> >  has intralata: true
>>>>>> >>> >  has npanxx:    true
>>>>>> >>> >  Reorder rate:  enabled
>>>>>> >>> >  Info in headers:       disabled
>>>>>> >>> >  Quote IN() List:       disabled
>>>>>> >>> >
>>>>>> >>> >
>>>>>> >>> >
>>>>>> >>> > lcr 617642  default  returns rate from the rate field table and
>>>>>> not
>>>>>> >>> > intra/inter state fields rates.
>>>>>> >>> >
>>>>>> >>> > Any ideas? thanks!
>>>>>> >>> >
>>>>>> >>> >
>>>>>> >>> >
>>>>>> >>> >
>>>>>> >>> > _______________________________________________
>>>>>> >>> > FreeSWITCH-users mailing list
>>>>>> >>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >>> >
>>>>>> >>> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >>> > http://www.freeswitch.org
>>>>>> >>> >
>>>>>> >>> >
>>>>>> >>>
>>>>>> >>>
>>>>>> >>>
>>>>>> >>> --
>>>>>> >>> -Rupa
>>>>>> >>>
>>>>>> >>> _______________________________________________
>>>>>> >>> FreeSWITCH-users mailing list
>>>>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >>> http://www.freeswitch.org
>>>>>> >>
>>>>>> >
>>>>>> >
>>>>>> > _______________________________________________
>>>>>> > FreeSWITCH-users mailing list
>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> > http://www.freeswitch.org
>>>>>> >
>>>>>> >
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From scottferri09 at gmail.com  Sun Jan 31 05:45:04 2010
From: scottferri09 at gmail.com (Scott Fernandez)
Date: Sun, 31 Jan 2010 19:15:04 +0530
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <874941.17255.qm@web33502.mail.mud.yahoo.com>
References: <874941.17255.qm@web33502.mail.mud.yahoo.com>
Message-ID: <df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>

Hi,

Thx for the information. Can I have some detailed steps to configure
mod_managed class call control and how do we write the API commands in .Net
applications?

In addition, how do we get the current STATE of the call when I use webapi?.
Because it is required for me to route the call to the user upon it is
answered or disconnect  it.

Thanks,
Scott

On Wed, Jan 20, 2010 at 8:47 PM, Diego Toro <dftoro at yahoo.com> wrote:

> Hi, the answer is yes, you can to use mod_managed wich offer C# managed
> class to call control http://wiki.freeswitch.org/wiki/Mod_managed. Or
> using managed ESL (libs/esl/managed) which offer C# managed class to receive
> and send events and commands to FreeSwitch.
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
> --- On Wed, 1/20/10, Scott Fernandez <scottferri09 at gmail.com> wrote:
>
> > From: Scott Fernandez <scottferri09 at gmail.com>
> > Subject: Re: [Freeswitch-users] Establishing a Call from .Net based
> application
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Wednesday, January 20, 2010, 2:17 AM
>  > Thanks Dome. Will try it out and get back to
> > you if I come across any issues.
> >
> > Regards,
> > Scott.
> >
> > On Wed, Jan 20, 2010 at 11:02 AM,
> > Dome Charoenyost <dome at tel.co.th>
> > wrote:
> >
> > Please try http://wiki.freeswitch.org/wiki/Webapi
> >
> >
> > you can create class and map to webapi.
> >
> >
> >
> > Dome C.
> >
> >
> >
> > 2010/1/19 Scott Fernandez <scottferri09 at gmail.com>:
> >
> > > Hi,
> >
> > >
> >
> > > Is there any API modules available for me to initiate
> > a call from .Net based
> >
> > > application?.
> >
> > >
> >
> > > The idea is to include the API modules if any with the
> > .NET base classes so
> >
> > > that the API commands will be made available on it. I
> > know it is doable when
> >
> > > I use socket programming in .NET in which Telnet
> > session is created.
> >
> > > However, this would potentially hamper the performance
> > of the application
> >
> > > because of multiple sessions that will be created for
> > each call.
> >
> > >
> >
> > > Other than that, Is there any Freeswitch API modules
> > (like plug-ins)
> >
> > > available in order to include it into the .Net classes
> > and start building
> >
> > > the customized application?
> >
> > >
> >
> > > Any help from any one is highly appreciated.
> >
> > >
> >
> > > Thanks,
> >
> > > Scott
> >
> > >
> >
> > >
> > _______________________________________________
> >
> > > FreeSWITCH-users mailing list
> >
> > > FreeSWITCH-users at lists.freeswitch.org
> >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > http://www.freeswitch.org
> >
> > >
> >
> > >
> >
> >
> >
> > _______________________________________________
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > http://www.freeswitch.org
> >
> >
> >
> >
> > -----Inline Attachment Follows-----
>  >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Sun Jan 31 06:21:35 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 31 Jan 2010 08:21:35 -0600
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <4B657FDC.5080109@puzzled.xs4all.nl>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<536A30FA-3494-40C2-9B9D-D8F63CA6BCC6@jerris.com>
	<4B657FDC.5080109@puzzled.xs4all.nl>
Message-ID: <191c3a031001310621s20264d79u27afbf8ff0ba1a64@mail.gmail.com>

Be careful with lua and sql
I have heard countless reports of the luasql leaking memory like a fire
hydrant.....

We may need to make our own odbc obj so every embedded lang can share it.
But it takes time and resources.

On Jan 31, 2010 7:11 AM, "Patrick" <freeswitch-list at puzzled.xs4all.nl>
wrote:

On 01/31/2010 06:58 AM, Michael Jerris wrote:
> http://www.google.com/search?q=cannot+restore+segmen...
To fix a similar error message this is what I had in an old spec file:
/sbin/restorecon -v /usr/lib64/somelib.so

Iirc this is not the proper way to fix this and one should use the chcon
command (chcon -t ...) or create an selinux policy. man chcon and google
has more info.

Regards,
Patrick





_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at l...
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From anthony.minessale at gmail.com  Sun Jan 31 06:24:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 31 Jan 2010 08:24:34 -0600
Subject: [Freeswitch-users] How to populate Voice recording metadata
	with custom CRM data?
In-Reply-To: <003c01caa23b$83ed0800$8bc71800$@co.za>
References: <003c01caa23b$83ed0800$8bc71800$@co.za>
Message-ID: <191c3a031001310624u34d02ccbtfd8db2766f263827@mail.gmail.com>

There are channel vars that all begin record_
Check the wiki and default config example in the dp

On Jan 31, 2010 12:18 AM, "Roly Maz" <rm at callrica.co.za> wrote:

 Hi All,



I am using Freeswitch 1.0.4 (Latest Windows Installer version) on Windows 7
Ultimate for testing



Assuming an outbound call, what would be the most sensible approach to pass
custom CRM data into the voice-recording?



I would like the voice recording of the call to include the customers social
security number in the file title, or even metadata.



In other words, is there a way to pass custom info, at time of call, to the
dialplan for use in the creating the voice recording.



Any pointers would be much appreciated



Rgds



Roly









_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From stevendt at primrosebank.net  Sun Jan 31 06:30:11 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sun, 31 Jan 2010 14:30:11 -0000
Subject: [Freeswitch-users] Trunk Version Number
Message-ID: <2360B060EDF44D368707DD12064BD416@bp1.ad.bp.com>

Hi,

Running the latest SVN (16453) under Windows, the console "Version" command displays :-

"FreeSWITCH Version 1.0.trunk (UNKNOWN)"

Should the version number not include a meaningful build version in the brackets ?

regards
Dave
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From sos at sokhapkin.dyndns.org  Sun Jan 31 06:33:17 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 31 Jan 2010 09:33:17 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <191c3a031001310621s20264d79u27afbf8ff0ba1a64@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<4B657FDC.5080109@puzzled.xs4all.nl>
	<191c3a031001310621s20264d79u27afbf8ff0ba1a64@mail.gmail.com>
Message-ID: <201001310933.17301.sos@sokhapkin.dyndns.org>

There is memory leak with luasql mysql module, but everything is fine with 
luasql odbc module.

Is someone working on lua interface to freeswitch core odbc functions?

On Sunday 31 January 2010, Anthony Minessale wrote:
> Be careful with lua and sql
> I have heard countless reports of the luasql leaking memory like a fire
> hydrant.....
>
> We may need to make our own odbc obj so every embedded lang can share it.
> But it takes time and resources.
>
> On Jan 31, 2010 7:11 AM, "Patrick" <freeswitch-list at puzzled.xs4all.nl>
> wrote:
>
> On 01/31/2010 06:58 AM, Michael Jerris wrote:
> > http://www.google.com/search?q=cannot+restore+segmen...
>
> To fix a similar error message this is what I had in an old spec file:
> /sbin/restorecon -v /usr/lib64/somelib.so
>
> Iirc this is not the proper way to fix this and one should use the chcon
> command (chcon -t ...) or create an selinux policy. man chcon and google
> has more info.
>
> Regards,
> Patrick
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at l...




From robin at swip.net  Sun Jan 31 06:42:54 2010
From: robin at swip.net (Robin Vleij)
Date: Sun, 31 Jan 2010 15:42:54 +0100
Subject: [Freeswitch-users] CPS per mod_sofia profile limited?
In-Reply-To: <201001301233.21516.errotan@gmail.com>
References: <4B62FF51.8070608@swip.net> <201001301233.21516.errotan@gmail.com>
Message-ID: <4B6596EE.8080103@swip.net>

On 1/30/10 12:33 PM, Pusk?s Zsolt wrote:

Hi!

> CPU usage is not the only thing that limit your calls. Have you set the
> recommended ulimit settings and / or started fs with the -waste option ?

Yes, I did read the wiki & docs and do have the right setup for max 
performance. The -waste option didn't seem to do much in the 1.05 setup 
I was testing in, but I can try again on 1.04 tomorrow when I continue 
testing.

/Robin


From robin at swip.net  Sun Jan 31 06:45:39 2010
From: robin at swip.net (Robin Vleij)
Date: Sun, 31 Jan 2010 15:45:39 +0100
Subject: [Freeswitch-users] CPS per mod_sofia profile limited?
In-Reply-To: <191c3a031001300857w36920b94r183f2e861db6baae@mail.gmail.com>
References: <4B62FF51.8070608@swip.net> <201001301233.21516.errotan@gmail.com>
	<191c3a031001300857w36920b94r183f2e861db6baae@mail.gmail.com>
Message-ID: <4B659793.70408@swip.net>

On 1/30/10 5:57 PM, Anthony Minessale wrote:

Hi Anthony,

> Also keep in mind that the industry standard is 50 which is the capacity
> to take over for the real standard of 25 in a fail-over scenario.  So
> you should be happy you even get 300cps for free.

Yeah, no discussion there. I think 300 on standard hardware seems really 
good. Especially compared to some commercial product we've seen / read 
about. :) It was just that it doesn't feel like the hardware is fully 
used, but some invisible wall we hit, ie bug maybe. That's why I'm 
asking if anyone has seen this. Whatever limit we'll have, we'll set in 
the system wide limits and we're done, so it's no show-stopper in 
putting it in production. I've also done a long term test with a few 
thousand ongoing calls at 50cps over the weekend, see how it goes with 
memory usage and such.

> and we are happy with what we have until we can get the lead dev to work
> on improving it with us when he has the time.

OK.

/robin


From robin at swip.net  Sun Jan 31 06:51:11 2010
From: robin at swip.net (Robin Vleij)
Date: Sun, 31 Jan 2010 15:51:11 +0100
Subject: [Freeswitch-users] CPS per mod_sofia profile limited?
In-Reply-To: <A1448B87-D14D-4181-892A-9ECCD5EB41F0@jerris.com>
References: <4B62FF51.8070608@swip.net>
	<201001301233.21516.errotan@gmail.com>	<191c3a031001300857w36920b94r183f2e861db6baae@mail.gmail.com>
	<A1448B87-D14D-4181-892A-9ECCD5EB41F0@jerris.com>
Message-ID: <4B6598DF.3020407@swip.net>

On 1/31/10 2:57 AM, Michael Jerris wrote:

Hi Mike!

> For clarification, is it correct that your getting worse numbers for
> sustainable cps on SVN then on 1.0.4? I would be interested in the

Yes, 1.04 offered a lot better performance in my test scenario, using 
the same call scripts. I agree that the scenario isn't really realistic, 
I should test with at least 30 second calls.

> numbers you would get with bypass_media=true instead of proxy_media=true

Thought about the same, and funnily enough that didn't help. But in the 
long list of things I tried I don't remember if I tried it on 1.04, 
without media proxying. I expect a much higher performance just 
bypassing the media. I'll try again tomorrow on 1.04 and post the results.

> and with neither setting set as well. Also, make sure your logging level
> is low and try putting the db dir on a ram disk. Thanks for the info. As

I check disk performance using iostat and they're on 2% usage. It's a 
10k rpm sas disk we're writing to, with ram cache on the raid 
controller. Also -nosql didn't have any influence at all on performance.

> that length of call matters very little. In proxy media or normal mode,
> the performance of the box is much more of a calculation on number of
> calls than cps as a result of the context switching from having to move

Exactly. I'll adjust the lengt of the call to 30 or 60 seconds and then 
try to see how many calls we can have. That's in the end what matters 
really and not cps on it's own. Have to create a 60 second pcap first. :)

> calls). Some other tips. While this extension may be trivial, what else
> is there in your dialplan context? Anything above that extension could

Nothing, it's a single entry dialplan and that entry is thus on the top.

> cause a significant impact. Do you have any of the presence features

All presence features disabled.

> enabled? These do significantly impact call handling performance even if
> your sipp scenarios do not send any of those packets.

Yep. :)

Thanks for the input so far!

Maybe after I get some final results I put it on the wiki for future 
reference to load tests?

/robin


From anthony.minessale at gmail.com  Sun Jan 31 10:40:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 31 Jan 2010 12:40:51 -0600
Subject: [Freeswitch-users] CPS per mod_sofia profile limited?
In-Reply-To: <191c3a031001311023w526a2c4cpbd93e2bcea8fd07e@mail.gmail.com>
References: <4B62FF51.8070608@swip.net> <201001301233.21516.errotan@gmail.com>
	<191c3a031001300857w36920b94r183f2e861db6baae@mail.gmail.com>
	<A1448B87-D14D-4181-892A-9ECCD5EB41F0@jerris.com>
	<4B6598DF.3020407@swip.net>
	<191c3a031001311021n1f9c89ffq1a13de517ea98fb2@mail.gmail.com>
	<191c3a031001311022k217817aanbdd6830672cc063b@mail.gmail.com>
	<191c3a031001311023w526a2c4cpbd93e2bcea8fd07e@mail.gmail.com>
Message-ID: <191c3a031001311040r3932da76m77bd2ccabe1689c3@mail.gmail.com>

There is a tradeoff between cps and audio quality and timing accuracy.

Try -vm -nocal if you want to mimic 1.0.4 we don't discuss performance here.

On Jan 31, 2010 8:56 AM, "Robin Vleij" <robin at swip.net> wrote:

On 1/31/10 2:57 AM, Michael Jerris wrote:

Hi Mike!


> For clarification, is it correct that your getting worse numbers for
> sustainable cps on SVN the...
Yes, 1.04 offered a lot better performance in my test scenario, using
the same call scripts. I agree that the scenario isn't really realistic,
I should test with at least 30 second calls.


> numbers you would get with bypass_media=true instead of proxy_media=true
Thought about the same, and funnily enough that didn't help. But in the
long list of things I tried I don't remember if I tried it on 1.04,
without media proxying. I expect a much higher performance just
bypassing the media. I'll try again tomorrow on 1.04 and post the results.


> and with neither setting set as well. Also, make sure your logging level
> is low and try putting...
I check disk performance using iostat and they're on 2% usage. It's a
10k rpm sas disk we're writing to, with ram cache on the raid
controller. Also -nosql didn't have any influence at all on performance.


> that length of call matters very little. In proxy media or normal mode,
> the performance of the ...
Exactly. I'll adjust the lengt of the call to 30 or 60 seconds and then
try to see how many calls we can have. That's in the end what matters
really and not cps on it's own. Have to create a 60 second pcap first. :)


> calls). Some other tips. While this extension may be trivial, what else
> is there in your dialpl...
Nothing, it's a single entry dialplan and that entry is thus on the top.


> cause a significant impact. Do you have any of the presence features
All presence features disabled.


> enabled? These do significantly impact call handling performance even if
> your sipp scenarios do...
Yep. :)

Thanks for the input so far!

Maybe after I get some final results I put it on the wiki for future
reference to load tests?

/robin


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at list...
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From mbsip at gazeta.pl  Sun Jan 31 09:05:28 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sun, 31 Jan 2010 18:05:28 +0100
Subject: [Freeswitch-users] voicemail_greeting_number - question
Message-ID: <28f27f5d1001310905r41b16ca7r5ef1f236f76a070c@mail.gmail.com>

Hi ALL,

I am playing around with VM and want to play user recorded greeting
instead of default one.
I've scaned wiki Mod_Voicemail and found proper parameter
"voicemail_greeting_number".
Unfortunately there is a lack of example hence i dont know if it is
already working.

Aforementioned param was placed in /conf/directory/default/1000.xml
file (param name="voicemail_greeting_number", i tried many values)
The effect is that the default greeting is played.

Is this param embeeded into FS right now?
How to use it?
Is there any other place I should do the changes?

I am running  FreeSWITCH Version 1.0.trunk (16456).

Thx in advance.
Maciej


From tayeb.meftah at gmail.com  Sun Jan 31 12:21:29 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 31 Jan 2010 21:21:29 +0100
Subject: [Freeswitch-users] determining the source of receyved call in
	public context
Message-ID: <4B65E649.2040007@gmail.com>

hi,
how do i determine the gateway or the Ip of a receyved call from ITSP's?
i am calling my did from my mobile, but i see is processing the the 
mobile number no the ITSP User or did
thanks


From codecomplete at free.fr  Sun Jan 31 12:41:11 2010
From: codecomplete at free.fr (Fred-145)
Date: Sun, 31 Jan 2010 21:41:11 +0100
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
References: <gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b@4ax.com>
	<gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b-e09XROE/p8c@public.gmane.org>
	<20100129164359.9B3C22B10D@cuneorg-email.cune.pri>
	<j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8@4ax.com>
	<j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8-e09XROE/p8c@public.gmane.org>
	<4468a6771001310247n4bf5d1a4rf4aea9f2d6c35a1f@mail.gmail.com>
Message-ID: <qiqbm55lueuqup1m6r6d0mff36b4l4fb9l@4ax.com>

On Sun, 31 Jan 2010 11:47:42 +0100, Ognjen Seslija
<oseslija at gmail.com> wrote:
>If FreeSWITCH is configured in bypass-media mode, and the endpoint behing
>NAT cannot use any of the NAT avoiding techiques to send public IP in the
>SDP (STUN etc.) then you'll have issues.

In this scenario, Freeswitch and the SIP end-points are configured to
handle NAT. I'm just curious to know what issues/drawbacks I should
expect if I decide to lower the CPU/network load on the Freeswitch
server if I decide to configure it in bypass-media mode.

Are there features that aren't available when the two end-points speak
RTP directly instead of having RTP packets go through Freeswitch?

>You can do what do I do, which is to make different sofia profiles for NATed
>and non-NATED endpoints (FS has many server-side nat traversal mechanisms).

If you have time, I'm interested in knowing more about your setup.



From mbsip at gazeta.pl  Sun Jan 31 12:46:34 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sun, 31 Jan 2010 21:46:34 +0100
Subject: [Freeswitch-users]  vm-disk-quota
Message-ID: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>

Hi ALL,

Maybe this question will be piece of cake for most of you, but it
makes me think.

I would like to configure "vm-disk-quota" for all users i have.
I followed the wiki page and provided:
<param name="vm-disk-quota" value="10"/>
to /conf/directory/default/1000.xml

After reloadxml, incoming call give me "mod_voicemail.c:3057 Voicemail
disk quota is exceeded" feedback
No surprise for me because i had more less 10 voice mails already
recorded (before the vm-disk-quota was set up).
Strange is that increasing value even to 100 does not change anything.
The same thing with deleting recordings from user directory.
The only wayout is to set it to default value=0 (even FS shutdown
doesn't change anything)

I am wondering why vm-disk-quota produces "Voicemail disk quota is
exceeded" all the time
Where the module is looking for stored voicemail recordings.

Below is part of my configuration.
1) /conf/autoload_configs/voicemail.conf.xml
<param name="storage-dir" value="/usr/local/freeswitch/vm"/>
2) /conf/directory/default/1000.xml
<param name="vm-disk-quota" value="100"/>
3) /vm/FS_ip_address/1000 is empty


Thanks in advance.
Maciej


From mbsip at gazeta.pl  Sun Jan 31 13:25:28 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sun, 31 Jan 2010 22:25:28 +0100
Subject: [Freeswitch-users] vm-disk-quota
In-Reply-To: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>
References: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>
Message-ID: <28f27f5d1001311325v5dc1fbeegc7ce27f21925a233@mail.gmail.com>

Small change after wiping out all db voicemail_msgs table in
voicemail_default.db - i am able to record just one voicemail.

Strange, isn't it?

Thx,
Maciej.


From mouncifbb at gmail.com  Sun Jan 31 14:18:26 2010
From: mouncifbb at gmail.com (Mouncifbb)
Date: Sun, 31 Jan 2010 17:18:26 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
	<b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
Message-ID: <9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>

So the CID must have 1 at front also? Usually people
Send only npa and nxx ex 6176427788 7817612233
Do I need to alter it?

Sent from my iPhone

On Jan 31, 2010, at 8:32 AM, Rupa Schomaker <rupa at rupa.com> wrote:

>
>
> On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane <mouncifbb at gmail.com 
> > wrote:
> OK going back to use default profile to keep things simple below 2  
> results
>
> Using:
>
> lcr 16179470890  default  19785223241  ( this one consult  
> npa_nxx_company_ocn)
>
> lcr 6179470890  default  9785223241 ( this one don't!! )
>
>
>
> Oh, right!  mod_lcr really expects you to normalize your prefix to  
> e164 format.  I thought there was discussion about this in the wiki,  
> but maybe not.  For simple prefix matching it doesn't matter, but  
> for things that make decisions based on the # (like the lata/state  
> stuff) it does.
>
> npanxx lookup only makes sense for NANPA numbers.  NANPA numbers  
> have a country code of "1" and a total length of 11 (including the 1).
>
> This is the only rational way to do it when you have a rate table  
> with both domestic (NANPA) and international prefixes.
>
>
> freeswitch> lcr 16179470890  default  19785223241
> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr  
> is [16179470890  default  19785223241]
> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to  
> [19785223241]
> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT  
> 'state', count(DISTINCT state) FROM npa_nxx_company_ocn WHERE  
> (npa=617 AND nxx=947) OR (npa=978 AND nxx=522) UNION SELECT 'lata',  
> count(DISTINCT lata) FROM npa_nxx_company_ocn WHERE (npa=617 AND  
> nxx=947) OR (npa=978 AND nxx=522)
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing  
> [state:1 lata:1] so rate field is [intralata_rate]
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT  
> l.digits, c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix,  
> cg.suffix AS gw_suffix, l.lead_strip, l.trail_strip, l.prefix,  
> l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON  
> l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id  
> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND  
> digits IN (16179470890, 1617947089, 161794708, 16179470, 1617947,  
> 161794, 16179, 1617, 161, 16, 1) AND CURRENT_TIMESTAMP BETWEEN  
> date_start AND date_end ORDER BY digits DESC,  intralata_rate, rand();
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning  
> Dialstring [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/ 
> carrier1/16179470890
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to  
> head of list
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning  
> Dialstring [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/ 
> 06179470890 at proxy.carrier2.net:5060
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to  
> end of list after carrier1
>
>
>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |  
> Dialstring  
>     
>     
>     
>     
>                                                                    |
>  | 1           | carrier1 | 0.00000  |       |            |  
> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/ 
> carrier1/16179470890                 |
>  | 1           | carrier2 | 0.00000  |       |            |  
> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/ 
> 06179470890 at proxy.carrier2.net:5060 |
>
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning  
> Dialstring [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/ 
> carrier1/16179470890
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning  
> Dialstring [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/ 
> 06179470890 at proxy.carrier2.net:5060
>
>
>
>
>
> freeswitch> lcr 6179470890  default  9785223241
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr  
> is [6179470890  default  9785223241]
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to  
> [9785223241]
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing  
> [state:0 lata:0] so rate field is [rate]
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT  
> l.digits, c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix  
> AS gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix ,  
> cg.codec , l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id  
> JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1'  
> AND cg.enabled = '1' AND l.enabled = '1' AND digits IN (6179470890, 617947089 
> , 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6) AND  
> CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits  
> DESC,  rate, rand();
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning  
> Dialstring [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/ 
> carrier1/16179470890
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to  
> head of list
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning  
> Dialstring [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/ 
> carrier1/16179470890
>
>
>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |  
> Dialstring  
>                                                                |
>  | 617947      | carrier1 | 0.09000  |       |            |  
> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/ 
> carrier1/16179470890 |
>
>
>
>
>
>
>
>
>
>
>
> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> turn up logging to debug again, and then reload mod_lcr.  It'll spit  
> out a bunch of crap when it tests out each profile you have  
> defined.  Give me the full log (here or in  
> pastebin.freeswitch.org).  That may show more useful info as to why  
> things are mucked up?
>
>
> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane  
> <mouncifbb at gmail.com> wrote:
> yes I use NANPA_STD profile instead of default cause I thought the  
> custom profile was causing issues, but looks like it's returning  
> same results.
>
> There is this line in thw wiki:
> intra lata/state selection is done manually by setting the channel  
> variables intrastate or intralata to the value true.
>
> do I have to set these ? if yes how?
>
> Thanks
>
>
> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> Stuff inline.
>
> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane  
> <mouncifbb at gmail.com> wrote:
> NPANXX,"LATA","OCN","NTER","INTRA"
> 201007,"224","7229","0.0059","0.0127"
> 201040,"224","9206","0.0036","0.0036"
>
>
> Looks like they give you the LATA and OCN values with the prefix.   
> We (should) look that up ourselves.
>
> FreeSWITCH Version 1.0.trunk (16540)
>
>
> Also I noticed the npa_nxx_ocn table never get consulted.
>
> I also see this now when making a real call instead of running  
> thorugh CLI
>
> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890 NANPA_STD)
> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 intrastate channel  
> var is [undef]
>
> This is fine.  it is a leftover from when you would tell mod_lcr via  
> a channel var that it should do intrastate.  I later had mod_lcr do  
> the lookup itself, but we still honor the old var.  There are no  
> channel vars associated with the cli, so you wouldn't see that msg.
>
>
> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes  
> based on interstate rates
> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on 16179470893 
>  using profile NANPA_STD
> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing  
> [state:0 lata:0] so rate field is [rate]
>
> called number 6179470890  caller ID: 6179472456
>
> any ideas??
>
>
> Only thing that jumps out at me.
>
> The output from lcr_admin show profiles showed only the default  
> one.  On the dialplan you use the NANPA_STD profile.  Can you check  
> lcr_admin list and see if that profile is defined and if so if it  
> says it is using the npanxx table?
>
>
>
>
>
>
>
> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker <rupa at rupa.com>  
> wrote:
> Something is still missing from the logs. Note the query of the npanxx
> table, the flags being set, and the rate field being chosen.  Umm..
> oh, what version of fs are you running?
>
> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
> (there is a link to that from mod_lcr's wiki page).
>
> An example from my own setup:
>
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
> is [12148267711 default 12148267712]
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to  
> [12148267712]
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR (npa=214
> AND nxx=826)
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count: 1
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
> [state:1 lata:1] so rate field is [intralata_rate]
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT l.digits
> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
> l.intralata_rate as lcr_rate_field,        cg.prefix AS lcr_gw_prefix,
> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
> random();
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/ 
> XXXX12148267711
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
> head of list
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/ 
> 12148267711
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to  
> end of list
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to  
> end of list
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/ 
> 12148267711
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to  
> end of list
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/ 
> YYYY12148267711
> [...]
>
> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <mouncifbb at gmail.com 
> > wrote:
> > Also the Provider has presented the rates in this format?
> > NPANXXLATA OCN INTER INTRA
> >
> >
> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <mouncifbb at gmail.com 
> >
> > wrote:
> >>
> >> Tried it and it's not giving me intralata  instead I get  
> interstate, does
> >> the npa_nxx_company_ocn table needs to be used in this case?,  
> also do I have
> >> to have the rate field in lcr table?
> >>
> >> lcr 617642  default 6176421212
> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |  
> Dialstring
> >>                                                        |
> >>  | 617642      | carrier1 | 0.00500  |       |            |
> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/ 
> carrier1/1617642 |
> >>
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to  
> lcr is
> >> [617642  default 6176421212]
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
> >> [6176421212]
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing  
> [state:0
> >> lata:0] so rate field is [rate]
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT  
> l.digits,
> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS  
> gw_suffix,
> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,  
> l.cid FROM lcr
> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND  
> l.enabled
> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND  
> CURRENT_TIMESTAMP
> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate, rand 
> ();
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning  
> Dialstring
> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/ 
> carrier1/1617642
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1  
> to head
> >> of list
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning  
> Dialstring
> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/ 
> carrier1/1617642
> >>
> >> Thank you Rupa!
> >>
> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com>  
> wrote:
> >>>
> >>> turn console logging up to debug and redo the lcr lookup.  The sql
> >>> statements along with status info will show up.  This should give
> >>> enough information to debug what is happening.
> >>>
> >>> I'm assuming the npanxx table is actually populated and not just
> >>> existing?
> >>>
> >>> When doing the lookup from the cli you have to tell lcr what CID  
> to
> >>> use (remember, it is relative to the src/dest number).  I'm pretty
> >>> sure you get something on the console log when you don't specify  
> a CID
> >>> when using the commandline.  Anyway:
> >>>
> >>> lcr 617642  default 6176421212
> >>>
> >>> should give you intralata.
> >>>
> >>> Note that the definition of intralata doesn't mean "local" for  
> some
> >>> providers.  Some providers define local to "same ratecenter"  
> which is
> >>> even more restrictive.
> >>>
> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <mouncifbb at gmail.com 
> >
> >>> wrote:
> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I  
> am using
> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
> >>> >
> >>> > lcr mysql table structure:
> >>> > CREATE TABLE `lcr` (
> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
> >>> >   `carrier_id` INT(11) NOT NULL,
> >>> >   `lead_strip` INT(11) NOT NULL,
> >>> >   `trail_strip` INT(11) NOT NULL,
> >>> >   `prefix` VARCHAR(16) NOT NULL,
> >>> >   `suffix` VARCHAR(16) NOT NULL,
> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
> >>> >   `quality` FLOAT(10,6) NOT NULL,
> >>> >   `reliability` FLOAT(10,6) NOT NULL,
> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
> >>> >   PRIMARY KEY  (`id`),
> >>> >   KEY `carrier_id` (`carrier_id`),
> >>> >   KEY `digits` (`digits`),
> >>> >   KEY `lcr_profile` (`lcr_profile`),
> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
> >>> > `carriers`
> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
> >>> >
> >>> >
> >>> > lcr_admin show profiles
> >>> > Name:           default
> >>> >  custom sql:    SELECT l.digits, c.carrier_name, l.$ 
> {lcr_rate_field},
> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
> >>> > l.trail_strip,
> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers  
> c ON
> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON  
> c.id=cg.carrier_id WHERE
> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND  
> digits IN
> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN  
> date_start
> >>> > AND
> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality  
> DESC,
> >>> >  reliability DESC, rand();
> >>> >  has %:         false
> >>> >  has vars:      true
> >>> >  has intrastate:        true
> >>> >  has intralata: true
> >>> >  has npanxx:    true
> >>> >  Reorder rate:  enabled
> >>> >  Info in headers:       disabled
> >>> >  Quote IN() List:       disabled
> >>> >
> >>> >
> >>> >
> >>> > lcr 617642  default  returns rate from the rate field table  
> and not
> >>> > intra/inter state fields rates.
> >>> >
> >>> > Any ideas? thanks!
> >>> >
> >>> >
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
> freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>>
> >>>
> >>> --
> >>> -Rupa
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
> freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> -- 
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> -- 
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> -- 
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From Russell.Mosemann at cune.org  Sun Jan 31 14:35:21 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sun, 31 Jan 2010 16:35:21 -0600
Subject: [Freeswitch-users] Equivalent to Asterisk's"directrtpsetup=yes"?
In-Reply-To: <qiqbm55lueuqup1m6r6d0mff36b4l4fb9l@4ax.com>
References: <gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b@4ax.com><gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b-e09XROE/p8c@public.gmane.org><20100129164359.9B3C22B10D@cuneorg-email.cune.pri><j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8@4ax.com><j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8-e09XROE/p8c@public.gmane.org><4468a6771001310247n4bf5d1a4rf4aea9f2d6c35a1f@mail.gmail.com>
	<qiqbm55lueuqup1m6r6d0mff36b4l4fb9l@4ax.com>
Message-ID: <15D48404014D48D19F85CFFFC4BBC76F@cune.pri>

Fred-145 asked:

> Are there features that aren't available when the two end-points speak
> RTP directly instead of having RTP packets go through Freeswitch?

I wasn't paying close attention, but in a recent discussion, someone wanted to have moh in a bypass-media situation. I think there was a way to do that, but then there was an issue with going back to bypass-media after the call was taken off hold. If you look through the list archive, you should be able to find it.

--
Russell Mosemann



From rupa at rupa.com  Sun Jan 31 15:07:29 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 31 Jan 2010 17:07:29 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
	<b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
	<9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>
Message-ID: <b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>

Yes, you need to normalize the values passed to lcr.  Otherwise, how could
it work?

You can normalize the CID by matching and adding a 1 for 10 digit #s, or
removing the leading + or other things you might need then setting it back
to the profile using the set_profile_var app (
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var).
 (mod_cidlookup will set it after doing a #->name/area lookup - but for now
you can set it yourself)

You can normalize the DID by doing similar matching rules as above and then
transfering to that normalized DID for the rest of your call plan
processing.

I'm pretty sure mod_cidlookup has an example of normalizing... yeah:
http://wiki.freeswitch.org/wiki/Mod_cidlookup#Dialplan_Application

On Sun, Jan 31, 2010 at 4:18 PM, Mouncifbb <mouncifbb at gmail.com> wrote:

> So the CID must have 1 at front also? Usually people
> Send only npa and nxx ex 6176427788 7817612233
> Do I need to alter it?
>
> Sent from my iPhone
>
> On Jan 31, 2010, at 8:32 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>
>
> On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane < <mouncifbb at gmail.com>
> mouncifbb at gmail.com> wrote:
>
>> OK going back to use default profile to keep things simple below 2 results
>>
>> Using:
>>
>> lcr 16179470890  default  19785223241  ( this one consult
>> npa_nxx_company_ocn)
>>
>> lcr 6179470890  default  9785223241 ( this one don't!! )
>>
>>
>>
> Oh, right!  mod_lcr really expects you to normalize your prefix to e164
> format.  I thought there was discussion about this in the wiki, but maybe
> not.  For simple prefix matching it doesn't matter, but for things that make
> decisions based on the # (like the lata/state stuff) it does.
>
> npanxx lookup only makes sense for NANPA numbers.  NANPA numbers have a
> country code of "1" and a total length of 11 (including the 1).
>
> This is the only rational way to do it when you have a rate table with both
> domestic (NANPA) and international prefixes.
>
>
>> freeswitch> lcr 16179470890  default  19785223241
>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>> [16179470890  default  19785223241]
>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>> [19785223241]
>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
>> OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
>> npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
>> lata:1] so rate field is [intralata_rate]
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>> c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
>> gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>> l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN
>> carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND
>> cg.enabled = '1' AND l.enabled = '1' AND digits IN (16179470890, 1617947089,
>> 161794708, 16179470, 1617947, 161794, 16179, 1617, 161, 16, 1) AND
>> CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>> intralata_rate, rand();
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>> of list
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>> 06179470890 at proxy.carrier2.net:5060
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end of
>> list after carrier1
>>
>>
>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>> Dialstring
>> |
>>  | 1           | carrier1 | 0.00000  |       |            |
>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>> |
>>  | 1           | carrier2 | 0.00000  |       |            |
>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>> 06179470890 at proxy.carrier2.net:5060 |
>>
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>> 06179470890 at proxy.carrier2.net:5060
>>
>>
>>
>>
>>
>> freeswitch> lcr 6179470890  default  9785223241
>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>> [6179470890  default  9785223241]
>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>> [9785223241]
>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
>> lata:0] so rate field is [rate]
>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>> (6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
>> AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>> rate, rand();
>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>> of list
>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>
>>
>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>> Dialstring                                                                |
>>   | 617947      | carrier1 | 0.09000  |       |            |
>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker < <rupa at rupa.com>
>> rupa at rupa.com> wrote:
>>
>>> turn up logging to debug again, and then reload mod_lcr.  It'll spit out
>>> a bunch of crap when it tests out each profile you have defined.  Give me
>>> the full log (here or in <http://pastebin.freeswitch.org>
>>> pastebin.freeswitch.org).  That may show more useful info as to why
>>> things are mucked up?
>>>
>>>
>>> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>> mouncifbb at gmail.com> wrote:
>>>
>>>> yes I use NANPA_STD profile instead of default cause I thought the
>>>> custom profile was causing issues, but looks like it's returning same
>>>> results.
>>>>
>>>> There is this line in thw wiki:
>>>> intra lata/state selection is done manually by setting the channel
>>>> variables *intrastate* or *intralata* to the value *true*.
>>>>
>>>> do I have to set these ? if yes how?
>>>>
>>>> Thanks
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker < <rupa at rupa.com>
>>>> rupa at rupa.com> wrote:
>>>>
>>>>> Stuff inline.
>>>>>
>>>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>> mouncifbb at gmail.com> wrote:
>>>>>
>>>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>>>> 201007,"224","7229","0.0059","0.0127"
>>>>>> 201040,"224","9206","0.0036","0.0036"
>>>>>>
>>>>>
>>>>> Looks like they give you the LATA and OCN values with the prefix.  We
>>>>> (should) look that up ourselves.
>>>>>
>>>>>
>>>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>>>
>>>>>>
>>>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>>>
>>>>>> I also see this now when making a real call instead of running thorugh
>>>>>> CLI
>>>>>>
>>>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890
>>>>>> NANPA_STD)
>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate channel
>>>>>> var is [undef]*
>>>>>
>>>>>
>>>>> This is fine.  it is a leftover from when you would tell mod_lcr via a
>>>>> channel var that it should do intrastate.  I later had mod_lcr do the lookup
>>>>> itself, but we still honor the old var.  There are no channel vars
>>>>> associated with the cli, so you wouldn't see that msg.
>>>>>
>>>>>
>>>>>>
>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based
>>>>>> on interstate rates
>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>>>> 16179470893 using profile NANPA_STD
>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing
>>>>>> [state:0 lata:0] so rate field is [rate]
>>>>>>
>>>>>> called number 6179470890  caller ID: 6179472456
>>>>>>
>>>>>> any ideas??
>>>>>>
>>>>>>
>>>>> Only thing that jumps out at me.
>>>>>
>>>>> The output from lcr_admin show profiles showed only the default one.
>>>>>  On the dialplan you use the NANPA_STD profile.  Can you check lcr_admin
>>>>> list and see if that profile is defined and if so if it says it is using the
>>>>> npanxx table?
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker < <rupa at rupa.com>
>>>>>> rupa at rupa.com> wrote:
>>>>>>
>>>>>>> Something is still missing from the logs. Note the query of the
>>>>>>> npanxx
>>>>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>>>>> oh, what version of fs are you running?
>>>>>>>
>>>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>>> <http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name>
>>>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>>>
>>>>>>> An example from my own setup:
>>>>>>>
>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>>>>>>> is [12148267711 default 12148267712]
>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>>>> [12148267712]
>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
>>>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
>>>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR
>>>>>>> (npa=214
>>>>>>> AND nxx=826)
>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count:
>>>>>>> 1
>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT
>>>>>>> l.digits
>>>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS
>>>>>>> lcr_gw_prefix,
>>>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
>>>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
>>>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>>>> random();
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>>
>>>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
>>>>>>> head of list
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>>
>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end
>>>>>>> of list
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end
>>>>>>> of list
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>>
>>>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to
>>>>>>> end of list
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>>
>>>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>>>> [...]
>>>>>>>
>>>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>> > Also the Provider has presented the rates in this format?
>>>>>>> > NPANXXLATA OCN INTER INTRA
>>>>>>> >
>>>>>>> >
>>>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>> mouncifbb at gmail.com>
>>>>>>> > wrote:
>>>>>>> >>
>>>>>>> >> Tried it and it's not giving me intralata  instead I get
>>>>>>> interstate, does
>>>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?, also
>>>>>>> do I have
>>>>>>> >> to have the rate field in lcr table?
>>>>>>> >>
>>>>>>> >> lcr 617642  default 6176421212
>>>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>>>> Dialstring
>>>>>>> >>                                                        |
>>>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>>>> >>
>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>>>> >>
>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to
>>>>>>> lcr is
>>>>>>> >> [617642  default 6176421212]
>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>>>>> >> [6176421212]
>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>> [state:0
>>>>>>> >> lata:0] so rate field is [rate]
>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>>>> l.digits,
>>>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>>>> gw_suffix,
>>>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid
>>>>>>> FROM lcr
>>>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
>>>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
>>>>>>> l.enabled
>>>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>>>> CURRENT_TIMESTAMP
>>>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate,
>>>>>>> rand();
>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>> Dialstring
>>>>>>> >>
>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1
>>>>>>> to head
>>>>>>> >> of list
>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>> Dialstring
>>>>>>> >>
>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>> >>
>>>>>>> >> Thank you Rupa!
>>>>>>> >>
>>>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker < <rupa at rupa.com>
>>>>>>> rupa at rupa.com> wrote:
>>>>>>> >>>
>>>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The
>>>>>>> sql
>>>>>>> >>> statements along with status info will show up.  This should give
>>>>>>> >>> enough information to debug what is happening.
>>>>>>> >>>
>>>>>>> >>> I'm assuming the npanxx table is actually populated and not just
>>>>>>> >>> existing?
>>>>>>> >>>
>>>>>>> >>> When doing the lookup from the cli you have to tell lcr what CID
>>>>>>> to
>>>>>>> >>> use (remember, it is relative to the src/dest number).  I'm
>>>>>>> pretty
>>>>>>> >>> sure you get something on the console log when you don't specify
>>>>>>> a CID
>>>>>>> >>> when using the commandline.  Anyway:
>>>>>>> >>>
>>>>>>> >>> lcr 617642  default 6176421212
>>>>>>> >>>
>>>>>>> >>> should give you intralata.
>>>>>>> >>>
>>>>>>> >>> Note that the definition of intralata doesn't mean "local" for
>>>>>>> some
>>>>>>> >>> providers.  Some providers define local to "same ratecenter"
>>>>>>> which is
>>>>>>> >>> even more restrictive.
>>>>>>> >>>
>>>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>> mouncifbb at gmail.com>
>>>>>>> >>> wrote:
>>>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I am
>>>>>>> using
>>>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>>>> >>> >
>>>>>>> >>> > lcr mysql table structure:
>>>>>>> >>> > CREATE TABLE `lcr` (
>>>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>>>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>>>> >>> >   KEY `digits` (`digits`),
>>>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>>>>>>> >>> > `carriers`
>>>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>>>> >>> >
>>>>>>> >>> >
>>>>>>> >>> > lcr_admin show profiles
>>>>>>> >>> > Name:           default
>>>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>>>> l.${lcr_rate_field},
>>>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>>>>> >>> > l.trail_strip,
>>>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers
>>>>>>> c ON
>>>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>> WHERE
>>>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>>>> digits IN
>>>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>>>> date_start
>>>>>>> >>> > AND
>>>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality
>>>>>>> DESC,
>>>>>>> >>> >  reliability DESC, rand();
>>>>>>> >>> >  has %:         false
>>>>>>> >>> >  has vars:      true
>>>>>>> >>> >  has intrastate:        true
>>>>>>> >>> >  has intralata: true
>>>>>>> >>> >  has npanxx:    true
>>>>>>> >>> >  Reorder rate:  enabled
>>>>>>> >>> >  Info in headers:       disabled
>>>>>>> >>> >  Quote IN() List:       disabled
>>>>>>> >>> >
>>>>>>> >>> >
>>>>>>> >>> >
>>>>>>> >>> > lcr 617642  default  returns rate from the rate field table and
>>>>>>> not
>>>>>>> >>> > intra/inter state fields rates.
>>>>>>> >>> >
>>>>>>> >>> > Any ideas? thanks!
>>>>>>> >>> >
>>>>>>> >>> >
>>>>>>> >>> >
>>>>>>> >>> >
>>>>>>> >>> > _______________________________________________
>>>>>>> >>> > FreeSWITCH-users mailing list
>>>>>>> >>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >>> >
>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >>> >
>>>>>>> >>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>> >>> >
>>>>>>> >>> >
>>>>>>> >>>
>>>>>>> >>>
>>>>>>> >>>
>>>>>>> >>> --
>>>>>>> >>> -Rupa
>>>>>>> >>>
>>>>>>> >>> _______________________________________________
>>>>>>> >>> FreeSWITCH-users mailing list
>>>>>>> >>> <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>> >>
>>>>>>> >
>>>>>>> >
>>>>>>> > _______________________________________________
>>>>>>> > FreeSWITCH-users mailing list
>>>>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>> >
>>>>>>> >
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> -Rupa
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From mailinglist at fribert.dk  Sun Jan 31 15:38:04 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Mon, 01 Feb 2010 00:38:04 +0100
Subject: [Freeswitch-users] Somebody help me understand the 'features' set
	up please :-)
Message-ID: <4B66226C020000E10000043C@mail.fribert.dk>

Ok, I've gotten the Freeswitch to register to my VoIP provider.
I've gotten my phones to register to Freeswitch, and I can receive and make calls, all very nice.
 
I've gotten a Sipura SPA901 and a Siemens Gigaset S68IP registered to the Freeswitch.
 
When I receive a call, I would like to be able to transfer the call to another phone, or change the call to a conference call with two local phones.
 
So I've been looking at the examples in the wiki, and I can't make them work, not as I understand them anyways. Especially the att_xfer seems to be able to do what I need.
 
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
 
As I understand Example1, I should answer the call, and then press *3 during the call, and either transfer it or change it to a threeway call.
 
I get the first part, create an extension in the dialplan called att_xfer.
But what is meant by the second par 'then bind this feature to DTMF 3', how do I enter that, and where?
 
I hope somebody can help me with this (again)?
 
 
 
Best regards
Fribse
 
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0
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From rupa at rupa.com  Sun Jan 31 16:45:15 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 31 Jan 2010 18:45:15 -0600
Subject: [Freeswitch-users] Somebody help me understand the 'features'
	set up please :-)
In-Reply-To: <4B66226C020000E10000043C@mail.fribert.dk>
References: <4B66226C020000E10000043C@mail.fribert.dk>
Message-ID: <b39273531001311645u618ea87am8a3d84ee6b1bcfcb@mail.gmail.com>

Look at bind_meta_app in the default dialplan.  It binds the dtmf to
the features context.

On Sun, Jan 31, 2010 at 5:38 PM, mailinglist <mailinglist at fribert.dk> wrote:
> Ok, I've gotten the Freeswitch to register to my VoIP provider.
> I've gotten my phones to register to Freeswitch, and I can receive and make
> calls, all very nice.
>
> I've gotten a Sipura SPA901 and a Siemens Gigaset S68IP registered to the
> Freeswitch.
>
> When I receive a call, I would like to be able to transfer the call to
> another phone, or change the call to a conference call with two local
> phones.
>
> So I've been looking at the examples in the wiki, and I can't make them
> work, not as I understand them anyways. Especially the att_xfer seems to be
> able to do what I need.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
>
> As I understand Example1, I should answer the call, and then press *3 during
> the call, and either transfer it or change it to a threeway call.
>
> I get the first part, create an extension in the dialplan called att_xfer.
> But what is meant by the second par 'then bind this feature to DTMF 3', how
> do I enter that, and where?
>
> I hope somebody can help me with this (again)?
>
>
>
> Best regards
> Fribse
>
> /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From mouncifbb at gmail.com  Sun Jan 31 19:13:10 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Sun, 31 Jan 2010 22:13:10 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
	<b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
	<9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>
	<b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
Message-ID: <ade741d61001311913mf2916e6g20796f8c457b28f6@mail.gmail.com>

Got it! I appreciate your help very much!

On Sun, Jan 31, 2010 at 6:07 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Yes, you need to normalize the values passed to lcr.  Otherwise, how could
> it work?
>
> You can normalize the CID by matching and adding a 1 for 10 digit #s, or
> removing the leading + or other things you might need then setting it back
> to the profile using the set_profile_var app (
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var).
>  (mod_cidlookup will set it after doing a #->name/area lookup - but for now
> you can set it yourself)
>
> You can normalize the DID by doing similar matching rules as above and then
> transfering to that normalized DID for the rest of your call plan
> processing.
>
> I'm pretty sure mod_cidlookup has an example of normalizing... yeah:
> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Dialplan_Application
>
> On Sun, Jan 31, 2010 at 4:18 PM, Mouncifbb <mouncifbb at gmail.com> wrote:
>
>> So the CID must have 1 at front also? Usually people
>> Send only npa and nxx ex 6176427788 7817612233
>> Do I need to alter it?
>>
>> Sent from my iPhone
>>
>> On Jan 31, 2010, at 8:32 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>
>>
>> On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>> mouncifbb at gmail.com> wrote:
>>
>>> OK going back to use default profile to keep things simple below 2
>>> results
>>>
>>> Using:
>>>
>>> lcr 16179470890  default  19785223241  ( this one consult
>>> npa_nxx_company_ocn)
>>>
>>> lcr 6179470890  default  9785223241 ( this one don't!! )
>>>
>>>
>>>
>> Oh, right!  mod_lcr really expects you to normalize your prefix to e164
>> format.  I thought there was discussion about this in the wiki, but maybe
>> not.  For simple prefix matching it doesn't matter, but for things that make
>> decisions based on the # (like the lata/state stuff) it does.
>>
>> npanxx lookup only makes sense for NANPA numbers.  NANPA numbers have a
>> country code of "1" and a total length of 11 (including the 1).
>>
>> This is the only rational way to do it when you have a rate table with
>> both domestic (NANPA) and international prefixes.
>>
>>
>>> freeswitch> lcr 16179470890  default  19785223241
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>> [16179470890  default  19785223241]
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>> [19785223241]
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
>>> OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
>>> npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
>>> lata:1] so rate field is [intralata_rate]
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>> c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
>>> gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>> l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN
>>> carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND
>>> cg.enabled = '1' AND l.enabled = '1' AND digits IN (16179470890, 1617947089,
>>> 161794708, 16179470, 1617947, 161794, 16179, 1617, 161, 16, 1) AND
>>> CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>> intralata_rate, rand();
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>> of list
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end
>>> of list after carrier1
>>>
>>>
>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>> Dialstring
>>> |
>>>  | 1           | carrier1 | 0.00000  |       |            |
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> |
>>>  | 1           | carrier2 | 0.00000  |       |            |
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060 |
>>>
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060
>>>
>>>
>>>
>>>
>>>
>>> freeswitch> lcr 6179470890  default  9785223241
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>> [6179470890  default  9785223241]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>> [9785223241]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>> lata:0] so rate field is [rate]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>>> (6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
>>> AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>> rate, rand();
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>> of list
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>
>>>
>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>> Dialstring                                                                |
>>>   | 617947      | carrier1 | 0.09000  |       |            |
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker < <rupa at rupa.com>
>>> rupa at rupa.com> wrote:
>>>
>>>> turn up logging to debug again, and then reload mod_lcr.  It'll spit out
>>>> a bunch of crap when it tests out each profile you have defined.  Give me
>>>> the full log (here or in <http://pastebin.freeswitch.org>
>>>> pastebin.freeswitch.org).  That may show more useful info as to why
>>>> things are mucked up?
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>> mouncifbb at gmail.com> wrote:
>>>>
>>>>> yes I use NANPA_STD profile instead of default cause I thought the
>>>>> custom profile was causing issues, but looks like it's returning same
>>>>> results.
>>>>>
>>>>> There is this line in thw wiki:
>>>>> intra lata/state selection is done manually by setting the channel
>>>>> variables *intrastate* or *intralata* to the value *true*.
>>>>>
>>>>> do I have to set these ? if yes how?
>>>>>
>>>>> Thanks
>>>>>
>>>>>
>>>>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker < <rupa at rupa.com>
>>>>> rupa at rupa.com> wrote:
>>>>>
>>>>>> Stuff inline.
>>>>>>
>>>>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>
>>>>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>>>>> 201007,"224","7229","0.0059","0.0127"
>>>>>>> 201040,"224","9206","0.0036","0.0036"
>>>>>>>
>>>>>>
>>>>>> Looks like they give you the LATA and OCN values with the prefix.  We
>>>>>> (should) look that up ourselves.
>>>>>>
>>>>>>
>>>>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>>>>
>>>>>>>
>>>>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>>>>
>>>>>>> I also see this now when making a real call instead of running
>>>>>>> thorugh CLI
>>>>>>>
>>>>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890
>>>>>>> NANPA_STD)
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate
>>>>>>> channel var is [undef]*
>>>>>>
>>>>>>
>>>>>> This is fine.  it is a leftover from when you would tell mod_lcr via a
>>>>>> channel var that it should do intrastate.  I later had mod_lcr do the lookup
>>>>>> itself, but we still honor the old var.  There are no channel vars
>>>>>> associated with the cli, so you wouldn't see that msg.
>>>>>>
>>>>>>
>>>>>>>
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based
>>>>>>> on interstate rates
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>>>>> 16179470893 using profile NANPA_STD
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>> [state:0 lata:0] so rate field is [rate]
>>>>>>>
>>>>>>> called number 6179470890  caller ID: 6179472456
>>>>>>>
>>>>>>> any ideas??
>>>>>>>
>>>>>>>
>>>>>> Only thing that jumps out at me.
>>>>>>
>>>>>> The output from lcr_admin show profiles showed only the default one.
>>>>>>  On the dialplan you use the NANPA_STD profile.  Can you check lcr_admin
>>>>>> list and see if that profile is defined and if so if it says it is using the
>>>>>> npanxx table?
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker < <rupa at rupa.com>
>>>>>>> rupa at rupa.com> wrote:
>>>>>>>
>>>>>>>> Something is still missing from the logs. Note the query of the
>>>>>>>> npanxx
>>>>>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>>>>>> oh, what version of fs are you running?
>>>>>>>>
>>>>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>>>> <http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name>
>>>>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>>>>
>>>>>>>> An example from my own setup:
>>>>>>>>
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>>>>>>>> is [12148267711 default 12148267712]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>>>>> [12148267712]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT
>>>>>>>> 'state',
>>>>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata',
>>>>>>>> count(DISTINCT
>>>>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR
>>>>>>>> (npa=214
>>>>>>>> AND nxx=826)
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count:
>>>>>>>> 1
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count:
>>>>>>>> 1
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT
>>>>>>>> l.digits
>>>>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS
>>>>>>>> lcr_gw_prefix,
>>>>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c
>>>>>>>> ON
>>>>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id
>>>>>>>> =cg.carrier_id
>>>>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP
>>>>>>>> BETWEEN
>>>>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>>>>> random();
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us
>>>>>>>> to
>>>>>>>> head of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>>>>> [...]
>>>>>>>>
>>>>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>> > Also the Provider has presented the rates in this format?
>>>>>>>> > NPANXXLATA OCN INTER INTRA
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com>
>>>>>>>> > wrote:
>>>>>>>> >>
>>>>>>>> >> Tried it and it's not giving me intralata  instead I get
>>>>>>>> interstate, does
>>>>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?,
>>>>>>>> also do I have
>>>>>>>> >> to have the rate field in lcr table?
>>>>>>>> >>
>>>>>>>> >> lcr 617642  default 6176421212
>>>>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>>>>> Dialstring
>>>>>>>> >>                                                        |
>>>>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>>>>> >>
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to
>>>>>>>> lcr is
>>>>>>>> >> [617642  default 6176421212]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID
>>>>>>>> to
>>>>>>>> >> [6176421212]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>> [state:0
>>>>>>>> >> lata:0] so rate field is [rate]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>>>>> l.digits,
>>>>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>>>>> gw_suffix,
>>>>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>>>>>> l.cid FROM lcr
>>>>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg
>>>>>>>> ON
>>>>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1'
>>>>>>>> AND l.enabled
>>>>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>>>>> CURRENT_TIMESTAMP
>>>>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate,
>>>>>>>> rand();
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>> Dialstring
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1
>>>>>>>> to head
>>>>>>>> >> of list
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>> Dialstring
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>> >>
>>>>>>>> >> Thank you Rupa!
>>>>>>>> >>
>>>>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <<rupa at rupa.com>
>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>> >>>
>>>>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The
>>>>>>>> sql
>>>>>>>> >>> statements along with status info will show up.  This should
>>>>>>>> give
>>>>>>>> >>> enough information to debug what is happening.
>>>>>>>> >>>
>>>>>>>> >>> I'm assuming the npanxx table is actually populated and not just
>>>>>>>> >>> existing?
>>>>>>>> >>>
>>>>>>>> >>> When doing the lookup from the cli you have to tell lcr what CID
>>>>>>>> to
>>>>>>>> >>> use (remember, it is relative to the src/dest number).  I'm
>>>>>>>> pretty
>>>>>>>> >>> sure you get something on the console log when you don't specify
>>>>>>>> a CID
>>>>>>>> >>> when using the commandline.  Anyway:
>>>>>>>> >>>
>>>>>>>> >>> lcr 617642  default 6176421212
>>>>>>>> >>>
>>>>>>>> >>> should give you intralata.
>>>>>>>> >>>
>>>>>>>> >>> Note that the definition of intralata doesn't mean "local" for
>>>>>>>> some
>>>>>>>> >>> providers.  Some providers define local to "same ratecenter"
>>>>>>>> which is
>>>>>>>> >>> even more restrictive.
>>>>>>>> >>>
>>>>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com>
>>>>>>>> >>> wrote:
>>>>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I
>>>>>>>> am using
>>>>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>>>>> >>> >
>>>>>>>> >>> > lcr mysql table structure:
>>>>>>>> >>> > CREATE TABLE `lcr` (
>>>>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01
>>>>>>>> 00:00:00',
>>>>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>>>>> >>> >   KEY `digits` (`digits`),
>>>>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`)
>>>>>>>> REFERENCES
>>>>>>>> >>> > `carriers`
>>>>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > lcr_admin show profiles
>>>>>>>> >>> > Name:           default
>>>>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>>>>> l.${lcr_rate_field},
>>>>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>>>>>> >>> > l.trail_strip,
>>>>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers
>>>>>>>> c ON
>>>>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>>> WHERE
>>>>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>>>>> digits IN
>>>>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>>>>> date_start
>>>>>>>> >>> > AND
>>>>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality
>>>>>>>> DESC,
>>>>>>>> >>> >  reliability DESC, rand();
>>>>>>>> >>> >  has %:         false
>>>>>>>> >>> >  has vars:      true
>>>>>>>> >>> >  has intrastate:        true
>>>>>>>> >>> >  has intralata: true
>>>>>>>> >>> >  has npanxx:    true
>>>>>>>> >>> >  Reorder rate:  enabled
>>>>>>>> >>> >  Info in headers:       disabled
>>>>>>>> >>> >  Quote IN() List:       disabled
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > lcr 617642  default  returns rate from the rate field table
>>>>>>>> and not
>>>>>>>> >>> > intra/inter state fields rates.
>>>>>>>> >>> >
>>>>>>>> >>> > Any ideas? thanks!
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > _______________________________________________
>>>>>>>> >>> > FreeSWITCH-users mailing list
>>>>>>>> >>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >>> >
>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >>> >
>>>>>>>> >>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>>
>>>>>>>> >>>
>>>>>>>> >>>
>>>>>>>> >>> --
>>>>>>>> >>> -Rupa
>>>>>>>> >>>
>>>>>>>> >>> _______________________________________________
>>>>>>>> >>> FreeSWITCH-users mailing list
>>>>>>>> >>> <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >>
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > _______________________________________________
>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >
>>>>>>>> >
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> -Rupa
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Sun Jan 31 20:36:25 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 31 Jan 2010 22:36:25 -0600
Subject: [Freeswitch-users] Equivalent to Asterisk's"directrtpsetup=yes"?
In-Reply-To: <15D48404014D48D19F85CFFFC4BBC76F@cune.pri>
References: <gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b@4ax.com><gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b-e09XROE/p8c@public.gmane.org><20100129164359.9B3C22B10D@cuneorg-email.cune.pri><j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8@4ax.com><j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8-e09XROE/p8c@public.gmane.org><4468a6771001310247n4bf5d1a4rf4aea9f2d6c35a1f@mail.gmail.com>
	<qiqbm55lueuqup1m6r6d0mff36b4l4fb9l@4ax.com>
	<15D48404014D48D19F85CFFFC4BBC76F@cune.pri>
Message-ID: <7FA9CDA6-EAB3-4C1D-9CD3-8D4B06B0F335@freeswitch.org>

Going back no_media after hold isn't supported yet.. Anthony said he would add it if someone really really wanted it and posted a bounty of $500 to cover his time to implement it.

/b

On Jan 31, 2010, at 4:35 PM, Russell Mosemann wrote:

> I wasn't paying close attention, but in a recent discussion, someone wanted to have moh in a bypass-media situation. I think there was a way to do that, but then there was an issue with going back to bypass-media after the call was taken off hold. If you look through the list archive, you should be able to find it.



From wasim at convergence.pk  Sun Jan 31 21:01:26 2010
From: wasim at convergence.pk (Wasim Baig)
Date: Mon, 1 Feb 2010 10:01:26 +0500
Subject: [Freeswitch-users] SS7 & MGCP support
In-Reply-To: <SNT106-W1023694FA10CC0F668ADCDF45A0@phx.gbl>
References: <SNT106-W1023694FA10CC0F668ADCDF45A0@phx.gbl>
Message-ID: <b8ad2a5b1001312101u68c2f5a9sa66fb45f114d6bac@mail.gmail.com>

On Sat, Jan 30, 2010 at 12:48 PM, Abid Saleem <abid_freeswitch at live.com>wrote:

> ?         Since it is a softswitch also, does it support SS7, MGCP and
> Megaco protocols to control media gateways?
>
I remember back in the day when FS was a glimmer in tony's eye and we had a
mostly working (but not stable) MGCP UA mode with it ... but then I don't
think any more work was done it. My requirement for it has phased away also,
although if anyone is really interested I'm sure we could come up with a
bounty for this, as MGCP is fairly basic.

The only option you have for SS7 is a commercial implementation from
Sangoma. No open source implementation for SS7 or SIGTRAN exist as of now,
although there are a couple of rumors in the air about diverse efforts.

> ?         Does it support call shops business model?
>
Technically yes, but the logic and billing etc is all up to you, so out of
the box, is it a preconfigured call shop system, no. Can it be used for one,
certainly. Will you have to work a bit for it, most cetainly, yes.

> ?         How to add new SIP user accounts into it that can be used to
> register to it. I know one way is to copy and paste          1000.xml file
> and edit it in the conf/directory folder. What is the optimal way to do this
> task?
>
You can use a DB for this as well and have the XML generated or used inline.
http://wiki.freeswitch.org/wiki/Mod_xml_curl

> o   Is there any GUI available. If yes how can I make it work and private
> label it.?
>
There are a couple of efforts. Read the wiki. You could use ASTPP with
reseller option to white label it.

-- 
wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
peace be upon you ...
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From mike at jerris.com  Sun Jan 31 21:53:07 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 00:53:07 -0500
Subject: [Freeswitch-users] Replace Internal IP with External IP in From
	Header
In-Reply-To: <4571ff701001261738w4f51b33dqee1d19b8d0e2236d@mail.gmail.com>
References: <4571ff701001261738w4f51b33dqee1d19b8d0e2236d@mail.gmail.com>
Message-ID: <27B2F566-0C42-4A55-8D62-D5E0D5D5EDE7@jerris.com>

Try current trunk, I think this is fixed now, if not, please open a bug on jira.FreeSWITCH.org


On Jan 26, 2010, at 8:38 PM, Code Ghar wrote:

> I followed the example in Freeswitch behind NAT (http://wiki.freeswitch.org/wiki/NAT_Traversal#Freeswitch_behind_NAT). In the Contact header of invite sent to an external gateway, I see sip:extension at ExternalIP:port but in the From header I see sip:extension at InternalIP. How can I change the From header of SIP message so that it displays the external IP instead of internal IP? The reason for doing this is that the external gateway authenticates and authorizes call based on the IP in From header. They expect an external IP and not an internal IP. 
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From mike at jerris.com  Sun Jan 31 21:58:21 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 00:58:21 -0500
Subject: [Freeswitch-users] Question about javascript
In-Reply-To: <20100127123533.602850e1.matsubara_tomoya@intec.co.jp>
References: <20100127123533.602850e1.matsubara_tomoya@intec.co.jp>
Message-ID: <7C99C342-AB78-4959-9508-9BB8CA5FE29F@jerris.com>

You can see if you have a memory leak by running in valgrind.  In the mean time, why would you do this in a lua script, all of this is easier and more clear to do right in dialplan.
also, your running ring ready, then answering, then setting ringback?  it seems like you have everything mixed up here.

This probably should just be a 5 line dialplan.  
set ringback
set continue on fail true
set hangup after bridge true
bridge (with api on answer)
playback test02.wav

Mike

On Jan 26, 2010, at 10:35 PM, Tomoya Matsubara wrote:

> Hello,
> 
> When the following scripts were tested, it seems to do the memory leak. 
> Please teach when there is a problem in this script. 
> 
> -- test script --
> session.execute("ring_ready");
> session.answer();
> session.setVariable("ringback", "%(1000, 2000, 440, 460)");
> 
> var bleg = new Session();
> var sound_wav = "sounds/test01.wav";
> var sound_leg = "both";
> var op = "api_on_answer=uuid_broadcast "+session.uuid+" "+sound_wav+" "+sound_leg;
> var ret = bleg.originate(session, "{"+op+"}" + "sofia/gateway/profile0_gateway1/1000");
> if(!session.ready()){
>  return;
> }
> 
> if(!ret){ // bleg not answered.
>  var sound_wav = "sounds/test02.wav";
>  session.streamFile(sound_wav);
>  if(session.ready()){
>    session.hangup();
>  }
>  return;
> }
> 
> if(bleg.ready()){
>  bridge(session, bleg);
> }
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Sun Jan 31 21:59:15 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 00:59:15 -0500
Subject: [Freeswitch-users] Wrong RTP port submitted?
In-Reply-To: <4B6011EF.6090706@gmx.net>
References: <4B6011EF.6090706@gmx.net>
Message-ID: <48056347-BFCC-4B38-8C29-9FB592033827@jerris.com>

please report this bug to jira.FreeSWITCH.org.

On Jan 27, 2010, at 5:14 AM, Peter P GMX wrote:

> I have defined the rtp port range for 12000-12100 in switch.conf.xml.
> However Freeswitch is offering a port 48320 in the invite message. The
> result is, that the incoming RTP stream is blocked by the firewall (I
> can see a reject for UDP 48320).
> Any hint how to solve this?



From mike at jerris.com  Sun Jan 31 22:04:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 01:04:20 -0500
Subject: [Freeswitch-users] Logging question
In-Reply-To: <da4ab26e1001292020o280f330ft62a278b93c7fb1fd@mail.gmail.com>
References: <da4ab26e1001292020o280f330ft62a278b93c7fb1fd@mail.gmail.com>
Message-ID: <8BEB200D-32AC-4D80-B59D-07C8228D7380@jerris.com>

sofia profile <profile_name> siptrace on

There is also a config param, it should be documented int he current default configs.

Mike


On Jan 29, 2010, at 11:20 PM, paul gore wrote:

> Hi there,
> I am running FS 1.0.trunk (14501) (I know it's old but we serve a small community and don't have time to upgrade/test the latest/greatest). I am having troubles understanding how to switch SIP trace in log files, I tried
> 
> fsctl loglevel debug
> sofia tracelevel debug
> 
> but it seem to have no effect, I only get sofia debug messages but no detailed SIP info.
> What also puzzling me is if I do 
> 
> console loglevel 0
> 
> I still get debug information on console.
> What am I doing wrong?



From mike at jerris.com  Sun Jan 31 22:05:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 01:05:37 -0500
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001300601r29530dcfm65c67e9121d3ddde@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<D56D65EB-B150-4A59-8B66-68BF4C58108B@jerris.com>
	<d361ca7b1001241004kcb73f11hc400bdb37c9d372b@mail.gmail.com>
	<3A27F063-E0C0-4178-A3AF-068956B55846@jerris.com>
	<d361ca7b1001241117q7f1379bewe1bc85279ef67c83@mail.gmail.com>
	<alpine.GSO.2.00.1001250716120.17161@h1.pp.uhwv.np.vy>
	<d361ca7b1001242150t29b36ebcy6c2e40ca3d360c00@mail.gmail.com>
	<d361ca7b1001251143p7f396dd8h11d372731a26f4d7@mail.gmail.com>
	<224C684A-B357-42E4-98AA-0EE238A27A49@jerris.com>
	<d361ca7b1001251206x21242589he3014081a7e110d1@mail.gmail.com>
	<d361ca7b1001300601r29530dcfm65c67e9121d3ddde@mail.gmail.com>
Message-ID: <08874DB8-D35F-47CA-8A5A-B6BF33C9D6B4@jerris.com>

Thanks again for you help reproducing this so we could chase this issue down.

Mike

On Jan 30, 2010, at 9:01 AM, Yehavi Bourvine wrote:

> It works ok now (fixed on r16534).
>  
>                 Thanks! __Yehavi:
> 



From mike at jerris.com  Sun Jan 31 22:08:01 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 01:08:01 -0500
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
References: <874941.17255.qm@web33502.mail.mud.yahoo.com>
	<df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
Message-ID: <35B856D0-81DB-4EF9-A376-D0B32780FD30@jerris.com>

api would not have a call associated with it at all.


On Jan 31, 2010, at 8:45 AM, Scott Fernandez wrote:

> Hi,
>  
> Thx for the information. Can I have some detailed steps to configure mod_managed class call control and how do we write the API commands in .Net applications?
>  
> In addition, how do we get the current STATE of the call when I use webapi?. Because it is required for me to route the call to the user upon it is answered or disconnect  it.
>  



From mike at jerris.com  Sun Jan 31 22:09:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 01:09:20 -0500
Subject: [Freeswitch-users] How to populate Voice recording metadata
	with custom CRM data?
In-Reply-To: <003c01caa23b$83ed0800$8bc71800$@co.za>
References: <003c01caa23b$83ed0800$8bc71800$@co.za>
Message-ID: <A86F32B9-6BF0-4344-B948-675AF39FF56F@jerris.com>

There is a later windows installer on the downloads site as well

http://files.freeswitch.org/windows_installer/

On Jan 31, 2010, at 1:05 AM, Roly Maz wrote:

> Hi All,  
>  
> I am using Freeswitch 1.0.4 (Latest Windows Installer version) on Windows 7 Ultimate for testing

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From mike at jerris.com  Sun Jan 31 22:11:58 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 01:11:58 -0500
Subject: [Freeswitch-users] Trunk Version Number
In-Reply-To: <2360B060EDF44D368707DD12064BD416@bp1.ad.bp.com>
References: <2360B060EDF44D368707DD12064BD416@bp1.ad.bp.com>
Message-ID: <8976E2C4-CF8C-433A-AC79-B58F65105D08@jerris.com>

it should.  This can happen if you build from an svn checkout and the svn client your using is newer than our static linked svnversion.exe.  If anyone can make me a newer stripped down version like that I would appreciate it I have not had the time.

On Jan 31, 2010, at 9:30 AM, Dave Stevenson wrote:

> Hi,
>  
> Running the latest SVN (16453) under Windows, the console "Version" command displays :-
>  
> "FreeSWITCH Version 1.0.trunk (UNKNOWN)"
>  
> Should the version number not include a meaningful build version in the brackets ?
>  
> regards
> Dave
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Sun Jan 31 22:14:14 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 01:14:14 -0500
Subject: [Freeswitch-users] determining the source of receyved call in
	public context
In-Reply-To: <4B65E649.2040007@gmail.com>
References: <4B65E649.2040007@gmail.com>
Message-ID: <46204926-1DCE-4C45-B31B-30CC698AE636@jerris.com>

What does the IP have to do with the numbers?  Can you rephrase the question?

On Jan 31, 2010, at 3:21 PM, Meftah Tayeb wrote:
> how do i determine the gateway or the Ip of a receyved call from ITSP's?
> i am calling my did from my mobile, but i see is processing the the 
> mobile number no the ITSP User or did



From mike at jerris.com  Sun Jan 31 22:26:36 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 01:26:36 -0500
Subject: [Freeswitch-users] vm-disk-quota
In-Reply-To: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>
References: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>
Message-ID: <7DFBD163-A72A-4186-BB89-6D468FD9ABA2@jerris.com>

If I read the code right (mod_voicemail.c:3051) it looks like it is measuring in seconds of vocicemail, but the wiki indicates number of voicemails, neither seems to match the name of the param.

Tamas-  Can you comment on how this was intended to work?

Mike

On Jan 31, 2010, at 3:46 PM, mbsip wrote:

> Hi ALL,
> 
> Maybe this question will be piece of cake for most of you, but it
> makes me think.
> 
> I would like to configure "vm-disk-quota" for all users i have.
> I followed the wiki page and provided:
> <param name="vm-disk-quota" value="10"/>
> to /conf/directory/default/1000.xml
> 
> After reloadxml, incoming call give me "mod_voicemail.c:3057 Voicemail
> disk quota is exceeded" feedback
> No surprise for me because i had more less 10 voice mails already
> recorded (before the vm-disk-quota was set up).
> Strange is that increasing value even to 100 does not change anything.
> The same thing with deleting recordings from user directory.
> The only wayout is to set it to default value=0 (even FS shutdown
> doesn't change anything)
> 
> I am wondering why vm-disk-quota produces "Voicemail disk quota is
> exceeded" all the time
> Where the module is looking for stored voicemail recordings.
> 
> Below is part of my configuration.
> 1) /conf/autoload_configs/voicemail.conf.xml
> <param name="storage-dir" value="/usr/local/freeswitch/vm"/>
> 2) /conf/directory/default/1000.xml
> <param name="vm-disk-quota" value="100"/>
> 3) /vm/FS_ip_address/1000 is empty



From ken at ukgb.net  Fri Jan  1 02:45:34 2010
From: ken at ukgb.net (Ken Gillett)
Date: Fri, 1 Jan 2010 10:45:34 +0000
Subject: [Freeswitch-users] video
Message-ID: <90AC63C4-48ED-43F7-A534-CB90C208604F@ukgb.net>

Sorry if this is a really basic question, but how do SIP video phones work and what capabilities does FreeSwitch have in this regard?

Does a PBX such as FS have to deal with the video stream, or does it just pass around a URL that the phone then uses as the source for the streaming video? Or is the video data all 'switched' along with the audio data? How would conference calls be handled?

Is this part of the SIP standard, or is it all proprietary and just down to the phone designer/manufacturer?


Ken G i l l e t t

_/_/_/_/_/_/_/_/






From tculjaga at gmail.com  Fri Jan  1 04:52:44 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 1 Jan 2010 13:52:44 +0100
Subject: [Freeswitch-users] freeswitch and H323
In-Reply-To: <87f2f3b90912310920q1821fe0eu5fd74e553e4cb12@mail.gmail.com>
References: <7aa8bd9d0912300514r68c90b12u7c631a649981cfa3@mail.gmail.com>
	<87f2f3b90912310920q1821fe0eu5fd74e553e4cb12@mail.gmail.com>
Message-ID: <65d96fc81001010452k37068e87sc0a229cdfe481c40@mail.gmail.com>

well, mod_h323 works for me... there are still some missing things and of
course bugs ... e.g. incorrect releaseCause mapping, no automatic codec
ptime sync... but it is usable ....


if you'd like to go mod_h323 way i can help you... it builds as a charm for
me...


T.




On Thu, Dec 31, 2009 at 6:20 PM, Michael Collins <msc at freeswitch.org> wrote:

> Are you trying to use mod_h323 or mod_opal? They are both works in
> progress, but the latter is farther along than the former. Use the latest
> FreeSWITCH trunk (or latest.freeswitch.org) and run the buildopal.sh
> script in the build directory. If you have any build issues then paste the
> log on pastebin.freeswitch.org and reply to this thread with the PB URL so
> that we can take a look.
> -MC
>
>
> On Wed, Dec 30, 2009 at 5:14 AM, Pete Kay <petedao at gmail.com> wrote:
>
>> Hi,
>>
>> has anyone been able to get H323 to work?
>>
>> I have problem trying to get it compiled with either 1.0.4 or 1.0.5.
>>
>> Thanks,
>> pete
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ken at ukgb.net  Fri Jan  1 05:03:31 2010
From: ken at ukgb.net (Ken Gillett)
Date: Fri, 1 Jan 2010 13:03:31 +0000
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <207e7a5e0912301103h17133fdfk22430bbce5a1718b@mail.gmail.com>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com>
	<B6763940-A3CB-4006-8DF1-ADEDD9C68294@ukgb.net>
	<0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org>
	<CA53F057-4A51-457C-A673-B0B6E4DF482A@ukgb.net>
	<207e7a5e0912301103h17133fdfk22430bbce5a1718b@mail.gmail.com>
Message-ID: <090E289D-76EE-4F4A-82DF-513626A915E6@ukgb.net>

Brilliant. Thank you. That's exactly what I needed to know.


On 30 Dec 2009, at 19:03, jonathan augenstine wrote:

> Ken,
> 
> configure
> make
> make install
> 
> This sequence of steps builds and installs the default configuration but without the audio files.  If you want the sound files installed also then:
> 
> make install sounds-install moh-install
> 
> Now the default sound files for conferencing, voicemail and music on hold are installed.
> 
> If you want to modify the default install to customize the build you can add and remove modules in modules.conf.  Then you run make/make install again to build those modules that are now included in the edited modules.conf file.
> 
> Jonathan
> 
> On Wed, Dec 30, 2009 at 10:45 AM, Ken Gillett <ken at ukgb.net> wrote:
> This is beginning to confuse me. Some say just:
> 
> > - configure
> > - make
> > - make install
> 
> is required, but the docs say more is needed for modules.conf. I'm still not sure if this only applies when modules.conf has been edited. Anyone help there?
> 
> On 28 Dec 2009, at 14:37, Brian West wrote:
> 
> > "all" is no longer needed.
> >
> > /b
> >
> > On Dec 25, 2009, at 3:07 AM, Ken Gillett wrote:
> >
> >>      make all install sounds-install moh-install.
> 
> So
> 
>        make install sounds-install moh-install.
> 
> is required? Always? Why?
> 
> Also, to bring this topic back to my original question (not that the diversity hasn't been interesting:-)
> 
> How can I best compile FS on one Mac and install it onto a different Mac?
> 
> 
> 
> Ken G i l l e t t
> 
> _/_/_/_/_/_/_/_/
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



Ken G i l l e t t

_/_/_/_/_/_/_/_/






From mike at jerris.com  Fri Jan  1 07:25:02 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 1 Jan 2010 10:25:02 -0500
Subject: [Freeswitch-users] Self alarm
In-Reply-To: <1262326847726-4238924.post@n2.nabble.com>
References: <1262250725607-4235713.post@n2.nabble.com>
	<1262326847726-4238924.post@n2.nabble.com>
Message-ID: <2A1E0DAF-A680-47B6-AC49-6A80FEC312A2@jerris.com>

The same what?

On Jan 1, 2010, at 1:20 AM, Sharad <sharad at coraltele.com> wrote:

>
> Hi
>
> I am also intresting in the same.
>
> Is there any script for this functionality.
>
> Regards
> --  
> View this message in context: http://n2.nabble.com/Self-alarm-tp4235713p4238924.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From mike at jerris.com  Fri Jan  1 07:27:26 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 1 Jan 2010 10:27:26 -0500
Subject: [Freeswitch-users] video
In-Reply-To: <90AC63C4-48ED-43F7-A534-CB90C208604F@ukgb.net>
References: <90AC63C4-48ED-43F7-A534-CB90C208604F@ukgb.net>
Message-ID: <C0A6C497-2073-4D09-8847-C078C4C81577@jerris.com>

Just like the audio, but in a different stream.  We don't do video  
transcoding at this time so it is passthrough only.  We have basic  
support for video follow audio in conference bit it is still rough on  
transitions.

Mike

On Jan 1, 2010, at 5:45 AM, Ken Gillett <ken at ukgb.net> wrote:

> Sorry if this is a really basic question, but how do SIP video  
> phones work and what capabilities does FreeSwitch have in this regard?
>
> Does a PBX such as FS have to deal with the video stream, or does it  
> just pass around a URL that the phone then uses as the source for  
> the streaming video? Or is the video data all 'switched' along with  
> the audio data? How would conference calls be handled?
>
> Is this part of the SIP standard, or is it all proprietary and just  
> down to the phone designer/manufacturer?
>
>
> Ken G i l l e t t
>
> _/_/_/_/_/_/_/_/
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From aep.lists at it46.se  Fri Jan  1 10:03:28 2010
From: aep.lists at it46.se (Alberto Escudero)
Date: Fri, 1 Jan 2010 19:03:28 +0100
Subject: [Freeswitch-users] XML IVR and Javascript menu-exec-app
Message-ID: <7d4c9553f55e0af32200eebdd332b2f9.squirrel@correo.nodo50.org>

Hi,

I am writing several IVRs using Freeswitch XML
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr

One of the nodes of the IVR is a Javascript application that records a
message.
e.g.:
<entry action="menu-exec-app" digits="3" param="javascript foo.js"/>

The Javascript application starts by issuing a
session.answer()

[records the voice message]

exit();

Once the Javascript exits, the channel is dropped and hence the IVR
terminates.
Is it possible to write a Javascript application that once is completed,
the channel returns back to the top menu of the IVR? I want to emulate the
same behavior that "menu-play-sound", that once the file is played, the
IVR logic returns to the top menu.

-aep

-- 
Stopping junk mailers is good for the environment






From jcasale at activenetwerx.com  Fri Jan  1 10:06:30 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 1 Jan 2010 18:06:30 +0000
Subject: [Freeswitch-users] Zap dialplan
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9068B44@prato.activenetwerx.local>

All the examples show bridging calls to an fx(s|o) port, but none show how
to handle an incoming call from the pstn on an fxo port. How do you distinguish
this call in your dialplan and begin routing it?

Does fs just place it in the public context, and if so iirc pstn calls don't have
a "destination_number" to match on, but a caller id only?

Thanks,
jlc



From codecomplete at free.fr  Fri Jan  1 14:18:05 2010
From: codecomplete at free.fr (Fred-145)
Date: Fri, 1 Jan 2010 14:18:05 -0800 (PST)
Subject: [Freeswitch-users] [Windows] Stable enough for production use?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C55540C9EAA@cooper>
References: <26807322.post@talk.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C55540C9EAA@cooper>
Message-ID: <26988383.post@talk.nabble.com>


Thanks everyone for the feedback.
-- 
View this message in context: http://old.nabble.com/-Windows--Stable-enough-for-production-use--tp26807322p26988383.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From mrene_lists at avgs.ca  Fri Jan  1 18:07:49 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sat, 2 Jan 2010 03:07:49 +0100
Subject: [Freeswitch-users] Zap dialplan
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9068B44@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E9068B44@prato.activenetwerx.local>
Message-ID: <3C85D953-8862-4006-9DE4-FFB40AF4BD8A@avgs.ca>

You define the destination number and context in the port's config  
section, in openzap.conf.xml

Sent from my iPhone

On 2010-01-01, at 7:06 PM, "Joseph L. Casale"  
<jcasale at activenetwerx.com> wrote:

> All the examples show bridging calls to an fx(s|o) port, but none  
> show how
> to handle an incoming call from the pstn on an fxo port. How do you  
> distinguish
> this call in your dialplan and begin routing it?
>
> Does fs just place it in the public context, and if so iirc pstn  
> calls don't have
> a "destination_number" to match on, but a caller id only?
>
> Thanks,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From jcasale at activenetwerx.com  Fri Jan  1 20:28:36 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 2 Jan 2010 04:28:36 +0000
Subject: [Freeswitch-users] More Zap issues
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906996A@prato.activenetwerx.local>

Now that the incoming zap is routed (and not hitting enum to get where it
needs to be) I thought I would just send it to the example ivr (as I am remote)
so I could see what happens at the cli.

I can see it enter the public context then find the extension which sends it to
5000 XML default. I see all the prompts being played at the cli but there is nothing
heard at the remote end?

Any ideas what's wrong? During initial setup when it was hitting enum, people on the
local end heard the ring and where able to answer and communicate with their sip phones?

Thanks!
jlc



From sharad at coraltele.com  Fri Jan  1 20:35:14 2010
From: sharad at coraltele.com (Sharad)
Date: Fri, 1 Jan 2010 20:35:14 -0800 (PST)
Subject: [Freeswitch-users] [!! SPAM]  Re:  Self alarm
In-Reply-To: <2A1E0DAF-A680-47B6-AC49-6A80FEC312A2@jerris.com>
References: <1262250725607-4235713.post@n2.nabble.com>
	<1262326847726-4238924.post@n2.nabble.com>
	<2A1E0DAF-A680-47B6-AC49-6A80FEC312A2@jerris.com>
Message-ID: <002b01ca8b65$16762640$0c04a8c0@compaq77db609e>


Self Alarm..
  ----- Original Message ----- 
  From: Michael Jerris [via freeswitch-users] 
  To: Sharad 
  Sent: Friday, January 01, 2010 9:04 PM
  Subject: [!! SPAM] Re: [Freeswitch-users] Self alarm


  The same what? 

  On Jan 1, 2010, at 1:20 AM, Sharad <[hidden email]> wrote: 


  > 
  > Hi 
  > 
  > I am also intresting in the same. 
  > 
  > Is there any script for this functionality. 
  > 
  > Regards 
  > --   
  > View this message in context: http://n2.nabble.com/Self-alarm-tp4235713p4238924.html
  > Sent from the freeswitch-users mailing list archive at Nabble.com. 
  > 
  > _______________________________________________ 
  > FreeSWITCH-users mailing list 
  > [hidden email] 
  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-  
  > users 
  > http://www.freeswitch.org

  _______________________________________________ 
  FreeSWITCH-users mailing list 
  [hidden email] 
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org



------------------------------------------------------------------------------

  View message @ http://n2.nabble.com/Self-alarm-tp4235713p4239714.html 
  To unsubscribe from Re: Self alarm, click here. 


-- 
View this message in context: http://n2.nabble.com/Self-alarm-tp4235713p4241557.html
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From jaugenstine at gmail.com  Fri Jan  1 20:50:55 2010
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Fri, 1 Jan 2010 20:50:55 -0800
Subject: [Freeswitch-users] [!! SPAM] Re: Self alarm
In-Reply-To: <002b01ca8b65$16762640$0c04a8c0@compaq77db609e>
References: <1262250725607-4235713.post@n2.nabble.com>
	<1262326847726-4238924.post@n2.nabble.com>
	<2A1E0DAF-A680-47B6-AC49-6A80FEC312A2@jerris.com>
	<002b01ca8b65$16762640$0c04a8c0@compaq77db609e>
Message-ID: <207e7a5e1001012050g18e5563dpe99e55b0509c5625@mail.gmail.com>

I believe that the question is, what do you want to alarm?  Do you want to
setup basic monitoring of the system?  Are you trying to track T1 alarms?
 Your question is too vague to answer.

On Fri, Jan 1, 2010 at 8:35 PM, Sharad <sharad at coraltele.com> wrote:

> Self Alarm..
>
> ----- Original Message -----
> *From:* [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4241557&i=0>
> *To:* [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4241557&i=1>
> *Sent:* Friday, January 01, 2010 9:04 PM
> *Subject:* [!! SPAM] Re: [Freeswitch-users] Self alarm
>
> The same what?
>
> On Jan 1, 2010, at 1:20 AM, Sharad <[hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4239714&i=0>>
> wrote:
>
> >
> > Hi
> >
> > I am also intresting in the same.
> >
> > Is there any script for this functionality.
> >
> > Regards
> > --
> > View this message in context:
> http://n2.nabble.com/Self-alarm-tp4235713p4238924.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4239714&i=1>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> [hidden email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4239714&i=2>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
> View this message in context: Re: [!! SPAM] Re: [Freeswitch-users] Self
> alarm <http://n2.nabble.com/Self-alarm-tp4235713p4241557.html>
>
> Sent from the freeswitch-users mailing list archive<http://n2.nabble.com/freeswitch-users-f2379917.html>at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Fri Jan  1 23:30:02 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sat, 2 Jan 2010 18:30:02 +1100
Subject: [Freeswitch-users] No audio after Remote SDP:
In-Reply-To: <7D75E6D6-8AB9-429E-A5A9-1639C0D5AD09@freeswitch.org>
References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com>
	<11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org>
	<33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com>
	<7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org>
	<33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com>
	<772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org>
	<33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com>
	<191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com>
	<33c87fa30912301213g3d24b773k86b95e7ce08f6761@mail.gmail.com>
	<7D75E6D6-8AB9-429E-A5A9-1639C0D5AD09@freeswitch.org>
Message-ID: <33c87fa31001012330g69fc8e88m61e648d893c4d8f7@mail.gmail.com>

Hi!

Both are auto-nat:
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>

FreeSWITCH Version 1.0.trunk (15490)

However, isn't it the IP address that is reported by the remote
SPA3102 that is incorrect?  Or?

On Thu, Dec 31, 2009 at 7:21 AM, Brian West <brian at freeswitch.org> wrote:
> show me the ext-rtp-ip and ext-sip-ip settings you're using along with SVN rev please.
>
> /b
>
> On Dec 30, 2009, at 2:13 PM, Mark Campbell-Smith wrote:
>
>> Hi Anthony,
>>
>> The only profiles I have defined are external and internal. ? These
>> should be using internal...
>>
>> 192.168.1.120 is the FS box, which is NAT'd. ?Never had any problems
>> with this being NAT'd though
>> 192.168.1.121 is a PAP2 ATA connected to FS
>>
>> I don't use proxy media.
>>
>> I am trying to call an SPA3102, which is on the internet and NAT'd
>> (external IP address 11.11.11.11 in the trace and internal/private ip
>> address of 192.168.1.3).
>>
>> Thanks!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From sharad at coraltele.com  Fri Jan  1 23:58:59 2010
From: sharad at coraltele.com (Sharad)
Date: Fri, 1 Jan 2010 23:58:59 -0800 (PST)
Subject: [Freeswitch-users]  User's Mailbox Password
Message-ID: <1262419139401-4241899.post@n2.nabble.com>


Hi

When a user changes his mailbox password from his phone using advance
options, the corresponding XML does not show the new password. 

Can someone tell me what is the use of vm-password parameter which is shown
in the XML of that user.

regards
Sharad
-- 
View this message in context: http://n2.nabble.com/User-s-Mailbox-Password-tp4241899p4241899.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From ken at ukgb.net  Sat Jan  2 05:17:57 2010
From: ken at ukgb.net (Ken Gillett)
Date: Sat, 2 Jan 2010 13:17:57 +0000
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <090E289D-76EE-4F4A-82DF-513626A915E6@ukgb.net>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com>
	<B6763940-A3CB-4006-8DF1-ADEDD9C68294@ukgb.net>
	<0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org>
	<CA53F057-4A51-457C-A673-B0B6E4DF482A@ukgb.net>
	<207e7a5e0912301103h17133fdfk22430bbce5a1718b@mail.gmail.com>
	<090E289D-76EE-4F4A-82DF-513626A915E6@ukgb.net>
Message-ID: <79B790D4-DE06-47FF-B9FD-52D9E1EB8BCB@ukgb.net>

One question still outstanding:-

How can I compile FS on one Mac and install it onto a different Mac? This means compiling on a MacPro running Snow Leopard and then installing onto a Snow Leopard Server which doesn't have the developer tools installed (and I don't want it to).

Advice on this would be appreciated.



Ken G i l l e t t

_/_/_/_/_/_/_/_/






From brian at freeswitch.org  Sat Jan  2 08:03:59 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 2 Jan 2010 10:03:59 -0600
Subject: [Freeswitch-users] No audio after Remote SDP:
In-Reply-To: <33c87fa31001012330g69fc8e88m61e648d893c4d8f7@mail.gmail.com>
References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com>
	<11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org>
	<33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com>
	<7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org>
	<33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com>
	<772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org>
	<33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com>
	<191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com>
	<33c87fa30912301213g3d24b773k86b95e7ce08f6761@mail.gmail.com>
	<7D75E6D6-8AB9-429E-A5A9-1639C0D5AD09@freeswitch.org>
	<33c87fa31001012330g69fc8e88m61e648d893c4d8f7@mail.gmail.com>
Message-ID: <180A53B1-EFB6-4B17-AA7C-19B007ADE04C@freeswitch.org>

Are you behind a nat-pmp/upnp router?

/b

On Jan 2, 2010, at 1:30 AM, Mark Campbell-Smith wrote:

> Hi!
> 
> Both are auto-nat:
>    <param name="ext-rtp-ip" value="auto-nat"/>
>    <param name="ext-sip-ip" value="auto-nat"/>
> 
> FreeSWITCH Version 1.0.trunk (15490)
> 
> However, isn't it the IP address that is reported by the remote
> SPA3102 that is incorrect?  Or?
> 
> On Thu, Dec 31, 2009 at 7:21 AM, Brian West <brian at freeswitch.org> wrote:
>> show me the ext-rtp-ip and ext-sip-ip settings you're using along with SVN rev please.
>> 
>> /b

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From brian at freeswitch.org  Sat Jan  2 08:04:52 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 2 Jan 2010 10:04:52 -0600
Subject: [Freeswitch-users] User's Mailbox Password
In-Reply-To: <1262419139401-4241899.post@n2.nabble.com>
References: <1262419139401-4241899.post@n2.nabble.com>
Message-ID: <3B8B47CA-3751-4FAE-BB17-3E482832869F@freeswitch.org>

The one from the XML will never change its stored in the db table in voicemail if you change it... but if you're using XML curl we do a request to let you know its updated so you can do what ever to update the db.

/b

On Jan 2, 2010, at 1:58 AM, Sharad wrote:

> 
> Hi
> 
> When a user changes his mailbox password from his phone using advance
> options, the corresponding XML does not show the new password. 
> 
> Can someone tell me what is the use of vm-password parameter which is shown
> in the XML of that user.
> 
> regards
> Sharad




From jaugenstine at gmail.com  Sat Jan  2 08:40:15 2010
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Sat, 2 Jan 2010 08:40:15 -0800
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <79B790D4-DE06-47FF-B9FD-52D9E1EB8BCB@ukgb.net>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com>
	<B6763940-A3CB-4006-8DF1-ADEDD9C68294@ukgb.net>
	<0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org>
	<CA53F057-4A51-457C-A673-B0B6E4DF482A@ukgb.net>
	<207e7a5e0912301103h17133fdfk22430bbce5a1718b@mail.gmail.com>
	<090E289D-76EE-4F4A-82DF-513626A915E6@ukgb.net>
	<79B790D4-DE06-47FF-B9FD-52D9E1EB8BCB@ukgb.net>
Message-ID: <207e7a5e1001020840y1bd3bae2y2b683a19659c9d7d@mail.gmail.com>

A dmg install package would need to be created.  A default package does not
currently exist and it would need to be created.  You would need to do this,
particularly if you are going to customize the build/install.

On Sat, Jan 2, 2010 at 5:17 AM, Ken Gillett <ken at ukgb.net> wrote:

> One question still outstanding:-
>
> How can I compile FS on one Mac and install it onto a different Mac? This
> means compiling on a MacPro running Snow Leopard and then installing onto a
> Snow Leopard Server which doesn't have the developer tools installed (and I
> don't want it to).
>
> Advice on this would be appreciated.
>
>
>
> Ken G i l l e t t
>
> _/_/_/_/_/_/_/_/
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jcasale at activenetwerx.com  Sat Jan  2 09:34:21 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 2 Jan 2010 17:34:21 +0000
Subject: [Freeswitch-users] Mixing zap and sip users in the directory
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9069EE7@prato.activenetwerx.local>

While trying to uncover my issues with no sound on a zap channel, somewhere
in the archive I came across the mention of mixing analog and digital extensions
in the directory, is there anything special that needs to be done for this?

Also, to answer an incoming call from an fxo channel, can I simply transfer it:

<include>
	<extension name="Openzap_in">
		<condition field="source" expression="mod_openzap">
			<action application="set" data="domain_name=$${domain}"/>
			<action application="transfer" data="2000 XML default"/>
		</condition>
	</extension>
</include>

Or do I have to answer and do other things?

Thanks!
jlc 



From aep.lists at it46.se  Sat Jan  2 10:08:38 2010
From: aep.lists at it46.se (Alberto Escudero)
Date: Sat, 2 Jan 2010 19:08:38 +0100
Subject: [Freeswitch-users] PHP ESL Problem
In-Reply-To: <285BD733E19541989B31B95871BF5642@fromage>
References: <285BD733E19541989B31B95871BF5642@fromage>
Message-ID: <6204a6f4ea0160933ba6493a292c1b9b.squirrel@correo.nodo50.org>

I do not know if really helps you but we are facing the same problem in
one of our implementations using the ESL.so for PHP.

We have only see this problem when subscribing to the CHANNEL_STATE

getType() should always match EventName... but it does not
./aep

-- 
Stopping junk mailers is good for the environment

> Would someone please take a look at this simple PHP event socket script
> and
> tell me what I am doing wrong - or tell me that this could be a bug
> elsewhere?  Any help would be appreciated.
>
> When I run the script without the call to execute(), everything seems
> fine.
> When I include the call to execute(), the calls to getType() return CUSTOM
> for a while, then later start to return the correct name.
>
> #!/usr/bin/php
> <?php
> require_once 'ESL.php';
> $endPoint = 'sofia/internal/695%192.168.100.132';
>
> $eventSocket = New ESLconnection('192.168.100.132', '8021', 'ClueCon');
> $event = $eventSocket->events('plain', 'ALL');
>
> // call endpoint, get uuid
> $event = $eventSocket->api('originate', $endPoint . ' &park');
> $serializedEvent = explode("\n", $event->serialize());
> foreach ($serializedEvent as $eventLine) {
>     list($dummy, $uuid) = explode('+OK ', $eventLine);
>     if ($uuid) { break; }
> }
>
> // play announcement to endpoint
> $event = $eventSocket->execute('playback', '/opt/ann/user-busy.wav',
> $uuid);
>
> // monitor events
> while (TRUE) {
>     echo "getType: " . $event->getType() . "\n";
>     $serializedEvent = explode("\n", $event->serialize());
>     foreach ($serializedEvent as $eventLine) {
>         list($header, $value) = explode(': ', $eventLine);
>         if ($header == "Event-Name")   { printf($eventLine . "\n"); }
>         if ($header == "Content-Type") { printf($eventLine . "\n"); }
>     }
>
> 	printf("\n");
>     $event = $eventSocket->recvEvent();
> }?>
>
>
> Run without the call to execute():
> ==================================
> getType: CUSTOM
> Content-Type: api/response
>
> getType: CHANNEL_CREATE
> Event-Name: CHANNEL_CREATE
>
> getType: CHANNEL_OUTGOING
> Event-Name: CHANNEL_OUTGOING
>
> getType: CHANNEL_ORIGINATE
> Event-Name: CHANNEL_ORIGINATE
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
> getType: PRESENCE_IN
> Event-Name: PRESENCE_IN
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
> getType: PRESENCE_IN
> Event-Name: PRESENCE_IN
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
> getType: CALL_UPDATE
> Event-Name: CALL_UPDATE
>
> getType: CHANNEL_PROGRESS
> Event-Name: CHANNEL_PROGRESS
>
> getType: HEARTBEAT
> Event-Name: HEARTBEAT
>
> getType: HEARTBEAT
> Event-Name: RE_SCHEDULE
>
> getType: CALL_UPDATE
> Event-Name: CALL_UPDATE
>
> getType: CODEC
> Event-Name: CODEC
>
> getType: CODEC
> Event-Name: CODEC
>
> getType: CHANNEL_ANSWER
> Event-Name: CHANNEL_ANSWER
>
> getType: PRESENCE_IN
> Event-Name: PRESENCE_IN
>
> getType: API
> Event-Name: API
>
> getType: PRESENCE_IN
> Event-Name: PRESENCE_IN
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
> getType: CHANNEL_EXECUTE
> Event-Name: CHANNEL_EXECUTE
>
> getType: CHANNEL_PARK
> Event-Name: CHANNEL_PARK
>
> getType: CHANNEL_HANGUP
> Event-Name: CHANNEL_HANGUP
>
> getType: CHANNEL_UNPARK
> Event-Name: CHANNEL_UNPARK
>
> getType: CHANNEL_EXECUTE_COMPLETE
> Event-Name: CHANNEL_EXECUTE_COMPLETE
>
> getType: PRESENCE_IN
> Event-Name: PRESENCE_IN
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
> getType: PRESENCE_IN
> Event-Name: PRESENCE_IN
>
> getType: CHANNEL_HANGUP_COMPLETE
> Event-Name: CHANNEL_HANGUP_COMPLETE
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
> getType: CHANNEL_DESTROY
> Event-Name: CHANNEL_DESTROY
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
>
> Run with the call to execute():
> ===============================
> getType: CUSTOM
> Content-Type: command/reply
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_CREATE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_OUTGOING
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_ORIGINATE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_STATE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: PRESENCE_IN
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_STATE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: PRESENCE_IN
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_STATE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CALL_UPDATE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_PROGRESS
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CALL_UPDATE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CODEC
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CODEC
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_ANSWER
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: PRESENCE_IN
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: API
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: PRESENCE_IN
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_STATE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_EXECUTE
>
> getType: CUSTOM
> Content-Type: text/event-plain
> Event-Name: CHANNEL_PARK
>
> getType: CHANNEL_EXECUTE
> Event-Name: CHANNEL_EXECUTE
>
> getType: CHANNEL_HANGUP
> Event-Name: CHANNEL_HANGUP
>
> getType: CHANNEL_EXECUTE_COMPLETE
> Event-Name: CHANNEL_EXECUTE_COMPLETE
>
> getType: COMMAND
> Event-Name: COMMAND
>
> getType: CHANNEL_UNPARK
> Event-Name: CHANNEL_UNPARK
>
> getType: CHANNEL_EXECUTE_COMPLETE
> Event-Name: CHANNEL_EXECUTE_COMPLETE
>
> getType: PRESENCE_IN
> Event-Name: PRESENCE_IN
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
> getType: PRESENCE_IN
> Event-Name: PRESENCE_IN
>
> getType: CHANNEL_HANGUP_COMPLETE
> Event-Name: CHANNEL_HANGUP_COMPLETE
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
> getType: CHANNEL_DESTROY
> Event-Name: CHANNEL_DESTROY
>
> getType: CHANNEL_STATE
> Event-Name: CHANNEL_STATE
>
>
> Thanks,
> Ron
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>





From jcasale at activenetwerx.com  Sat Jan  2 14:17:50 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 2 Jan 2010 22:17:50 +0000
Subject: [Freeswitch-users] system functions silently ignored
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906A04C@prato.activenetwerx.local>

What could cause this? I have a fax script which just stopped working.
I see the logs display the complete diaplan which involves the system
call, I can manually execute the very command from the cli, but during
the call, it just skips the system application and hangups without printing
anything in the logs?

I even tried something simple like:
<action application="system" data="touch /tmp/rxfax"/>

Thanks,
jlc


From Russell.Mosemann at cune.org  Sat Jan  2 14:32:40 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sat, 2 Jan 2010 16:32:40 -0600
Subject: [Freeswitch-users] system functions silently ignored
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E906A04C@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E906A04C@prato.activenetwerx.local>
Message-ID: <CFF81042AFE645CCA2D6AE3276D63283@cune.pri>

Joseph L. Casale asked:

> What could cause this

Have you updated to the latest release in SVN?

--
Russell Mosemann




From mcampbellsmith at gmail.com  Sat Jan  2 15:19:39 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 3 Jan 2010 10:19:39 +1100
Subject: [Freeswitch-users] No audio after Remote SDP:
In-Reply-To: <180A53B1-EFB6-4B17-AA7C-19B007ADE04C@freeswitch.org>
References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com>
	<7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org>
	<33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com>
	<772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org>
	<33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com>
	<191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com>
	<33c87fa30912301213g3d24b773k86b95e7ce08f6761@mail.gmail.com>
	<7D75E6D6-8AB9-429E-A5A9-1639C0D5AD09@freeswitch.org>
	<33c87fa31001012330g69fc8e88m61e648d893c4d8f7@mail.gmail.com>
	<180A53B1-EFB6-4B17-AA7C-19B007ADE04C@freeswitch.org>
Message-ID: <33c87fa31001021519j41ead649rb4a85e6be6236254@mail.gmail.com>

I have a Linksys SPA3102, NAT'd on the internet (remotely) and
connected to my FS on the otherside of the world, which is also
natted.  A PAP2T is connected on the same subnet as the FS.  The 3102
registers successfully and a call can be set up from the PAP2 to the
3102.

However, after FS receives the Remote SDP the audio stops (ring tone
stops in my case)

The FS router is upnp enabled.  The SPA3102 router is NOT upnp enabled
(SPA3102 does not support upnp anyway I think).



On Sun, Jan 3, 2010 at 3:03 AM, Brian West <brian at freeswitch.org> wrote:
> Are you behind a nat-pmp/upnp router?
> /b
> On Jan 2, 2010, at 1:30 AM, Mark Campbell-Smith wrote:
>
> Hi!
>
> Both are auto-nat:
> ???<param name="ext-rtp-ip" value="auto-nat"/>
> ???<param name="ext-sip-ip" value="auto-nat"/>
>
> FreeSWITCH Version 1.0.trunk (15490)
>
> However, isn't it the IP address that is reported by the remote
> SPA3102 that is incorrect? ?Or?
>
> On Thu, Dec 31, 2009 at 7:21 AM, Brian West <brian at freeswitch.org> wrote:
>
> show me the ext-rtp-ip and ext-sip-ip settings you're using along with SVN
> rev please.
>
> /b
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From jcasale at activenetwerx.com  Sat Jan  2 15:48:09 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 2 Jan 2010 23:48:09 +0000
Subject: [Freeswitch-users] system functions silently ignored
In-Reply-To: <CFF81042AFE645CCA2D6AE3276D63283@cune.pri>
References: <CA5A491E9DEFBE4CB777DE97E21575E906A04C@prato.activenetwerx.local>,
	<CFF81042AFE645CCA2D6AE3276D63283@cune.pri>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906A13E@prato.activenetwerx.local>

>> What could cause this
>
>Have you updated to the latest release in SVN?

Yeah, sorry that was on latest, I just recompiled pre10 and same behaviour now?



From mcampbellsmith at gmail.com  Sat Jan  2 16:09:23 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 3 Jan 2010 11:09:23 +1100
Subject: [Freeswitch-users] Dropped calls
Message-ID: <33c87fa31001021609t60ed802eyd6d6d4db7d314e1d@mail.gmail.com>

Hi!

I just got a couple of dropped calls.  Unfortunately I lost the sip
traces, but I do have the debug logs... in both cases FS shows a
Duplicate SDP received.  I'm not sure if this is a cause - do these
show anything to anyone as to why the calls dropped?

FS version is FreeSWITCH Version 1.0.trunk (15490)

Drop 1:
2010-01-02 17:18:50.806686 [DEBUG] switch_core_io.c:234
sofia/internal/2001 at myddns.dydns.org:442 receive message
[TRANSCODING_NECESSARY]
2010-01-02 17:18:50.826355 [DEBUG] switch_rtp.c:1972 Correct ip/port confirmed.
2010-01-02 17:18:51.645569 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state [ready][200]
2010-01-02 17:18:51.666115 [DEBUG] switch_core_session.c:704 Send
signal sofia/internal/sip:1000 at 192.168.1.121:5060 [BREAK]
2010-01-02 17:18:51.666115 [DEBUG] switch_core_session.c:704 Send
signal sofia/internal/2001 at myddns.dydns.org:442 [BREAK]
2010-01-02 17:18:51.685576 [DEBUG] switch_ivr_bridge.c:122
sofia/internal/2001 at myddns.dydns.org:442 receive message [DISPLAY]
2010-01-02 17:18:51.685576 [DEBUG] switch_ivr_bridge.c:122
sofia/internal/sip:1000 at 192.168.1.121:5060 receive message [DISPLAY]
2010-01-02 17:19:40.169692 [DEBUG] switch_rtp.c:2344 RTP RECV DTMF 1:404
2010-01-02 17:19:40.189926 [DEBUG] switch_rtp.c:1641 Send start packet
for [1] ts=24065640 dur=160/160/404 seq=15337
2010-01-02 17:19:40.205843 [DEBUG] switch_rtp.c:1577 Send middle
packet for [1] ts=24065640 dur=320/320/404 seq=15338
2010-01-02 17:19:40.225669 [DEBUG] switch_rtp.c:1577 Send end packet
for [1] ts=24065640 dur=480/480/404 seq=15339
2010-01-02 17:19:40.225669 [DEBUG] switch_rtp.c:1577 Send end packet
for [1] ts=24065640 dur=480/480/404 seq=15340
2010-01-02 17:19:40.225669 [DEBUG] switch_rtp.c:1577 Send end packet
for [1] ts=24065640 dur=480/480/404 seq=15341
2010-01-02 17:19:48.942440 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state [calling][0]
2010-01-02 17:19:52.093594 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state [ready][200]
2010-01-02 17:19:52.093594 [DEBUG] sofia.c:3654 Duplicate SDP
v=0
o=- 238296 238296 IN IP4 192.168.1.3
s=-
c=IN IP4 192.168.1.3
t=0 0
m=audio 19428 RTP/AVP 2 0 8 4 96 97 98 100 101
a=rtpmap:2 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2010-01-02 17:19:59.713933 [DEBUG] switch_rtp.c:2344 RTP RECV DTMF 1:404
2010-01-02 17:19:59.725822 [DEBUG] switch_rtp.c:1641 Send start packet
for [1] ts=24225880 dur=160/160/404 seq=16315
2010-01-02 17:19:59.745728 [DEBUG] switch_rtp.c:1577 Send middle
packet for [1] ts=24225880 dur=320/320/404 seq=16316
2010-01-02 17:19:59.765676 [DEBUG] switch_rtp.c:1577 Send end packet
for [1] ts=24225880 dur=480/480/404 seq=16317
2010-01-02 17:19:59.765676 [DEBUG] switch_rtp.c:1577 Send end packet
for [1] ts=24225880 dur=480/480/404 seq=16318
2010-01-02 17:19:59.765676 [DEBUG] switch_rtp.c:1577 Send end packet
for [1] ts=24225880 dur=480/480/404 seq=16319
2010-01-02 17:20:49.969621 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state [calling][0]
2010-01-02 17:20:54.036198 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state
[terminating][503]
2010-01-02 17:20:54.036198 [NOTICE] sofia.c:4247 Hangup
sofia/internal/2001 at myddns.dydns.org:442 [CS_EXECUTE]
[NORMAL_TEMPORARY_FAILURE]
2010-01-02 17:20:54.040419 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/2001 at myddns.dydns.org:442 [KILL]
2010-01-02 17:20:54.040419 [DEBUG] switch_core_session.c:982 Send
signal sofia/internal/2001 at myddns.dydns.org:442 [BREAK]

Drop 2:
2010-01-02 17:10:42.177817 [DEBUG] switch_core_session.c:704 Send
signal sofia/internal/sip:1000 at 192.168.1.121:5060 [BREAK]
2010-01-02 17:10:42.177817 [DEBUG] sofia.c:411
sofia/internal/sip:1000 at 192.168.1.121:5060 receive message [DISPLAY]
2010-01-02 17:10:42.177817 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state
[completed][200]
2010-01-02 17:10:42.185735 [DEBUG] switch_core_session.c:643 Send
signal sofia/internal/2001 at myddns.dydns.org:442 [BREAK]
2010-01-02 17:10:42.189574 [NOTICE] switch_ivr_originate.c:2836
Channel [sofia/internal/2001 at myddns.dydns.org:442] has been answered
2010-01-02 17:10:42.193802 [DEBUG] switch_channel.c:182
sofia/internal/2001 at myddns.dydns.org:442 receive message [AUDIO_SYNC]
2010-01-02 17:10:42.193802 [DEBUG] switch_ivr_originate.c:2881
Originate Resulted in Success:
[sofia/internal/sip:1000 at 192.168.1.121:5060]
2010-01-02 17:10:42.197882 [DEBUG] switch_channel.c:182
sofia/internal/sip:1000 at 192.168.1.121:5060 receive message
[AUDIO_SYNC]
2010-01-02 17:10:42.201876 [DEBUG] switch_channel.c:182
sofia/internal/2001 at myddns.dydns.org:442 receive message [AUDIO_SYNC]
2010-01-02 17:10:42.217738 [DEBUG] switch_ivr_originate.c:2881
Originate Resulted in Success:
[sofia/internal/sip:1000 at 192.168.1.121:5060]
2010-01-02 17:10:42.221921 [DEBUG] switch_channel.c:182
sofia/internal/sip:1000 at 192.168.1.121:5060 receive message
[AUDIO_SYNC]
2010-01-02 17:10:42.221921 [DEBUG] switch_channel.c:182
sofia/internal/2001 at myddns.dydns.org:442 receive message [AUDIO_SYNC]
2010-01-02 17:10:42.237732 [DEBUG] switch_ivr_bridge.c:1004
sofia/internal/sip:1000 at 192.168.1.121:5060 receive message [BRIDGE]
2010-01-02 17:10:42.237732 [DEBUG] switch_core_session.c:643 Send
signal sofia/internal/sip:1000 at 192.168.1.121:5060 [BREAK]
2010-01-02 17:10:42.241599 [DEBUG] switch_ivr_bridge.c:1011
sofia/internal/2001 at myddns.dydns.org:442 receive message [BRIDGE]
2010-01-02 17:10:42.241599 [DEBUG] switch_core_session.c:643 Send
signal sofia/internal/2001 at myddns.dydns.org:442 [BREAK]
2010-01-02 17:10:42.241599 [DEBUG] switch_ivr_bridge.c:1055
(sofia/internal/sip:1000 at 192.168.1.121:5060) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2010-01-02 17:10:42.246345 [DEBUG] switch_core_session.c:982 Send
signal sofia/internal/sip:1000 at 192.168.1.121:5060 [BREAK]
2010-01-02 17:10:42.246345 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1000 at 192.168.1.121:5060) Running State Change
CS_EXCHANGE_MEDIA
2010-01-02 17:10:42.249596 [DEBUG] switch_core_state_machine.c:351
(sofia/internal/sip:1000 at 192.168.1.121:5060) State EXCHANGE_MEDIA
2010-01-02 17:10:42.249596 [DEBUG] mod_sofia.c:464 SOFIA LOOPBACK
2010-01-02 17:10:42.265702 [DEBUG] switch_core_io.c:234
sofia/internal/2001 at myddns.dydns.org:442 receive message
[TRANSCODING_NECESSARY]
2010-01-02 17:10:42.285925 [DEBUG] switch_rtp.c:1972 Correct ip/port confirmed.
2010-01-02 17:10:44.402422 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state [ready][200]
2010-01-02 17:10:44.413899 [DEBUG] switch_core_session.c:704 Send
signal sofia/internal/sip:1000 at 192.168.1.121:5060 [BREAK]
2010-01-02 17:10:44.419743 [DEBUG] switch_core_session.c:704 Send
signal sofia/internal/2001 at myddns.dydns.org:442 [BREAK]
2010-01-02 17:10:44.425721 [DEBUG] switch_ivr_bridge.c:122
sofia/internal/sip:1000 at 192.168.1.121:5060 receive message [DISPLAY]
2010-01-02 17:10:44.433714 [DEBUG] switch_ivr_bridge.c:122
sofia/internal/2001 at myddns.dydns.org:442 receive message [DISPLAY]
2010-01-02 17:11:44.822897 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state [calling][0]
2010-01-02 17:11:45.353266 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state [ready][200]
2010-01-02 17:11:45.353266 [DEBUG] sofia.c:3654 Duplicate SDP
v=0
o=- 188941 188941 IN IP4 192.168.1.3
s=-
c=IN IP4 192.168.1.3
t=0 0
m=audio 19424 RTP/AVP 2 0 8 4 96 97 98 100 101
a=rtpmap:2 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2010-01-02 17:12:44.834908 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state [calling][0]
2010-01-02 17:12:45.125887 [DEBUG] sofia.c:3646 Channel
sofia/internal/2001 at myddns.dydns.org:442 entering state
[terminating][503]
2010-01-02 17:12:45.125887 [NOTICE] sofia.c:4247 Hangup
sofia/internal/2001 at myddns.dydns.org:442 [CS_EXECUTE]
[NORMAL_TEMPORARY_FAILURE]
2010-01-02 17:12:45.125887 [DEBUG] switch_channel.c:1912 Send signal
sofia/internal/2001 at myddns.dydns.org:442 [KILL]
2010-01-02 17:12:45.125887 [DEBUG] switch_core_session.c:982 Send
signal sofia/internal/2001 at myddns.dydns.org:442 [BREAK]



From jcasale at activenetwerx.com  Sat Jan  2 21:21:42 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sun, 3 Jan 2010 05:21:42 +0000
Subject: [Freeswitch-users] Zap dialplan characteristics
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906A406@prato.activenetwerx.local>

It seems it was permissions problems which were causing the
audio issues for me, I was attempting to run a manually
compiled instance of freeswitch with the stock init scripts.

Apparently, setting the udev rules to freeswitch/daemon as
fs runs won't work. I got it running finally tonight as a
user and group 'freeswitch' but not till after I tried the
zaptel release from the wiki's reco which didn't work until
the perms issue was discovered. I am sure I can go back to
using Digiums Dahdi package for Centos.

So last question. W/ Asterisk, I had to answer the dahdi line
so the far end didn't activate the call forward on no answer.
Would it be safe in assuming this needs to be replicated here
as well. If so, do I understand this right if I do this:

<include>
  <extension name="openzap_in">
    <condition field="source" expression="mod_openzap">
      <action application="set" data="domain_name=$${domain}"/>
      <action application="ring_ready"/>
      <action application="transfer" data="2000 XML default"/>
    </condition>
  </extension>
</include>

So that I can ring the call group for its preset time which surely
exceeds that of the call-forward from the telco?

Thanks!
jlc





From vmknott at gmail.com  Sun Jan  3 06:06:16 2010
From: vmknott at gmail.com (VM Knott)
Date: Sun, 3 Jan 2010 09:06:16 -0500
Subject: [Freeswitch-users] Voicemail Question (using multiple servers)
Message-ID: <ed9046461001030606w679e2d9cx60b54fa8607b6a9f@mail.gmail.com>

Bill,

The original plan was to share the db across multiple switches.

However, if I have to implement functionality to track records that
identify a specific IP address for every switch in the cluster, I was
thinking that a temporary work-around for our application would be a
standard greeting for all mailboxes.  We do not want the default TTS
sounding greeting that identifies the mailbox number, so I was going
to assign a standard greeting for all new mailboxes, and then if the
owner submits a custom greeting to their mailbox, it will override the
default.

We can implement logic to synchronize the db, not a huge deal.  But
this just feels like I?m over-complicating the solution.

- VMK




---------- Forwarded message ----------
From: "Bill W." <freeswitch at aastral.net>
To: freeswitch-users at lists.freeswitch.org
Date: Fri, 01 Jan 2010 01:31:14 -0500
Subject: Re: [Freeswitch-users] Voicemail Question (using multiple servers)
So is the problem that you're having to replicate the voicemail database
 across switches in the cluster or is the problem the content of the
entries in voicemail database?

Because in your original post you're speaking of trying to share the
voicemail db over NFS.

Thanks,
Bill

VM Knott wrote:
> Thank you Bill for the input, but changing how I access the data
> source does not solve my problem.
> I was hoping to avoid the management of IP Addresses for every
> voicemail box on the system.
>
> Is there a way for me to set a default greeting to all voicemail boxes
> globally, without having to go to a repository (regardless of means of
> access) for each mailbox?
>
>
>



From freeswitch at aastral.net  Sun Jan  3 07:24:49 2010
From: freeswitch at aastral.net (Bill W.)
Date: Sun, 03 Jan 2010 10:24:49 -0500
Subject: [Freeswitch-users] Voicemail Question (using multiple servers)
In-Reply-To: <ed9046461001030606w679e2d9cx60b54fa8607b6a9f@mail.gmail.com>
References: <ed9046461001030606w679e2d9cx60b54fa8607b6a9f@mail.gmail.com>
Message-ID: <4B40B6C1.1020009@aastral.net>

Hey VM,

I'm interested in your issue because I will need to implement this
feature in probably 6 months or so.  But I'm not currently familiar with
the VM database records or how they're used.

The weird thing is I'm using a shared sofia database for registrations
across a cluster and it works fine.  I know the IP address of the switch
the UA registered to gets stored in the registration database, but any
switch can use that registration record.

Looking at the sql in the voicemail module, it shows a column for
'domain'.  Is this where the IP address is being stored?   If so, maybe
you can find a way to change that to a domain name as Tony suggested.
Is the domain name/ip address being used in the filesystem path?

In any case I'm willing to help you solve this because I need to solve
this issue as well.   If you don't want to clog up the list with all the
troubleshooting,  we can take this off-list and then post our results to
the thread when it's all done.

Thanks,
Bill




VM Knott wrote:
> Bill,
> 
> The original plan was to share the db across multiple switches.
> 
> However, if I have to implement functionality to track records that
> identify a specific IP address for every switch in the cluster, I was
> thinking that a temporary work-around for our application would be a
> standard greeting for all mailboxes.  We do not want the default TTS
> sounding greeting that identifies the mailbox number, so I was going
> to assign a standard greeting for all new mailboxes, and then if the
> owner submits a custom greeting to their mailbox, it will override the
> default.
> 
> We can implement logic to synchronize the db, not a huge deal.  But
> this just feels like I?m over-complicating the solution.
> 
> - VMK
> 
> 



From yehavi.bourvine at gmail.com  Sun Jan  3 07:57:20 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 3 Jan 2010 17:57:20 +0200
Subject: [Freeswitch-users] Polycom & TLS - help please...
Message-ID: <d361ca7b1001030757m2f55a400n3e2b3b74185c4202@mail.gmail.com>

Hello,

  I am writing again because I am quite desparate... I fail to enable TLS on
Polycom while on SNOM I can make it work with the same configuration. From
TCPDUMP I see the following difference between the two handshakes:


   - SNOM: The protocol suite used is TLS_RSA_WITH_RC4_128_SHA
   - Polyco: The protocol suite used is TLS_RSA_WITH_AES_256_CBC_SHA and
   sessionTicketTLS also appears there. After the key exchange the phone
   disconnects the connection.

Any idea how to debug it?

                               Thanks! __Yehavi:
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From kdjakovic at hotmail.com  Sun Jan  3 08:50:21 2010
From: kdjakovic at hotmail.com (katarina djakovic)
Date: Sun, 3 Jan 2010 17:50:21 +0100
Subject: [Freeswitch-users] How to suspend certain users from the system
	using flush_inbound_reg
Message-ID: <SNT137-w54C41FF82F0E62B26415E9B3750@phx.gbl>


Hi,

 

we are trying to figure out how to suspend certain subscribers from our system and we have some problems with removing thier 

registrations. The UAs are ATAs.

 

This is what we do:

 

1) We remove the subscriber extension from the conf\directory .xml files
2) We do reloadxml
3) We flush user's registration with flush_inbound_reg 

 

but, the users are still able to make calls as if they were still registered. To make it clearer, their registrations are removed from 

the registration list (checked with sofia status), but they system still accepts the calls from them. From this, it seems that if ATA is never rebooted - we are not able to ban these users from the system.

 

Only after the ATA is rebooted user is not able to make calls any more, as the ATA can not register any more - since they users are 

removed from the directory. But before we reboot ATA everything works as nothing had been done.

 

Does anyone have an idea what are we doing wrong? We expect that after the registration is removed from the FS the UA should not be 

able to make a call but this is not what happnes.

 

Can anybody help please?

Thanks,
Katarina
 		 	   		  
_________________________________________________________________
Keep your friends updated?even when you?re not signed in.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010
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From linux4michelle at tamay-dogan.net  Sun Jan  3 10:46:36 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Sun, 3 Jan 2010 19:46:36 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
Message-ID: <20100103184636.GW5547@tamay-dogan.net>

Hello *,

I am owner and developer of the  enterprise  "electronica at tdnet UG"  and
currently I design a  GSM  router  with  VoIP   gateway  using  a  Texas
Instruments Sitara or OMAP  with  an  attached  CologneChip  4port  ISDN
Controller and a Silicon Laboratories Quad ProSLIC.  Also it will have a
GSM/HSPA modem.

My questionis, does someone use FreeSwitch with a GSM/HSPA Modem and can
use internet connectivity, telephonie and SMS?

If yes, which GSM Modem do you use?

I have this question since in theorie my cell-phone "Nokia 6120 classic"
can do this, but I was not able to get the streams...

Otherwise I would develop a GSM/HSPA Modem which CAN DO THIS my own.

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From a.alalousi at gmail.com  Sun Jan  3 12:25:50 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Sun, 3 Jan 2010 20:25:50 +0000
Subject: [Freeswitch-users] How to suspend certain users from the system
	using flush_inbound_reg
In-Reply-To: <SNT137-w54C41FF82F0E62B26415E9B3750@phx.gbl>
References: <SNT137-w54C41FF82F0E62B26415E9B3750@phx.gbl>
Message-ID: <cd93b3701001031225p44cd0d0fk79ed7ae05639417c@mail.gmail.com>

Hi Katarina,

Sounds like you have enabled ipauth by having cidr attributes within the
extension file. E.g:

<user id="test" cidr="192.168.0.0/16">

If this is the case, then username/password tuplles will fail (because you
have disabled them) but ipauth will work, and FS will allow unregistered
calls.

Also, check you conf/autoload_configs/acl.conf.xml to see if your default
domains acl state is one of allow rather than deny. E.g:

<list name="domains" default="allow">
    <node type="allow" domain="$${domain}"/>
</list>

This type of list is bad news anyway. I've seen it allow unregistered calls
from anyone simply by them knowing your domain.

What we do in our set-up is remove the default ACLs altogether, and apply
our own custom ones + firewalling at the border.

Hope this helps you a little.




2010/1/3 katarina djakovic <kdjakovic at hotmail.com>

>  Hi,
>
> we are trying to figure out how to suspend certain subscribers from our
> system and we have some problems with removing thier
> registrations. The UAs are ATAs.
>
> This is what we do:
>
> 1) We remove the subscriber extension from the conf\directory .xml files
> 2) We do reloadxml
> 3) We flush user's registration with flush_inbound_reg
>
> but, the users are still able to make calls as if they were still
> registered. To make it clearer, their registrations are removed from
> the registration list (checked with sofia status), but they system still
> accepts the calls from them. From this, it seems that if ATA is never
> rebooted - we are not able to ban these users from the system.
>
> Only after the ATA is rebooted user is not able to make calls any more, as
> the ATA can not register any more - since they users are
> removed from the directory. But before we reboot ATA everything works as
> nothing had been done.
>
> Does anyone have an idea what are we doing wrong? We expect that after the
> registration is removed from the FS the UA should not be
> able to make a call but this is not what happnes.
>
> Can anybody help please?
> Thanks,
> Katarina
>
> ------------------------------
> Keep your friends updated? even when you?re not signed in.<http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From freeswitch at aastral.net  Sun Jan  3 15:27:04 2010
From: freeswitch at aastral.net (Bill W.)
Date: Sun, 03 Jan 2010 18:27:04 -0500
Subject: [Freeswitch-users] How to suspend certain users from the system
 using flush_inbound_reg
In-Reply-To: <SNT137-w54C41FF82F0E62B26415E9B3750@phx.gbl>
References: <SNT137-w54C41FF82F0E62B26415E9B3750@phx.gbl>
Message-ID: <4B4127C8.30404@aastral.net>

Hello Katarina,

You could do this several ways, but it has to be more than just removing
their extension.

* You could use mod_nibblebill to enforce  a zero balance so they can't
make calls.

* You could use dialplan logic and xml_curl where a variable set in the
database gets populated in the user's directory entry and enforced in
the dialplan condition.

* You could use ACL logic and xml_curl where a variable set in the
database gets populated in the user's directory entry and enforced via
ACLs.   (auth-calls in sofia combined with auth-acl in the directory)

Hope this helps,
-Bill



katarina djakovic wrote:
> Hi,
>  
> we are trying to figure out how to suspend certain subscribers from our
> system and we have some problems with removing thier
> registrations. The UAs are ATAs.
>  
> This is what we do:
>  
> 1) We remove the subscriber extension from the conf\directory .xml files
> 2) We do reloadxml
> 3) We flush user's registration with flush_inbound_reg
>  
> but, the users are still able to make calls as if they were still
> registered. To make it clearer, their registrations are removed from
> the registration list (checked with sofia status), but they system still
> accepts the calls from them. From this, it seems that if ATA is never
> rebooted - we are not able to ban these users from the system.
>  
> Only after the ATA is rebooted user is not able to make calls any more,
> as the ATA can not register any more - since they users are
> removed from the directory. But before we reboot ATA everything works as
> nothing had been done.
>  
> Does anyone have an idea what are we doing wrong? We expect that after
> the registration is removed from the FS the UA should not be
> able to make a call but this is not what happnes.
>  
> Can anybody help please?
> Thanks,
> Katarina
> 
> ------------------------------------------------------------------------
> Keep your friends updated? even when you?re not signed in.
> <http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010>
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From saeedahmad1981 at gmail.com  Sun Jan  3 16:36:34 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Mon, 4 Jan 2010 01:36:34 +0100
Subject: [Freeswitch-users] freeswitch and H323
In-Reply-To: <65d96fc81001010452k37068e87sc0a229cdfe481c40@mail.gmail.com>
References: <7aa8bd9d0912300514r68c90b12u7c631a649981cfa3@mail.gmail.com> 
	<87f2f3b90912310920q1821fe0eu5fd74e553e4cb12@mail.gmail.com> 
	<65d96fc81001010452k37068e87sc0a229cdfe481c40@mail.gmail.com>
Message-ID: <c96b95651001031636q1ecfec18x2bba1c22ff6ca9bb@mail.gmail.com>

HI,

It would be really nice if you can create a wiki page.

Thanks

On Fri, Jan 1, 2010 at 1:52 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> well, mod_h323 works for me... there are still some missing things and of
> course bugs ... e.g. incorrect releaseCause mapping, no automatic codec
> ptime sync... but it is usable ....
>
>
> if you'd like to go mod_h323 way i can help you... it builds as a charm for
> me...
>
>
> T.
>
>
>
>
>
> On Thu, Dec 31, 2009 at 6:20 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Are you trying to use mod_h323 or mod_opal? They are both works in
>> progress, but the latter is farther along than the former. Use the latest
>> FreeSWITCH trunk (or latest.freeswitch.org) and run the buildopal.sh
>> script in the build directory. If you have any build issues then paste the
>> log on pastebin.freeswitch.org and reply to this thread with the PB URL
>> so that we can take a look.
>>  -MC
>>
>>
>> On Wed, Dec 30, 2009 at 5:14 AM, Pete Kay <petedao at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> has anyone been able to get H323 to work?
>>>
>>> I have problem trying to get it compiled with either 1.0.4 or 1.0.5.
>>>
>>> Thanks,
>>> pete
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sun Jan  3 16:43:24 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 3 Jan 2010 19:43:24 -0500
Subject: [Freeswitch-users] Zap dialplan characteristics
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E906A406@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E906A406@prato.activenetwerx.local>
Message-ID: <4145E5D0-AF8C-4113-B9E6-D87B23E7CA97@jerris.com>

If you want the call to "ring" longer than the telco would allow you  
will need to answer the call first but I find this approach very  
suboptimal.  I would adjust the call forward no answer times on the  
carrier side so you can have consistant cdrs and normal flow of  
progress instead of hacks.

Mike

On Jan 3, 2010, at 12:21 AM, "Joseph L. Casale" <jcasale at activenetwerx.com 
 > wrote:

> It seems it was permissions problems which were causing the
> audio issues for me, I was attempting to run a manually
> compiled instance of freeswitch with the stock init scripts.
>
> Apparently, setting the udev rules to freeswitch/daemon as
> fs runs won't work. I got it running finally tonight as a
> user and group 'freeswitch' but not till after I tried the
> zaptel release from the wiki's reco which didn't work until
> the perms issue was discovered. I am sure I can go back to
> using Digiums Dahdi package for Centos.
>
> So last question. W/ Asterisk, I had to answer the dahdi line
> so the far end didn't activate the call forward on no answer.
> Would it be safe in assuming this needs to be replicated here
> as well. If so, do I understand this right if I do this:
>
> <include>
>  <extension name="openzap_in">
>    <condition field="source" expression="mod_openzap">
>      <action application="set" data="domain_name=$${domain}"/>
>      <action application="ring_ready"/>
>      <action application="transfer" data="2000 XML default"/>
>    </condition>
>  </extension>
> </include>
>
> So that I can ring the call group for its preset time which surely
> exceeds that of the call-forward from the telco?
>
> Thanks!
> jlc
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From tom at tomcarlson.com  Sun Jan  3 12:31:35 2010
From: tom at tomcarlson.com (Tom Carlson)
Date: Sun, 3 Jan 2010 12:31:35 -0800
Subject: [Freeswitch-users] jitter buffer effecting input callback in lua
Message-ID: <21e9d36c1001031231yed4a914u5c8e7e8568208c45@mail.gmail.com>

I have a very simple lua script (shown below my message)

This script plays a greeting, lets the caller record a message, detecting
when caller is done by sensing a keypress.  It then plays the message back
to the caller.

This works perfectly, except the audio quality of the recorded message is
less than I had hoped.  To try to fix this, I have added a line to activate
the jitter buffer.  This single line keeps the script from detecting the
dtmf tones that end the recording, so the script just stays locked in record
mode forever, until you hang up.

The log shows no problems.

How can I activate the jitter buffer, and still detect dtmf events?

Thanks for your help.

Tom

-- ---------------------------------------------------------------------
function key_press(session, input_type, data, args)
  if input_type == "dtmf" then
    freeswitch.consoleLog("info", "Key pressed: " .. data["digit"])
    return "break"
  end
end

session:setVariable("jitterbuffer_msec", "200");
if session:ready() then
  session:answer();
  while (session:ready() == true) do
    session:setAutoHangup(false);
    session:sleep(1000);


session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-record_message.wav");


    session:setInputCallback("key_press", "");
    session:recordFile("/tmp/blah.wav", 5000, 10, 10); -- pressing key ends
the recording
    session:streamFile("/tmp/blah.wav");
  end
end
--
--------------------------------------------------------------------------------------------------
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From mike at jerris.com  Sun Jan  3 17:24:15 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 3 Jan 2010 20:24:15 -0500
Subject: [Freeswitch-users] How to suspend certain users from the system
	using flush_inbound_reg
In-Reply-To: <SNT137-w54C41FF82F0E62B26415E9B3750@phx.gbl>
References: <SNT137-w54C41FF82F0E62B26415E9B3750@phx.gbl>
Message-ID: <59CAAB6B-5259-4B78-9C9C-676ACF04D6B1@jerris.com>

First a note.  Registration and authentication are completely  
different. Removing the registration has to do with the switch knowin  
where to send the calls and nothing to do with auth for receiving  
calls.  There is one caveat to this.  We do support nonce count, and  
it could be using the auth from the previous registration that is  
still valid.  Double check the nc from the registrations and the call  
and see if that rings true.  We may want to add something to  
explicitly expire the nonce when youflush reg but I need some  
confirmation on that first.  Otherwise the other responces seem to  
cover the possibilities.  Crank up the debug and check sip trace for  
more details on what is allowing the call through and report back.

Mike

On Jan 3, 2010, at 11:50 AM, katarina djakovic <kdjakovic at hotmail.com>  
wrote:

> Hi,
>
> we are trying to figure out how to suspend certain subscribers from  
> our system and we have some problems with removing thier
> registrations. The UAs are ATAs.
>
> This is what we do:
>
> 1) We remove the subscriber extension from the conf\directory .xml  
> files
> 2) We do reloadxml
> 3) We flush user's registration with flush_inbound_reg
>
> but, the users are still able to make calls as if they were still  
> registered. To make it clearer, their registrations are removed from
> the registration list (checked with sofia status), but they system  
> still accepts the calls from them. From this, it seems that if ATA  
> is never rebooted - we are not able to ban these users from the  
> system.
>
> Only after the ATA is rebooted user is not able to make calls any  
> more, as the ATA can not register any more - since they users are
> removed from the directory. But before we reboot ATA everything  
> works as nothing had been done.
>
> Does anyone have an idea what are we doing wrong? We expect that  
> after the registration is removed from the FS the UA should not be
> able to make a call but this is not what happnes.
>
> Can anybody help please?
> Thanks,
> Katarina
>
> Keep your friends updated? even when you?re not signed in.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From mike at jerris.com  Sun Jan  3 17:25:17 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 3 Jan 2010 20:25:17 -0500
Subject: [Freeswitch-users] Polycom & TLS - help please...
In-Reply-To: <d361ca7b1001030757m2f55a400n3e2b3b74185c4202@mail.gmail.com>
References: <d361ca7b1001030757m2f55a400n3e2b3b74185c4202@mail.gmail.com>
Message-ID: <BFB88CA0-8319-4D05-B3CD-39E3450DE157@jerris.com>

Perhapse cranking up the Sofia tport log to level 9 may help.

Mike

On Jan 3, 2010, at 10:57 AM, Yehavi Bourvine  
<yehavi.bourvine at gmail.com> wrote:

> Hello,
>
>   I am writing again because I am quite desparate... I fail to  
> enable TLS on Polycom while on SNOM I can make it work with the same  
> configuration. From TCPDUMP I see the following difference between  
> the two handshakes:
>
> SNOM: The protocol suite used is TLS_RSA_WITH_RC4_128_SHA
> Polyco: The protocol suite used is TLS_RSA_WITH_AES_256_CBC_SHA and  
> sessionTicketTLS also appears there. After the key exchange the  
> phone disconnects the connection.
> Any idea how to debug it?
>
>                                Thanks! __Yehavi:
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From mike at jerris.com  Sun Jan  3 17:29:40 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 3 Jan 2010 20:29:40 -0500
Subject: [Freeswitch-users] jitter buffer effecting input callback in lua
In-Reply-To: <21e9d36c1001031231yed4a914u5c8e7e8568208c45@mail.gmail.com>
References: <21e9d36c1001031231yed4a914u5c8e7e8568208c45@mail.gmail.com>
Message-ID: <F46682E5-43C4-4B1F-A318-ECAC7608353E@jerris.com>

There is no reason I can think of that would cause this.  Have you  
tried different phones to eliminate if it is an issue just with one  
type of phone?  Please open a bug on jira.freeswitch.org with a  
minimal script example to reproduce and details of the devices it has  
been reproduced with.

Mike

On Jan 3, 2010, at 3:31 PM, Tom Carlson <tom at tomcarlson.com> wrote:

> I have a very simple lua script (shown below my message)
>
> This script plays a greeting, lets the caller record a message,  
> detecting when caller is done by sensing a keypress.  It then plays  
> the message back to the caller.
>
> This works perfectly, except the audio quality of the recorded  
> message is less than I had hoped.  To try to fix this, I have added  
> a line to activate the jitter buffer.  This single line keeps the  
> script from detecting the dtmf tones that end the recording, so the  
> script just stays locked in record mode forever, until you hang up.
>
> The log shows no problems.
>
> How can I activate the jitter buffer, and still detect dtmf events?
>
> Thanks for your help.
>
> Tom
>
> --  
> ---------------------------------------------------------------------
> function key_press(session, input_type, data, args)
>   if input_type == "dtmf" then
>     freeswitch.consoleLog("info", "Key pressed: " .. data["digit"])
>     return "break"
>   end
> end
>
> session:setVariable("jitterbuffer_msec", "200");
> if session:ready() then
>   session:answer();
>   while (session:ready() == true) do
>     session:setAutoHangup(false);
>     session:sleep(1000);
>
>     session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/ 
> voicemail/8000/vm-record_message.wav");
>
>     session:setInputCallback("key_press", "");
>     session:recordFile("/tmp/blah.wav", 5000, 10, 10); -- pressing  
> key ends the recording
>     session:streamFile("/tmp/blah.wav");
>   end
> end
> --   
> --- 
> --- 
> --- 
> --- 
> --- 
> --- 
> --- 
> --- 
> --- 
> --- 
> --------------------------------------------------------------------
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From vmknott at gmail.com  Sun Jan  3 18:01:17 2010
From: vmknott at gmail.com (VM Knott)
Date: Sun, 3 Jan 2010 21:01:17 -0500
Subject: [Freeswitch-users] Voicemail Question (using multiple servers)
Message-ID: <ed9046461001031801p24542862p3e7b0409957c713a@mail.gmail.com>

Bill,

The "domain name" solution will not work for me.
My FreeSWITCH servers are spread across multiple domain names, so I
would prefer a solution that encompasses a more generic model.

I'm guessing that my situation is more uncommon than I originally thought.

I can setup something that is more specific to my architecture, and
create a service-component that manages the central database to
identify the greeting messages on all servers in the cluster.

A less desirable approach, but in consideration of all of the other
inherent features of FreeSWITCH, a small price to pay.

- VMK



---------- Forwarded message ----------
From: "Bill W." <freeswitch at aastral.net>
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 03 Jan 2010 10:24:49 -0500
Subject: Re: [Freeswitch-users] Voicemail Question (using multiple servers)
Hey VM,

I'm interested in your issue because I will need to implement this
feature in probably 6 months or so.  But I'm not currently familiar with
the VM database records or how they're used.

The weird thing is I'm using a shared sofia database for registrations
across a cluster and it works fine.  I know the IP address of the switch
the UA registered to gets stored in the registration database, but any
switch can use that registration record.

Looking at the sql in the voicemail module, it shows a column for
'domain'.  Is this where the IP address is being stored?   If so, maybe
you can find a way to change that to a domain name as Tony suggested.
Is the domain name/ip address being used in the filesystem path?

In any case I'm willing to help you solve this because I need to solve
this issue as well.   If you don't want to clog up the list with all the
troubleshooting,  we can take this off-list and then post our results to
the thread when it's all done.

Thanks,
Bill



From mike at jerris.com  Sun Jan  3 18:22:50 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 3 Jan 2010 21:22:50 -0500
Subject: [Freeswitch-users] Voicemail Question (using multiple servers)
In-Reply-To: <ed9046461001031801p24542862p3e7b0409957c713a@mail.gmail.com>
References: <ed9046461001031801p24542862p3e7b0409957c713a@mail.gmail.com>
Message-ID: <AC15E5EE-FDC1-434C-ADB0-FC0A576E78CD@jerris.com>

You misunderstand the domain name suggestion.  Look a bit closer at  
how the call to voicemail looks in the dialplan.  You can pass  
whatever you like.  Also, you can force domain for registration and  
the like as well and that may be the right solution for you as well  
but depends on full details of your setup.
Mike

On Jan 3, 2010, at 9:01 PM, VM Knott <vmknott at gmail.com> wrote:

> Bill,
>
> The "domain name" solution will not work for me.
> My FreeSWITCH servers are spread across multiple domain names, so I
> would prefer a solution that encompasses a more generic model.
>
> I'm guessing that my situation is more uncommon than I originally  
> thought.
>
> I can setup something that is more specific to my architecture, and
> create a service-component that manages the central database to
> identify the greeting messages on all servers in the cluster.
>
> A less desirable approach, but in consideration of all of the other
> inherent features of FreeSWITCH, a small price to pay.
>
> - VMK
>
>
>
> ---------- Forwarded message ----------
> From: "Bill W." <freeswitch at aastral.net>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sun, 03 Jan 2010 10:24:49 -0500
> Subject: Re: [Freeswitch-users] Voicemail Question (using multiple  
> servers)
> Hey VM,
>
> I'm interested in your issue because I will need to implement this
> feature in probably 6 months or so.  But I'm not currently familiar  
> with
> the VM database records or how they're used.
>
> The weird thing is I'm using a shared sofia database for registrations
> across a cluster and it works fine.  I know the IP address of the  
> switch
> the UA registered to gets stored in the registration database, but any
> switch can use that registration record.
>
> Looking at the sql in the voicemail module, it shows a column for
> 'domain'.  Is this where the IP address is being stored?   If so,  
> maybe
> you can find a way to change that to a domain name as Tony suggested.
> Is the domain name/ip address being used in the filesystem path?
>
> In any case I'm willing to help you solve this because I need to solve
> this issue as well.   If you don't want to clog up the list with all  
> the
> troubleshooting,  we can take this off-list and then post our  
> results to
> the thread when it's all done.
>
> Thanks,
> Bill
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From brian at freeswitch.org  Sun Jan  3 18:32:22 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 3 Jan 2010 20:32:22 -0600
Subject: [Freeswitch-users] Polycom & TLS - help please...
In-Reply-To: <BFB88CA0-8319-4D05-B3CD-39E3450DE157@jerris.com>
References: <d361ca7b1001030757m2f55a400n3e2b3b74185c4202@mail.gmail.com>
	<BFB88CA0-8319-4D05-B3CD-39E3450DE157@jerris.com>
Message-ID: <D950E561-1B94-48E0-9221-3739BAB8F91A@freeswitch.org>

If you didn't manually install the CA cert into the phone as per the wiki it won't ever work.

/b

On Jan 3, 2010, at 7:25 PM, Michael Jerris wrote:

> Perhapse cranking up the Sofia tport log to level 9 may help.
> 
> Mike
> 
> On Jan 3, 2010, at 10:57 AM, Yehavi Bourvine  
> <yehavi.bourvine at gmail.com> wrote:
> 
>> Hello,
>> 
>>  I am writing again because I am quite desparate... I fail to  
>> enable TLS on Polycom while on SNOM I can make it work with the same  
>> configuration. From TCPDUMP I see the following difference between  
>> the two handshakes:
>> 
>> SNOM: The protocol suite used is TLS_RSA_WITH_RC4_128_SHA
>> Polyco: The protocol suite used is TLS_RSA_WITH_AES_256_CBC_SHA and  
>> sessionTicketTLS also appears there. After the key exchange the  
>> phone disconnects the connection.
>> Any idea how to debug it?
>> 
>>                               Thanks! __Yehavi:

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From brian at freeswitch.org  Sun Jan  3 18:36:15 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 3 Jan 2010 20:36:15 -0600
Subject: [Freeswitch-users] system functions silently ignored
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E906A13E@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E906A04C@prato.activenetwerx.local>,
	<CFF81042AFE645CCA2D6AE3276D63283@cune.pri>
	<CA5A491E9DEFBE4CB777DE97E21575E906A13E@prato.activenetwerx.local>
Message-ID: <78ED9C83-CC46-41D5-AA2F-85291381B5C3@freeswitch.org>

Funny Pre's no longer exist use this:

http://latest.freeswitch.org/freeswitch-1.0.5-latest.tar.gz

/b

On Jan 2, 2010, at 5:48 PM, Joseph L. Casale wrote:

>>> What could cause this
>> 
>> Have you updated to the latest release in SVN?
> 
> Yeah, sorry that was on latest, I just recompiled pre10 and same behaviour now?




From brian at freeswitch.org  Sun Jan  3 18:36:45 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 3 Jan 2010 20:36:45 -0600
Subject: [Freeswitch-users] Dropped calls
In-Reply-To: <33c87fa31001021609t60ed802eyd6d6d4db7d314e1d@mail.gmail.com>
References: <33c87fa31001021609t60ed802eyd6d6d4db7d314e1d@mail.gmail.com>
Message-ID: <159CEFD5-4E6C-42EC-AB2B-29A9C6649A50@freeswitch.org>

Fairly old SVN rev please update, try again.. then post if it's not fixed.

http://latest.freeswitch.org/freeswitch-1.0.5-latest.tar.gz

/b

On Jan 2, 2010, at 6:09 PM, Mark Campbell-Smith wrote:

> FS version is FreeSWITCH Version 1.0.trunk (15490)




From brian at freeswitch.org  Sun Jan  3 18:40:52 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 3 Jan 2010 20:40:52 -0600
Subject: [Freeswitch-users] jitter buffer effecting input callback in lua
In-Reply-To: <F46682E5-43C4-4B1F-A318-ECAC7608353E@jerris.com>
References: <21e9d36c1001031231yed4a914u5c8e7e8568208c45@mail.gmail.com>
	<F46682E5-43C4-4B1F-A318-ECAC7608353E@jerris.com>
Message-ID: <7073E494-191F-4AAD-90EF-10CA6BEDA7C8@freeswitch.org>

Also please check collect an RTP packet cap if possible this will help too.

/b

On Jan 3, 2010, at 7:29 PM, Michael Jerris wrote:

> There is no reason I can think of that would cause this.  Have you  
> tried different phones to eliminate if it is an issue just with one  
> type of phone?  Please open a bug on jira.freeswitch.org with a  
> minimal script example to reproduce and details of the devices it has  
> been reproduced with.
> 
> Mike

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From jcasale at activenetwerx.com  Sun Jan  3 19:09:18 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Mon, 4 Jan 2010 03:09:18 +0000
Subject: [Freeswitch-users] system functions silently ignored
In-Reply-To: <78ED9C83-CC46-41D5-AA2F-85291381B5C3@freeswitch.org>
References: <CA5A491E9DEFBE4CB777DE97E21575E906A04C@prato.activenetwerx.local>,
	<CFF81042AFE645CCA2D6AE3276D63283@cune.pri>
	<CA5A491E9DEFBE4CB777DE97E21575E906A13E@prato.activenetwerx.local>
	<78ED9C83-CC46-41D5-AA2F-85291381B5C3@freeswitch.org>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906AC73@prato.activenetwerx.local>

>Funny Pre's no longer exist use this:
>
>http://latest.freeswitch.org/freeswitch-1.0.5-latest.tar.gz

That's what I was on, I just tried pre10 to see if it was a bug
in that "latest" build.

This is on hold for a day or two now, when this server shuts off
it panic's with a fault related to dahdi, so given that, I am not
going to trouble shoot this until obviously that's resolved.

I'll get back to this once we have some new hardware...



From yehavi.bourvine at gmail.com  Sun Jan  3 19:32:35 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 4 Jan 2010 05:32:35 +0200
Subject: [Freeswitch-users] Polycom & TLS - help please...
In-Reply-To: <D950E561-1B94-48E0-9221-3739BAB8F91A@freeswitch.org>
References: <d361ca7b1001030757m2f55a400n3e2b3b74185c4202@mail.gmail.com>
	<BFB88CA0-8319-4D05-B3CD-39E3450DE157@jerris.com>
	<D950E561-1B94-48E0-9221-3739BAB8F91A@freeswitch.org>
Message-ID: <d361ca7b1001031932m7695150fv967c0df683083af9@mail.gmail.com>

I've built the slef-signed root certificate and server;s certificate per the
TLS wiki, and installed the root certificate on the phone (both manually and
via the config files). I did not enter the "== untrusted ==" instead of the
cerificate (as the Polycom's wiki suggests) as the pone doesn't allow this.
It accepted the certificate.

I've tried this on 501 (running 3.1.3 which is the last supported version on
it), and 550 & 650 running 3.2.2.

                            Thans, __Yehavi:

 2010/1/4 Brian West <brian at freeswitch.org>

>  If you didn't manually install the CA cert into the phone as per the wiki
> it won't ever work.
>
> /b
>
>  On Jan 3, 2010, at 7:25 PM, Michael Jerris wrote:
>
> Perhapse cranking up the Sofia tport log to level 9 may help.
>
> Mike
>
> On Jan 3, 2010, at 10:57 AM, Yehavi Bourvine
> <yehavi.bourvine at gmail.com> wrote:
>
> Hello,
>
>
>  I am writing again because I am quite desparate... I fail to
>
> enable TLS on Polycom while on SNOM I can make it work with the same
>
> configuration. From TCPDUMP I see the following difference between
>
> the two handshakes:
>
>
> SNOM: The protocol suite used is TLS_RSA_WITH_RC4_128_SHA
>
> Polyco: The protocol suite used is TLS_RSA_WITH_AES_256_CBC_SHA and
>
> sessionTicketTLS also appears there. After the key exchange the
>
> phone disconnects the connection.
>
> Any idea how to debug it?
>
>
>                               Thanks! __Yehavi:
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nicolas at medularis.com  Sun Jan  3 22:45:20 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 4 Jan 2010 03:45:20 -0300
Subject: [Freeswitch-users] How to control call volume?
Message-ID: <1b46b4e81001032245k7d10e027p46b21e50bfb2264f@mail.gmail.com>

Hi, is there a way of controlling the volume of a call? I'm bridging
calls with a JS script. Sometimes the people getting the calls
complain the volume is too low. I've recorded a few of the calls and
most of the times, while playing the recorded wav files, the volume of
LegB (second leg of the bridge) is pretty hard to hear, even with the
computer and player volume to the max (ok, it's a laptop, but even
with headphones). I saw there are some volume control options for
conferences, but I couldn't find anything for regular "originate
calls" or bridges. I am doing transcoding, so that might help (?).

Thank you very much for your help.

Best,

Nicolas



From ken at ukgb.net  Mon Jan  4 01:14:17 2010
From: ken at ukgb.net (Ken Gillett)
Date: Mon, 4 Jan 2010 09:14:17 +0000
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <207e7a5e1001020840y1bd3bae2y2b683a19659c9d7d@mail.gmail.com>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com>
	<B6763940-A3CB-4006-8DF1-ADEDD9C68294@ukgb.net>
	<0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org>
	<CA53F057-4A51-457C-A673-B0B6E4DF482A@ukgb.net>
	<207e7a5e0912301103h17133fdfk22430bbce5a1718b@mail.gmail.com>
	<090E289D-76EE-4F4A-82DF-513626A915E6@ukgb.net>
	<79B790D4-DE06-47FF-B9FD-52D9E1EB8BCB@ukgb.net>
	<207e7a5e1001020840y1bd3bae2y2b683a19659c9d7d@mail.gmail.com>
Message-ID: <6E6E6D36-6E93-49BF-9516-FC028E73D279@ukgb.net>

I will look into this, but in the meantime, would it not be possible to simply copy the installed files? I realise that basically this must be possible, but practically it depends on what is installed where. I believe the default prefix is /usr/local/freeswitch, but is EVERYTHING in there or is other stuff scattered in other directories?

If it's all together I'm thinking I could simply tar the folder and copy the tarball. Wouldn't this be possible?


On 2 Jan 2010, at 16:40, jonathan augenstine wrote:

> A dmg install package would need to be created.  A default package does not currently exist and it would need to be created.  You would need to do this, particularly if you are going to customize the build/install.
> 
> On Sat, Jan 2, 2010 at 5:17 AM, Ken Gillett <ken at ukgb.net> wrote:
> One question still outstanding:-
> 
> How can I compile FS on one Mac and install it onto a different Mac? This means compiling on a MacPro running Snow Leopard and then installing onto a Snow Leopard Server which doesn't have the developer tools installed (and I don't want it to).
> 
> Advice on this would be appreciated.
> 



Ken G i l l e t t

_/_/_/_/_/_/_/_/






From sharad at coraltele.com  Mon Jan  4 01:54:51 2010
From: sharad at coraltele.com (Sharad)
Date: Mon, 4 Jan 2010 01:54:51 -0800 (PST)
Subject: [Freeswitch-users]  time stamp for sched_api
Message-ID: <1262598891498-4249284.post@n2.nabble.com>


I am trying the following API. I want this API to run on 3rd Jan 2010 at
15:13 hours.

sched_api 010315132010  none originate sofia/external/1006 at 192.168.4.106
&playback(ivr/alarm.wav)

But it is not getting activated on the maturity of this time.

Can someone let me know the error in the time format.

Regards 
Sharad


-- 
View this message in context: http://n2.nabble.com/time-stamp-for-sched-api-tp4249284p4249284.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From wasim at convergence.pk  Mon Jan  4 02:23:07 2010
From: wasim at convergence.pk (Wasim Baig)
Date: Mon, 4 Jan 2010 15:23:07 +0500
Subject: [Freeswitch-users] time stamp for sched_api
In-Reply-To: <1262598891498-4249284.post@n2.nabble.com>
References: <1262598891498-4249284.post@n2.nabble.com>
Message-ID: <b8ad2a5b1001040223yc7ab19dm57e0101ee335a2f1@mail.gmail.com>

On Mon, Jan 4, 2010 at 2:54 PM, Sharad <sharad at coraltele.com> wrote:

>
> I am trying the following API. I want this API to run on 3rd Jan 2010 at
> 15:13 hours.
>
> sched_api 010315132010  none originate sofia/external/1006 at 192.168.4.106
> &playback(ivr/alarm.wav)
>
> But it is not getting activated on the maturity of this time.
>
> Can someone let me know the error in the time format.
>

unixtimestamp ......seconds since Jan 01 1970.

-- 
wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
peace be upon you ...
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From sharad at coraltele.com  Mon Jan  4 02:51:39 2010
From: sharad at coraltele.com (Sharad)
Date: Mon, 4 Jan 2010 02:51:39 -0800 (PST)
Subject: [Freeswitch-users] time stamp for sched_api
In-Reply-To: <b8ad2a5b1001040223yc7ab19dm57e0101ee335a2f1@mail.gmail.com>
References: <1262598891498-4249284.post@n2.nabble.com>
	<b8ad2a5b1001040223yc7ab19dm57e0101ee335a2f1@mail.gmail.com>
Message-ID: <1262602299063-4249417.post@n2.nabble.com>


Thanks Mr. Baig..

So can you plz write the syntax for the same API if we want this to get
executed on 4th Jan 2010 at 1800 hours.

Regards



Wasim Baig wrote:
> 
> On Mon, Jan 4, 2010 at 2:54 PM, Sharad <sharad at coraltele.com> wrote:
> 
>>
>> I am trying the following API. I want this API to run on 3rd Jan 2010 at
>> 15:13 hours.
>>
>> sched_api 010315132010  none originate sofia/external/1006 at 192.168.4.106
>> &playback(ivr/alarm.wav)
>>
>> But it is not getting activated on the maturity of this time.
>>
>> Can someone let me know the error in the time format.
>>
> 
> unixtimestamp ......seconds since Jan 01 1970.
> 
> -- 
> wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
> peace be upon you ...
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/time-stamp-for-sched-api-tp4249284p4249417.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From nicolas at medularis.com  Mon Jan  4 07:10:42 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 4 Jan 2010 12:10:42 -0300
Subject: [Freeswitch-users] Hangup on silence?
Message-ID: <1b46b4e81001040710x4886d57ci8227c80c7fe3b796@mail.gmail.com>

Hi, is there a way to hangup after a certain amount of silence? my
problem is that with regular PSTN calls I don't get a BYE from my
provider until around 1 minute after the phone has been hanged up.
This is pretty standard, and the provider is not at fault, but I would
like to hangup the call after I detect a certain amount of silence
time. The only thing I could find was wait_for_silence on the wiki,
but that application only delays dialplan execution, and I would need
something like: hangup_on_silence=30, where 30 means to hangup if 30
seconds of silence have passed.

Thanks for your help!

Nicolas



From brian at freeswitch.org  Mon Jan  4 07:31:22 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 4 Jan 2010 09:31:22 -0600
Subject: [Freeswitch-users] Polycom & TLS - help please...
In-Reply-To: <d361ca7b1001031932m7695150fv967c0df683083af9@mail.gmail.com>
References: <d361ca7b1001030757m2f55a400n3e2b3b74185c4202@mail.gmail.com>
	<BFB88CA0-8319-4D05-B3CD-39E3450DE157@jerris.com>
	<D950E561-1B94-48E0-9221-3739BAB8F91A@freeswitch.org>
	<d361ca7b1001031932m7695150fv967c0df683083af9@mail.gmail.com>
Message-ID: <3968BF39-7A15-4F9F-8FA2-A78BC4F0F38E@freeswitch.org>

OK to properly use TLS you have to setup NAPTR and SRV records and the DNS domain has to match the cert.  Did you do that?

/b

On Jan 3, 2010, at 9:32 PM, Yehavi Bourvine wrote:

> I've built the slef-signed root certificate and server;s certificate per the TLS wiki, and installed the root certificate on the phone (both manually and via the config files). I did not enter the "== untrusted ==" instead of the cerificate (as the Polycom's wiki suggests) as the pone doesn't allow this. It accepted the certificate.
>  
> I've tried this on 501 (running 3.1.3 which is the last supported version on it), and 550 & 650 running 3.2.2.
>  
>                             Thans, __Yehavi:




From brian at freeswitch.org  Mon Jan  4 07:35:15 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 4 Jan 2010 09:35:15 -0600
Subject: [Freeswitch-users] time stamp for sched_api
In-Reply-To: <1262602299063-4249417.post@n2.nabble.com>
References: <1262598891498-4249284.post@n2.nabble.com>
	<b8ad2a5b1001040223yc7ab19dm57e0101ee335a2f1@mail.gmail.com>
	<1262602299063-4249417.post@n2.nabble.com>
Message-ID: <B3DC0945-8C81-4BD5-8821-F48A3B894E83@freeswitch.org>

You could open a bounty and pay someone to write it...  also you can prepend it with a +sign... IE +60 and make it execute 60 seconds from NOW.

/b

On Jan 4, 2010, at 4:51 AM, Sharad wrote:

> 
> Thanks Mr. Baig..
> 
> So can you plz write the syntax for the same API if we want this to get
> executed on 4th Jan 2010 at 1800 hours.
> 
> Regards




From vinuth.madinur at gmail.com  Mon Jan  4 07:41:30 2010
From: vinuth.madinur at gmail.com (Vinuth Madinur)
Date: Mon, 4 Jan 2010 21:11:30 +0530
Subject: [Freeswitch-users] Hangup on silence?
In-Reply-To: <1b46b4e81001040710x4886d57ci8227c80c7fe3b796@mail.gmail.com>
References: <1b46b4e81001040710x4886d57ci8227c80c7fe3b796@mail.gmail.com>
Message-ID: <910309031001040741x219d4085k5821045f5b5edce8@mail.gmail.com>

You can call wait_for_silence with timeout as 1 minute. It'll wait for max 1
minute, but it'll return immediately when there is silence of mentioned
length. You can next invoke hangup in the dialplan.

Or, you can use event socket library to call "wait_for_silence" and wait for
"CHANNEL_EXECUTE_COMPLETE" with
variable_current_application=wait_for_silence, upon which you can hangup.

Helps?


On Mon, Jan 4, 2010 at 8:40 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Hi, is there a way to hangup after a certain amount of silence? my
> problem is that with regular PSTN calls I don't get a BYE from my
> provider until around 1 minute after the phone has been hanged up.
> This is pretty standard, and the provider is not at fault, but I would
> like to hangup the call after I detect a certain amount of silence
> time. The only thing I could find was wait_for_silence on the wiki,
> but that application only delays dialplan execution, and I would need
> something like: hangup_on_silence=30, where 30 means to hangup if 30
> seconds of silence have passed.
>
> Thanks for your help!
>
> Nicolas
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nicolas at medularis.com  Mon Jan  4 07:59:29 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 4 Jan 2010 12:59:29 -0300
Subject: [Freeswitch-users] Hangup on silence?
In-Reply-To: <910309031001040741x219d4085k5821045f5b5edce8@mail.gmail.com>
References: <1b46b4e81001040710x4886d57ci8227c80c7fe3b796@mail.gmail.com>
	<910309031001040741x219d4085k5821045f5b5edce8@mail.gmail.com>
Message-ID: <1b46b4e81001040759n1cb84b18n1cab0138a6ed05cf@mail.gmail.com>

Vinuth, thank you very much for your response. I guess the event
socket solution is the best for me, since I'm not using the dialplan.
for these calls Do you think there's an alternative for Javascript
though? hehe.


On Mon, Jan 4, 2010 at 12:41 PM, Vinuth Madinur
<vinuth.madinur at gmail.com> wrote:
> You can call wait_for_silence with timeout as 1 minute. It'll wait for max 1
> minute, but it'll return immediately when there is silence of mentioned
> length. You can next invoke hangup in the dialplan.
> Or, you can use event socket library to call "wait_for_silence" and wait for
> "CHANNEL_EXECUTE_COMPLETE" with
> variable_current_application=wait_for_silence, upon which you can hangup.
> Helps?
>



From sos at sokhapkin.dyndns.org  Mon Jan  4 05:51:23 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 4 Jan 2010 08:51:23 -0500
Subject: [Freeswitch-users] Was call answered or not?
Message-ID: <201001040851.23977.sos@sokhapkin.dyndns.org>

Which channel variable accessible from mod_cdr_csv can be used to reliable 
find out if the call was answered or not? "billsec" can't be used - it is 
equal to 0 if the call has been answered for less than 1 second.



From mike at jerris.com  Mon Jan  4 08:01:11 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Jan 2010 11:01:11 -0500
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <6E6E6D36-6E93-49BF-9516-FC028E73D279@ukgb.net>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com>
	<B6763940-A3CB-4006-8DF1-ADEDD9C68294@ukgb.net>
	<0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org>
	<CA53F057-4A51-457C-A673-B0B6E4DF482A@ukgb.net>
	<207e7a5e0912301103h17133fdfk22430bbce5a1718b@mail.gmail.com>
	<090E289D-76EE-4F4A-82DF-513626A915E6@ukgb.net>
	<79B790D4-DE06-47FF-B9FD-52D9E1EB8BCB@ukgb.net>
	<207e7a5e1001020840y1bd3bae2y2b683a19659c9d7d@mail.gmail.com>
	<6E6E6D36-6E93-49BF-9516-FC028E73D279@ukgb.net>
Message-ID: <5BED3371-0023-4D53-BBE1-37D241B4AAFB@jerris.com>

its all in there.

Mike

On Jan 4, 2010, at 4:14 AM, Ken Gillett wrote:

> I will look into this, but in the meantime, would it not be possible to simply copy the installed files? I realise that basically this must be possible, but practically it depends on what is installed where. I believe the default prefix is /usr/local/freeswitch, but is EVERYTHING in there or is other stuff scattered in other directories?
> 
> If it's all together I'm thinking I could simply tar the folder and copy the tarball. Wouldn't this be possible?
> 
> 
> On 2 Jan 2010, at 16:40, jonathan augenstine wrote:
> 
>> A dmg install package would need to be created.  A default package does not currently exist and it would need to be created.  You would need to do this, particularly if you are going to customize the build/install.
>> 
>> On Sat, Jan 2, 2010 at 5:17 AM, Ken Gillett <ken at ukgb.net> wrote:
>> One question still outstanding:-
>> 
>> How can I compile FS on one Mac and install it onto a different Mac? This means compiling on a MacPro running Snow Leopard and then installing onto a Snow Leopard Server which doesn't have the developer tools installed (and I don't want it to).
>> 
>> Advice on this would be appreciated.
>> 




From brian at freeswitch.org  Mon Jan  4 08:05:05 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 4 Jan 2010 10:05:05 -0600
Subject: [Freeswitch-users] Was call answered or not?
In-Reply-To: <201001040851.23977.sos@sokhapkin.dyndns.org>
References: <201001040851.23977.sos@sokhapkin.dyndns.org>
Message-ID: <AA7E74A2-3C0E-44F8-A585-0BE706D763C7@freeswitch.org>

Less than one second will usually mean ZERO... you can use 


Caller-Profile-Created-Time: 1262619753748917
Caller-Channel-Created-Time: 1262619753748917
Caller-Channel-Answered-Time: 1262619754069545
Caller-Channel-Progress-Time: 0
Caller-Channel-Progress-Media-Time: 1262619754069545
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0


These will give you in nano second I think.. maybe micro second.

/b

On Jan 4, 2010, at 7:51 AM, Sergey Okhapkin wrote:

> Which channel variable accessible from mod_cdr_csv can be used to reliable 
> find out if the call was answered or not? "billsec" can't be used - it is 
> equal to 0 if the call has been answered for less than 1 second.




From devel at thom.fr.eu.org  Mon Jan  4 08:07:37 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Mon, 04 Jan 2010 17:07:37 +0100
Subject: [Freeswitch-users] Zap channel not released when voicemail starts
Message-ID: <ea085d8b43f8105108d9c1063fb56534@thom.fr.eu.org>



Hello, 

I have an issue with voicemail and openzap channels. 

When an
incoming call on an openzap channel is bridged to voicemail, if that
channel is hung up before the beginning of voicemail recording, that
channel is kept open open until 3 or 4 seconds after the voicemail started
to record the message. 

What should I do to make freeswitch/voicemail
release the channel immediately when the caller hang up ? 

Thanks in
advance 

Fran?ois
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From sos at sokhapkin.dyndns.org  Mon Jan  4 08:19:14 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 4 Jan 2010 11:19:14 -0500
Subject: [Freeswitch-users] Was call answered or not?
In-Reply-To: <AA7E74A2-3C0E-44F8-A585-0BE706D763C7@freeswitch.org>
References: <201001040851.23977.sos@sokhapkin.dyndns.org>
	<AA7E74A2-3C0E-44F8-A585-0BE706D763C7@freeswitch.org>
Message-ID: <201001041119.15011.sos@sokhapkin.dyndns.org>

Thanks for the idea, if Caller-Channel-Answered-Time is not 0, then the call 
has been answered.

On Monday 04 January 2010, Brian West wrote:
> Less than one second will usually mean ZERO... you can use
>
>
> Caller-Profile-Created-Time: 1262619753748917
> Caller-Channel-Created-Time: 1262619753748917
> Caller-Channel-Answered-Time: 1262619754069545
> Caller-Channel-Progress-Time: 0
> Caller-Channel-Progress-Media-Time: 1262619754069545
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
>
>
> These will give you in nano second I think.. maybe micro second.
>
> /b
>
> On Jan 4, 2010, at 7:51 AM, Sergey Okhapkin wrote:
> > Which channel variable accessible from mod_cdr_csv can be used to
> > reliable find out if the call was answered or not? "billsec" can't be
> > used - it is equal to 0 if the call has been answered for less than 1
> > second.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From anthony.minessale at gmail.com  Mon Jan  4 08:30:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Jan 2010 10:30:51 -0600
Subject: [Freeswitch-users] How to control call volume?
In-Reply-To: <1b46b4e81001032245k7d10e027p46b21e50bfb2264f@mail.gmail.com>
References: <1b46b4e81001032245k7d10e027p46b21e50bfb2264f@mail.gmail.com>
Message-ID: <191c3a031001040830j527bcfd4g37797edcab2466f8@mail.gmail.com>

The volume should really be set on the devices who are originally encoding
the audio (the phone or analog card)
Digital audio never changes so the server is not the right place to mess
with the volume because you will have to actually manipulate the digital
signal to do it.  We have a way but I recommend you find the real source of
your problem.

<action application="set_audio_level" data="read 1"/>

change read to write if you want to do it going the other way


On Mon, Jan 4, 2010 at 12:45 AM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Hi, is there a way of controlling the volume of a call? I'm bridging
> calls with a JS script. Sometimes the people getting the calls
> complain the volume is too low. I've recorded a few of the calls and
> most of the times, while playing the recorded wav files, the volume of
> LegB (second leg of the bridge) is pretty hard to hear, even with the
> computer and player volume to the max (ok, it's a laptop, but even
> with headphones). I saw there are some volume control options for
> conferences, but I couldn't find anything for regular "originate
> calls" or bridges. I am doing transcoding, so that might help (?).
>
> Thank you very much for your help.
>
> Best,
>
> Nicolas
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at jerris.com  Mon Jan  4 08:41:45 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Jan 2010 11:41:45 -0500
Subject: [Freeswitch-users] Hangup on silence?
In-Reply-To: <1b46b4e81001040759n1cb84b18n1cab0138a6ed05cf@mail.gmail.com>
References: <1b46b4e81001040710x4886d57ci8227c80c7fe3b796@mail.gmail.com>
	<910309031001040741x219d4085k5821045f5b5edce8@mail.gmail.com>
	<1b46b4e81001040759n1cb84b18n1cab0138a6ed05cf@mail.gmail.com>
Message-ID: <36FB269F-4416-4CFE-AF7F-0E7FBC2BC7E3@jerris.com>

you can bind events in javascript in much the same way.  Why is it normal to not get a hangup for 60 seconds?  

Mike

On Jan 4, 2010, at 10:59 AM, Nicolas Brenner wrote:

> Vinuth, thank you very much for your response. I guess the event
> socket solution is the best for me, since I'm not using the dialplan.
> for these calls Do you think there's an alternative for Javascript
> though? hehe.
> 
> 
> On Mon, Jan 4, 2010 at 12:41 PM, Vinuth Madinur
> <vinuth.madinur at gmail.com> wrote:
>> You can call wait_for_silence with timeout as 1 minute. It'll wait for max 1
>> minute, but it'll return immediately when there is silence of mentioned
>> length. You can next invoke hangup in the dialplan.
>> Or, you can use event socket library to call "wait_for_silence" and wait for
>> "CHANNEL_EXECUTE_COMPLETE" with
>> variable_current_application=wait_for_silence, upon which you can hangup.
>> Helps?
>> 




From anthony.minessale at gmail.com  Mon Jan  4 08:53:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Jan 2010 10:53:17 -0600
Subject: [Freeswitch-users] system functions silently ignored
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E906AC73@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E906A04C@prato.activenetwerx.local>
	<CFF81042AFE645CCA2D6AE3276D63283@cune.pri>
	<CA5A491E9DEFBE4CB777DE97E21575E906A13E@prato.activenetwerx.local>
	<78ED9C83-CC46-41D5-AA2F-85291381B5C3@freeswitch.org>
	<CA5A491E9DEFBE4CB777DE97E21575E906AC73@prato.activenetwerx.local>
Message-ID: <191c3a031001040853s648a774bsc254ce616465e8dd@mail.gmail.com>

The best bet is to use Sangoma cards so you do not need any Dahdi,
That makes you an Emancipated Minor I guess.


On Sun, Jan 3, 2010 at 9:09 PM, Joseph L. Casale
<jcasale at activenetwerx.com>wrote:

> >Funny Pre's no longer exist use this:
> >
> >http://latest.freeswitch.org/freeswitch-1.0.5-latest.tar.gz
>
> That's what I was on, I just tried pre10 to see if it was a bug
> in that "latest" build.
>
> This is on hold for a day or two now, when this server shuts off
> it panic's with a fault related to dahdi, so given that, I am not
> going to trouble shoot this until obviously that's resolved.
>
> I'll get back to this once we have some new hardware...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jcasale at activenetwerx.com  Mon Jan  4 09:05:00 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Mon, 4 Jan 2010 17:05:00 +0000
Subject: [Freeswitch-users] system functions silently ignored
In-Reply-To: <191c3a031001040853s648a774bsc254ce616465e8dd@mail.gmail.com>
References: <CA5A491E9DEFBE4CB777DE97E21575E906A04C@prato.activenetwerx.local>
	<CFF81042AFE645CCA2D6AE3276D63283@cune.pri>
	<CA5A491E9DEFBE4CB777DE97E21575E906A13E@prato.activenetwerx.local>
	<78ED9C83-CC46-41D5-AA2F-85291381B5C3@freeswitch.org>
	<CA5A491E9DEFBE4CB777DE97E21575E906AC73@prato.activenetwerx.local>
	<191c3a031001040853s648a774bsc254ce616465e8dd@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906B34F@prato.activenetwerx.local>

>The best bet is to use Sangoma cards so you do not need any Dahdi, 
>That makes you an Emancipated Minor I guess.

I'll make a note of this point. I soon decided after this initial purchase
ages ago that the next time I ever needed FXO/S ports, I would use an ip gateway.

Thanks,
jlc



From william.suffill at gmail.com  Mon Jan  4 09:44:29 2010
From: william.suffill at gmail.com (William Suffill)
Date: Mon, 4 Jan 2010 12:44:29 -0500
Subject: [Freeswitch-users] time stamp for sched_api
In-Reply-To: <B3DC0945-8C81-4BD5-8821-F48A3B894E83@freeswitch.org>
References: <1262598891498-4249284.post@n2.nabble.com>
	<b8ad2a5b1001040223yc7ab19dm57e0101ee335a2f1@mail.gmail.com> 
	<1262602299063-4249417.post@n2.nabble.com>
	<B3DC0945-8C81-4BD5-8821-F48A3B894E83@freeswitch.org>
Message-ID: <6b65470d1001040944u508f3b25mc08c40a93242c232@mail.gmail.com>

Many programming languages already have functions to handle converting
date/time to a unix timestamp. For example:
http://www.php.net/manual/en/function.mktime.php

takes date/time assuming the local timezone and returns a unix timestamp
that could be used with sched_api.

-- W
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From anthony.minessale at gmail.com  Mon Jan  4 09:47:39 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Jan 2010 11:47:39 -0600
Subject: [Freeswitch-users] PHP ESL Problem
In-Reply-To: <6204a6f4ea0160933ba6493a292c1b9b.squirrel@correo.nodo50.org>
References: <285BD733E19541989B31B95871BF5642@fromage>
	<6204a6f4ea0160933ba6493a292c1b9b.squirrel@correo.nodo50.org>
Message-ID: <191c3a031001040947t6cc3ea8h6a3cb2bc7c06109e@mail.gmail.com>

What you are missing is that you are parsing the results incorrectly.

The events that are showing up as CUSTOM only do so because the id of CUSTOM
is 0
there are no types in the events received with recvEvent the are only used
for transport.

When you say "events plain all"

you are asking for events to be delivered, the FS events are not the same as
the events you are using
to communicate at the lowest level.

What you should be doing is checking for content-type of text/event-plain
and then and only then, get the payload with getBody.
This will contain a serialized event in it's entirety from FS.

for clarity sake I have added a new event SOCKET_DATA to tree and from now
on you will
see those low level events with that type.



On Sat, Jan 2, 2010 at 12:08 PM, Alberto Escudero <aep.lists at it46.se> wrote:

> I do not know if really helps you but we are facing the same problem in
> one of our implementations using the ESL.so for PHP.
>
> We have only see this problem when subscribing to the CHANNEL_STATE
>
> getType() should always match EventName... but it does not
> ./aep
>
> --
> Stopping junk mailers is good for the environment
>
> > Would someone please take a look at this simple PHP event socket script
> > and
> > tell me what I am doing wrong - or tell me that this could be a bug
> > elsewhere?  Any help would be appreciated.
> >
> > When I run the script without the call to execute(), everything seems
> > fine.
> > When I include the call to execute(), the calls to getType() return
> CUSTOM
> > for a while, then later start to return the correct name.
> >
> > #!/usr/bin/php
> > <?php
> > require_once 'ESL.php';
> > $endPoint = 'sofia/internal/695%192.168.100.132';
> >
> > $eventSocket = New ESLconnection('192.168.100.132', '8021', 'ClueCon');
> > $event = $eventSocket->events('plain', 'ALL');
> >
> > // call endpoint, get uuid
> > $event = $eventSocket->api('originate', $endPoint . ' &park');
> > $serializedEvent = explode("\n", $event->serialize());
> > foreach ($serializedEvent as $eventLine) {
> >     list($dummy, $uuid) = explode('+OK ', $eventLine);
> >     if ($uuid) { break; }
> > }
> >
> > // play announcement to endpoint
> > $event = $eventSocket->execute('playback', '/opt/ann/user-busy.wav',
> > $uuid);
> >
> > // monitor events
> > while (TRUE) {
> >     echo "getType: " . $event->getType() . "\n";
> >     $serializedEvent = explode("\n", $event->serialize());
> >     foreach ($serializedEvent as $eventLine) {
> >         list($header, $value) = explode(': ', $eventLine);
> >         if ($header == "Event-Name")   { printf($eventLine . "\n"); }
> >         if ($header == "Content-Type") { printf($eventLine . "\n"); }
> >     }
> >
> >       printf("\n");
> >     $event = $eventSocket->recvEvent();
> > }?>
> >
> >
> > Run without the call to execute():
> > ==================================
> > getType: CUSTOM
> > Content-Type: api/response
> >
> > getType: CHANNEL_CREATE
> > Event-Name: CHANNEL_CREATE
> >
> > getType: CHANNEL_OUTGOING
> > Event-Name: CHANNEL_OUTGOING
> >
> > getType: CHANNEL_ORIGINATE
> > Event-Name: CHANNEL_ORIGINATE
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> > getType: PRESENCE_IN
> > Event-Name: PRESENCE_IN
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> > getType: PRESENCE_IN
> > Event-Name: PRESENCE_IN
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> > getType: CALL_UPDATE
> > Event-Name: CALL_UPDATE
> >
> > getType: CHANNEL_PROGRESS
> > Event-Name: CHANNEL_PROGRESS
> >
> > getType: HEARTBEAT
> > Event-Name: HEARTBEAT
> >
> > getType: HEARTBEAT
> > Event-Name: RE_SCHEDULE
> >
> > getType: CALL_UPDATE
> > Event-Name: CALL_UPDATE
> >
> > getType: CODEC
> > Event-Name: CODEC
> >
> > getType: CODEC
> > Event-Name: CODEC
> >
> > getType: CHANNEL_ANSWER
> > Event-Name: CHANNEL_ANSWER
> >
> > getType: PRESENCE_IN
> > Event-Name: PRESENCE_IN
> >
> > getType: API
> > Event-Name: API
> >
> > getType: PRESENCE_IN
> > Event-Name: PRESENCE_IN
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> > getType: CHANNEL_EXECUTE
> > Event-Name: CHANNEL_EXECUTE
> >
> > getType: CHANNEL_PARK
> > Event-Name: CHANNEL_PARK
> >
> > getType: CHANNEL_HANGUP
> > Event-Name: CHANNEL_HANGUP
> >
> > getType: CHANNEL_UNPARK
> > Event-Name: CHANNEL_UNPARK
> >
> > getType: CHANNEL_EXECUTE_COMPLETE
> > Event-Name: CHANNEL_EXECUTE_COMPLETE
> >
> > getType: PRESENCE_IN
> > Event-Name: PRESENCE_IN
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> > getType: PRESENCE_IN
> > Event-Name: PRESENCE_IN
> >
> > getType: CHANNEL_HANGUP_COMPLETE
> > Event-Name: CHANNEL_HANGUP_COMPLETE
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> > getType: CHANNEL_DESTROY
> > Event-Name: CHANNEL_DESTROY
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> >
> > Run with the call to execute():
> > ===============================
> > getType: CUSTOM
> > Content-Type: command/reply
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_CREATE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_OUTGOING
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_ORIGINATE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_STATE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: PRESENCE_IN
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_STATE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: PRESENCE_IN
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_STATE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CALL_UPDATE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_PROGRESS
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CALL_UPDATE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CODEC
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CODEC
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_ANSWER
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: PRESENCE_IN
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: API
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: PRESENCE_IN
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_STATE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_EXECUTE
> >
> > getType: CUSTOM
> > Content-Type: text/event-plain
> > Event-Name: CHANNEL_PARK
> >
> > getType: CHANNEL_EXECUTE
> > Event-Name: CHANNEL_EXECUTE
> >
> > getType: CHANNEL_HANGUP
> > Event-Name: CHANNEL_HANGUP
> >
> > getType: CHANNEL_EXECUTE_COMPLETE
> > Event-Name: CHANNEL_EXECUTE_COMPLETE
> >
> > getType: COMMAND
> > Event-Name: COMMAND
> >
> > getType: CHANNEL_UNPARK
> > Event-Name: CHANNEL_UNPARK
> >
> > getType: CHANNEL_EXECUTE_COMPLETE
> > Event-Name: CHANNEL_EXECUTE_COMPLETE
> >
> > getType: PRESENCE_IN
> > Event-Name: PRESENCE_IN
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> > getType: PRESENCE_IN
> > Event-Name: PRESENCE_IN
> >
> > getType: CHANNEL_HANGUP_COMPLETE
> > Event-Name: CHANNEL_HANGUP_COMPLETE
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> > getType: CHANNEL_DESTROY
> > Event-Name: CHANNEL_DESTROY
> >
> > getType: CHANNEL_STATE
> > Event-Name: CHANNEL_STATE
> >
> >
> > Thanks,
> > Ron
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From nicolas at medularis.com  Mon Jan  4 10:15:14 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 4 Jan 2010 15:15:14 -0300
Subject: [Freeswitch-users] How to control call volume?
In-Reply-To: <191c3a031001040830j527bcfd4g37797edcab2466f8@mail.gmail.com>
References: <1b46b4e81001032245k7d10e027p46b21e50bfb2264f@mail.gmail.com>
	<191c3a031001040830j527bcfd4g37797edcab2466f8@mail.gmail.com>
Message-ID: <1b46b4e81001041015g74d550b3q3714d3e1f9cd6719@mail.gmail.com>

Great! Thanks! I'll play around with those setting to see how it goes.
Can I set that variable "per leg"? (instead of globally for a call).
The devices getting the calls are regular phones and the termination
service is provided by a few different VoIP companies. I'd say the
volume problem has to do with bad or poorly configured GSM gateways
(that's how they make calls to cellphones), plus maybe some problems
in the GSM network relating to poor signal or something like that. I
can't really control the PSTN or the GSM network and have almost zero
influence with the VoIP companies, so my best bet now is to mess with
the transcoding.

Thank you very much, we'll see how it goes.

Nicolas


On Mon, Jan 4, 2010 at 1:30 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> The volume should really be set on the devices who are originally encoding
> the audio (the phone or analog card)
> Digital audio never changes so the server is not the right place to mess
> with the volume because you will have to actually manipulate the digital
> signal to do it.? We have a way but I recommend you find the real source of
> your problem.
>
> <action application="set_audio_level" data="read 1"/>
>
> change read to write if you want to do it going the other way
>



From nicolas at medularis.com  Mon Jan  4 10:21:37 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 4 Jan 2010 15:21:37 -0300
Subject: [Freeswitch-users] Hangup on silence?
In-Reply-To: <36FB269F-4416-4CFE-AF7F-0E7FBC2BC7E3@jerris.com>
References: <1b46b4e81001040710x4886d57ci8227c80c7fe3b796@mail.gmail.com>
	<910309031001040741x219d4085k5821045f5b5edce8@mail.gmail.com>
	<1b46b4e81001040759n1cb84b18n1cab0138a6ed05cf@mail.gmail.com>
	<36FB269F-4416-4CFE-AF7F-0E7FBC2BC7E3@jerris.com>
Message-ID: <1b46b4e81001041021o7bf52b3bp5f1e4c2a5bf19d80@mail.gmail.com>

Event-based javascript? like Node.js? how would I bind the end of a
wait_for_silence to some script or callback function?

Here, when you call from a regular landline to another, the person
receiving the call may hangup and the call will still be alive, this
is so the person can "transfer" the call between phones connected to
the same line (e.g. kitchen and room phones). Only if the person
originating the call hangs up, the call is terminated right away.
Maybe 60 seconds is too much, but that's the way the PSTN works here.


On Mon, Jan 4, 2010 at 1:41 PM, Michael Jerris <mike at jerris.com> wrote:
> you can bind events in javascript in much the same way. ?Why is it normal to not get a hangup for 60 seconds?
>
> Mike
>



From anthony.minessale at gmail.com  Mon Jan  4 11:16:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Jan 2010 13:16:23 -0600
Subject: [Freeswitch-users] How to control call volume?
In-Reply-To: <1b46b4e81001041015g74d550b3q3714d3e1f9cd6719@mail.gmail.com>
References: <1b46b4e81001032245k7d10e027p46b21e50bfb2264f@mail.gmail.com>
	<191c3a031001040830j527bcfd4g37797edcab2466f8@mail.gmail.com>
	<1b46b4e81001041015g74d550b3q3714d3e1f9cd6719@mail.gmail.com>
Message-ID: <191c3a031001041116x56a102f3y96294c493a30ce67@mail.gmail.com>

yes the example is per_leg per_direction

On Mon, Jan 4, 2010 at 12:15 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Great! Thanks! I'll play around with those setting to see how it goes.
> Can I set that variable "per leg"? (instead of globally for a call).
> The devices getting the calls are regular phones and the termination
> service is provided by a few different VoIP companies. I'd say the
> volume problem has to do with bad or poorly configured GSM gateways
> (that's how they make calls to cellphones), plus maybe some problems
> in the GSM network relating to poor signal or something like that. I
> can't really control the PSTN or the GSM network and have almost zero
> influence with the VoIP companies, so my best bet now is to mess with
> the transcoding.
>
> Thank you very much, we'll see how it goes.
>
> Nicolas
>
>
> On Mon, Jan 4, 2010 at 1:30 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > The volume should really be set on the devices who are originally
> encoding
> > the audio (the phone or analog card)
> > Digital audio never changes so the server is not the right place to mess
> > with the volume because you will have to actually manipulate the digital
> > signal to do it.  We have a way but I recommend you find the real source
> of
> > your problem.
> >
> > <action application="set_audio_level" data="read 1"/>
> >
> > change read to write if you want to do it going the other way
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From msc at freeswitch.org  Mon Jan  4 13:23:21 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Jan 2010 13:23:21 -0800
Subject: [Freeswitch-users] How to control call volume?
In-Reply-To: <191c3a031001041116x56a102f3y96294c493a30ce67@mail.gmail.com>
References: <1b46b4e81001032245k7d10e027p46b21e50bfb2264f@mail.gmail.com>
	<191c3a031001040830j527bcfd4g37797edcab2466f8@mail.gmail.com>
	<1b46b4e81001041015g74d550b3q3714d3e1f9cd6719@mail.gmail.com>
	<191c3a031001041116x56a102f3y96294c493a30ce67@mail.gmail.com>
Message-ID: <87f2f3b91001041323r5984942ete1f326bac4d15cc3@mail.gmail.com>

FYI, I just wikified this dp app:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_audio_level

Feel free to add/edit as needed.
-MC

On Mon, Jan 4, 2010 at 11:16 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> yes the example is per_leg per_direction
>
>
> On Mon, Jan 4, 2010 at 12:15 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> Great! Thanks! I'll play around with those setting to see how it goes.
>> Can I set that variable "per leg"? (instead of globally for a call).
>> The devices getting the calls are regular phones and the termination
>> service is provided by a few different VoIP companies. I'd say the
>> volume problem has to do with bad or poorly configured GSM gateways
>> (that's how they make calls to cellphones), plus maybe some problems
>> in the GSM network relating to poor signal or something like that. I
>> can't really control the PSTN or the GSM network and have almost zero
>> influence with the VoIP companies, so my best bet now is to mess with
>> the transcoding.
>>
>> Thank you very much, we'll see how it goes.
>>
>> Nicolas
>>
>>
>> On Mon, Jan 4, 2010 at 1:30 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > The volume should really be set on the devices who are originally
>> encoding
>> > the audio (the phone or analog card)
>> > Digital audio never changes so the server is not the right place to mess
>> > with the volume because you will have to actually manipulate the digital
>> > signal to do it.  We have a way but I recommend you find the real source
>> of
>> > your problem.
>> >
>> > <action application="set_audio_level" data="read 1"/>
>> >
>> > change read to write if you want to do it going the other way
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jerry.richards at teotech.com  Mon Jan  4 15:49:32 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 4 Jan 2010 15:49:32 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
Message-ID: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>


Hello,

I have one FS instance that is working well with a PRI and running FS
version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
upgraded it.

Now, I am trying to bring up another FS instance (basically a clone of the
first), but the PRI does not work.  When I attempt to make an
internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
both servers are using the latest Sangoma Wanpipe driver, and I copied the
conf XML file tree from the old server to the new one.  I think the problem
has to do with the openzap module, but I'm having difficulty isolating the
problem.  Could it have built the openzap module incorrectly?  Another
difference is that I installed 1.0.5pre9 from scratch on the new server
(i.e. it never had 1.0.4 running on it).

I put the FS log into the pastebin when an outbound call attempt is made:  

http://pastebin.freeswitch.org/11675

Could someone give me a pointer on what to try next?

Thanks and Best Regards,
Jerry




From anthony.minessale at gmail.com  Mon Jan  4 16:00:05 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Jan 2010 18:00:05 -0600
Subject: [Freeswitch-users] Zap channel not released when voicemail
	starts
In-Reply-To: <ea085d8b43f8105108d9c1063fb56534@thom.fr.eu.org>
References: <ea085d8b43f8105108d9c1063fb56534@thom.fr.eu.org>
Message-ID: <191c3a031001041600y1c98465cifcc522056096f746@mail.gmail.com>

hangup detection on TDM is a bitch.


On Mon, Jan 4, 2010 at 10:07 AM, Fran?ois Legal <devel at thom.fr.eu.org>wrote:

> Hello,
>
>
>
> I have an issue with voicemail and openzap channels.
>
> When an incoming call on an openzap channel is bridged to voicemail, if
> that channel is hung up before the beginning of voicemail recording, that
> channel is kept open open until 3 or 4 seconds after the voicemail started
> to record the message.
>
> What should I do to make freeswitch/voicemail release the channel
> immediately when the caller hang up ?
>
>
>
> Thanks in advance
>
>
>
> Fran?ois
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From timuckun at gmail.com  Mon Jan  4 18:42:57 2010
From: timuckun at gmail.com (Tim Uckun)
Date: Tue, 5 Jan 2010 15:42:57 +1300
Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS
	based callcenter)
In-Reply-To: <20091219014359.GA21798@hijacked.us>
References: <20091219014359.GA21798@hijacked.us>
Message-ID: <855e4dcf1001041842u46c6d36q9c2e4ece2ced7380@mail.gmail.com>

> http://opencsm.org/downloads/spice-telephony-0.9.6.tar.gz
>

There seems to be something wrong with both opencsm.org and wiki.opencsm.org.

Just thought I'd let you know.



From help at pdscc.com  Mon Jan  4 19:08:15 2010
From: help at pdscc.com (Harondel J. Sibble)
Date: Mon, 04 Jan 2010 19:08:15 -0800
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <C2EB0262-CB8A-44F5-9978-1C5F8073832F@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20091228233838.75E611694@sinclaire.sibble.net>,
	<C2EB0262-CB8A-44F5-9978-1C5F8073832F@freeswitch.org>
Message-ID: <20100105030813.8B90012F5@sinclaire.sibble.net>

Brian, Just following up on this, any news?

On 28 Dec 2009 at 17:49, Brian West wrote:

> I'm still not done with this I think we found a bug in the lib... Viktor
> fixed it today and I'm going to retry after I get done testing G729 more
> today!  ;)
> 
> /b
> 
> On Dec 28, 2009, at 5:38 PM, Harondel J. Sibble wrote:
> 
> > Hmm, okay, I went back to basics and did a full rebuild for 1.0.4 svn
> trunk, 
> > I am able to now enroll my nokia e61i running the beta 2.0.7 Tiviphone 
> > client, however I am not seeing the enrollment option popup in zfone 0.92
> > build 218 on windows in front of an x-lite client.
> > 
> > Any suggestions on what I should look at to troubleshoot this?
> > 
> > I am waiting for the Tivi folks to send a 2.0.7 beta for windows mobile,
> but 
> > until then....

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From max.bridgewater at gmail.com  Mon Jan  4 19:34:45 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 4 Jan 2010 22:34:45 -0500
Subject: [Freeswitch-users] Unable to start mod_java
Message-ID: <c55d46951001041934n49bfcbf4h42564e4548aa301a@mail.gmail.com>

Hi,

I built Freeswitch with mod_java enabled. But now, when Freeswitch starts, I
get the following error message:


2010-01-04 22:32:46.574770 [NOTICE] switch_loadable_module.c:271 Adding API
Function 'jsrun'
2010-01-04 22:32:46.574811 [NOTICE] switch_loadable_module.c:271 Adding API
Function 'jsapi'
2010-01-04 22:32:46.575306 [NOTICE] modjava.c:244 Java Framework Loading...
2010-01-04 22:32:46.575721 [ERR] modjava.c:133 Error loading
/usr/local/java/jdk1.6.0_17/jre/lib/i386/client/libjvm.so
2010-01-04 22:32:46.575743 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_java.so
**Module load routine returned an error**
2010-01-04 22:32:46.576742 [CONSOLE] switch_loadable_module.c:890
Successfully Loaded [mod_lua]
2010-01-04 22:32:46.576748 [NOTICE] switch_loadable_module.c:209 Adding
Dialplan 'LUA'
2010-01-04 22:32:46.576795 [NOTICE] switch_loadable_module.c:249 Adding
Application 'lua'


As far as I can tell the
/usr/local/java/jdk1.6.0_17/jre/lib/i386/client/libjvm.so is the correct
path.
Any idea?

max.
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From help at pdscc.com  Mon Jan  4 20:23:30 2010
From: help at pdscc.com (Harondel J. Sibble)
Date: Mon, 04 Jan 2010 20:23:30 -0800
Subject: [Freeswitch-users] encrypt voicemail emails and attachments?
Message-ID: <20100105042327.7CBF412DD@sinclaire.sibble.net>

Looking throught the wiki, I see various configs for having FS email you a 
copy of received voicemail messages, has anyone done any work with having the 
voicemail messages gpg encrypted with public prior to sending?  Or is that 
something that should pretty much be handled at the mta level leaving FS out 
of the mix altogether? I'm thinking probably so, but before I try to do this, 
I figured i'd ask first.


-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From jason at jasonjgw.net  Mon Jan  4 20:25:08 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 5 Jan 2010 15:25:08 +1100
Subject: [Freeswitch-users] Unable to start mod_java
In-Reply-To: <c55d46951001041934n49bfcbf4h42564e4548aa301a@mail.gmail.com>
References: <c55d46951001041934n49bfcbf4h42564e4548aa301a@mail.gmail.com>
Message-ID: <20100105042508.GA25483@jdc.jasonjgw.net>

Max Bridgewater <max.bridgewater at gmail.com> wrote:
 
> As far as I can tell the
> /usr/local/java/jdk1.6.0_17/jre/lib/i386/client/libjvm.so is the correct
> path.
> Any idea?

Permissions?




From mike at jerris.com  Mon Jan  4 20:51:33 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Jan 2010 23:51:33 -0500
Subject: [Freeswitch-users] encrypt voicemail emails and attachments?
In-Reply-To: <20100105042327.7CBF412DD@sinclaire.sibble.net>
References: <20100105042327.7CBF412DD@sinclaire.sibble.net>
Message-ID: <03DDC308-7EC5-46EC-8AFB-CC73417E20F5@jerris.com>

you can just make a shell script (or perl or whatever) that is called as the mailer that does this.

Mike

On Jan 4, 2010, at 11:23 PM, Harondel J. Sibble wrote:

> Looking throught the wiki, I see various configs for having FS email you a 
> copy of received voicemail messages, has anyone done any work with having the 
> voicemail messages gpg encrypted with public prior to sending?  Or is that 
> something that should pretty much be handled at the mta level leaving FS out 
> of the mix altogether? I'm thinking probably so, but before I try to do this, 
> I figured i'd ask first.
> 




From sharad at coraltele.com  Mon Jan  4 20:56:36 2010
From: sharad at coraltele.com (Sharad)
Date: Mon, 4 Jan 2010 20:56:36 -0800 (PST)
Subject: [Freeswitch-users] time stamp for sched_api
In-Reply-To: <6b65470d1001040944u508f3b25mc08c40a93242c232@mail.gmail.com>
References: <1262598891498-4249284.post@n2.nabble.com>
	<b8ad2a5b1001040223yc7ab19dm57e0101ee335a2f1@mail.gmail.com>
	<1262602299063-4249417.post@n2.nabble.com>
	<B3DC0945-8C81-4BD5-8821-F48A3B894E83@freeswitch.org>
	<6b65470d1001040944u508f3b25mc08c40a93242c232@mail.gmail.com>
Message-ID: <1262667396468-4253597.post@n2.nabble.com>




William Suffill wrote:
> 
> Many programming languages already have functions to handle converting
> date/time to a unix timestamp. For example:
> http://www.php.net/manual/en/function.mktime.php
> 
> takes date/time assuming the local timezone and returns a unix timestamp
> that could be used with sched_api.
> 
> -- W
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 


thanks /b n william for your kind answer.

regards
sharad
-- 
View this message in context: http://n2.nabble.com/time-stamp-for-sched-api-tp4249284p4253597.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From sharad at coraltele.com  Mon Jan  4 20:59:18 2010
From: sharad at coraltele.com (Sharad)
Date: Mon, 4 Jan 2010 20:59:18 -0800 (PST)
Subject: [Freeswitch-users] [!! SPAM] Re: Self alarm
In-Reply-To: <207e7a5e1001012050g18e5563dpe99e55b0509c5625@mail.gmail.com>
References: <1262250725607-4235713.post@n2.nabble.com>
	<1262326847726-4238924.post@n2.nabble.com>
	<2A1E0DAF-A680-47B6-AC49-6A80FEC312A2@jerris.com>
	<002b01ca8b65$16762640$0c04a8c0@compaq77db609e>
	<207e7a5e1001012050g18e5563dpe99e55b0509c5625@mail.gmail.com>
Message-ID: <1262667558154-4253610.post@n2.nabble.com>




jonathan augenstine wrote:
> 
> I believe that the question is, what do you want to alarm?  Do you want to
> setup basic monitoring of the system?  Are you trying to track T1 alarms?
>  Your question is too vague to answer.
> 
> On Fri, Jan 1, 2010 at 8:35 PM, Sharad <sharad at coraltele.com> wrote:
> 
>> Self Alarm..
>>
>> ----- Original Message -----
>> *From:* [hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4241557&i=0>
>> *To:* [hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4241557&i=1>
>> *Sent:* Friday, January 01, 2010 9:04 PM
>> *Subject:* [!! SPAM] Re: [Freeswitch-users] Self alarm
>>
>> The same what?
>>
>> On Jan 1, 2010, at 1:20 AM, Sharad <[hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4239714&i=0>>
>> wrote:
>>
>> >
>> > Hi
>> >
>> > I am also intresting in the same.
>> >
>> > Is there any script for this functionality.
>> >
>> > Regards
>> > --
>> > View this message in context:
>> http://n2.nabble.com/Self-alarm-tp4235713p4238924.html
>> > Sent from the freeswitch-users mailing list archive at Nabble.com.
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > [hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4239714&i=1>
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> > users
>> > http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> [hidden
>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=4239714&i=2>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> ------------------------------
>> View this message in context: Re: [!! SPAM] Re: [Freeswitch-users] Self
>> alarm <http://n2.nabble.com/Self-alarm-tp4235713p4241557.html>
>>
>> Sent from the freeswitch-users mailing list
>> archive<http://n2.nabble.com/freeswitch-users-f2379917.html>at
>> Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

we are writing a script for the self alarm application which will make the
reminder call to the predefined user at the defined time. It is under
testing& will take some more days.

Once it is done, I will upload the same for everyone.

regards
sharad

-- 
View this message in context: http://n2.nabble.com/Self-alarm-tp4235713p4253610.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From help at pdscc.com  Mon Jan  4 22:54:01 2010
From: help at pdscc.com (Harondel J. Sibble)
Date: Mon, 04 Jan 2010 22:54:01 -0800
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <C2EB0262-CB8A-44F5-9978-1C5F8073832F@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20091228233838.75E611694@sinclaire.sibble.net>,
	<C2EB0262-CB8A-44F5-9978-1C5F8073832F@freeswitch.org>
Message-ID: <20100105065356.AEE0612F5@sinclaire.sibble.net>

Maybe that's what's affecting me now.....

I've both phones registered (confirmed by calling 9787) on both devices and 
it says each device is already enrolled. (how does one un-enroll????). Both 
phones are running the Tivi 2.0.7 beta. 

Now however, other than the first call I made between devices after 
enrollment, the sas is not matching anymore.

I set both these options in the console

global_action application="set" data="zrtp_enrollment=true"
global_setvar zrtp_secure_media=true

What should I be looking for in the console output

On 28 Dec 2009 at 17:49, Brian West wrote:

> I'm still not done with this I think we found a bug in the lib... Viktor
> fixed it today and I'm going to retry after I get done testing G729 more
> today!  ;)
> 
> /b
> 
> On Dec 28, 2009, at 5:38 PM, Harondel J. Sibble wrote:
> 
> > Hmm, okay, I went back to basics and did a full rebuild for 1.0.4 svn
> trunk, 
> > I am able to now enroll my nokia e61i running the beta 2.0.7 Tiviphone 
> > client, however I am not seeing the enrollment option popup in zfone 0.92
> > build 218 on windows in front of an x-lite client.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From yehavi.bourvine at gmail.com  Tue Jan  5 00:38:22 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 5 Jan 2010 10:38:22 +0200
Subject: [Freeswitch-users] Polycom & TLS - help please...
In-Reply-To: <3968BF39-7A15-4F9F-8FA2-A78BC4F0F38E@freeswitch.org>
References: <d361ca7b1001030757m2f55a400n3e2b3b74185c4202@mail.gmail.com>
	<BFB88CA0-8319-4D05-B3CD-39E3450DE157@jerris.com>
	<D950E561-1B94-48E0-9221-3739BAB8F91A@freeswitch.org>
	<d361ca7b1001031932m7695150fv967c0df683083af9@mail.gmail.com>
	<3968BF39-7A15-4F9F-8FA2-A78BC4F0F38E@freeswitch.org>
Message-ID: <d361ca7b1001050038x4e7103bdke322fd8741403172@mail.gmail.com>

Thanks, I have a partial success which involved two steps:

   - The wiki says to create a root certifcate with *gentls_cert setup *with
   no other parameters; I had to add my domain's data to this command. The new
   certificate has been downloaded to the phone.
   - Replaced the registrar definitions in the phone's config files from IP
   address to the server's name.
   - The above setup worked as-is. To be sure I've added the NAPTR records
   to the DNS after the above two steps worked.
      - BTW, the wiki says that the NAPTR records are not mandatory, thus I
      did not add them at the first place.

I said "partial" because now I have a phenomenon similar (not the same, but
close to) to the one I have with the SNOMs: The TLS link is reseted after a
while and then a new (additional) registration is done. I'll continue search
it per the tips I got on the other topic.

                          Thanks! __Yehavi:

2010/1/4 Brian West <brian at freeswitch.org>

> OK to properly use TLS you have to setup NAPTR and SRV records and the DNS
> domain has to match the cert.  Did you do that?
>
> /b
>
> On Jan 3, 2010, at 9:32 PM, Yehavi Bourvine wrote:
>
> > I've built the slef-signed root certificate and server;s certificate per
> the TLS wiki, and installed the root certificate on the phone (both manually
> and via the config files). I did not enter the "== untrusted ==" instead of
> the cerificate (as the Polycom's wiki suggests) as the pone doesn't allow
> this. It accepted the certificate.
> >
> > I've tried this on 501 (running 3.1.3 which is the last supported version
> on it), and 550 & 650 running 3.2.2.
> >
> >                             Thans, __Yehavi:
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From a.alalousi at gmail.com  Tue Jan  5 01:40:14 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Tue, 5 Jan 2010 09:40:14 +0000
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
Message-ID: <cd93b3701001050140q14182667o8e90dba27fa3bc85@mail.gmail.com>

Hi Jerry,

Looked at your log and your problem is quiet simple to resolve, but first
here's what's happenning:

You copied the conf/ subtree to your new server. As such, you have also
duplicated your vars.xml. By doing so, you have set the domain on the *new
server* to the same domain used on your old server which would be fine, *but
* the default domain settings used by FS is to use your primary IPv4 IP
address as your domain.

By duplicating the conf subtree from the old server, you have effectively
bound the new instance of FS to a domain that is the IP address of the old
server, if this makes sense.

You can see this on third line of your log:

*192.168.72.29 Rejected by acl "domains"*

To resolve this, modify your vars.xml on the new server to reflect whatever
domain it is you want to route, or set the domain to the new server's IP
address like so:

 <X-PRE-PROCESS cmd="set" data="domain=IP_ADDRESS_OF_NEW_SERVER"/>

You also need to check that any other files (e.g. the conf/sip_profiles,
conf/directory/ and conf/dialplan/ hierarchy) are modified to reflect the
new server settings as well.

In the limit, resolving those conflicts will also resolve your issues,
unless there is something else that's wrong.

Let's know how you get along.

Regards,

Ahmed.


2010/1/4 Jerry Richards <jerry.richards at teotech.com>

>
> Hello,
>
> I have one FS instance that is working well with a PRI and running FS
> version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
> upgraded it.
>
> Now, I am trying to bring up another FS instance (basically a clone of the
> first), but the PRI does not work.  When I attempt to make an
> internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
> both servers are using the latest Sangoma Wanpipe driver, and I copied the
> conf XML file tree from the old server to the new one.  I think the problem
> has to do with the openzap module, but I'm having difficulty isolating the
> problem.  Could it have built the openzap module incorrectly?  Another
> difference is that I installed 1.0.5pre9 from scratch on the new server
> (i.e. it never had 1.0.4 running on it).
>
> I put the FS log into the pastebin when an outbound call attempt is made:
>
> http://pastebin.freeswitch.org/11675
>
> Could someone give me a pointer on what to try next?
>
> Thanks and Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From a.alalousi at gmail.com  Tue Jan  5 01:44:20 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Tue, 5 Jan 2010 09:44:20 +0000
Subject: [Freeswitch-users] Zap channel not released when voicemail
	starts
In-Reply-To: <191c3a031001041600y1c98465cifcc522056096f746@mail.gmail.com>
References: <ea085d8b43f8105108d9c1063fb56534@thom.fr.eu.org>
	<191c3a031001041600y1c98465cifcc522056096f746@mail.gmail.com>
Message-ID: <cd93b3701001050144m648905e0i76a0f59fa3b01db1@mail.gmail.com>

I'll second that.

My way of dealing with it has been to write a little script to detect
hangups on the TDM end, then force release the corresponding "B-leg" that is
hooked up to VM. In the process of converting this to an FS module.

Not clean .. but works. Would have liked to see the same code within FS core
and, if appropriate, the VM subsystem to achieve the same end.

Regards,

Ahmed.


2010/1/5 Anthony Minessale <anthony.minessale at gmail.com>

> hangup detection on TDM is a bitch.
>
>
> On Mon, Jan 4, 2010 at 10:07 AM, Fran?ois Legal <devel at thom.fr.eu.org>wrote:
>
>> Hello,
>>
>>
>>
>> I have an issue with voicemail and openzap channels.
>>
>> When an incoming call on an openzap channel is bridged to voicemail, if
>> that channel is hung up before the beginning of voicemail recording, that
>> channel is kept open open until 3 or 4 seconds after the voicemail started
>> to record the message.
>>
>> What should I do to make freeswitch/voicemail release the channel
>> immediately when the caller hang up ?
>>
>>
>>
>> Thanks in advance
>>
>>
>>
>> Fran?ois
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From mcampbellsmith at gmail.com  Tue Jan  5 03:07:50 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 5 Jan 2010 22:07:50 +1100
Subject: [Freeswitch-users] No audio after Remote SDP:
In-Reply-To: <33c87fa31001021519j41ead649rb4a85e6be6236254@mail.gmail.com>
References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com>
	<33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com>
	<772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org>
	<33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com>
	<191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com>
	<33c87fa30912301213g3d24b773k86b95e7ce08f6761@mail.gmail.com>
	<7D75E6D6-8AB9-429E-A5A9-1639C0D5AD09@freeswitch.org>
	<33c87fa31001012330g69fc8e88m61e648d893c4d8f7@mail.gmail.com>
	<180A53B1-EFB6-4B17-AA7C-19B007ADE04C@freeswitch.org>
	<33c87fa31001021519j41ead649rb4a85e6be6236254@mail.gmail.com>
Message-ID: <33c87fa31001050307v3cbcae01q9d7d47383a63ad33@mail.gmail.com>

OK.. I have looked at this some more...   Below is the syslog from the SPA3102:

THIS WORKS
Jan  5 21:18:17 92.xx.xx.xx [0]->124.xxx.xxx.xxx:442(544)
Jan  5 21:18:17 92.xx.xx.xx REGISTER sip:myddns.dydns.org:442 SIP/2.0
Via: SIP/2.0/TLS 92.xx.xx.xx:5069;branch=z9hG4bK-7b7ebbf9  From: 2001
<sip:2001 at myddns.dydns.org:442>;tag=b69c38c549e24c42o0  To: 2001
<sip:2001 at myddns.dydns.org:442>  Call-ID:
e6b918dc-71d58fe3 at 192.168.1.3  CSeq: 44358 REGISTER  Max-Forwards: 70
Contact: 2001 <sip:2001 at 92.xx.xx.xx:5069;transport=tls>;expires=600
User-Agent: Linksys/SPA3102-5.1.10(GW)  Content-Length: 0  Allow: ACK,
BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER  Supported:
x-sipura, replaces
Jan  5 21:18:17 92.xx.xx.xx
Jan  5 21:18:17 92.xx.xx.xx
Jan  5 21:18:17 92.xx.xx.xx [0]<<124.xxx.xxx.xxx:442(658)
Jan  5 21:18:17 92.xx.xx.xx [0]<<124.xxx.xxx.xxx:442(658)
Jan  5 21:18:17 92.xx.xx.xx SIP/2.0 401 Unauthorized  Via: SIP/2.0/TLS
92.xx.xx.xx:5069;branch=z9hG4bK-7b7ebbf9;rport=5069  From: 2001
<sip:2001 at myddns.dydns.org:442>;tag=b69c38c549e24c42o0  To: 2001
<sip:2001 at myddns.dydns.org:442>;tag=Bav1HeBr3jm3B  Call-ID:
e6b918dc-71d58fe3 at 192.168.1.3  CSeq: 44358 REGISTER  User-Agent:
FreeSWITCH-mod_sofia/1.0.trunk-16131  Allow: INVITE, ACK, BYE, CANCEL,
OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH,
SUBSCRIBE  Supported: timer, precondition, path, replaces
WWW-Authenticate: Digest realm="myddns.dydns.org",
nonce="a4128380-f9e3-11de-99eb-53ce5686ac9a", algorithm=MD5,
qop="auth"  Content-Length: 0
Jan  5 21:18:17 92.xx.xx.xx
Jan  5 21:18:17 92.xx.xx.xx
Jan  5 21:18:17 92.xx.xx.xx [0]->124.xxx.xxx.xxx:442(782)
Jan  5 21:18:17 92.xx.xx.xx [0]->124.xxx.xxx.xxx:442(782)
Jan  5 21:18:17 92.xx.xx.xx REGISTER sip:myddns.dydns.org:442 SIP/2.0
Via: SIP/2.0/TLS 92.xx.xx.xx:5069;branch=z9hG4bK-18f00822  From: 2001
<sip:2001 at myddns.dydns.org:442>;tag=b69c38c549e24c42o0  To: 2001
<sip:2001 at myddns.dydns.org:442>  Call-ID:
e6b918dc-71d58fe3 at 192.168.1.3  CSeq: 44359 REGISTER  Max-Forwards: 70
Authorization: Digest
username="2001",realm="myddns.dydns.org",nonce="a4128380-f9e3-11de-99eb-53ce5686ac9a",uri="sip:myddns.dydns.org:442",algorithm=MD5,response="324ee93184ae202be4a209f5a9255229",qop=auth,nc=00000001,cnonce="804844b"
 Contact: 2001 <sip:2001 at 92.xx.xx.xx:5069;transport=tls>;expires=600
User-Agent: Linksys/SPA3102-5.1.10(GW)  Content-Length: 0  Allow: ACK,
BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER  Supported:
x-sipura, replaces

THIS DOES NOT WORK
Jan  5 22:05:48 92.xxx.xxx.xxx REGISTER sip:myddns.dydns.org:442
SIP/2.0  Via: SIP/2.0/TLS 192.168.1.3:5070;branch=z9hG4bK-faf8477a
From: 2001 <sip:2001 at myddns.dydns.org:442>;tag=b065057e3ed0befdo0  To:
2001 <sip:2001 at myddns.dydns.org:442>  Call-ID:
e6b918dc-71d58fe3 at 192.168.1.3  CSeq: 44435 REGISTER  Max-Forwards: 70
Contact: 2001 <sip:2001 at 192.168.1.3:5070;transport=tls>;expires=600
User-Agent: Linksys/SPA3102-5.1.10(GW)  Content-Length: 0  Allow: ACK,
BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER  Supported:
x-sipura, replaces
Jan  5 22:05:48 92.xxx.xxx.xxx SIP/2.0 401 Unauthorized  Via:
SIP/2.0/TLS 192.168.1.3:5070;branch=z9hG4bK-faf8477a;received=92.xxx.xxx.xxx;rport=5070
 From: 2001 <sip:2001 at myddns.dydns.org:442>;tag=b065057e3ed0befdo0
To: 2001 <sip:2001 at myddns.dydns.org:442>;tag=N51jvyeca9Umj  Call-ID:
e6b918dc-71d58fe3 at 192.168.1.3  CSeq: 44435 REGISTER  User-Agent:
FreeSWITCH-mod_sofia/1.0.trunk-16131  Allow: INVITE, ACK, BYE, CANCEL,
OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH,
SUBSCRIBE  Supported: timer, precondition, path, replaces
WWW-Authenticate: Digest realm="myddns.dydns.org",
nonce="476097b0-f9ea-11de-99fd-53ce5686ac9a", algorithm=MD5,
qop="auth"  Content-Length: 0
Jan  5 22:05:48 92.xxx.xxx.xxx
Jan  5 22:05:48 92.xxx.xxx.xxx
Jan  5 22:05:58 92.xxx.xxx.xxx [0]->124.xxx.xxx.xxx:442(783)
Jan  5 22:05:58 92.xxx.xxx.xxx [0]->124.xxx.xxx.xxx:442(783)
Jan  5 22:05:58 92.xxx.xxx.xxx REGISTER sip:myddns.dydns.org:442
SIP/2.0  Via: SIP/2.0/TLS 92.xxx.xxx.xxx:5070;branch=z9hG4bK-bfc992b3
From: 2001 <sip:2001 at myddns.dydns.org:442>;tag=b065057e3ed0befdo0  To:
2001 <sip:2001 at myddns.dydns.org:442>  Call-ID:
e6b918dc-71d58fe3 at 192.168.1.3  CSeq: 44436 REGISTER  Max-Forwards: 70
Authorization: Digest
username="2001",realm="myddns.dydns.org",nonce="476097b0-f9ea-11de-99fd-53ce5686ac9a",uri="sip:myddns.dydns.org:442",algorithm=MD5,response="466ea78ac2ccddca991a5c3d4d021bed",qop=auth,nc=00000001,cnonce="ebd80711"
 Contact: 2001 <sip:2001 at 92.xxx.xxx.xxx:5070;transport=tls>;expires=600
 User-Agent: Linksys/SPA3102-5.1.10(GW)  Content-Length: 0  Allow:
ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER  Supported:
x-sipura, replaces


The difference was a hard coded external IP address in the first
session that worked.  However I can't have it set like this as the IP
address is not static.

The second REGISTER in the Not Working session seems to be ignored by
FreeSwitch (I don't see it in the logs of FS either).   Is there
something in this Register that causes FS to ignore it?

Thanks
On Sun, Jan 3, 2010 at 10:19 AM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> I have a Linksys SPA3102, NAT'd on the internet (remotely) and
> connected to my FS on the otherside of the world, which is also
> natted. ?A PAP2T is connected on the same subnet as the FS. ?The 3102
> registers successfully and a call can be set up from the PAP2 to the
> 3102.
>
> However, after FS receives the Remote SDP the audio stops (ring tone
> stops in my case)
>
> The FS router is upnp enabled. ?The SPA3102 router is NOT upnp enabled
> (SPA3102 does not support upnp anyway I think).
>
>
>
> On Sun, Jan 3, 2010 at 3:03 AM, Brian West <brian at freeswitch.org> wrote:
>> Are you behind a nat-pmp/upnp router?
>> /b
>> On Jan 2, 2010, at 1:30 AM, Mark Campbell-Smith wrote:
>>
>> Hi!
>>
>> Both are auto-nat:
>> ???<param name="ext-rtp-ip" value="auto-nat"/>
>> ???<param name="ext-sip-ip" value="auto-nat"/>
>>
>> FreeSWITCH Version 1.0.trunk (15490)
>>
>> However, isn't it the IP address that is reported by the remote
>> SPA3102 that is incorrect? ?Or?
>>
>> On Thu, Dec 31, 2009 at 7:21 AM, Brian West <brian at freeswitch.org> wrote:
>>
>> show me the ext-rtp-ip and ext-sip-ip settings you're using along with SVN
>> rev please.
>>
>> /b
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>



From Prometheus001 at gmx.net  Tue Jan  5 04:40:29 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 05 Jan 2010 13:40:29 +0100
Subject: [Freeswitch-users] Force endpoint to use rfc2833 for dtmf
In-Reply-To: <FD8E5927-9DE2-41D0-B072-5051AF7C1907@jerris.com>
References: <4B30B01B.30809@gmx.net>
	<FD8E5927-9DE2-41D0-B072-5051AF7C1907@jerris.com>
Message-ID: <4B43333D.8020801@gmx.net>

Hello Michael,

I have opened a Jira for this.

Best rgerads
Peter


Michael Jerris schrieb:
> Not sure if we have an option to disable info.  Even without this,  
> dtmf should go across the bridge fine.  Please open up a bug on jira  
> about this
>
> Mike
>
> On Dec 22, 2009, at 6:40 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>
>   
>> Hello,
>>
>> in a bigger installation with some thousand endpoints in the field we
>> see, that the endpoint equipment is always using INFO messages  
>> (standard
>> setting is auto, so the endpoint decides which method to use). I  
>> have 2
>> questions to that scenario:
>>
>>   1. Is there a way that Freeswitch forces/restricts the endpoint to
>>      use rfc2833 or not to send to allow INFO in the invite message?
>>   2. Currently INFO messages do not get forwarded from the caller
>>      through freeswitch to called endpoint. How can we enable that FS
>>      is fowarding the INFO messages?
>>
>> Best regards
>> Peter
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From aep.lists at it46.se  Tue Jan  5 06:18:47 2010
From: aep.lists at it46.se (Alberto Escudero)
Date: Tue, 5 Jan 2010 15:18:47 +0100
Subject: [Freeswitch-users] encrypt voicemail emails and attachments?
In-Reply-To: <20100105042327.7CBF412DD@sinclaire.sibble.net>
References: <20100105042327.7CBF412DD@sinclaire.sibble.net>
Message-ID: <2f9531f86fc90f8d3f4cafab0cba4eae.squirrel@correo.nodo50.org>

One way to do it, it is to use procmail to handle the mails locally before
forwarding them to a final destination. You can take a similar approach
that Spam/Antivirus software and use procmailrc to add one more filter *in
your case gpg*.

/aep
-- 
Stopping junk mailers is good for the environment

> Looking throught the wiki, I see various configs for having FS email you a
> copy of received voicemail messages, has anyone done any work with having
> the
> voicemail messages gpg encrypted with public prior to sending?  Or is that
> something that should pretty much be handled at the mta level leaving FS
> out
> of the mix altogether? I'm thinking probably so, but before I try to do
> this,
> I figured i'd ask first.
>
>
> --
> Harondel J. Sibble
> Sibble Computer Consulting
> Creating Solutions for the small and medium business computer user.
> help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
> (604) 739-3709 (voice)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>





From brian at freeswitch.org  Tue Jan  5 07:15:15 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Jan 2010 09:15:15 -0600
Subject: [Freeswitch-users] No audio after Remote SDP:
In-Reply-To: <33c87fa31001050307v3cbcae01q9d7d47383a63ad33@mail.gmail.com>
References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com>
	<33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com>
	<772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org>
	<33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com>
	<191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com>
	<33c87fa30912301213g3d24b773k86b95e7ce08f6761@mail.gmail.com>
	<7D75E6D6-8AB9-429E-A5A9-1639C0D5AD09@freeswitch.org>
	<33c87fa31001012330g69fc8e88m61e648d893c4d8f7@mail.gmail.com>
	<180A53B1-EFB6-4B17-AA7C-19B007ADE04C@freeswitch.org>
	<33c87fa31001021519j41ead649rb4a85e6be6236254@mail.gmail.com>
	<33c87fa31001050307v3cbcae01q9d7d47383a63ad33@mail.gmail.com>
Message-ID: <117A38AB-C1AE-4AF0-AD36-4165FAA94816@freeswitch.org>

This is why you set up stun correctly on the SPA.

/b

On Jan 5, 2010, at 5:07 AM, Mark Campbell-Smith wrote:

> The difference was a hard coded external IP address in the first
> session that worked.  However I can't have it set like this as the IP
> address is not static.




From david.varnes at gmail.com  Tue Jan  5 05:41:42 2010
From: david.varnes at gmail.com (david varnes)
Date: Wed, 6 Jan 2010 00:41:42 +1100
Subject: [Freeswitch-users] [contrib] a new java ESL inbound/outbound client
Message-ID: <74a861001001050541v4e02f487xd21fe8fc13e8ed0a@mail.gmail.com>

Hi all,
I had a basic java inbound ESL client kicking around that I have used
in a couple of small projects over the last year.  I needed something
a little more complete for a new project so I dusted it off and made it
less incomplete.

It still needs more work and testing, but it certainly is usable right now.
I have tested it against FS 1.0.4 and latest trunk.

I have put it in my contrib area in svn in hopes that some may find it
useful:
http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/dvarnes/java/esl-client
I would be interested in any feedback ...

Features
 * Apache License (ASL) version 2
 * Standalone Inbound client
 * Framework classes to easily create an Outbound socket client
 * based on Netty [1] nio library version 3.1.5.GA (previously was
   using Apache MINA, but this is easier)
 * logging via slf4j
 * only dependencies are slf4j-api and netty (both Apache licensed)
 * single jar which is a valid OSGi bundle
 * built using maven
 * eclipse projects
 * reasonable level of java docs

Still todo
 * Docs
 * Simple example apps
 * .. more in TODO.txt in project root.

There is no binary jar available right now since I don't know how/if
I can put files up to file.freeswitch.org.

In the meantime to build you need maven [2] installed.  If you are
unfamiliar with maven usage, I can post a simple howto.

davidv

[1] http://www.jboss.org/netty/downloads.html
[2] http://maven.apache.org



From brian at freeswitch.org  Tue Jan  5 07:53:52 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Jan 2010 09:53:52 -0600
Subject: [Freeswitch-users] Unable to start mod_java
In-Reply-To: <c55d46951001041934n49bfcbf4h42564e4548aa301a@mail.gmail.com>
References: <c55d46951001041934n49bfcbf4h42564e4548aa301a@mail.gmail.com>
Message-ID: <BC685614-4CE4-46AB-9263-DB5F7FC815A3@freeswitch.org>

Are you on a 64bit platform?  If so then you're jdk is wrong.

/b

On Jan 4, 2010, at 9:34 PM, Max Bridgewater wrote:

> Hi,
> 
> I built Freeswitch with mod_java enabled. But now, when Freeswitch starts, I get the following error message:
> 
> 
> 2010-01-04 22:32:46.574770 [NOTICE] switch_loadable_module.c:271 Adding API Function 'jsrun'
> 2010-01-04 22:32:46.574811 [NOTICE] switch_loadable_module.c:271 Adding API Function 'jsapi'
> 2010-01-04 22:32:46.575306 [NOTICE] modjava.c:244 Java Framework Loading...
> 2010-01-04 22:32:46.575721 [ERR] modjava.c:133 Error loading /usr/local/java/jdk1.6.0_17/jre/lib/i386/client/libjvm.so
> 2010-01-04 22:32:46.575743 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_java.so
> **Module load routine returned an error**
> 2010-01-04 22:32:46.576742 [CONSOLE] switch_loadable_module.c:890 Successfully Loaded [mod_lua]
> 2010-01-04 22:32:46.576748 [NOTICE] switch_loadable_module.c:209 Adding Dialplan 'LUA'
> 2010-01-04 22:32:46.576795 [NOTICE] switch_loadable_module.c:249 Adding Application 'lua'
> 
> 
> As far as I can tell the /usr/local/java/jdk1.6.0_17/jre/lib/i386/client/libjvm.so is the correct path.
> Any idea?
> 
> max.




From anthony.minessale at gmail.com  Tue Jan  5 08:11:05 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Jan 2010 10:11:05 -0600
Subject: [Freeswitch-users] Zap channel not released when voicemail
	starts
In-Reply-To: <cd93b3701001050144m648905e0i76a0f59fa3b01db1@mail.gmail.com>
References: <ea085d8b43f8105108d9c1063fb56534@thom.fr.eu.org>
	<191c3a031001041600y1c98465cifcc522056096f746@mail.gmail.com>
	<cd93b3701001050144m648905e0i76a0f59fa3b01db1@mail.gmail.com>
Message-ID: <191c3a031001050811w5c33d5f6na52bcedc039ab188@mail.gmail.com>

one way is to run tone_detect on the busy signal and map it to the hangup
app


On Tue, Jan 5, 2010 at 3:44 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:

> I'll second that.
>
> My way of dealing with it has been to write a little script to detect
> hangups on the TDM end, then force release the corresponding "B-leg" that is
> hooked up to VM. In the process of converting this to an FS module.
>
> Not clean .. but works. Would have liked to see the same code within FS
> core and, if appropriate, the VM subsystem to achieve the same end.
>
> Regards,
>
> Ahmed.
>
>
> 2010/1/5 Anthony Minessale <anthony.minessale at gmail.com>
>
> hangup detection on TDM is a bitch.
>>
>>
>> On Mon, Jan 4, 2010 at 10:07 AM, Fran?ois Legal <devel at thom.fr.eu.org>wrote:
>>
>>> Hello,
>>>
>>>
>>>
>>> I have an issue with voicemail and openzap channels.
>>>
>>> When an incoming call on an openzap channel is bridged to voicemail, if
>>> that channel is hung up before the beginning of voicemail recording, that
>>> channel is kept open open until 3 or 4 seconds after the voicemail started
>>> to record the message.
>>>
>>> What should I do to make freeswitch/voicemail release the channel
>>> immediately when the caller hang up ?
>>>
>>>
>>>
>>> Thanks in advance
>>>
>>>
>>>
>>> Fran?ois
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From devel at thom.fr.eu.org  Tue Jan  5 08:11:27 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Tue, 05 Jan 2010 17:11:27 +0100
Subject: [Freeswitch-users] How to konw who picks up in group bridge
Message-ID: <06bbe9d9d06077a04e9245c84d4cb013@thom.fr.eu.org>



Hello, 

In my diaplan, when a call arrives on some specific channel, it
is routed to an extension that tries to bridge it on multiple channels
using coma separated list (either openzap, sofia or both). 

I would like
to see in my CDR which channel did pick up which channel did pick up the
call. Which variable can I use ? 

Thanks 

Fran?ois
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From help at pdscc.com  Tue Jan  5 09:11:37 2010
From: help at pdscc.com (Harondel J. Sibble)
Date: Tue, 05 Jan 2010 09:11:37 -0800
Subject: [Freeswitch-users] encrypt voicemail emails and attachments?
In-Reply-To: <03DDC308-7EC5-46EC-8AFB-CC73417E20F5@jerris.com>
References: <20100105042327.7CBF412DD@sinclaire.sibble.net>,
	<03DDC308-7EC5-46EC-8AFB-CC73417E20F5@jerris.com>
Message-ID: <20100105171137.A433E1DB501@sinclaire.sibble.net>

That's what I suspected, thanks!

On 4 Jan 2010 at 23:51, Michael Jerris wrote:

> you can just make a shell script (or perl or whatever) that is called as the
> mailer that does this.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From ken at ukgb.net  Tue Jan  5 10:06:52 2010
From: ken at ukgb.net (Ken Gillett)
Date: Tue, 5 Jan 2010 18:06:52 +0000
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <5BED3371-0023-4D53-BBE1-37D241B4AAFB@jerris.com>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com>
	<B6763940-A3CB-4006-8DF1-ADEDD9C68294@ukgb.net>
	<0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org>
	<CA53F057-4A51-457C-A673-B0B6E4DF482A@ukgb.net>
	<207e7a5e0912301103h17133fdfk22430bbce5a1718b@mail.gmail.com>
	<090E289D-76EE-4F4A-82DF-513626A915E6@ukgb.net>
	<79B790D4-DE06-47FF-B9FD-52D9E1EB8BCB@ukgb.net>
	<207e7a5e1001020840y1bd3bae2y2b683a19659c9d7d@mail.gmail.com>
	<6E6E6D36-6E93-49BF-9516-FC028E73D279@ukgb.net>
	<5BED3371-0023-4D53-BBE1-37D241B4AAFB@jerris.com>
Message-ID: <A301DB14-66F4-4154-B9F1-17704B51BD75@ukgb.net>

Thanks for that. Now I know how to proceed.


On 4 Jan 2010, at 16:01, Michael Jerris wrote:

> its all in there.
> 
> Mike
> 
> On Jan 4, 2010, at 4:14 AM, Ken Gillett wrote:
> 
>> I will look into this, but in the meantime, would it not be possible to simply copy the installed files? I realise that basically this must be possible, but practically it depends on what is installed where. I believe the default prefix is /usr/local/freeswitch, but is EVERYTHING in there or is other stuff scattered in other directories?
>> 
>> If it's all together I'm thinking I could simply tar the folder and copy the tarball. Wouldn't this be possible?
>> 
> 



Ken G i l l e t t

_/_/_/_/_/_/_/_/






From jerry.richards at teotech.com  Tue Jan  5 10:24:18 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 5 Jan 2010 10:24:18 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <cd93b3701001050140q14182667o8e90dba27fa3bc85@mail.gmail.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<cd93b3701001050140q14182667o8e90dba27fa3bc85@mail.gmail.com>
Message-ID: <8FBE2BC8AF8C486B8E569C08D4941064@greyhawk.tonecommander.com>

Hi Ahmed,
 
My vars.xml file does not set the literal IP address (nor the server's DNS
name), rather it uses the following line (so this should not cause a
problem):
 
<X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
 
Also, there are no references to hard-coded addresses anywhere in the other
XML files.  Also, line #3 of the pastebin "IP 192.168.72.29 Rejected by acl
"domains" Falling back to Digest auth." just means it is authenticating the
INVITE and then it does proceed with the call.
 
The problem is down at line #71: "zap_io.c:1197 outgoing_call method not
implemented!".  What does this error mean?
 
Thank you and Best Regards,
Jerry
 

  _____  

From: Ahmed Naji [mailto:a.alalousi at gmail.com] 
Sent: Tuesday, January 05, 2010 1:40 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER


Hi Jerry,

Looked at your log and your problem is quiet simple to resolve, but first
here's what's happenning:

You copied the conf/ subtree to your new server. As such, you have also
duplicated your vars.xml. By doing so, you have set the domain on the new
server to the same domain used on your old server which would be fine, but
the default domain settings used by FS is to use your primary IPv4 IP
address as your domain.

By duplicating the conf subtree from the old server, you have effectively
bound the new instance of FS to a domain that is the IP address of the old
server, if this makes sense.

You can see this on third line of your log:

192.168.72.29 Rejected by acl "domains"

To resolve this, modify your vars.xml on the new server to reflect whatever
domain it is you want to route, or set the domain to the new server's IP
address like so:

 <X-PRE-PROCESS cmd="set" data="domain=IP_ADDRESS_OF_NEW_SERVER"/>

You also need to check that any other files (e.g. the conf/sip_profiles,
conf/directory/ and conf/dialplan/ hierarchy) are modified to reflect the
new server settings as well.

In the limit, resolving those conflicts will also resolve your issues,
unless there is something else that's wrong.

Let's know how you get along.

Regards,

Ahmed.
 


2010/1/4 Jerry Richards <jerry.richards at teotech.com>



Hello,

I have one FS instance that is working well with a PRI and running FS
version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
upgraded it.

Now, I am trying to bring up another FS instance (basically a clone of the
first), but the PRI does not work.  When I attempt to make an
internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
both servers are using the latest Sangoma Wanpipe driver, and I copied the
conf XML file tree from the old server to the new one.  I think the problem
has to do with the openzap module, but I'm having difficulty isolating the
problem.  Could it have built the openzap module incorrectly?  Another
difference is that I installed 1.0.5pre9 from scratch on the new server
(i.e. it never had 1.0.4 running on it).

I put the FS log into the pastebin when an outbound call attempt is made:

http://pastebin.freeswitch.org/11675

Could someone give me a pointer on what to try next?

Thanks and Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
Ahmed Naji


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From tculjaga at gmail.com  Tue Jan  5 11:25:13 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 5 Jan 2010 20:25:13 +0100
Subject: [Freeswitch-users] freeswitch and H323
In-Reply-To: <c96b95651001031636q1ecfec18x2bba1c22ff6ca9bb@mail.gmail.com>
References: <7aa8bd9d0912300514r68c90b12u7c631a649981cfa3@mail.gmail.com>
	<87f2f3b90912310920q1821fe0eu5fd74e553e4cb12@mail.gmail.com>
	<65d96fc81001010452k37068e87sc0a229cdfe481c40@mail.gmail.com>
	<c96b95651001031636q1ecfec18x2bba1c22ff6ca9bb@mail.gmail.com>
Message-ID: <65d96fc81001051125u3a6c7d96q5ed39d39c0026107@mail.gmail.com>

its already there: http://wiki.freeswitch.org/wiki/Mod_h323

T.

On Mon, Jan 4, 2010 at 1:36 AM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:

> HI,
>
> It would be really nice if you can create a wiki page.
>
> Thanks
>
>
> On Fri, Jan 1, 2010 at 1:52 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> well, mod_h323 works for me... there are still some missing things and of
>> course bugs ... e.g. incorrect releaseCause mapping, no automatic codec
>> ptime sync... but it is usable ....
>>
>>
>> if you'd like to go mod_h323 way i can help you... it builds as a charm
>> for me...
>>
>>
>> T.
>>
>>
>>
>>
>>
>> On Thu, Dec 31, 2009 at 6:20 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Are you trying to use mod_h323 or mod_opal? They are both works in
>>> progress, but the latter is farther along than the former. Use the latest
>>> FreeSWITCH trunk (or latest.freeswitch.org) and run the buildopal.sh
>>> script in the build directory. If you have any build issues then paste the
>>> log on pastebin.freeswitch.org and reply to this thread with the PB URL
>>> so that we can take a look.
>>>  -MC
>>>
>>>
>>> On Wed, Dec 30, 2009 at 5:14 AM, Pete Kay <petedao at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> has anyone been able to get H323 to work?
>>>>
>>>> I have problem trying to get it compiled with either 1.0.4 or 1.0.5.
>>>>
>>>> Thanks,
>>>> pete
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Jan  5 13:02:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jan 2010 13:02:37 -0800
Subject: [Freeswitch-users] XML IVR and Javascript menu-exec-app
In-Reply-To: <7d4c9553f55e0af32200eebdd332b2f9.squirrel@correo.nodo50.org>
References: <7d4c9553f55e0af32200eebdd332b2f9.squirrel@correo.nodo50.org>
Message-ID: <87f2f3b91001051302n70d4462nc667e2c14fded790@mail.gmail.com>

On Fri, Jan 1, 2010 at 10:03 AM, Alberto Escudero <aep.lists at it46.se> wrote:

> Hi,
>
> I am writing several IVRs using Freeswitch XML
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr
>
> One of the nodes of the IVR is a Javascript application that records a
> message.
> e.g.:
> <entry action="menu-exec-app" digits="3" param="javascript foo.js"/>
>
> The Javascript application starts by issuing a
> session.answer()
>
> [records the voice message]
>
> exit();
>
> Once the Javascript exits, the channel is dropped and hence the IVR
> terminates.
> Is it possible to write a Javascript application that once is completed,
> the channel returns back to the top menu of the IVR? I want to emulate the
> same behavior that "menu-play-sound", that once the file is played, the
> IVR logic returns to the top menu.
>
> Just transfer the call to an extension in the dialplan that in turn sends
the call to the IVR in question...
-MC
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From msc at freeswitch.org  Tue Jan  5 13:11:01 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jan 2010 13:11:01 -0800
Subject: [Freeswitch-users] Polycom & TLS - help please...
In-Reply-To: <d361ca7b1001050038x4e7103bdke322fd8741403172@mail.gmail.com>
References: <d361ca7b1001030757m2f55a400n3e2b3b74185c4202@mail.gmail.com>
	<BFB88CA0-8319-4D05-B3CD-39E3450DE157@jerris.com>
	<D950E561-1B94-48E0-9221-3739BAB8F91A@freeswitch.org>
	<d361ca7b1001031932m7695150fv967c0df683083af9@mail.gmail.com>
	<3968BF39-7A15-4F9F-8FA2-A78BC4F0F38E@freeswitch.org>
	<d361ca7b1001050038x4e7103bdke322fd8741403172@mail.gmail.com>
Message-ID: <87f2f3b91001051311o3f348035x86f6383e84680291@mail.gmail.com>

On Tue, Jan 5, 2010 at 12:38 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Thanks, I have a partial success which involved two steps:
>
>    - The wiki says to create a root certifcate with *gentls_cert setup *with
>    no other parameters; I had to add my domain's data to this command. The new
>    certificate has been downloaded to the phone.
>    - Replaced the registrar definitions in the phone's config files from
>    IP address to the server's name.
>    - The above setup worked as-is. To be sure I've added the NAPTR records
>    to the DNS after the above two steps worked.
>       - BTW, the wiki says that the NAPTR records are not mandatory, thus
>       I did not add them at the first place.
>
> Did you add your specific information to the wiki? If not please do so.
Thanks!
-MC
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From anthony.minessale at gmail.com  Tue Jan  5 13:18:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Jan 2010 15:18:00 -0600
Subject: [Freeswitch-users] XML IVR and Javascript menu-exec-app
In-Reply-To: <87f2f3b91001051302n70d4462nc667e2c14fded790@mail.gmail.com>
References: <7d4c9553f55e0af32200eebdd332b2f9.squirrel@correo.nodo50.org>
	<87f2f3b91001051302n70d4462nc667e2c14fded790@mail.gmail.com>
Message-ID: <191c3a031001051318q79737971l39c6acbbc3bab818@mail.gmail.com>

or call:
 session.setAutoHangup(0);

so exiting the script will not hangup the channel

if it still happens get the log line "its blue" that shows where the call is
being hungup from"



On Tue, Jan 5, 2010 at 3:02 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Jan 1, 2010 at 10:03 AM, Alberto Escudero <aep.lists at it46.se>wrote:
>
>> Hi,
>>
>> I am writing several IVRs using Freeswitch XML
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr
>>
>> One of the nodes of the IVR is a Javascript application that records a
>> message.
>> e.g.:
>> <entry action="menu-exec-app" digits="3" param="javascript foo.js"/>
>>
>> The Javascript application starts by issuing a
>> session.answer()
>>
>> [records the voice message]
>>
>> exit();
>>
>> Once the Javascript exits, the channel is dropped and hence the IVR
>> terminates.
>> Is it possible to write a Javascript application that once is completed,
>> the channel returns back to the top menu of the IVR? I want to emulate the
>> same behavior that "menu-play-sound", that once the file is played, the
>> IVR logic returns to the top menu.
>>
>> Just transfer the call to an extension in the dialplan that in turn sends
> the call to the IVR in question...
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From a.alalousi at gmail.com  Tue Jan  5 13:58:41 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Tue, 5 Jan 2010 21:58:41 +0000
Subject: [Freeswitch-users] Zap channel not released when voicemail
	starts
In-Reply-To: <191c3a031001050811w5c33d5f6na52bcedc039ab188@mail.gmail.com>
References: <ea085d8b43f8105108d9c1063fb56534@thom.fr.eu.org>
	<191c3a031001041600y1c98465cifcc522056096f746@mail.gmail.com>
	<cd93b3701001050144m648905e0i76a0f59fa3b01db1@mail.gmail.com>
	<191c3a031001050811w5c33d5f6na52bcedc039ab188@mail.gmail.com>
Message-ID: <cd93b3701001051358h51931809o56a2ce3cfb8f7d2e@mail.gmail.com>

Good method, though this isn't this assuming too much, in the sense that we
are assuming a hangup with cause 17 or cause 16 with a forced busy tone ?

2010/1/5 Anthony Minessale <anthony.minessale at gmail.com>

> one way is to run tone_detect on the busy signal and map it to the hangup
> app
>
>
>
> On Tue, Jan 5, 2010 at 3:44 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>
>> I'll second that.
>>
>> My way of dealing with it has been to write a little script to detect
>> hangups on the TDM end, then force release the corresponding "B-leg" that is
>> hooked up to VM. In the process of converting this to an FS module.
>>
>> Not clean .. but works. Would have liked to see the same code within FS
>> core and, if appropriate, the VM subsystem to achieve the same end.
>>
>> Regards,
>>
>> Ahmed.
>>
>>
>> 2010/1/5 Anthony Minessale <anthony.minessale at gmail.com>
>>
>> hangup detection on TDM is a bitch.
>>>
>>>
>>> On Mon, Jan 4, 2010 at 10:07 AM, Fran?ois Legal <devel at thom.fr.eu.org>wrote:
>>>
>>>> Hello,
>>>>
>>>>
>>>>
>>>> I have an issue with voicemail and openzap channels.
>>>>
>>>> When an incoming call on an openzap channel is bridged to voicemail, if
>>>> that channel is hung up before the beginning of voicemail recording, that
>>>> channel is kept open open until 3 or 4 seconds after the voicemail started
>>>> to record the message.
>>>>
>>>> What should I do to make freeswitch/voicemail release the channel
>>>> immediately when the caller hang up ?
>>>>
>>>>
>>>>
>>>> Thanks in advance
>>>>
>>>>
>>>>
>>>> Fran?ois
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Ahmed Naji
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From anthony.minessale at gmail.com  Tue Jan  5 14:06:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Jan 2010 16:06:03 -0600
Subject: [Freeswitch-users] Zap channel not released when voicemail
	starts
In-Reply-To: <cd93b3701001051358h51931809o56a2ce3cfb8f7d2e@mail.gmail.com>
References: <ea085d8b43f8105108d9c1063fb56534@thom.fr.eu.org>
	<191c3a031001041600y1c98465cifcc522056096f746@mail.gmail.com>
	<cd93b3701001050144m648905e0i76a0f59fa3b01db1@mail.gmail.com>
	<191c3a031001050811w5c33d5f6na52bcedc039ab188@mail.gmail.com>
	<cd93b3701001051358h51931809o56a2ce3cfb8f7d2e@mail.gmail.com>
Message-ID: <191c3a031001051406o58ff93adk2654b475de971743@mail.gmail.com>

depends,
some telco actually expect you to react to this tone for hangup and charge
you for real hangup signaling which is probably the case here.


On Tue, Jan 5, 2010 at 3:58 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:

> Good method, though this isn't this assuming too much, in the sense that we
> are assuming a hangup with cause 17 or cause 16 with a forced busy tone ?
>
>
> 2010/1/5 Anthony Minessale <anthony.minessale at gmail.com>
>
>> one way is to run tone_detect on the busy signal and map it to the hangup
>> app
>>
>>
>>
>> On Tue, Jan 5, 2010 at 3:44 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>>
>>> I'll second that.
>>>
>>> My way of dealing with it has been to write a little script to detect
>>> hangups on the TDM end, then force release the corresponding "B-leg" that is
>>> hooked up to VM. In the process of converting this to an FS module.
>>>
>>> Not clean .. but works. Would have liked to see the same code within FS
>>> core and, if appropriate, the VM subsystem to achieve the same end.
>>>
>>> Regards,
>>>
>>> Ahmed.
>>>
>>>
>>> 2010/1/5 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>> hangup detection on TDM is a bitch.
>>>>
>>>>
>>>> On Mon, Jan 4, 2010 at 10:07 AM, Fran?ois Legal <devel at thom.fr.eu.org>wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>>
>>>>>
>>>>> I have an issue with voicemail and openzap channels.
>>>>>
>>>>> When an incoming call on an openzap channel is bridged to voicemail, if
>>>>> that channel is hung up before the beginning of voicemail recording, that
>>>>> channel is kept open open until 3 or 4 seconds after the voicemail started
>>>>> to record the message.
>>>>>
>>>>> What should I do to make freeswitch/voicemail release the channel
>>>>> immediately when the caller hang up ?
>>>>>
>>>>>
>>>>>
>>>>> Thanks in advance
>>>>>
>>>>>
>>>>>
>>>>> Fran?ois
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Ahmed Naji
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From djbinter at yahoo.com  Tue Jan  5 14:42:58 2010
From: djbinter at yahoo.com (DJB)
Date: Tue, 5 Jan 2010 14:42:58 -0800 (PST)
Subject: [Freeswitch-users] Min-SE Header
Message-ID: <77858.66861.qm@web37507.mail.mud.yahoo.com>

I am wondering whether it is possible to suppress the Min-SE Header Field in SIP INVITE message.

Thank you,
Dorn B.


      
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From nik.middleton at noblesolutions.co.uk  Tue Jan  5 15:02:15 2010
From: nik.middleton at noblesolutions.co.uk (Nik Middleton)
Date: Tue, 5 Jan 2010 23:02:15 -0000
Subject: [Freeswitch-users] Call limits (time)
Message-ID: <A2B4F48489739E439D8845296A2FD9C4147E1F@sun.Noble.local>

Hi Guys,

 

I'm looking to migrate my billing platform to use FS.  So far so good,
however, if a user is low on credit I need to limit the call length.  In
other words, from the rate card, prior to connecting the call, I know
they are going to call a mobile at say $0.10/min, but they only have $2
of credit, so I want to terminate the call after 20 mins, preferably
with a message to the originator saying they only have X mins left.  Is
there a way of achieving this with the originate command?

 

Regards

 

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From sos at sokhapkin.dyndns.org  Tue Jan  5 15:14:13 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 5 Jan 2010 18:14:13 -0500
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C4147E1F@sun.Noble.local>
References: <A2B4F48489739E439D8845296A2FD9C4147E1F@sun.Noble.local>
Message-ID: <201001051814.13887.sos@sokhapkin.dyndns.org>

See sched_hangup dialplan application.

On Tuesday 05 January 2010, Nik Middleton wrote:
> Hi Guys,
>
>
>
> I'm looking to migrate my billing platform to use FS.  So far so good,
> however, if a user is low on credit I need to limit the call length.  In
> other words, from the rate card, prior to connecting the call, I know
> they are going to call a mobile at say $0.10/min, but they only have $2
> of credit, so I want to terminate the call after 20 mins, preferably
> with a message to the originator saying they only have X mins left.  Is
> there a way of achieving this with the originate command?
>
>
>
> Regards





From ron.freeswitch at mcleodnet.com  Tue Jan  5 15:25:26 2010
From: ron.freeswitch at mcleodnet.com (Ron McLeod)
Date: Tue, 5 Jan 2010 15:25:26 -0800
Subject: [Freeswitch-users] unable to call out troug siemens hie9200 in pur
	SIP
Message-ID: <B1789BC477C54AF9AD96A33F051FBAB6@fromage>

Posted on behalf of Tayeb Meftah...


hi dear friends,
we have a siemens hie9200 softswitch
we want to interconnect freeswitch with it
to use it for service, like media, voicemail, audio conferencing and ...
call trace is atached belo
if we calls from tdm, call is passed but without rtp
i routed a number like 021000000 is a tdm number to 3001 the default
8khz fs conference
so inbound call is passed but without audio
for outbound calls, i wanted to call a tdm number, like 021298235
the call tack some long time but return a 500 internal server error
from the hie9200 softswitch:
sip.Reason == "Q.850 ;cause=47 ;text=\"Resource unavailable, unspecified\""

from fs i get normal temporary failur
so please see the trace and return for me any reply ;)
thanks for your helps and times.


Traces at:
http://siplabs.net/tracebin/fs-siemens-500.pcap
http://siplabs.net/tracebin/fs-siemens-rtp.pcap





From nik.middleton at noblesolutions.co.uk  Tue Jan  5 15:43:26 2010
From: nik.middleton at noblesolutions.co.uk (Nik Middleton)
Date: Tue, 5 Jan 2010 23:43:26 -0000
Subject: [Freeswitch-users] Migrating from asterisk to FS
Message-ID: <A2B4F48489739E439D8845296A2FD9C4147E22@sun.Noble.local>

HI Guys

 

While I've been using FS for around 18 months now, and love it to bits,
it's been a specific solution.  I'm now looking to move my customer base
across, and have on the base of it some basic and perhaps dumb
questions.

 

I currently have around 150 Sip phones attached to my systems   These
are all geographically spread, so re-configuring them is out of the
question.  They all register on port 5060.  Given that FS uses port 5080
for external clients, do I simply need to do a Port translate on my
firewall or is there a simpler solution?

 

Further, how does FS handle a call FWD?  In other words, if a SIP phone
has a divert on busy set will it account for the redirect?   Currently
in Asterisk I use the 'I' option to disable this as I can't account for
the call.

 

Regards,

 

 

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From msc at freeswitch.org  Tue Jan  5 16:33:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jan 2010 16:33:26 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>

On Mon, Jan 4, 2010 at 3:49 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> Hello,
>
> I have one FS instance that is working well with a PRI and running FS
> version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
> upgraded it.
>
> Now, I am trying to bring up another FS instance (basically a clone of the
> first), but the PRI does not work.  When I attempt to make an
> internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
> both servers are using the latest Sangoma Wanpipe driver, and I copied the
> conf XML file tree from the old server to the new one.  I think the problem
> has to do with the openzap module, but I'm having difficulty isolating the
> problem.  Could it have built the openzap module incorrectly?  Another
> difference is that I installed 1.0.5pre9 from scratch on the new server
> (i.e. it never had 1.0.4 running on it).
>
> I put the FS log into the pastebin when an outbound call attempt is made:
>
> http://pastebin.freeswitch.org/11675
>
> Could someone give me a pointer on what to try next?
>

Jerry, I noticed this line:
(OpenZAP/1:1/3491028 at g1)

Is your carrier wanting full ten digit phone numbers? Try adding the area
code on this and see what happens. The error usually would be something like
"invalid number format" but I've seen carriers do stupid things like this.
Try that first and see if it makes a difference. If not you'll need to turn
on Q931 debugging as per the Sangoma wiki. (See
http://wiki.freeswitch.org/wiki/OpenZAP#Debugging_PRI_With_wanpipemon for
the link.)

-MC
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From msc at freeswitch.org  Tue Jan  5 16:37:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jan 2010 16:37:14 -0800
Subject: [Freeswitch-users] Zap channel not released when voicemail
	starts
In-Reply-To: <191c3a031001051406o58ff93adk2654b475de971743@mail.gmail.com>
References: <ea085d8b43f8105108d9c1063fb56534@thom.fr.eu.org>
	<191c3a031001041600y1c98465cifcc522056096f746@mail.gmail.com>
	<cd93b3701001050144m648905e0i76a0f59fa3b01db1@mail.gmail.com>
	<191c3a031001050811w5c33d5f6na52bcedc039ab188@mail.gmail.com>
	<cd93b3701001051358h51931809o56a2ce3cfb8f7d2e@mail.gmail.com>
	<191c3a031001051406o58ff93adk2654b475de971743@mail.gmail.com>
Message-ID: <87f2f3b91001051637p43ad3350l7bbf3074212be845@mail.gmail.com>

On Tue, Jan 5, 2010 at 2:06 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> depends,
> some telco actually expect you to react to this tone for hangup and charge
> you for real hangup signaling which is probably the case here.


Example: CenturyLink charges $5 per line per month for "disconnect after
hangup." No joke. It's an actual "feature" that they "sell" to the people
using their analog lines. I'm sure other scumbag telcos do the same thing.

-MC
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From msc at freeswitch.org  Tue Jan  5 16:46:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jan 2010 16:46:07 -0800
Subject: [Freeswitch-users] Migrating from asterisk to FS
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C4147E22@sun.Noble.local>
References: <A2B4F48489739E439D8845296A2FD9C4147E22@sun.Noble.local>
Message-ID: <87f2f3b91001051646h481e598fp3cfe2a5c605c5de0@mail.gmail.com>

On Tue, Jan 5, 2010 at 3:43 PM, Nik Middleton <
nik.middleton at noblesolutions.co.uk> wrote:

>  HI Guys
>
>
>
> While I?ve been using FS for around 18 months now, and love it to bits,
> it?s been a specific solution.  I?m now looking to move my customer base
> across, and have on the base of it some basic and perhaps dumb questions.
>
>
>
> I currently have around 150 Sip phones attached to my systems   These are
> all geographically spread, so re-configuring them is out of the question.
> They all register on port 5060.  Given that FS uses port 5080 for external
> clients, do I simply need to do a Port translate on my firewall or is there
> a simpler solution?
>
>
>
FreeSWITCH doesn't *force* you to use port 5080 for inbound registrations.
You can, but it's not a requirement. Personally I just use the internal SIP
profile for those external phones wherever possible. The real issue is
whether or not you have a horrible NAT device in between FS and the Internet
connection. In my experience, if you have a decent NAT device that supports
UPnP (like the WRT54GL running Tomato firmware) then connecting external
phones to a FS box behind NAT using port 5060 just works. Give it a try and
let us know how it works.

> Further, how does FS handle a call FWD?  In other words, if a SIP phone has
> a divert on busy set will it account for the redirect?   Currently in
> Asterisk I use the ?I? option to disable this as I can?t account for the
> call.
>
When calling to a FS box or when FS calls another server?
-MC
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From rupa at rupa.com  Tue Jan  5 18:13:25 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 5 Jan 2010 20:13:25 -0600
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C4147E1F@sun.Noble.local>
References: <AcqOWx9O9EDYsJ/ZQJSSOHSvWxhecA==>
	<A2B4F48489739E439D8845296A2FD9C4147E1F@sun.Noble.local>
Message-ID: <b39273531001051813v5271858fpd133326559238b5e@mail.gmail.com>

Look at using mod_nibblebill

On Tue, Jan 5, 2010 at 5:02 PM, Nik Middleton <
nik.middleton at noblesolutions.co.uk> wrote:

>  Hi Guys,
>
>
>
> I?m looking to migrate my billing platform to use FS.  So far so good,
> however, if a user is low on credit I need to limit the call length.  In
> other words, from the rate card, prior to connecting the call, I know they
> are going to call a mobile at say $0.10/min, but they only have $2 of
> credit, so I want to terminate the call after 20 mins, preferably with a
> message to the originator saying they only have X mins left.  Is there a way
> of achieving this with the originate command?
>
>
>
> Regards
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From sos at sokhapkin.dyndns.org  Tue Jan  5 18:25:06 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 5 Jan 2010 21:25:06 -0500
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <b39273531001051813v5271858fpd133326559238b5e@mail.gmail.com>
References: <AcqOWx9O9EDYsJ/ZQJSSOHSvWxhecA==>
	<A2B4F48489739E439D8845296A2FD9C4147E1F@sun.Noble.local>
	<b39273531001051813v5271858fpd133326559238b5e@mail.gmail.com>
Message-ID: <201001052125.06909.sos@sokhapkin.dyndns.org>

Unfortunalely, mod_nibblebill doesn't take billing increments into account.

On Tuesday 05 January 2010, Rupa Schomaker wrote:
> Look at using mod_nibblebill
>
> On Tue, Jan 5, 2010 at 5:02 PM, Nik Middleton <
>
> nik.middleton at noblesolutions.co.uk> wrote:
> >  Hi Guys,
> >
> >
> >
> > I?m looking to migrate my billing platform to use FS.  So far so good,
> > however, if a user is low on credit I need to limit the call length.  In
> > other words, from the rate card, prior to connecting the call, I know
> > they are going to call a mobile at say $0.10/min, but they only have $2
> > of credit, so I want to terminate the call after 20 mins, preferably with
> > a message to the originator saying they only have X mins left.  Is there
> > a way of achieving this with the originate command?
> >
> >
> >
> > Regards
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From mcampbellsmith at gmail.com  Tue Jan  5 19:36:25 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 6 Jan 2010 14:36:25 +1100
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
Message-ID: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>

Hi!

If I try to call out on one of my voip providers I get
INCOMPATIBLE_DESTINATION.  Something is going wrong with codec
negotiation:

2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP:
v=0
o=Sippy 257534956 1 IN IP4 80.232.37.178
s=-
t=0 0
m=audio 47904 RTP/AVP 2 101 13
c=IN IP4 213.50.90.3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [G721:2:8000:20]/[G726-32:2:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [G721:2:8000:20]/[G726-32:2:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [G721:2:8000:20]/[PCMU:0:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [G721:2:8000:20]/[PCMA:8:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3262 Set 2833 dtmf
payload to 101
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [telephone-event:101:8000:20]/[G726-32:2:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [telephone-event:101:8000:20]/[G726-32:2:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [telephone-event:101:8000:20]/[PCMU:0:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [CN:13:8000:20]/[G726-32:2:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [CN:13:8000:20]/[G726-32:2:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [CN:13:8000:20]/[PCMU:0:8000:20]
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [CN:13:8000:20]/[PCMA:8:8000:20]
2010-01-05 18:57:56.845029 [NOTICE] sofia.c:3937 Hangup
sofia/external/020555500 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]

According to their website, the codec they us is G711 a-law or G729.
What codecs have name CN and telephone-event?

How do I get these to match?  I assume the format of the debug output
is: Audio Codec compare [Codec Name:Media Format:Media Name:Rate:RTP
Size?]

Thanks



From brian at freeswitch.org  Tue Jan  5 19:45:20 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Jan 2010 21:45:20 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
Message-ID: <DE3CBF01-9E0B-46FC-95C3-B27B824DC7F1@freeswitch.org>

your invite says G726-32  (thats what the 2 is in the audio line)

/b

On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:

> 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP:
> v=0
> o=Sippy 257534956 1 IN IP4 80.232.37.178
> s=-
> t=0 0
> m=audio 47904 RTP/AVP 2 101 13
> c=IN IP4 213.50.90.3
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15




From mcampbellsmith at gmail.com  Tue Jan  5 19:59:19 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 6 Jan 2010 14:59:19 +1100
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <DE3CBF01-9E0B-46FC-95C3-B27B824DC7F1@freeswitch.org>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<DE3CBF01-9E0B-46FC-95C3-B27B824DC7F1@freeswitch.org>
Message-ID: <33c87fa31001051959q42627fekc2c6a85fb68792af@mail.gmail.com>

I thought the Remote SDP comes from my SIP provider?  o=Sippy
257534956 1 IN IP4 80.232.37.178.  80.232.37.178 is not my IP address,
its the ip address of my voip provider?   But now you mention that
2=G726-32, its what I have as default.

Anyway, even if G726-32 is default, I also have PCMU and PCMA ... why
isn't negotiation used to select these codecs and isn't FS comparing
codecs?

Sorry, I guess these are basic questions ...


On Wed, Jan 6, 2010 at 2:45 PM, Brian West <brian at freeswitch.org> wrote:
> your invite says G726-32 ?(thats what the 2 is in the audio line)
>
> /b
>
> On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
>
>> 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP:
>> v=0
>> o=Sippy 257534956 1 IN IP4 80.232.37.178
>> s=-
>> t=0 0
>> m=audio 47904 RTP/AVP 2 101 13
>> c=IN IP4 213.50.90.3
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Tue Jan  5 20:02:59 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Jan 2010 22:02:59 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001051959q42627fekc2c6a85fb68792af@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<DE3CBF01-9E0B-46FC-95C3-B27B824DC7F1@freeswitch.org>
	<33c87fa31001051959q42627fekc2c6a85fb68792af@mail.gmail.com>
Message-ID: <A42B7C6C-F40B-453F-B895-EE33307DAB81@freeswitch.org>

because they aren't in the invite... it can't negotiate things that aren't in the invite... its clearly NOT in that sdp... its only CN, G726-32 and Telephony event.

/b

On Jan 5, 2010, at 9:59 PM, Mark Campbell-Smith wrote:

> Anyway, even if G726-32 is default, I also have PCMU and PCMA ... why
> isn't negotiation used to select these codecs and isn't FS comparing
> codecs?
> 
> Sorry, I guess these are basic questions ...




From mcampbellsmith at gmail.com  Tue Jan  5 20:16:40 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 6 Jan 2010 15:16:40 +1100
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <A42B7C6C-F40B-453F-B895-EE33307DAB81@freeswitch.org>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<DE3CBF01-9E0B-46FC-95C3-B27B824DC7F1@freeswitch.org>
	<33c87fa31001051959q42627fekc2c6a85fb68792af@mail.gmail.com>
	<A42B7C6C-F40B-453F-B895-EE33307DAB81@freeswitch.org>
Message-ID: <33c87fa31001052016r6d2ae072g8a210e244dfcc268@mail.gmail.com>

For my codec prefs I have:
  <X-PRE-PROCESS cmd="set"
data="global_codec_prefs=iLBC at 30i,G726-32,G722,PCMU,PCMA"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G726-32,PCMU,PCMA"/>

How many codecs are sent in an invite?  Is it only the top three of
global_codec_prefs?  Is CN = iLBC and telephony-event = G722?

Thanks

On Wed, Jan 6, 2010 at 3:02 PM, Brian West <brian at freeswitch.org> wrote:
> because they aren't in the invite... it can't negotiate things that aren't in the invite... its clearly NOT in that sdp... its only CN, G726-32 and Telephony event.
>
> /b
>
> On Jan 5, 2010, at 9:59 PM, Mark Campbell-Smith wrote:
>
>> Anyway, even if G726-32 is default, I also have PCMU and PCMA ... why
>> isn't negotiation used to select these codecs and isn't FS comparing
>> codecs?
>>
>> Sorry, I guess these are basic questions ...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Tue Jan  5 20:20:25 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Jan 2010 22:20:25 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
Message-ID: <1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>

Lets try this again.. THIS SDP IS NOT from FreeSWITCH its an inbound invite to FreeSWITCH... fix that and you'll be golden.

/b

On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:

> 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP:
> v=0
> o=Sippy 257534956 1 IN IP4 80.232.37.178
> s=-
> t=0 0
> m=audio 47904 RTP/AVP 2 101 13
> c=IN IP4 213.50.90.3
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15




From freeswitch at aastral.net  Tue Jan  5 20:55:15 2010
From: freeswitch at aastral.net (Bill W.)
Date: Tue, 05 Jan 2010 23:55:15 -0500
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <201001052125.06909.sos@sokhapkin.dyndns.org>
References: <AcqOWx9O9EDYsJ/ZQJSSOHSvWxhecA==>	<A2B4F48489739E439D8845296A2FD9C4147E1F@sun.Noble.local>	<b39273531001051813v5271858fpd133326559238b5e@mail.gmail.com>
	<201001052125.06909.sos@sokhapkin.dyndns.org>
Message-ID: <4B4417B3.9090807@aastral.net>

Hey Sergey,

But nibblebill will transfer to an extension of your choice when the
balance reaches $0. So if you set the nibble heartbeat to 60 seconds or
whatever, nibblebill will deduct the appropriate amount every
<heartbeat> seconds.   So after about 20 minutes, the call will execute
the nobal_action specified in nibblebill.conf.xml.

So that should meet your needs.

Bill W.


Sergey Okhapkin wrote:
> Unfortunalely, mod_nibblebill doesn't take billing increments into account.
> 
> On Tuesday 05 January 2010, Rupa Schomaker wrote:
>> Look at using mod_nibblebill
>>



From mcampbellsmith at gmail.com  Tue Jan  5 21:03:33 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 6 Jan 2010 16:03:33 +1100
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
Message-ID: <33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>

ahh... thats what I thought originally.

So what codecs match CN and telephony-event?  Where can I find these mappings?

m=audio 47904 RTP/AVP 2 101 13

Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20]
Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20]

Thanks Brian!

On Wed, Jan 6, 2010 at 3:20 PM, Brian West <brian at freeswitch.org> wrote:
> Lets try this again.. THIS SDP IS NOT from FreeSWITCH its an inbound invite to FreeSWITCH... fix that and you'll be golden.
>
> /b
>
> On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
>
>> 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP:
>> v=0
>> o=Sippy 257534956 1 IN IP4 80.232.37.178
>> s=-
>> t=0 0
>> m=audio 47904 RTP/AVP 2 101 13
>> c=IN IP4 213.50.90.3
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Tue Jan  5 21:19:28 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Jan 2010 23:19:28 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
Message-ID: <895DF0A8-334E-4214-916D-16446AAED1F7@freeswitch.org>

13 = CN
101 = Telephony Event
2 = G723-32

/b

On Jan 5, 2010, at 11:03 PM, Mark Campbell-Smith wrote:

> ahh... thats what I thought originally.
> 
> So what codecs match CN and telephony-event?  Where can I find these mappings?
> 
> m=audio 47904 RTP/AVP 2 101 13
> 
> Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20]
> Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20]
> 
> Thanks Brian!




From anthony.minessale at gmail.com  Tue Jan  5 21:20:04 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Jan 2010 23:20:04 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
Message-ID: <191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>

those are not audio codecs CN is just comfort noise and telephone-event is
dtmf


On Tue, Jan 5, 2010 at 11:03 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> ahh... thats what I thought originally.
>
> So what codecs match CN and telephony-event?  Where can I find these
> mappings?
>
> m=audio 47904 RTP/AVP 2 101 13
>
> Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20]
> Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20]
>
> Thanks Brian!
>
> On Wed, Jan 6, 2010 at 3:20 PM, Brian West <brian at freeswitch.org> wrote:
> > Lets try this again.. THIS SDP IS NOT from FreeSWITCH its an inbound
> invite to FreeSWITCH... fix that and you'll be golden.
> >
> > /b
> >
> > On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
> >
> >> 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP:
> >> v=0
> >> o=Sippy 257534956 1 IN IP4 80.232.37.178
> >> s=-
> >> t=0 0
> >> m=audio 47904 RTP/AVP 2 101 13
> >> c=IN IP4 213.50.90.3
> >> a=rtpmap:101 telephone-event/8000
> >> a=fmtp:101 0-15
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mcampbellsmith at gmail.com  Tue Jan  5 21:40:10 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 6 Jan 2010 16:40:10 +1100
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
	<191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
Message-ID: <33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>

Thanks Brian and Anthony.

Brian: 2 = G723-32 Do you mean G721 instead of G723-32 ?

2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
Compare [G721:2:8000:20]/[G726-32:2:8000:20]

So the only codec they have offered is G723-32 (or G721), which FS
only supports as a passthrough codec.  Is that correct?


On Wed, Jan 6, 2010 at 4:20 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> those are not audio codecs CN is just comfort noise and telephone-event is
> dtmf
>
>
> On Tue, Jan 5, 2010 at 11:03 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>>
>> ahh... thats what I thought originally.
>>
>> So what codecs match CN and telephony-event? ?Where can I find these
>> mappings?
>>
>> m=audio 47904 RTP/AVP 2 101 13
>>
>> Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20]
>> Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20]
>>
>> Thanks Brian!
>>
>> On Wed, Jan 6, 2010 at 3:20 PM, Brian West <brian at freeswitch.org> wrote:
>> > Lets try this again.. THIS SDP IS NOT from FreeSWITCH its an inbound
>> > invite to FreeSWITCH... fix that and you'll be golden.
>> >
>> > /b
>> >
>> > On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
>> >
>> >> 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP:
>> >> v=0
>> >> o=Sippy 257534956 1 IN IP4 80.232.37.178
>> >> s=-
>> >> t=0 0
>> >> m=audio 47904 RTP/AVP 2 101 13
>> >> c=IN IP4 213.50.90.3
>> >> a=rtpmap:101 telephone-event/8000
>> >> a=fmtp:101 0-15
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From jon at radel.com  Tue Jan  5 21:45:37 2010
From: jon at radel.com (Jon Radel)
Date: Wed, 06 Jan 2010 00:45:37 -0500
Subject: [Freeswitch-users] Codec Negotiation:
	Codec	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
Message-ID: <4B442381.9020705@radel.com>

Mark Campbell-Smith wrote:
> ahh... thats what I thought originally.
> 
> So what codecs match CN and telephony-event?  Where can I find these mappings?

http://www.iana.org/assignments/rtp-parameters

is one place to start your journey, but there's enough dynamic 
assignment, convention, and general etc. to make it an interesting 
journey.  If somebody can set me straight and provide the definitive 
documentation on payload types, should such actually exist, I'd be most 
appreciative.

-- 

--Jon Radel
jon at radel.com
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From anthony.minessale at gmail.com  Tue Jan  5 21:50:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Jan 2010 23:50:10 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
	<191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
	<33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
Message-ID: <191c3a031001052150x1fd22ed0p86d3a6f307df4b32@mail.gmail.com>

no we fully support it, just add it to your config

google for iana codec sdp for the reserved numbers and what they mean.


On Tue, Jan 5, 2010 at 11:40 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Thanks Brian and Anthony.
>
> Brian: 2 = G723-32 Do you mean G721 instead of G723-32 ?
>
> 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
> Compare [G721:2:8000:20]/[G726-32:2:8000:20]
>
> So the only codec they have offered is G723-32 (or G721), which FS
> only supports as a passthrough codec.  Is that correct?
>
>
> On Wed, Jan 6, 2010 at 4:20 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > those are not audio codecs CN is just comfort noise and telephone-event
> is
> > dtmf
> >
> >
> > On Tue, Jan 5, 2010 at 11:03 PM, Mark Campbell-Smith
> > <mcampbellsmith at gmail.com> wrote:
> >>
> >> ahh... thats what I thought originally.
> >>
> >> So what codecs match CN and telephony-event?  Where can I find these
> >> mappings?
> >>
> >> m=audio 47904 RTP/AVP 2 101 13
> >>
> >> Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20]
> >> Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20]
> >>
> >> Thanks Brian!
> >>
> >> On Wed, Jan 6, 2010 at 3:20 PM, Brian West <brian at freeswitch.org>
> wrote:
> >> > Lets try this again.. THIS SDP IS NOT from FreeSWITCH its an inbound
> >> > invite to FreeSWITCH... fix that and you'll be golden.
> >> >
> >> > /b
> >> >
> >> > On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
> >> >
> >> >> 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP:
> >> >> v=0
> >> >> o=Sippy 257534956 1 IN IP4 80.232.37.178
> >> >> s=-
> >> >> t=0 0
> >> >> m=audio 47904 RTP/AVP 2 101 13
> >> >> c=IN IP4 213.50.90.3
> >> >> a=rtpmap:101 telephone-event/8000
> >> >> a=fmtp:101 0-15
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Tue Jan  5 21:50:46 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Jan 2010 23:50:46 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
	<191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
	<33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
Message-ID: <0A8CD6A7-CA51-44F2-B530-15AE623EBA5D@freeswitch.org>

They are in fact one in the same please see ITU.

/b

On Jan 5, 2010, at 11:40 PM, Mark Campbell-Smith wrote:

> Thanks Brian and Anthony.
> 
> Brian: 2 = G723-32 Do you mean G721 instead of G723-32 ?
> 
> 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
> Compare [G721:2:8000:20]/[G726-32:2:8000:20]
> 
> So the only codec they have offered is G723-32 (or G721), which FS
> only supports as a passthrough codec.  Is that correct?




From achaloyan at yahoo.com  Tue Jan  5 23:29:10 2010
From: achaloyan at yahoo.com (Arsen Chaloyan)
Date: Tue, 5 Jan 2010 23:29:10 -0800 (PST)
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <0A8CD6A7-CA51-44F2-B530-15AE623EBA5D@freeswitch.org>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
	<191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
	<33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
	<0A8CD6A7-CA51-44F2-B530-15AE623EBA5D@freeswitch.org>
Message-ID: <637054.57565.qm@web111313.mail.gq1.yahoo.com>

The following section in RFC3551 states the same
http://tools.ietf.org/html/rfc3551#section-4.5.4


   The encodings of G.726 are here denoted as G726-40, G726-32, G726-24,
   and G726-16.  Prior to 1990, G721 described the 32 kbit/s ADPCM
   encoding, and G723 described the 40, 32, and 16 kbit/s encodings.
   Thus, G726-32 designates the same algorithm as G721 in RFC 1890.





________________________________
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Wed, January 6, 2010 9:50:46 AM
Subject: Re: [Freeswitch-users] Codec Negotiation: Codec telephone-event:101:8000:20

They are in fact one in the same please see ITU.

/b

On Jan 5, 2010, at 11:40 PM, Mark Campbell-Smith wrote:

> Thanks Brian and Anthony.
> 
> Brian: 2 = G723-32 Do you mean G721 instead of G723-32 ?
> 
> 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec
> Compare [G721:2:8000:20]/[G726-32:2:8000:20]
> 
> So the only codec they have offered is G723-32 (or G721), which FS
> only supports as a passthrough codec.  Is that correct?


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From durk.debeer at isp.solcon.nl  Wed Jan  6 01:04:26 2010
From: durk.debeer at isp.solcon.nl (Durk.de Beer)
Date: Wed, 06 Jan 2010 10:04:26 +0100
Subject: [Freeswitch-users]  Detecting status of User Agent
Message-ID: <mailman.15.1269221067.3324.freeswitch-users@lists.freeswitch.org>



Ok I want to do the following thing. If an user agent (UA), some SIP-client for instance X-lite, is of line the call is to be forwarded to an phone number provided by the user how was called.
I succeed doing this if there is no registration for the number dialled. FS is then reporting an USER_NOT_REGISTERED so I am able to alter the dial plan accordingly and redirect the call. So far so good. Now my problem arises when there is an registration on FS but the UA is not online for what ever reason (my cat seems to like to chew on CAT5 cable). If this occurs FS is trying to bridge but it is unable to because there is no response from the UA called. It ends by a RECOVERY_ON_TIMER_EXPIRE but the time it takes for this to happen is far to long. I've tried to set this variable but failed to do so. Other channel variables seem to be only effective if there's an successful bridge. So how do I set the RECOVERY_ON_TIMER_EXPIRE?




Durk 


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From sos at sokhapkin.dyndns.org  Wed Jan  6 03:20:13 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 6 Jan 2010 06:20:13 -0500
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <4B4417B3.9090807@aastral.net>
References: <AcqOWx9O9EDYsJ/ZQJSSOHSvWxhecA==>
	<201001052125.06909.sos@sokhapkin.dyndns.org>
	<4B4417B3.9090807@aastral.net>
Message-ID: <201001060620.13735.sos@sokhapkin.dyndns.org>

nibblebill has no concept of billing blocks. What if I want to bill customer 
30 seconds minimum and 6 seconds increment thereafter?

On Tuesday 05 January 2010, Bill W. wrote:
> Hey Sergey,
>
> But nibblebill will transfer to an extension of your choice when the
> balance reaches $0. So if you set the nibble heartbeat to 60 seconds or
> whatever, nibblebill will deduct the appropriate amount every
> <heartbeat> seconds.   So after about 20 minutes, the call will execute
> the nobal_action specified in nibblebill.conf.xml.
>
> So that should meet your needs.
>
> Bill W.
>
> Sergey Okhapkin wrote:
> > Unfortunalely, mod_nibblebill doesn't take billing increments into
> > account.
> >
> > On Tuesday 05 January 2010, Rupa Schomaker wrote:
> >> Look at using mod_nibblebill
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From jbr at consiglia.dk  Wed Jan  6 03:55:14 2010
From: jbr at consiglia.dk (Jon Bruel)
Date: Wed, 6 Jan 2010 12:55:14 +0100
Subject: [Freeswitch-users] Is there support for custom fields for all events
Message-ID: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE09AC568438@SBS2008SERVER.consiglia.local>

I would like to add a custom field to all events sent from the FS. The value of the field should be the value of a global variable, the name should preferably be the name of this variable. Is there any way to set this up? /Jon
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From jcasale at activenetwerx.com  Wed Jan  6 03:56:14 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 6 Jan 2010 11:56:14 +0000
Subject: [Freeswitch-users] Dahdi Saga continues
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906D5C2@prato.activenetwerx.local>

So when the dahdi config is left empty, just defaultzone=us and
loadzone=us, it works except obviously there is no echo canceller.

Once I add an echo canceller and set fxsks=1 then callerid fails,
it just says OpenZap on the handsets and the audio doesn't start
working after some time after the call is answered.

Any ideas?

Thanks!
jlc



From Russell.Mosemann at cune.org  Wed Jan  6 04:17:15 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Wed, 6 Jan 2010 06:17:15 -0600
Subject: [Freeswitch-users] Dahdi Saga continues
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E906D5C2@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E906D5C2@prato.activenetwerx.local>
Message-ID: <45529145881D44A89C1B0437D78B2712@cune.pri>

Joseph L. Casale wrote:

> Once I add an echo canceller and set fxsks=1 then callerid fails,

You could try building DAHDI with OSLEC. Don't put any echo cancel statements in the config file. The steps you want are under "Install OSLEC with DAHDI".

http://www.rowetel.com/ucasterisk/oslec.html

--
Russell Mosemann




From jbr at consiglia.dk  Wed Jan  6 05:33:07 2010
From: jbr at consiglia.dk (Jon Bruel)
Date: Wed, 6 Jan 2010 14:33:07 +0100
Subject: [Freeswitch-users] Detecting status of User Agent
In-Reply-To: <9ad1f5d7-ad02-49df-81dd-42e4f7d5f1cd@SBS2008SERVER.consiglia.local>
References: <9ad1f5d7-ad02-49df-81dd-42e4f7d5f1cd@SBS2008SERVER.consiglia.local>
Message-ID: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE09AC568439@SBS2008SERVER.consiglia.local>

Hi Durk

I tried the situation mentioned by unplugging the UA. FS then reports back (via the channel variable originate_disposition): NORMAL_TEMPORARY_FAILURE, which can then be used to take action. Before bridging to the phone, I have set <action application="set" data="continue_on_fail=true"/>. Hope it assists you.

Anyhow, your cat must be very intelligent since it knows it's a CAT cable.

/Jon

Jon Br?el
Consiglia Telecommunications
DK-2960 Rungsted Kyst
Tel: +45 45 16 1000
Mob: +45 26 15 30 60
CVR: 27047882

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Durk.de Beer
Sent: 6. januar 2010 10:04
To: Freeswitch-Users
Subject: [Freeswitch-users] Detecting status of User Agent




Ok I want to do the following thing. If an user agent (UA), some SIP-client for instance X-lite, is of line the call is to be forwarded to an phone number provided by the user how was called.

I succeed doing this if there is no registration for the number dialled. FS is then reporting an USER_NOT_REGISTERED so I am able to alter the dial plan accordingly and redirect the call. So far so good. Now my problem arises when there is an registration on FS but the UA is not online for what ever reason (my cat seems to like to chew on CAT5 cable). If this occurs FS is trying to bridge but it is unable to because there is no response from the UA called. It ends by a RECOVERY_ON_TIMER_EXPIRE but the time it takes for this to happen is far to long. I've tried to set this variable but failed to do so. Other channel variables seem to be only effective if there's an successful bridge. So how do I set the RECOVERY_ON_TIMER_EXPIRE?





Durk

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From linux4michelle at tamay-dogan.net  Wed Jan  6 05:52:10 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Wed, 6 Jan 2010 14:52:10 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <20100103184636.GW5547@tamay-dogan.net>
References: <20100103184636.GW5547@tamay-dogan.net>
Message-ID: <20100106135210.GG5547@tamay-dogan.net>

Realy no one who use FreeSwitch as GSM PBX?


Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From jcasale at activenetwerx.com  Wed Jan  6 06:29:38 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 6 Jan 2010 14:29:38 +0000
Subject: [Freeswitch-users] Dahdi Saga continues
In-Reply-To: <45529145881D44A89C1B0437D78B2712@cune.pri>
References: <CA5A491E9DEFBE4CB777DE97E21575E906D5C2@prato.activenetwerx.local>
	<45529145881D44A89C1B0437D78B2712@cune.pri>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906E0F1@prato.activenetwerx.local>

>You could try building DAHDI with OSLEC. Don't put any echo cancel statements in the config file. The steps you want >are under "Install OSLEC with DAHDI".
>
>http://www.rowetel.com/ucasterisk/oslec.html

You still need to specify the echo canceller w/ oslec as well
though. What do you think the lack of configuration making it
work is indicative of? I think for for the price, I might just
buy an SPA3102 and be done w/ this nightmare...



From Russell.Mosemann at cune.org  Wed Jan  6 06:38:52 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Wed, 6 Jan 2010 14:38:52 -0000
Subject: [Freeswitch-users] Dahdi Saga continues
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E906E0F1@prato.activenetwerx.local>
Message-ID: <20100106143852.7E98C2C83EC@cuneorg-email.cune.pri>

"Joseph L. Casale" <jcasale at activenetwerx.com> said:

> You still need to specify the echo canceller w/ oslec as well
> though.

Are you sure? I was under the impression that OSLEC was built in and that
there was no choice to turn it on or off.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From brian at freeswitch.org  Wed Jan  6 07:04:39 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jan 2010 09:04:39 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <637054.57565.qm@web111313.mail.gq1.yahoo.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
	<191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
	<33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
	<0A8CD6A7-CA51-44F2-B530-15AE623EBA5D@freeswitch.org>
	<637054.57565.qm@web111313.mail.gq1.yahoo.com>
Message-ID: <965454EF-F779-4563-AC64-082CAB07CCA5@freeswitch.org>

w00t! :)

/b

On Jan 6, 2010, at 1:29 AM, Arsen Chaloyan wrote:

> The following section in RFC3551 states the same
> http://tools.ietf.org/html/rfc3551#section-4.5.4
> 
> 
>    The encodings of G.726 are here denoted as G726-40, G726-32, G726-24,
>    and G726-16.  Prior to 1990, G721 described the 32 kbit/s ADPCM
>    encoding, and G723 described the 40, 32, and 16 kbit/s encodings.
>    Thus, G726-32 designates the same algorithm as G721 in RFC 1890.
> 
> 
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wed, January 6, 2010 9:50:46 AM
> Subject: Re: [Freeswitch-users] Codec Negotiation: Codec telephone-event:101:8000:20
> 
> They are in fact one in the same please see ITU.
> 
> /b

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From mike at jerris.com  Wed Jan  6 07:19:19 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 6 Jan 2010 10:19:19 -0500
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <20100106135210.GG5547@tamay-dogan.net>
References: <20100103184636.GW5547@tamay-dogan.net>
	<20100106135210.GG5547@tamay-dogan.net>
Message-ID: <CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>

I was hoping Giovanni, who is working on code as an endpoint module for gsm devices, would answer this.  But he has not

Mie

On Jan 6, 2010, at 8:52 AM, Michelle Konzack wrote:

> Realy no one who use FreeSwitch as GSM PBX?




From gmaruzz at celliax.org  Wed Jan  6 07:59:14 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 6 Jan 2010 16:59:14 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>
References: <20100103184636.GW5547@tamay-dogan.net>
	<20100106135210.GG5547@tamay-dogan.net> 
	<CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>
Message-ID: <7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com>

hehehe,

MikeJ, you're right!

I'm taking opportunity from holidays (in Italy we still in holidays
until tomorrow), to make ready for testing the endpoint for GSM
devices.

Short blurb:
- can use as phisical interface high end GSM modules, or gsm modems,
or gsm cellphones (with cables)
- can send/receive SMSs and voice calls
- SMSs generates events
- will use standard CHAT api interface for SMSs (like Jingle and sofia/SIMPLE)
- will be possible to compile it with sound support and a c++ library
for PDU access (for who that wants maximum SMS details plus voice
calls)
-will be possible to compile it as bare C (no c++) without sound
support and without PDU support for maximum embeddability in low end
machines (will act as an SMS gateway, sending/receiving SMSs, without
voice calls)

will soon be ported to work on windoz too (at the moment, works only
on Linux, maybe on *BSD too)

You can see old wikipage (no more reliable, to be thoroughly updated)
here: http://wiki.freeswitch.org/wiki/GSMopen

-giovanni


On Wed, Jan 6, 2010 at 4:19 PM, Michael Jerris <mike at jerris.com> wrote:
> I was hoping Giovanni, who is working on code as an endpoint module for gsm devices, would answer this. ?But he has not
>
> Mie
>
> On Jan 6, 2010, at 8:52 AM, Michelle Konzack wrote:
>
>> Realy no one who use FreeSwitch as GSM PBX?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From gmaruzz at celliax.org  Wed Jan  6 08:08:42 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 6 Jan 2010 17:08:42 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com>
References: <20100103184636.GW5547@tamay-dogan.net>
	<20100106135210.GG5547@tamay-dogan.net> 
	<CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>
	<7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com>
Message-ID: <7b197bef1001060808m1139bd96o3d86bfe89601399d@mail.gmail.com>

Michelle,

sorry, after thorough search I find your first two messages in the
Spam box (while MikeJ was passing through).

(I've now set the filter on Freeswitch-users to "never" go in Spam)

If you need any info, don't hesitate to write to the mailing list or
catch me in IRC @freeswitch channel as gmaruzz (or gmaruzz1, at the
will of net splits).

-giovanni


On Wed, Jan 6, 2010 at 4:59 PM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
> hehehe,
>
> MikeJ, you're right!
>
> I'm taking opportunity from holidays (in Italy we still in holidays
> until tomorrow), to make ready for testing the endpoint for GSM
> devices.
>
> Short blurb:
> - can use as phisical interface high end GSM modules, or gsm modems,
> or gsm cellphones (with cables)
> - can send/receive SMSs and voice calls
> - SMSs generates events
> - will use standard CHAT api interface for SMSs (like Jingle and sofia/SIMPLE)
> - will be possible to compile it with sound support and a c++ library
> for PDU access (for who that wants maximum SMS details plus voice
> calls)
> -will be possible to compile it as bare C (no c++) without sound
> support and without PDU support for maximum embeddability in low end
> machines (will act as an SMS gateway, sending/receiving SMSs, without
> voice calls)
>
> will soon be ported to work on windoz too (at the moment, works only
> on Linux, maybe on *BSD too)
>
> You can see old wikipage (no more reliable, to be thoroughly updated)
> here: http://wiki.freeswitch.org/wiki/GSMopen
>
> -giovanni
>
>
> On Wed, Jan 6, 2010 at 4:19 PM, Michael Jerris <mike at jerris.com> wrote:
>> I was hoping Giovanni, who is working on code as an endpoint module for gsm devices, would answer this. ?But he has not
>>
>> Mie
>>
>> On Jan 6, 2010, at 8:52 AM, Michelle Konzack wrote:
>>
>>> Realy no one who use FreeSwitch as GSM PBX?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From freeswitch at aastral.net  Wed Jan  6 08:25:18 2010
From: freeswitch at aastral.net (Bill W)
Date: Wed, 06 Jan 2010 11:25:18 -0500
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <201001060620.13735.sos@sokhapkin.dyndns.org>
References: <AcqOWx9O9EDYsJ/ZQJSSOHSvWxhecA==>	<201001052125.06909.sos@sokhapkin.dyndns.org>	<4B4417B3.9090807@aastral.net>
	<201001060620.13735.sos@sokhapkin.dyndns.org>
Message-ID: <4B44B96E.2040408@aastral.net>

True, there is no inherent support for that, but you might be able to 
get close by doing it in the dialplan.  Establish the call, pause 
nibblebill, deduct a specific amount (nibblebill adjust), and when 30 
seconds are up, unpause nibblebill.

More than likely you'd have to do this in a script rather than in XML.

The issue would be the last interval after the last heartbeat. 
Nibblebill won't round up to the next 6 seconds.  It will just bill for 
the actual call time. (If I understand things correctly).

Or you could put in a bounty to have billing block support added to 
nibblebill.


Hope this helps.
Bill


Sergey Okhapkin wrote:
> nibblebill has no concept of billing blocks. What if I want to bill customer 
> 30 seconds minimum and 6 seconds increment thereafter?
> 



From oscav at hotmail.fr  Wed Jan  6 08:37:21 2010
From: oscav at hotmail.fr (Oscav)
Date: Wed, 6 Jan 2010 08:37:21 -0800 (PST)
Subject: [Freeswitch-users] re lease an outbound call when caller sends
 digits like ##
Message-ID: <27026910.post@talk.nabble.com>


Hi,

How can we cancel an outbound call if the caller digits some DTMF like ## ??
There is the bind_meta_app but it only handles 1 digit.

Thanks.
-- 
View this message in context: http://old.nabble.com/release-an-outbound-call-when-caller-sends-digits-like----tp27026910p27026910.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From mike at jerris.com  Wed Jan  6 08:41:08 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 6 Jan 2010 11:41:08 -0500
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <201001060620.13735.sos@sokhapkin.dyndns.org>
References: <AcqOWx9O9EDYsJ/ZQJSSOHSvWxhecA==>
	<201001052125.06909.sos@sokhapkin.dyndns.org>
	<4B4417B3.9090807@aastral.net>
	<201001060620.13735.sos@sokhapkin.dyndns.org>
Message-ID: <EF51CEC4-232C-4026-AAEF-451AB18CAB01@jerris.com>

This is open source, you can create a patch to add this functionality and contribute it back.

Mike

On Jan 6, 2010, at 6:20 AM, Sergey Okhapkin wrote:

> nibblebill has no concept of billing blocks. What if I want to bill customer 
> 30 seconds minimum and 6 seconds increment thereafter?
> 




From dome at tel.co.th  Wed Jan  6 08:59:44 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 6 Jan 2010 23:59:44 +0700
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <201001060620.13735.sos@sokhapkin.dyndns.org>
References: <201001052125.06909.sos@sokhapkin.dyndns.org>
	<4B4417B3.9090807@aastral.net>
	<201001060620.13735.sos@sokhapkin.dyndns.org>
Message-ID: <8ccbff061001060859p59aa6b5bw6f52275650f9138@mail.gmail.com>

2010/1/6 Sergey Okhapkin <sos at sokhapkin.dyndns.org>:
> nibblebill has no concept of billing blocks. What if I want to bill customer
> 30 seconds minimum and 6 seconds increment thereafter?

I have billing (in house develop) and customize  nibble_bill update
cdr table (in my billing) i use postgresql trigger to update account
balance. i have many increment rule 1/1 30/6 60/60

It's work well for me :)


BG

Dome C.
>
> On Tuesday 05 January 2010, Bill W. wrote:
>> Hey Sergey,
>>
>> But nibblebill will transfer to an extension of your choice when the
>> balance reaches $0. So if you set the nibble heartbeat to 60 seconds or
>> whatever, nibblebill will deduct the appropriate amount every
>> <heartbeat> seconds. ? So after about 20 minutes, the call will execute
>> the nobal_action specified in nibblebill.conf.xml.
>>
>> So that should meet your needs.
>>
>> Bill W.
>>
>> Sergey Okhapkin wrote:
>> > Unfortunalely, mod_nibblebill doesn't take billing increments into
>> > account.
>> >
>> > On Tuesday 05 January 2010, Rupa Schomaker wrote:
>> >> Look at using mod_nibblebill
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From linux4michelle at tamay-dogan.net  Wed Jan  6 09:14:13 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Wed, 6 Jan 2010 18:14:13 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com>
References: <20100103184636.GW5547@tamay-dogan.net>
	<20100106135210.GG5547@tamay-dogan.net>
	<CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>
	<7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com>
Message-ID: <20100106171413.GI5547@tamay-dogan.net>

Chiao Giovanni,

thankyoufor you efforts to get this running...

Do you have a list of High-End GSM-Modules?

If not my co-worker and me would develop one but we are not sure, which
End-Points we need in the Modem...  The biggest  problem  is,  that  we
need the HD*PA part too, to get the Internet connection.

If I see it right, the "Nokia 76120 classic" has  tonns  of  End-Points
and it seems, irt support Data+Voice in the same time...

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant


Am 2010-01-06 16:59:14, schrieb Giovanni Maruzzelli:
> hehehe,
> 
> MikeJ, you're right!
> 
> I'm taking opportunity from holidays (in Italy we still in holidays
> until tomorrow), to make ready for testing the endpoint for GSM
> devices.
> 
> Short blurb:
> - can use as phisical interface high end GSM modules, or gsm modems,
> or gsm cellphones (with cables)
> - can send/receive SMSs and voice calls
> - SMSs generates events
> - will use standard CHAT api interface for SMSs (like Jingle and sofia/SIMPLE)
> - will be possible to compile it with sound support and a c++ library
> for PDU access (for who that wants maximum SMS details plus voice
> calls)
> -will be possible to compile it as bare C (no c++) without sound
> support and without PDU support for maximum embeddability in low end
> machines (will act as an SMS gateway, sending/receiving SMSs, without
> voice calls)
> 
> will soon be ported to work on windoz too (at the moment, works only
> on Linux, maybe on *BSD too)
> 
> You can see old wikipage (no more reliable, to be thoroughly updated)
> here: http://wiki.freeswitch.org/wiki/GSMopen
> 
> -giovanni
------------------------ END OF REPLIED MESSAGE ------------------------



-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From jcasale at activenetwerx.com  Wed Jan  6 09:26:02 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 6 Jan 2010 17:26:02 +0000
Subject: [Freeswitch-users] Dahdi Saga continues
In-Reply-To: <20100106143852.7E98C2C83EC@cuneorg-email.cune.pri>
References: <CA5A491E9DEFBE4CB777DE97E21575E906E0F1@prato.activenetwerx.local>
	<20100106143852.7E98C2C83EC@cuneorg-email.cune.pri>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906E729@prato.activenetwerx.local>

>Are you sure? I was under the impression that OSLEC was built in and that
>there was no choice to turn it on or off.

All the digium dahdi docs and my previous experience with it reference it
being activated like any other canceller.

Dahdi specifically states all echo cancellers must be manually activated
per channel where dahdi_echocan_* => echocanceller=*,1... and the oslec
module builds with the same naming convention...



From anthony.minessale at gmail.com  Wed Jan  6 09:34:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Jan 2010 11:34:37 -0600
Subject: [Freeswitch-users] re lease an outbound call when caller sends
	digits like ##
In-Reply-To: <27026910.post@talk.nabble.com>
References: <27026910.post@talk.nabble.com>
Message-ID: <191c3a031001060934n3f33a977m1820d6e491364f16@mail.gmail.com>

well it's 1 digit plus the meta so its really 2

*0 for instance


On Wed, Jan 6, 2010 at 10:37 AM, Oscav <oscav at hotmail.fr> wrote:

>
> Hi,
>
> How can we cancel an outbound call if the caller digits some DTMF like ##
> ??
> There is the bind_meta_app but it only handles 1 digit.
>
> Thanks.
> --
> View this message in context:
> http://old.nabble.com/release-an-outbound-call-when-caller-sends-digits-like----tp27026910p27026910.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From gmaruzz at celliax.org  Wed Jan  6 09:39:22 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 6 Jan 2010 18:39:22 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <20100106171413.GI5547@tamay-dogan.net>
References: <20100103184636.GW5547@tamay-dogan.net>
	<20100106135210.GG5547@tamay-dogan.net> 
	<CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>
	<7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com> 
	<20100106171413.GI5547@tamay-dogan.net>
Message-ID: <7b197bef1001060939rd1e08f3h2cb82cd89e50225e@mail.gmail.com>

Hello Michelle,

it supports all modules that accepts standard ETSI AT-GSM commands
(so, let's say all of them).

Maybe I do not understand the second question, what do you means for Endpoints?

If you're talking usb endpoints, you'll need a modem endpoint (that
can be seen as a serial port), and (if you need audio, eg not just
SMSs but voice calls too) you need an audio endpoint (that can be seen
as a soundcard).

Many modules and cellphones can be seen as HDSPA or GPRS modems, just
check their specs.

For audio, if the module/cellphone/modem does not offer an audio usb
endpoint (eg cannot be seen as a soundcard) one trick is to connect
the headset jack to an usb soundcard (you can find soundcard with for
factor like a dongle based on cm-108 chipset for under $10). I'll
publish the schema of the cable needed from hadset jack in the
phone/module to the usb soundcard).

If I have not get what you asked, please explain more your question.

-giovanni

On Wed, Jan 6, 2010 at 6:14 PM, Michelle Konzack
<linux4michelle at tamay-dogan.net> wrote:
> Chiao Giovanni,
>
> thankyoufor you efforts to get this running...
>
> Do you have a list of High-End GSM-Modules?
>
> If not my co-worker and me would develop one but we are not sure, which
> End-Points we need in the Modem... ?The biggest ?problem ?is, ?that ?we
> need the HD*PA part too, to get the Internet connection.
>
> If I see it right, the "Nokia 76120 classic" has ?tonns ?of ?End-Points
> and it seems, irt support Data+Voice in the same time...
>
> Thanks, Greetings and nice Day/Evening
> ? ?Michelle Konzack
> ? ?Electronic Engineer
> ? ?Tamay Dogan Network
> ? ?Debian GNU/Linux Consultant
>
>
> Am 2010-01-06 16:59:14, schrieb Giovanni Maruzzelli:
>> hehehe,
>>
>> MikeJ, you're right!
>>
>> I'm taking opportunity from holidays (in Italy we still in holidays
>> until tomorrow), to make ready for testing the endpoint for GSM
>> devices.
>>
>> Short blurb:
>> - can use as phisical interface high end GSM modules, or gsm modems,
>> or gsm cellphones (with cables)
>> - can send/receive SMSs and voice calls
>> - SMSs generates events
>> - will use standard CHAT api interface for SMSs (like Jingle and sofia/SIMPLE)
>> - will be possible to compile it with sound support and a c++ library
>> for PDU access (for who that wants maximum SMS details plus voice
>> calls)
>> -will be possible to compile it as bare C (no c++) without sound
>> support and without PDU support for maximum embeddability in low end
>> machines (will act as an SMS gateway, sending/receiving SMSs, without
>> voice calls)
>>
>> will soon be ported to work on windoz too (at the moment, works only
>> on Linux, maybe on *BSD too)
>>
>> You can see old wikipage (no more reliable, to be thoroughly updated)
>> here: http://wiki.freeswitch.org/wiki/GSMopen
>>
>> -giovanni
> ------------------------ END OF REPLIED MESSAGE ------------------------
>
>
>
> --
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
> ##################### Debian GNU/Linux Consultant #####################
> <http://www.tamay-dogan.net/> ? ? ? ? ? ? ? ? Michelle Konzack
> <http://www.can4linux.org/> ? ? ? ? ? ? ? ? ? Apt. 917
> <http://www.flexray4linux.org/> ? ? ? ? ? ? ? 50, rue de Soultz
> Jabber linux4michelle at jabber.ccc.de ? ? ? ? ? 67100 Strabourg/France
> IRC ? ?#Debian (irc.icq.com) ? ? ? ? ? ? ? ? ?Tel. DE: +49 177 9351947
> ICQ ? ?#328449886 ? ? ? ? ? ? ? ? ? ? ? ? ? ? Tel. FR: +33 ?6 ?61925193
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From jerry.richards at teotech.com  Wed Jan  6 09:59:35 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 6 Jan 2010 09:59:35 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>
Message-ID: <BE608CAF54594CA48BB8B4DB82A8BFD6@greyhawk.tonecommander.com>

When I attempt an internal-to-PSTN call, there are no Q931 packets sent out
the PRI (I confirmed this using the Sangoma wanpipemon utility).  I suspect
this has something to do with my XML configuration.  Below are my
openzap/wanpipe configurations (which should all be defaulted).  Do you seen
anything wrong with these defaults, which might cause the following FS
console error?
 
zap_io.c:1197 outgoing_call method not implemented!
 
openzap.conf:
[span wanpipe smg_prid]
name => smg_prid
trunk_type =>t1
b-channel => 1:1-23
 
autoload_configs/openzap.conf.xml:
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="9"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
  <boost_spans>
    <span id="smg_prid">
      <!--<param name="hold-music" value="$${moh_uri}"/>-->
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <!-- regex to stop dialing when it matches -->
      <!--<param name="dial-regex" value="5555"/>-->
      <!-- regex to stop dialing when it does not match -->
      <!--<param name="fail-dial-regex" value="^5"/>-->
    </span>
  </boost_spans>
  <analog_spans>
  </analog_spans>
</configuration>
 
wanpipe.conf:
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
 
Thanks and Best Regards,
Jerry
 


  _____  

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Tuesday, January 05, 2010 4:33 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER




On Mon, Jan 4, 2010 at 3:49 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:



Hello,

I have one FS instance that is working well with a PRI and running FS
version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
upgraded it.

Now, I am trying to bring up another FS instance (basically a clone of the
first), but the PRI does not work.  When I attempt to make an
internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
both servers are using the latest Sangoma Wanpipe driver, and I copied the
conf XML file tree from the old server to the new one.  I think the problem
has to do with the openzap module, but I'm having difficulty isolating the
problem.  Could it have built the openzap module incorrectly?  Another
difference is that I installed 1.0.5pre9 from scratch on the new server
(i.e. it never had 1.0.4 running on it).

I put the FS log into the pastebin when an outbound call attempt is made:

http://pastebin.freeswitch.org/11675

Could someone give me a pointer on what to try next?



Jerry, I noticed this line:
(OpenZAP/1:1/3491028 at g1)

Is your carrier wanting full ten digit phone numbers? Try adding the area
code on this and see what happens. The error usually would be something like
"invalid number format" but I've seen carriers do stupid things like this.
Try that first and see if it makes a difference. If not you'll need to turn
on Q931 debugging as per the Sangoma wiki. (See
http://wiki.freeswitch.org/wiki/OpenZAP#Debugging_PRI_With_wanpipemon for
the link.)

-MC



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From anthony.minessale at gmail.com  Wed Jan  6 11:51:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Jan 2010 13:51:23 -0600
Subject: [Freeswitch-users] Is there support for custom fields for all
	events
In-Reply-To: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE09AC568438@SBS2008SERVER.consiglia.local>
References: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE09AC568438@SBS2008SERVER.consiglia.local>
Message-ID: <191c3a031001061151o756149d4x9e08ceea5cb263ff@mail.gmail.com>

not exactly, but you can set channel variables in the dialplan and those
will be in every event about that channel.
so you can make a global exten in your dialplan to set the variable before
anything else.

the hostname of the box is also in every event.
finally you could post a bounty request for some list of special variables
to add to every event on the bounty page or in jira under bounties. probably
about $500 by my estimation.



On Wed, Jan 6, 2010 at 5:55 AM, Jon Bruel <jbr at consiglia.dk> wrote:

>  *I would like to add a custom field to all events sent from the FS. The
> value of the field should be the value of a global variable, the name should
> preferably be the name of this variable. Is there any way to set this up?
> /Jon*
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jerry.richards at teotech.com  Wed Jan  6 11:52:21 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 6 Jan 2010 11:52:21 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com> 
Message-ID: <8441EC83BCD54223A5DAE91BEEC9C1FF@greyhawk.tonecommander.com>

I capured logs of my FS startup and put them into the pastebin
(http://pastebin.freeswitch.org/11692).  At line 722, I see some errors:
 
2010-01-06 11:42:49.907861 [ERR] zap_io.c:2562 Error loading
/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
[/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
file: No such file or directory]
2010-01-06 11:42:49.907883 [ERR] zap_io.c:2722 can't find 'sangoma_boost
2010-01-06 11:42:49.907902 [ERR] mod_openzap.c:2379 Error starting OpenZAP
span 1 error
2010-01-06 11:42:49.907929 [CONSOLE] switch_loadable_module.c:890
Successfully Loaded [mod_openzap]
 
Do you know why I would get this?  Where is the ozmod_sangoma_boost.so file
supposed to come from?
 
Thanks,
Jerry
 
 


  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, January 06, 2010 10:00 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] DESTINATION_OUT_OF_ORDER


When I attempt an internal-to-PSTN call, there are no Q931 packets sent out
the PRI (I confirmed this using the Sangoma wanpipemon utility).  I suspect
this has something to do with my XML configuration.  Below are my
openzap/wanpipe configurations (which should all be defaulted).  Do you seen
anything wrong with these defaults, which might cause the following FS
console error?
 
zap_io.c:1197 outgoing_call method not implemented!
 
openzap.conf:
[span wanpipe smg_prid]
name => smg_prid
trunk_type =>t1
b-channel => 1:1-23
 
autoload_configs/openzap.conf.xml:
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="9"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
  <boost_spans>
    <span id="smg_prid">
      <!--<param name="hold-music" value="$${moh_uri}"/>-->
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <!-- regex to stop dialing when it matches -->
      <!--<param name="dial-regex" value="5555"/>-->
      <!-- regex to stop dialing when it does not match -->
      <!--<param name="fail-dial-regex" value="^5"/>-->
    </span>
  </boost_spans>
  <analog_spans>
  </analog_spans>
</configuration>
 
wanpipe.conf:
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
 
Thanks and Best Regards,
Jerry
 


  _____  

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Tuesday, January 05, 2010 4:33 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER




On Mon, Jan 4, 2010 at 3:49 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:



Hello,

I have one FS instance that is working well with a PRI and running FS
version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
upgraded it.

Now, I am trying to bring up another FS instance (basically a clone of the
first), but the PRI does not work.  When I attempt to make an
internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
both servers are using the latest Sangoma Wanpipe driver, and I copied the
conf XML file tree from the old server to the new one.  I think the problem
has to do with the openzap module, but I'm having difficulty isolating the
problem.  Could it have built the openzap module incorrectly?  Another
difference is that I installed 1.0.5pre9 from scratch on the new server
(i.e. it never had 1.0.4 running on it).

I put the FS log into the pastebin when an outbound call attempt is made:

http://pastebin.freeswitch.org/11675

Could someone give me a pointer on what to try next?



Jerry, I noticed this line:
(OpenZAP/1:1/3491028 at g1)

Is your carrier wanting full ten digit phone numbers? Try adding the area
code on this and see what happens. The error usually would be something like
"invalid number format" but I've seen carriers do stupid things like this.
Try that first and see if it makes a difference. If not you'll need to turn
on Q931 debugging as per the Sangoma wiki. (See
http://wiki.freeswitch.org/wiki/OpenZAP#Debugging_PRI_With_wanpipemon for
the link.)

-MC



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From msc at freeswitch.org  Wed Jan  6 11:58:06 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Jan 2010 11:58:06 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <BE608CAF54594CA48BB8B4DB82A8BFD6@greyhawk.tonecommander.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>
	<BE608CAF54594CA48BB8B4DB82A8BFD6@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91001061158h4420961aj2878dc813d5637aa@mail.gmail.com>

I have an A104D on a box here. I'll need some time but I will see if I can
reproduce your symptoms. I'll let you know what I find out.

-MC

On Wed, Jan 6, 2010 at 9:59 AM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  When I attempt an internal-to-PSTN call, there are no Q931 packets sent
> out the PRI (I confirmed this using the Sangoma wanpipemon utility).  I
> suspect this has something to do with my XML configuration.  Below are my
> openzap/wanpipe configurations (which should all be defaulted).  Do you
> seen anything wrong with these defaults, which might cause the following FS
> console error?
>
> zap_io.c:1197 outgoing_call method not implemented!
>

>
> *openzap.conf:
> *[span wanpipe smg_prid]
> name => smg_prid
> trunk_type =>t1
> b-channel => 1:1-23
>
> *autoload_configs/openzap.conf.xml:*
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>   <settings>
>     <param name="debug" value="9"/>
>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>     <!--<param name="enable-analog-option" value="3-way"/>-->
>   </settings>
>   <boost_spans>
>     <span id="smg_prid">
>       <!--<param name="hold-music" value="$${moh_uri}"/>-->
>       <param name="dialplan" value="XML"/>
>       <param name="context" value="default"/>
>       <!-- regex to stop dialing when it matches -->
>       <!--<param name="dial-regex" value="5555"/>-->
>       <!-- regex to stop dialing when it does not match -->
>       <!--<param name="fail-dial-regex" value="^5"/>-->
>     </span>
>   </boost_spans>
>   <analog_spans>
>   </analog_spans>
> </configuration>
>
> *wanpipe.conf:*
> [defaults]
> codec_ms => 20
> wink_ms => 150
> flash_ms => 750
>
> Thanks and Best Regards,
> Jerry
>
>
>  ------------------------------
> *From:* Michael Collins [mailto:msc at freeswitch.org]
> *Sent:* Tuesday, January 05, 2010 4:33 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
>
>
>
> On Mon, Jan 4, 2010 at 3:49 PM, Jerry Richards <jerry.richards at teotech.com
> > wrote:
>
>>
>> Hello,
>>
>> I have one FS instance that is working well with a PRI and running FS
>> version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
>> upgraded it.
>>
>> Now, I am trying to bring up another FS instance (basically a clone of the
>> first), but the PRI does not work.  When I attempt to make an
>> internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
>> both servers are using the latest Sangoma Wanpipe driver, and I copied the
>> conf XML file tree from the old server to the new one.  I think the
>> problem
>> has to do with the openzap module, but I'm having difficulty isolating the
>> problem.  Could it have built the openzap module incorrectly?  Another
>> difference is that I installed 1.0.5pre9 from scratch on the new server
>> (i.e. it never had 1.0.4 running on it).
>>
>> I put the FS log into the pastebin when an outbound call attempt is made:
>>
>> http://pastebin.freeswitch.org/11675
>>
>> Could someone give me a pointer on what to try next?
>>
>
> Jerry, I noticed this line:
> (OpenZAP/1:1/3491028 at g1)
>
> Is your carrier wanting full ten digit phone numbers? Try adding the area
> code on this and see what happens. The error usually would be something like
> "invalid number format" but I've seen carriers do stupid things like this.
> Try that first and see if it makes a difference. If not you'll need to turn
> on Q931 debugging as per the Sangoma wiki. (See
> http://wiki.freeswitch.org/wiki/OpenZAP#Debugging_PRI_With_wanpipemon for
> the link.)
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Jan  6 11:58:08 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jan 2010 13:58:08 -0600
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <8441EC83BCD54223A5DAE91BEEC9C1FF@greyhawk.tonecommander.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>
	<8441EC83BCD54223A5DAE91BEEC9C1FF@greyhawk.tonecommander.com>
Message-ID: <7C331461-5692-47E7-B86F-81DD19904185@freeswitch.org>

You need to make sure you have SCTP libs installed and dev env to compile it.

/b

On Jan 6, 2010, at 1:52 PM, Jerry Richards wrote:

> I capured logs of my FS startup and put them into the pastebin (http://pastebin.freeswitch.org/11692).  At line 722, I see some errors:
>  
> 2010-01-06 11:42:49.907861 [ERR] zap_io.c:2562 Error loading /usr/local/freeswitch/mod/ozmod_sangoma_boost.so [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object file: No such file or directory]
> 2010-01-06 11:42:49.907883 [ERR] zap_io.c:2722 can't find 'sangoma_boost
> 2010-01-06 11:42:49.907902 [ERR] mod_openzap.c:2379 Error starting OpenZAP span 1 error
> 2010-01-06 11:42:49.907929 [CONSOLE] switch_loadable_module.c:890 Successfully Loaded [mod_openzap]
>  
> Do you know why I would get this?  Where is the ozmod_sangoma_boost.so file supposed to come from?
>  
> Thanks,
> Jerry
>  

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From Russell.Mosemann at cune.org  Wed Jan  6 12:04:26 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Wed, 6 Jan 2010 14:04:26 -0600
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <87f2f3b91001061158h4420961aj2878dc813d5637aa@mail.gmail.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com><87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com><BE608CAF54594CA48BB8B4DB82A8BFD6@greyhawk.tonecommander.com>
	<87f2f3b91001061158h4420961aj2878dc813d5637aa@mail.gmail.com>
Message-ID: <C2934F2F571143B5BD99457FB3C0475D@cune.pri>

> 	openzap.conf:
> 	[span wanpipe smg_prid]
> 	name => smg_prid
> 	trunk_type =>t1
> 	b-channel => 1:1-23

Do you miss a line when you copied the lines, or is a D channel not defined?

--
Russell Mosemann




From jmesquita at freeswitch.org  Wed Jan  6 14:16:44 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 6 Jan 2010 20:16:44 -0200
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <8ccbff061001060859p59aa6b5bw6f52275650f9138@mail.gmail.com>
References: <201001052125.06909.sos@sokhapkin.dyndns.org>
	<4B4417B3.9090807@aastral.net>
	<201001060620.13735.sos@sokhapkin.dyndns.org>
	<8ccbff061001060859p59aa6b5bw6f52275650f9138@mail.gmail.com>
Message-ID: <ebbc2d551001061416l25c17143gafe62de35cc6dee0@mail.gmail.com>

Why can't someone just sponsor some love to the module? The author has his
email on the header or open a Jira asking stuff so we have nibblebill more
mature.

Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927


On Wed, Jan 6, 2010 at 2:59 PM, Dome Charoenyost <dome at tel.co.th> wrote:

> 2010/1/6 Sergey Okhapkin <sos at sokhapkin.dyndns.org>:
> > nibblebill has no concept of billing blocks. What if I want to bill
> customer
> > 30 seconds minimum and 6 seconds increment thereafter?
>
> I have billing (in house develop) and customize  nibble_bill update
> cdr table (in my billing) i use postgresql trigger to update account
> balance. i have many increment rule 1/1 30/6 60/60
>
> It's work well for me :)
>
>
> BG
>
> Dome C.
> >
> > On Tuesday 05 January 2010, Bill W. wrote:
> >> Hey Sergey,
> >>
> >> But nibblebill will transfer to an extension of your choice when the
> >> balance reaches $0. So if you set the nibble heartbeat to 60 seconds or
> >> whatever, nibblebill will deduct the appropriate amount every
> >> <heartbeat> seconds.   So after about 20 minutes, the call will execute
> >> the nobal_action specified in nibblebill.conf.xml.
> >>
> >> So that should meet your needs.
> >>
> >> Bill W.
> >>
> >> Sergey Okhapkin wrote:
> >> > Unfortunalely, mod_nibblebill doesn't take billing increments into
> >> > account.
> >> >
> >> > On Tuesday 05 January 2010, Rupa Schomaker wrote:
> >> >> Look at using mod_nibblebill
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jerry.richards at teotech.com  Wed Jan  6 14:22:21 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 6 Jan 2010 14:22:21 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com> 
Message-ID: <4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>

By the way, I posted another pastebin log
(http://pastebin.freeswitch.org/11694) that shows the output of the "make"
and "make install sounds-install moh-install..." commands.  Just prior to
these makes I executed "make clean" and "make distclean".  You will notice
in the makefile output that ozmod_sangoma_boost.c never appears to get
compiled.  Shouldn't everything be compiled in this case?  I also confirmed
that my working server does have the ozmod_sangoma_boost.so file located in
the right place, which is why it's working.
 
Could version 1.0.5pre9 introduced this bug?  Could that be why the older
server (which original ran with 1.0.4) works and the new one doesn't?
 
Thanks and Best Regards,
Jerry
 
 


  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, January 06, 2010 11:52 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] DESTINATION_OUT_OF_ORDER


I capured logs of my FS startup and put them into the pastebin
(http://pastebin.freeswitch.org/11692).  At line 722, I see some errors:
 
2010-01-06 11:42:49.907861 [ERR] zap_io.c:2562 Error loading
/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
[/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
file: No such file or directory]
2010-01-06 11:42:49.907883 [ERR] zap_io.c:2722 can't find 'sangoma_boost
2010-01-06 11:42:49.907902 [ERR] mod_openzap.c:2379 Error starting OpenZAP
span 1 error
2010-01-06 11:42:49.907929 [CONSOLE] switch_loadable_module.c:890
Successfully Loaded [mod_openzap]
 
Do you know why I would get this?  Where is the ozmod_sangoma_boost.so file
supposed to come from?
 
Thanks,
Jerry
 
 


  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, January 06, 2010 10:00 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] DESTINATION_OUT_OF_ORDER


When I attempt an internal-to-PSTN call, there are no Q931 packets sent out
the PRI (I confirmed this using the Sangoma wanpipemon utility).  I suspect
this has something to do with my XML configuration.  Below are my
openzap/wanpipe configurations (which should all be defaulted).  Do you seen
anything wrong with these defaults, which might cause the following FS
console error?
 
zap_io.c:1197 outgoing_call method not implemented!
 
openzap.conf:
[span wanpipe smg_prid]
name => smg_prid
trunk_type =>t1
b-channel => 1:1-23
 
autoload_configs/openzap.conf.xml:
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="9"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
  <boost_spans>
    <span id="smg_prid">
      <!--<param name="hold-music" value="$${moh_uri}"/>-->
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <!-- regex to stop dialing when it matches -->
      <!--<param name="dial-regex" value="5555"/>-->
      <!-- regex to stop dialing when it does not match -->
      <!--<param name="fail-dial-regex" value="^5"/>-->
    </span>
  </boost_spans>
  <analog_spans>
  </analog_spans>
</configuration>
 
wanpipe.conf:
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
 
Thanks and Best Regards,
Jerry
 


  _____  

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Tuesday, January 05, 2010 4:33 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER




On Mon, Jan 4, 2010 at 3:49 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:



Hello,

I have one FS instance that is working well with a PRI and running FS
version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
upgraded it.

Now, I am trying to bring up another FS instance (basically a clone of the
first), but the PRI does not work.  When I attempt to make an
internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
both servers are using the latest Sangoma Wanpipe driver, and I copied the
conf XML file tree from the old server to the new one.  I think the problem
has to do with the openzap module, but I'm having difficulty isolating the
problem.  Could it have built the openzap module incorrectly?  Another
difference is that I installed 1.0.5pre9 from scratch on the new server
(i.e. it never had 1.0.4 running on it).

I put the FS log into the pastebin when an outbound call attempt is made:

http://pastebin.freeswitch.org/11675

Could someone give me a pointer on what to try next?



Jerry, I noticed this line:
(OpenZAP/1:1/3491028 at g1)

Is your carrier wanting full ten digit phone numbers? Try adding the area
code on this and see what happens. The error usually would be something like
"invalid number format" but I've seen carriers do stupid things like this.
Try that first and see if it makes a difference. If not you'll need to turn
on Q931 debugging as per the Sangoma wiki. (See
http://wiki.freeswitch.org/wiki/OpenZAP#Debugging_PRI_With_wanpipemon for
the link.)

-MC



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From brian at freeswitch.org  Wed Jan  6 14:32:37 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jan 2010 16:32:37 -0600
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>
	<4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
Message-ID: <407CE59D-5591-4838-B3D7-A0EC674844F7@freeswitch.org>

Again if you did not have the SCTP libs and dev headers installed when you did ./configure you'll never get this working SCTP is required to have boost work.  Also don't use Pre9 get the latest from latest.freeswitch.org and run with that please.

/b

On Jan 6, 2010, at 4:22 PM, Jerry Richards wrote:

> By the way, I posted another pastebin log (http://pastebin.freeswitch.org/11694) that shows the output of the "make" and "make install sounds-install moh-install..." commands.  Just prior to these makes I executed "make clean" and "make distclean".  You will notice in the makefile output that ozmod_sangoma_boost.c never appears to get compiled.  Shouldn't everything be compiled in this case?  I also confirmed that my working server does have the ozmod_sangoma_boost.so file located in the right place, which is why it's working.
>  
> Could version 1.0.5pre9 introduced this bug?  Could that be why the older server (which original ran with 1.0.4) works and the new one doesn't?
>  
> Thanks and Best Regards,
> Jerry
>  

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From brian at freeswitch.org  Wed Jan  6 14:33:34 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jan 2010 16:33:34 -0600
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <ebbc2d551001061416l25c17143gafe62de35cc6dee0@mail.gmail.com>
References: <201001052125.06909.sos@sokhapkin.dyndns.org>
	<4B4417B3.9090807@aastral.net>
	<201001060620.13735.sos@sokhapkin.dyndns.org>
	<8ccbff061001060859p59aa6b5bw6f52275650f9138@mail.gmail.com>
	<ebbc2d551001061416l25c17143gafe62de35cc6dee0@mail.gmail.com>
Message-ID: <5F37CA2E-129F-4082-B304-D50D0E1A4FAF@freeswitch.org>

Or better yet take over the maint. of the module if its making you money give back by providing some help to the project... its the ultimate way to give back.

/b

On Jan 6, 2010, at 4:16 PM, Jo?o Mesquita wrote:

> Why can't someone just sponsor some love to the module? The author has his email on the header or open a Jira asking stuff so we have nibblebill more mature.
> 
> Jo?o Mesquita
> FreeSWITCH? Solutions
> t: +1 (646) 4959927
> 




From jerry.richards at teotech.com  Wed Jan  6 15:37:48 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 6 Jan 2010 15:37:48 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com><87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>
	<4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
Message-ID: <536764FFF8C1495DB281DFABC9F02438@greyhawk.tonecommander.com>

Thank you for your suggestions.  Yes, I have the series of three
lksctp-tools-1.0.6 (SCTP) packages installed, and I do have the Development
Tools installed because the rest of the system is building okay.
 
Regarding the openzap.conf file, we have only 1 D-channel and 23 B-channels
and the openzap.conf file is auto-generated without a D-channel line, so I
don't think this is the issue.  Just to be sure, I did try adding a
D-channel line (d-channel => 1:1), but this produced another error (failed
to open wanpipe device span 1 channel 1) and the original error still
exists.
 
So I am trying to figure out why the openzap make is not compiling the
ozmod_sangoma_boost.c source file (http://pastebin.freeswitch.org/11694).
And I don't know why yet?
 
Thank You and Regards,
Jerry
 


  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, January 06, 2010 2:22 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER


By the way, I posted another pastebin log
(http://pastebin.freeswitch.org/11694) that shows the output of the "make"
and "make install sounds-install moh-install..." commands.  Just prior to
these makes I executed "make clean" and "make distclean".  You will notice
in the makefile output that ozmod_sangoma_boost.c never appears to get
compiled.  Shouldn't everything be compiled in this case?  I also confirmed
that my working server does have the ozmod_sangoma_boost.so file located in
the right place, which is why it's working.
 
Could version 1.0.5pre9 introduced this bug?  Could that be why the older
server (which original ran with 1.0.4) works and the new one doesn't?
 
Thanks and Best Regards,
Jerry
 
 


  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, January 06, 2010 11:52 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] DESTINATION_OUT_OF_ORDER


I capured logs of my FS startup and put them into the pastebin
(http://pastebin.freeswitch.org/11692).  At line 722, I see some errors:
 
2010-01-06 11:42:49.907861 [ERR] zap_io.c:2562 Error loading
/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
[/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
file: No such file or directory]
2010-01-06 11:42:49.907883 [ERR] zap_io.c:2722 can't find 'sangoma_boost
2010-01-06 11:42:49.907902 [ERR] mod_openzap.c:2379 Error starting OpenZAP
span 1 error
2010-01-06 11:42:49.907929 [CONSOLE] switch_loadable_module.c:890
Successfully Loaded [mod_openzap]
 
Do you know why I would get this?  Where is the ozmod_sangoma_boost.so file
supposed to come from?
 
Thanks,
Jerry
 
 


  _____  

From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, January 06, 2010 10:00 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] DESTINATION_OUT_OF_ORDER


When I attempt an internal-to-PSTN call, there are no Q931 packets sent out
the PRI (I confirmed this using the Sangoma wanpipemon utility).  I suspect
this has something to do with my XML configuration.  Below are my
openzap/wanpipe configurations (which should all be defaulted).  Do you seen
anything wrong with these defaults, which might cause the following FS
console error?
 
zap_io.c:1197 outgoing_call method not implemented!
 
openzap.conf:
[span wanpipe smg_prid]
name => smg_prid
trunk_type =>t1
b-channel => 1:1-23
 
autoload_configs/openzap.conf.xml:
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="9"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
  <boost_spans>
    <span id="smg_prid">
      <!--<param name="hold-music" value="$${moh_uri}"/>-->
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <!-- regex to stop dialing when it matches -->
      <!--<param name="dial-regex" value="5555"/>-->
      <!-- regex to stop dialing when it does not match -->
      <!--<param name="fail-dial-regex" value="^5"/>-->
    </span>
  </boost_spans>
  <analog_spans>
  </analog_spans>
</configuration>
 
wanpipe.conf:
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
 
Thanks and Best Regards,
Jerry
 


  _____  

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Tuesday, January 05, 2010 4:33 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER




On Mon, Jan 4, 2010 at 3:49 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:



Hello,

I have one FS instance that is working well with a PRI and running FS
version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
upgraded it.

Now, I am trying to bring up another FS instance (basically a clone of the
first), but the PRI does not work.  When I attempt to make an
internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
both servers are using the latest Sangoma Wanpipe driver, and I copied the
conf XML file tree from the old server to the new one.  I think the problem
has to do with the openzap module, but I'm having difficulty isolating the
problem.  Could it have built the openzap module incorrectly?  Another
difference is that I installed 1.0.5pre9 from scratch on the new server
(i.e. it never had 1.0.4 running on it).

I put the FS log into the pastebin when an outbound call attempt is made:

http://pastebin.freeswitch.org/11675

Could someone give me a pointer on what to try next?



Jerry, I noticed this line:
(OpenZAP/1:1/3491028 at g1)

Is your carrier wanting full ten digit phone numbers? Try adding the area
code on this and see what happens. The error usually would be something like
"invalid number format" but I've seen carriers do stupid things like this.
Try that first and see if it makes a difference. If not you'll need to turn
on Q931 debugging as per the Sangoma wiki. (See
http://wiki.freeswitch.org/wiki/OpenZAP#Debugging_PRI_With_wanpipemon for
the link.)

-MC



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From anthony.minessale at gmail.com  Wed Jan  6 15:49:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Jan 2010 17:49:24 -0600
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <536764FFF8C1495DB281DFABC9F02438@greyhawk.tonecommander.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>
	<4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
	<536764FFF8C1495DB281DFABC9F02438@greyhawk.tonecommander.com>
Message-ID: <191c3a031001061549n2b67cbcax89d93a89fc62b55e@mail.gmail.com>

ya, you should not use make distclean with any of our code, we do not have
it properly implemented since we have a very large and challenging build
system/


On Wed, Jan 6, 2010 at 5:37 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  Thank you for your suggestions.  Yes, I have the series of three
> lksctp-tools-1.0.6 (SCTP) packages installed, and I do have the
> Development Tools installed because the rest of the system is building
> okay.
>
> Regarding the openzap.conf file, we have only 1 D-channel and 23 B-channels
> and the openzap.conf file is auto-generated without a D-channel line, so I
> don't think this is the issue.  Just to be sure, I did try adding a
> D-channel line (d-channel => 1:1), but this produced another error (failed
> to open wanpipe device span 1 channel 1) and the original error still
> exists.
>
> So I am trying to figure out why the openzap make is not compiling the
> ozmod_sangoma_boost.c source file (http://pastebin.freeswitch.org/11694).
> And I don't know why yet?
>
> Thank You and Regards,
> Jerry
>
>
>  ------------------------------
> *From:* Jerry Richards [mailto:jerry.richards at teotech.com]
> *Sent:* Wednesday, January 06, 2010 2:22 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
>
>  By the way, I posted another pastebin log (
> http://pastebin.freeswitch.org/11694) that shows the output of the "make"
> and "make install sounds-install moh-install..." commands.  Just prior to
> these makes I executed "make clean" and "make distclean".  You will notice
> in the makefile output that ozmod_sangoma_boost.c never appears to get
> compiled.  Shouldn't everything be compiled in this case?  I also confirmed
> that my working server does have the ozmod_sangoma_boost.so file located in
> the right place, which is why it's working.
>
> Could version 1.0.5pre9 introduced this bug?  Could that be why the older
> server (which original ran with 1.0.4) works and the new one doesn't?
>
> Thanks and Best Regards,
> Jerry
>
>
>
>  ------------------------------
> *From:* Jerry Richards [mailto:jerry.richards at teotech.com]
> *Sent:* Wednesday, January 06, 2010 11:52 AM
> *To:* 'freeswitch-users at lists.freeswitch.org'
> *Subject:* RE: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
>
>  I capured logs of my FS startup and put them into the pastebin (
> http://pastebin.freeswitch.org/11692).  At line 722, I see some errors:
>
> 2010-01-06 11:42:49.907861 [ERR] zap_io.c:2562 Error loading
> /usr/local/freeswitch/mod/ozmod_sangoma_boost.so [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so:
> cannot open shared object file: No such file or directory]
> 2010-01-06 11:42:49.907883 [ERR] zap_io.c:2722 can't find 'sangoma_boost
> 2010-01-06 11:42:49.907902 [ERR] mod_openzap.c:2379 Error starting OpenZAP
> span 1 error
> 2010-01-06 11:42:49.907929 [CONSOLE] switch_loadable_module.c:890Successfully Loaded
> [mod_openzap]
>
> Do you know why I would get this?  Where is the ozmod_sangoma_boost.so file
> supposed to come from?
>
> Thanks,
> Jerry
>
>
>
>  ------------------------------
> *From:* Jerry Richards [mailto:jerry.richards at teotech.com]
> *Sent:* Wednesday, January 06, 2010 10:00 AM
> *To:* 'freeswitch-users at lists.freeswitch.org'
> *Subject:* RE: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
>
>  When I attempt an internal-to-PSTN call, there are no Q931 packets sent
> out the PRI (I confirmed this using the Sangoma wanpipemon utility).  I
> suspect this has something to do with my XML configuration.  Below are my
> openzap/wanpipe configurations (which should all be defaulted).  Do you
> seen anything wrong with these defaults, which might cause the following FS
> console error?
>
> zap_io.c:1197 outgoing_call method not implemented!
>
> *openzap.conf:
> *[span wanpipe smg_prid]
> name => smg_prid
> trunk_type =>t1
> b-channel => 1:1-23
>
> *autoload_configs/openzap.conf.xml:*
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>   <settings>
>     <param name="debug" value="9"/>
>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>     <!--<param name="enable-analog-option" value="3-way"/>-->
>   </settings>
>   <boost_spans>
>     <span id="smg_prid">
>       <!--<param name="hold-music" value="$${moh_uri}"/>-->
>       <param name="dialplan" value="XML"/>
>       <param name="context" value="default"/>
>       <!-- regex to stop dialing when it matches -->
>       <!--<param name="dial-regex" value="5555"/>-->
>       <!-- regex to stop dialing when it does not match -->
>       <!--<param name="fail-dial-regex" value="^5"/>-->
>     </span>
>   </boost_spans>
>   <analog_spans>
>   </analog_spans>
> </configuration>
>
> *wanpipe.conf:*
> [defaults]
> codec_ms => 20
> wink_ms => 150
> flash_ms => 750
>
> Thanks and Best Regards,
> Jerry
>
>
>  ------------------------------
> *From:* Michael Collins [mailto:msc at freeswitch.org]
> *Sent:* Tuesday, January 05, 2010 4:33 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
>
>
>
> On Mon, Jan 4, 2010 at 3:49 PM, Jerry Richards <jerry.richards at teotech.com
> > wrote:
>
>>
>> Hello,
>>
>> I have one FS instance that is working well with a PRI and running FS
>> version 1.0.5pre9.  Originally, it was running FS version 1.0.4 and I then
>> upgraded it.
>>
>> Now, I am trying to bring up another FS instance (basically a clone of the
>> first), but the PRI does not work.  When I attempt to make an
>> internal-to-PSTN call, I get a "502 Bad Gateway" reply.  I verified that
>> both servers are using the latest Sangoma Wanpipe driver, and I copied the
>> conf XML file tree from the old server to the new one.  I think the
>> problem
>> has to do with the openzap module, but I'm having difficulty isolating the
>> problem.  Could it have built the openzap module incorrectly?  Another
>> difference is that I installed 1.0.5pre9 from scratch on the new server
>> (i.e. it never had 1.0.4 running on it).
>>
>> I put the FS log into the pastebin when an outbound call attempt is made:
>>
>> http://pastebin.freeswitch.org/11675
>>
>> Could someone give me a pointer on what to try next?
>>
>
> Jerry, I noticed this line:
> (OpenZAP/1:1/3491028 at g1)
>
> Is your carrier wanting full ten digit phone numbers? Try adding the area
> code on this and see what happens. The error usually would be something like
> "invalid number format" but I've seen carriers do stupid things like this.
> Try that first and see if it makes a difference. If not you'll need to turn
> on Q931 debugging as per the Sangoma wiki. (See
> http://wiki.freeswitch.org/wiki/OpenZAP#Debugging_PRI_With_wanpipemon for
> the link.)
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mcampbellsmith at gmail.com  Wed Jan  6 16:43:42 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 7 Jan 2010 11:43:42 +1100
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <965454EF-F779-4563-AC64-082CAB07CCA5@freeswitch.org>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
	<191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
	<33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
	<0A8CD6A7-CA51-44F2-B530-15AE623EBA5D@freeswitch.org>
	<637054.57565.qm@web111313.mail.gq1.yahoo.com>
	<965454EF-F779-4563-AC64-082CAB07CCA5@freeswitch.org>
Message-ID: <33c87fa31001061643p4829e176leb294bec11e8b6c9@mail.gmail.com>

Thanks guys for your response.  I'll have a read through the links sent to me.

On the INVITE my FS box sends to Phonzo (the SIP provider) the codecs
it supports are G726-32, PCMU, PCMA.

 v=0
 o=FreeSWITCH 1262742568 1262742569 IN IP4 xxx.xxx.xxx.xx
 s=FreeSWITCH
 c=IN IP4 xxx.xxx.xxx.xxx
 t=0 0
 m=audio 31566 RTP/AVP 2 0 8 101 13
 a=rtpmap:2 G726-32/8000
 a=rtpmap:0 PCMU/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=rtpmap:13 CN/8000
 a=ptime:20

However, Phonzo responds with the following in the Session Progress message:

  v=0
  o=Sippy 158698636 1 IN IP4 80.232.37.178
  s=-
  t=0 0
  m=audio 61812 RTP/AVP 2 101 13
  c=IN IP4 213.50.91.3
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-15

Does that make sense?   The codec rtpmap only included DTMF ...

On Thu, Jan 7, 2010 at 2:04 AM, Brian West <brian at freeswitch.org> wrote:
> w00t! :)
> /b
> On Jan 6, 2010, at 1:29 AM, Arsen Chaloyan wrote:
>
> The following section in RFC3551 states the same
> http://tools.ietf.org/html/rfc3551#section-4.5.4
>
>
>    The encodings of G.726 are here denoted as G726-40, G726-32, G726-24,
>    and G726-16.  Prior to 1990, G721 described the 32 kbit/s ADPCM
>    encoding, and G723 described the 40, 32, and 16 kbit/s encodings.
>    Thus, G726-32 designates the same algorithm as G721 in RFC 1890.
>
>
> ________________________________
> From:?Brian West <brian at freeswitch.org>
> To:?freeswitch-users at lists.freeswitch.org
> Sent:?Wed, January 6, 2010 9:50:46 AM
> Subject:?Re: [Freeswitch-users] Codec Negotiation: Codec
> telephone-event:101:8000:20
>
> They are in fact one in the same please see ITU.
>
> /b
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Wed Jan  6 16:57:48 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jan 2010 18:57:48 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <33c87fa31001061643p4829e176leb294bec11e8b6c9@mail.gmail.com>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
	<191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
	<33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
	<0A8CD6A7-CA51-44F2-B530-15AE623EBA5D@freeswitch.org>
	<637054.57565.qm@web111313.mail.gq1.yahoo.com>
	<965454EF-F779-4563-AC64-082CAB07CCA5@freeswitch.org>
	<33c87fa31001061643p4829e176leb294bec11e8b6c9@mail.gmail.com>
Message-ID: <4DDDB005-6482-4495-881E-7242DB157082@freeswitch.org>

NO the codec map includes 2 aka g726-32 which you don't have to list in the map if its defined in the standard... what I need is a pcap of the whole process.

/b

On Jan 6, 2010, at 6:43 PM, Mark Campbell-Smith wrote:

> Thanks guys for your response.  I'll have a read through the links sent to me.
> 
> On the INVITE my FS box sends to Phonzo (the SIP provider) the codecs
> it supports are G726-32, PCMU, PCMA.
> 
> v=0
> o=FreeSWITCH 1262742568 1262742569 IN IP4 xxx.xxx.xxx.xx
> s=FreeSWITCH
> c=IN IP4 xxx.xxx.xxx.xxx
> t=0 0
> m=audio 31566 RTP/AVP 2 0 8 101 13
> a=rtpmap:2 G726-32/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
> 
> However, Phonzo responds with the following in the Session Progress message:
> 
>  v=0
>  o=Sippy 158698636 1 IN IP4 80.232.37.178
>  s=-
>  t=0 0
>  m=audio 61812 RTP/AVP 2 101 13
>  c=IN IP4 213.50.91.3
>  a=rtpmap:101 telephone-event/8000
>  a=fmtp:101 0-15
> 
> Does that make sense?   The codec rtpmap only included DTMF ...




From anthony.minessale at gmail.com  Wed Jan  6 17:51:12 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Jan 2010 19:51:12 -0600
Subject: [Freeswitch-users] Codec Negotiation: Codec
	telephone-event:101:8000:20
In-Reply-To: <4DDDB005-6482-4495-881E-7242DB157082@freeswitch.org>
References: <33c87fa31001051936g356920c1ve6f8d5d7d5d7568b@mail.gmail.com>
	<1C061ADA-25CC-4B6B-A4D0-711448E9A5F2@freeswitch.org>
	<33c87fa31001052103j55127d96j97cb21594c9e27e9@mail.gmail.com>
	<191c3a031001052120m3a2307fdy6facd50b7a01c8dd@mail.gmail.com>
	<33c87fa31001052140u81d1ff6se195541919f240a6@mail.gmail.com>
	<0A8CD6A7-CA51-44F2-B530-15AE623EBA5D@freeswitch.org>
	<637054.57565.qm@web111313.mail.gq1.yahoo.com>
	<965454EF-F779-4563-AC64-082CAB07CCA5@freeswitch.org>
	<33c87fa31001061643p4829e176leb294bec11e8b6c9@mail.gmail.com>
	<4DDDB005-6482-4495-881E-7242DB157082@freeswitch.org>
Message-ID: <191c3a031001061751w445324ebqbb34fd4b0fba08cf@mail.gmail.com>

when they only return 1, that means that is what they want you to use.
The call should establish in that case with g726-32


On Wed, Jan 6, 2010 at 6:57 PM, Brian West <brian at freeswitch.org> wrote:

> NO the codec map includes 2 aka g726-32 which you don't have to list in the
> map if its defined in the standard... what I need is a pcap of the whole
> process.
>
> /b
>
> On Jan 6, 2010, at 6:43 PM, Mark Campbell-Smith wrote:
>
> > Thanks guys for your response.  I'll have a read through the links sent
> to me.
> >
> > On the INVITE my FS box sends to Phonzo (the SIP provider) the codecs
> > it supports are G726-32, PCMU, PCMA.
> >
> > v=0
> > o=FreeSWITCH 1262742568 1262742569 IN IP4 xxx.xxx.xxx.xx
> > s=FreeSWITCH
> > c=IN IP4 xxx.xxx.xxx.xxx
> > t=0 0
> > m=audio 31566 RTP/AVP 2 0 8 101 13
> > a=rtpmap:2 G726-32/8000
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=rtpmap:13 CN/8000
> > a=ptime:20
> >
> > However, Phonzo responds with the following in the Session Progress
> message:
> >
> >  v=0
> >  o=Sippy 158698636 1 IN IP4 80.232.37.178
> >  s=-
> >  t=0 0
> >  m=audio 61812 RTP/AVP 2 101 13
> >  c=IN IP4 213.50.91.3
> >  a=rtpmap:101 telephone-event/8000
> >  a=fmtp:101 0-15
> >
> > Does that make sense?   The codec rtpmap only included DTMF ...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From Russell.Mosemann at cune.org  Wed Jan  6 18:16:33 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Wed, 6 Jan 2010 20:16:33 -0600
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <536764FFF8C1495DB281DFABC9F02438@greyhawk.tonecommander.com>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com><87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com><4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
	<536764FFF8C1495DB281DFABC9F02438@greyhawk.tonecommander.com>
Message-ID: <F677BB8D0A424932889541C56C96C78A@cune.pri>

Jerry Richards wrote:

> Regarding the openzap.conf file, we have only 1 D-channel and 23 B-
> channels and the openzap.conf file is auto-generated without a D-channel
> line, so I don't think this is the issue.  Just to be sure, I did try
> adding a D-channel line (d-channel => 1:1), but this produced another
> error (failed to open wanpipe device span 1 channel 1) and the original
> error still exists.

Channel 1 is already assigned to a B channel.

> b-channel => 1:1-23

You can't also assign it to a D channel. You can assign it to channel 24.

d-channel => 1:24

That's the same example that you will find on the wiki under "Wanpipe mode".

http://wiki.freeswitch.org/wiki/OpenZAP

--
Russell Mosemann




From brian at freeswitch.org  Wed Jan  6 18:25:53 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jan 2010 20:25:53 -0600
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <F677BB8D0A424932889541C56C96C78A@cune.pri>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com><87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com><4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
	<536764FFF8C1495DB281DFABC9F02438@greyhawk.tonecommander.com>
	<F677BB8D0A424932889541C56C96C78A@cune.pri>
Message-ID: <33827AB6-8B14-4F07-9C3E-36E155939ED2@freeswitch.org>

You don't define any d-channels when using boost.

/b

On Jan 6, 2010, at 8:16 PM, Russell Mosemann wrote:

> You can't also assign it to a D channel. You can assign it to channel 24.
> 
> d-channel => 1:24




From Russell.Mosemann at cune.org  Wed Jan  6 18:30:18 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Wed, 6 Jan 2010 20:30:18 -0600
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <33827AB6-8B14-4F07-9C3E-36E155939ED2@freeswitch.org>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com><87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com><4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com><536764FFF8C1495DB281DFABC9F02438@greyhawk.tonecommander.com><F677BB8D0A424932889541C56C96C78A@cune.pri>
	<33827AB6-8B14-4F07-9C3E-36E155939ED2@freeswitch.org>
Message-ID: <F5A693D296B54119AB9F51B5C05BEB21@cune.pri>

> You don't define any d-channels when using boost.

Thanks for the clarification.

--
Russell Mosemann




From darklion11 at yahoo.com  Wed Jan  6 21:04:31 2010
From: darklion11 at yahoo.com (Edmar Cruz)
Date: Wed, 6 Jan 2010 21:04:31 -0800 (PST)
Subject: [Freeswitch-users]  Script for Presence using PHP
Message-ID: <27050100.post@talk.nabble.com>


Dear Sir,

       Is there a script to trigger whether a user is online or offline
using a script or a PHP code?

       Presence for freeswitch is for command line only?


       Can you give me an example?

Thanks,
Edmar
-- 
View this message in context: http://old.nabble.com/Script-for-Presence-using-PHP-tp27050100p27050100.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From brian at freeswitch.org  Wed Jan  6 21:18:17 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jan 2010 23:18:17 -0600
Subject: [Freeswitch-users] Script for Presence using PHP
In-Reply-To: <27050100.post@talk.nabble.com>
References: <27050100.post@talk.nabble.com>
Message-ID: <A15C41BC-3548-4A2E-9BB6-FC79EB7A15C6@freeswitch.org>

see libs/esl/perl/*.pl should be similar to php in esl.

/b

On Jan 6, 2010, at 11:04 PM, Edmar Cruz wrote:

> 
> Dear Sir,
> 
>       Is there a script to trigger whether a user is online or offline
> using a script or a PHP code?
> 
>       Presence for freeswitch is for command line only?
> 
> 
>       Can you give me an example?
> 
> Thanks,
> Edmar




From msc at freeswitch.org  Wed Jan  6 22:59:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Jan 2010 22:59:10 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <F5A693D296B54119AB9F51B5C05BEB21@cune.pri>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com>
	<87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com>
	<4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
	<536764FFF8C1495DB281DFABC9F02438@greyhawk.tonecommander.com>
	<F677BB8D0A424932889541C56C96C78A@cune.pri>
	<33827AB6-8B14-4F07-9C3E-36E155939ED2@freeswitch.org>
	<F5A693D296B54119AB9F51B5C05BEB21@cune.pri>
Message-ID: <87f2f3b91001062259j7edbf029tacb7cdfa9156350@mail.gmail.com>

On Wed, Jan 6, 2010 at 6:30 PM, Russell Mosemann
<Russell.Mosemann at cune.org>wrote:

> > You don't define any d-channels when using boost.
>
> Thanks for the clarification.
>
FYI, I started the boost section under Sangoma on the wiki but got
distracted before finishing. I will get on that ASAP as soon as I have a
minute to get a boost setup all configured and running. Also, a tip o' the
hat to Moy at Sangoma who has been extremely responsive to my boost
questions in the past.

Thanks,
MC
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From msc at freeswitch.org  Wed Jan  6 23:03:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Jan 2010 23:03:16 -0800
Subject: [Freeswitch-users] encrypt voicemail emails and attachments?
In-Reply-To: <20100105171137.A433E1DB501@sinclaire.sibble.net>
References: <20100105042327.7CBF412DD@sinclaire.sibble.net>
	<03DDC308-7EC5-46EC-8AFB-CC73417E20F5@jerris.com>
	<20100105171137.A433E1DB501@sinclaire.sibble.net>
Message-ID: <87f2f3b91001062303n41871ff4x8c220d0564b27a50@mail.gmail.com>

On Tue, Jan 5, 2010 at 9:11 AM, Harondel J. Sibble <help at pdscc.com> wrote:

> That's what I suspected, thanks!
>

Let us know what you do so we can document it on the wiki.
Thanks,
MC
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From ken at ukgb.net  Thu Jan  7 01:17:21 2010
From: ken at ukgb.net (Ken Gillett)
Date: Thu, 7 Jan 2010 09:17:21 +0000
Subject: [Freeswitch-users] video
In-Reply-To: <90AC63C4-48ED-43F7-A534-CB90C208604F@ukgb.net>
References: <90AC63C4-48ED-43F7-A534-CB90C208604F@ukgb.net>
Message-ID: <9BFCBAD9-742A-41CD-8AB2-18E93B4A2A14@ukgb.net>

I'm still rather in the dark about what might loosely be termed 'SIP Video Phone' and in particular how it can relate to FreeSwitch.

	How does a video call work?
	Is it really a standard that governs this?
	What happens if the destination has no display?
	Does the originating camera only start streaming video when the call is started?
	Is the above hardware dependent?
	What would be a good video softphone for the Mac?

I have no prior knowledge of the working of such a video phone, but am trying to gain a better understanding. At least I want to know in what direction to research further, so hope someone can offer some basic ideas here.



Ken G i l l e t t

_/_/_/_/_/_/_/_/






From devel at thom.fr.eu.org  Thu Jan  7 01:55:17 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Thu, 07 Jan 2010 10:55:17 +0100
Subject: [Freeswitch-users] Zap channel not released when voicemail
	starts
Message-ID: <045f8df05054638c4a8e62e87026c060@thom.fr.eu.org>

 

Thanks. This works perfectly. 

Fran?ois 

On Tue, 5 Jan 2010 10:11:05
-0600, Anthony Minessale wrote:  

one way is to run tone_detect on the
busy signal and map it to the hangup app

 On Tue, Jan 5, 2010 at 3:44 AM,
Ahmed Naji  wrote:
 I'll second that.

My way of dealing with it has been
to write a little script to detect hangups on the TDM end, then force
release the corresponding "B-leg" that is hooked up to VM. In the process
of converting this to an FS module.

Not clean .. but works. Would have
liked to see the same code within FS core and, if appropriate, the VM
subsystem to achieve the same end.

Regards,

Ahmed.

 2010/1/5 Anthony
Minessale   
 hangup detection on TDM is a bitch.

 On Mon, Jan 4, 2010 at
10:07 AM, Fran?ois Legal  wrote:

Hello, 

I have an issue with voicemail
and openzap channels. 

When an incoming call on an openzap channel is
bridged to voicemail, if that channel is hung up before the beginning of
voicemail recording, that channel is kept open open until 3 or 4 seconds
after the voicemail started to record the message. 

What should I do to
make freeswitch/voicemail release the channel immediately when the caller
hang up ? 

Thanks in advance 

Fran?ois

_______________________________________________
 FreeSWITCH-users mailing
list
FreeSWITCH-users at lists.freeswitch.org
[4]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [5]

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[6]
http://www.freeswitch.org [7]

-- 
Anthony Minessale II

FreeSWITCH
http://www.freeswitch.org/ [8]
ClueCon http://www.cluecon.com/ [9]

Twitter: http://twitter.com/FreeSWITCH_wire [10]

AIM:
anthm
MSN:anthony_minessale at hotmail.com [11]

GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [12]
 IRC: irc.freenode.net
[13] #freeswitch

FreeSWITCH Developer
Conference
sip:888 at conference.freeswitch.org
[14]
iax:guest at conference.freeswitch.org/888
[15]
googletalk:conf+888 at conference.freeswitch.org
[16]
pstn:+19193869900

_______________________________________________

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
[17]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [18]

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[19]
http://www.freeswitch.org [20]

-- 
Ahmed
Naji

_______________________________________________
 FreeSWITCH-users
mailing list
FreeSWITCH-users at lists.freeswitch.org
[21]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [22]

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[23]
http://www.freeswitch.org [24]

-- 
Anthony Minessale II

FreeSWITCH
http://www.freeswitch.org/ [25]
ClueCon http://www.cluecon.com/ [26]

Twitter: http://twitter.com/FreeSWITCH_wire [27]

AIM:
anthm
MSN:anthony_minessale at hotmail.com
[28]
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [29]
 IRC:
irc.freenode.net [30] #freeswitch

FreeSWITCH Developer
Conference
sip:888 at conference.freeswitch.org
[31]
iax:guest at conference.freeswitch.org/888
[32]
googletalk:conf+888 at conference.freeswitch.org [33]
pstn:+19193869900




Links:
------
[1] mailto:a.alalousi at gmail.com
[2]
mailto:anthony.minessale at gmail.com
[3] mailto:devel at thom.fr.eu.org
[4]
mailto:FreeSWITCH-users at lists.freeswitch.org
[5]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[6]
http://lists.freeswitch.org/mailman/options/freeswitch-users
[7]
http://www.freeswitch.org
[8] http://www.freeswitch.org/
[9]
http://www.cluecon.com/
[10] http://twitter.com/FreeSWITCH_wire
[11]
mailto:MSN%3Aanthony_minessale at hotmail.com
[12]
mailto:PAYPAL%3Aanthony.minessale at gmail.com
[13]
http://irc.freenode.net
[14]
mailto:sip%3A888 at conference.freeswitch.org
[15]
http://iax:guest at conference.freeswitch.org/888
[16]
mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
[17]
mailto:FreeSWITCH-users at lists.freeswitch.org
[18]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[19]
http://lists.freeswitch.org/mailman/options/freeswitch-users
[20]
http://www.freeswitch.org
[21]
mailto:FreeSWITCH-users at lists.freeswitch.org
[22]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[23]
http://lists.freeswitch.org/mailman/options/freeswitch-users
[24]
http://www.freeswitch.org
[25] http://www.freeswitch.org/
[26]
http://www.cluecon.com/
[27] http://twitter.com/FreeSWITCH_wire
[28]
mailto:MSN%3Aanthony_minessale at hotmail.com
[29]
mailto:PAYPAL%3Aanthony.minessale at gmail.com
[30]
http://irc.freenode.net
[31]
mailto:sip%3A888 at conference.freeswitch.org
[32]
http://iax:guest at conference.freeswitch.org/888
[33]
mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
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From tzury.by at reguluslabs.com  Thu Jan  7 02:18:07 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Thu, 7 Jan 2010 12:18:07 +0200
Subject: [Freeswitch-users] a weired problem when calling pstn number(s)
	with openzap and libpri
Message-ID: <10128ef11001070218g1367dc9bvcb5576c9d1a1b4dd@mail.gmail.com>

Hi all,

I got a weired problem and I suspect it has to do with either openzap or
libpri.

My setup is simple FS box, and a Sangoma E1 (A101) card.

Everything seems to be well configured and working. Yet, sometimes, when I
call from the SIP the call failed, on the client I get "call ended" message.

On the server when looking at the log I see UNALLOCATED_NUMBER (line:274)

I can certainly tell that this number exists and alive and it is not a wrong
number.

to make it even more complicated, calling that problematic number over and
over, yields a statistics of working rate 5/1
that is every 5th or 4th call works

I placed here the logs of the working case and the not working case

http://gist.github.com/raw/271113/b8f13ef292669a1ab69878bfe205fc41d3722d0a/libpri-openzap-bridge-outbound-calls-working.cs<http://gist.github.com/raw/271113/b8f13ef292669a1ab69878bfe205fc41d3722d0a/libpri-openzap-bridge-outbound-calls-working.log>
http://gist.github.com/raw/271113/8b8b55922ebd8faffe1a850681f6a642dbdf4f04/libpri-openzap-bridge-outbound-calls-not-working.cs


please help,
Tzury
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From saeedahmad1981 at gmail.com  Thu Jan  7 02:35:31 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Thu, 7 Jan 2010 11:35:31 +0100
Subject: [Freeswitch-users] freeswitch and H323
In-Reply-To: <65d96fc81001051125u3a6c7d96q5ed39d39c0026107@mail.gmail.com>
References: <7aa8bd9d0912300514r68c90b12u7c631a649981cfa3@mail.gmail.com> 
	<87f2f3b90912310920q1821fe0eu5fd74e553e4cb12@mail.gmail.com> 
	<65d96fc81001010452k37068e87sc0a229cdfe481c40@mail.gmail.com> 
	<c96b95651001031636q1ecfec18x2bba1c22ff6ca9bb@mail.gmail.com> 
	<65d96fc81001051125u3a6c7d96q5ed39d39c0026107@mail.gmail.com>
Message-ID: <c96b95651001070235j2fe1a4b4m26da30aeb1861bb4@mail.gmail.com>

Thanks

On Tue, Jan 5, 2010 at 8:25 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> its already there: http://wiki.freeswitch.org/wiki/Mod_h323
>
> T.
>
>
> On Mon, Jan 4, 2010 at 1:36 AM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:
>
>> HI,
>>
>> It would be really nice if you can create a wiki page.
>>
>> Thanks
>>
>>
>> On Fri, Jan 1, 2010 at 1:52 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> well, mod_h323 works for me... there are still some missing things and of
>>> course bugs ... e.g. incorrect releaseCause mapping, no automatic codec
>>> ptime sync... but it is usable ....
>>>
>>>
>>> if you'd like to go mod_h323 way i can help you... it builds as a charm
>>> for me...
>>>
>>>
>>> T.
>>>
>>>
>>>
>>>
>>>
>>> On Thu, Dec 31, 2009 at 6:20 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Are you trying to use mod_h323 or mod_opal? They are both works in
>>>> progress, but the latter is farther along than the former. Use the latest
>>>> FreeSWITCH trunk (or latest.freeswitch.org) and run the buildopal.sh
>>>> script in the build directory. If you have any build issues then paste the
>>>> log on pastebin.freeswitch.org and reply to this thread with the PB URL
>>>> so that we can take a look.
>>>>  -MC
>>>>
>>>>
>>>> On Wed, Dec 30, 2009 at 5:14 AM, Pete Kay <petedao at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> has anyone been able to get H323 to work?
>>>>>
>>>>> I have problem trying to get it compiled with either 1.0.4 or 1.0.5.
>>>>>
>>>>> Thanks,
>>>>> pete
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From saeedahmad1981 at gmail.com  Thu Jan  7 03:00:50 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Thu, 7 Jan 2010 12:00:50 +0100
Subject: [Freeswitch-users] Installing freeswitch on CentOS
Message-ID: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>

Hi,

Since CentOS is recommend for FS but i can't see a CentOS specific
installation guide on wiki as we have a separate guide for Ubuntu.

Do we have similar guide like this one
http://wiki.freeswitch.org/wiki/SBC_Setup* *actually its for debian but good
thing is that it also explains which extra services should be stopped or
removed for better performance.

Do we have similar for CentOS?

Thanks
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From jcasale at activenetwerx.com  Thu Jan  7 04:02:56 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Thu, 7 Jan 2010 12:02:56 +0000
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>

>Since CentOS is recommend for FS but i can't see a CentOS specific installation guide on wiki as we have a separate >guide for Ubuntu.?
>
>Do we have similar guide like this one?http://wiki.freeswitch.org/wiki/SBC_Setup?actually its for debian but good thing >is that it also explains which extra >services should be stopped or removed for better performance.?
>
>Do we have similar for CentOS?

Check out the http://wiki.freeswitch.org/wiki/Installation_Guide





From freeswitch at peely.com  Thu Jan  7 06:56:00 2010
From: freeswitch at peely.com (peely)
Date: Thu, 7 Jan 2010 06:56:00 -0800 (PST)
Subject: [Freeswitch-users] Call through gateway without register > sends to
 gateway name?
Message-ID: <26979541.post@talk.nabble.com>


Hi,

I have a problem where I'm trying to send calls to a gateway that does not
support registration.

In my external sip profile directory I have a file containing:

<include>
    <gateway name="mygateway">
      
      
      
      
      
    </gateway>
</include>

Then in my dialplan I have a dialplan with an action of:

<action application="bridge" data="sofia/gateway/mygateway/1234567890"/>

However, when I call out, the Sofia diag shows:

sres_send_dns_query(0x7f19ac011150, 0x7f19a401cdf0) id=20898 NAPTR mygateway
(to [172.16.1.1]:53)

Could somebody please tell me how I get the gateway config to send INVITEs
to a specific IP?



Thanks,


Neil.

-- 
View this message in context: http://old.nabble.com/Call-through-gateway-without-register-%3E-sends-to-gateway-name--tp26979541p26979541.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From anthony.minessale at gmail.com  Thu Jan  7 07:14:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Jan 2010 09:14:50 -0600
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
Message-ID: <191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>

the big rule for the time being is stick with 5.3 5.4 appears to have some
bugs in the toolchain and libc

On Thu, Jan 7, 2010 at 6:02 AM, Joseph L. Casale
<jcasale at activenetwerx.com>wrote:

> >Since CentOS is recommend for FS but i can't see a CentOS specific
> installation guide on wiki as we have a separate >guide for Ubuntu.
> >
> >Do we have similar guide like this one
> http://wiki.freeswitch.org/wiki/SBC_Setup actually its for debian but good
> thing >is that it also explains which extra >services should be stopped or
> removed for better performance.
> >
> >Do we have similar for CentOS?
>
> Check out the http://wiki.freeswitch.org/wiki/Installation_Guide
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From nicolas at medularis.com  Thu Jan  7 07:43:11 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Thu, 7 Jan 2010 12:43:11 -0300
Subject: [Freeswitch-users] Calls getting queued?
Message-ID: <1b46b4e81001070743v5389ef3ewb75dbeb06b402c9a@mail.gmail.com>

Hi, I'm having a strange problem with FS. I'm using a few JS scripts
to generate calls and bridge them together. Usually everything works
just fine, but them at some point it's like if FS choked, calls for
the first leg of the bridges are apparently made, but the second leg
is never called. The call is not hanged up for several minutes and the
system keeps opening new channels but never connecting a call.

For example, right now, doing 'show channels' on the console, I get a
list of 72 open channels (it's adding up, it was 40 a couple minutes
ago), but doing a 'show calls' gives me 0 active calls. The usual
behavior, when everything's working fine, is to get twice as many
channels as there are active calls and no channels at all when there
are no calls, unless they haven't been bridged yet.

The originate string is something like this:

var stUsRing = "%(2000,4000,440,480)";
var timeout = 45;
originate_str1 = "{api_hangup_hook=jsapi::callback.js
l1,execute_on_answer=lua answered.lua 1
c2c_call,ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,ringback='"+stUsRing+"',medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]"+dialstr1;

Where diasltr1 has the phonenumber and and gateway info. The
callback.js has a curl request to update some call info on an external
database and answered.lua calls a ruby script through the os.execute()
function (I know, I should be doing all this through the event socket,
I was doing that but had trouble and had to come up with a quick
solution).

The system is not loaded at all, at least not for what I think and
read that FS can handle. We are having at most 10 concurrent calls (20
channels), with maybe 5 to 10 calls per minute.

What worries me is not only that I don't know where the problem is,
but that I have no clue how to debug it or send you guys more
"lowlevel" and detailed information to give you an insight about
what's going on. Any help would be greatly appreciated!

Thanks!

Nico



From dujinfang at gmail.com  Thu Jan  7 08:11:21 2010
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 8 Jan 2010 00:11:21 +0800
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org>
	<2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org>
	<B732197E-1123-4738-A69F-F84206FCE38A@freeswitch.org>
	<0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org>
	<0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org>
	<23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com>
	<60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org>
	<23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com>
	<ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com>
Message-ID: <23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com>

:lol. I do like to involve into this. I saw you have done a lot of
works. I read some code and here are some questions:

1) what's your nick on IRC? I'm seven(or seven_ ?)
2) Are you developing on Windows? How can I compile on Mac(I have no
experience on QT)?
3) Does is needs fsGUI? I downloaded fsGUI dmg file and it complains
no mods on start.


2009/12/30, Jo?o Mesquita <jmesquita at freeswitch.org>:
> What is JM is not the question but rather WHO is JM and that would be me.
> :-)
>
> I have already stripped down the config handler based on mod_xml_curl. I
> have been discussing with Brian how to make it happen and I am conducting a
> couple of tests with Qt. Today I might be able to have it properly linked
> with Qt and the core spawn on its own thread inside the Qt event loop. I'll
> keep you posted.
>
> Jo?o Mesquita A.K.A -> JM
>
>
> On Wed, Dec 30, 2009 at 12:17 PM, Seven Du <dujinfang at gmail.com> wrote:
>
>> I rarely joined in IRC, becuase I live in China, timezone +8000 ....
>> I really would like to start the official softphone, btw, what is JM?
>>
>> 2009/12/30, Brian West <brian at freeswitch.org>:
>> > You should join IRC and join in JM and really start the official
>> softphone
>> > project.
>> >
>> > /b
>> >
>> > On Dec 29, 2009, at 7:36 PM, Seven Du wrote:
>> >
>> >> I had wrote a Air based GUI, is it make sense?
>> >>
>> >> http://wiki.freeswitch.org/wiki/FsAir
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>



From brian at freeswitch.org  Thu Jan  7 08:25:11 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jan 2010 10:25:11 -0600
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org>
	<2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org>
	<B732197E-1123-4738-A69F-F84206FCE38A@freeswitch.org>
	<0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org>
	<0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org>
	<23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com>
	<60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org>
	<23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com>
	<ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com>
	<23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com>
Message-ID: <B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>

Lets schedule FSComm on the weekly conference call... We need people to step up and take some roles in both FreeSWITCH and FSComm projects... Even if its just testing bugs and collecting info.

Thanks,
Brian

On Jan 7, 2010, at 10:11 AM, Seven Du wrote:

> :lol. I do like to involve into this. I saw you have done a lot of
> works. I read some code and here are some questions:
> 
> 1) what's your nick on IRC? I'm seven(or seven_ ?)
> 2) Are you developing on Windows? How can I compile on Mac(I have no
> experience on QT)?
> 3) Does is needs fsGUI? I downloaded fsGUI dmg file and it complains
> no mods on start.




From oscav at hotmail.fr  Thu Jan  7 08:34:22 2010
From: oscav at hotmail.fr (Oscav)
Date: Thu, 7 Jan 2010 08:34:22 -0800 (PST)
Subject: [Freeswitch-users]  Failed to connect to a SKYPE API
Message-ID: <27062783.post@talk.nabble.com>


Hi,

I'm trying to use to SkypeIAX. When I load the mod_skypiax, I got the
following error :

 Failed to connect to a SKYPE API for interface_id=1, no SKYPE client
running, please (re)start Skype client. Skypiax exiting

Skype is running with the correct account and skypiax.conf use the same
account. I was expecting a permission request from the Skype user but
nothing happens.

Somebody knows how I can solve this ??

Many thanks.
-- 
View this message in context: http://old.nabble.com/Failed-to-connect-to-a-SKYPE-API-tp27062783p27062783.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From Claudio.Cavalera at italtel.it  Thu Jan  7 08:46:24 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Thu, 7 Jan 2010 17:46:24 +0100
Subject: [Freeswitch-users] Codecs and things
In-Reply-To: <4B3229C2.4080109@coppice.org>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F362B8@BESONE.corp.dom>

> The G.729 codec for FS is in testing, and should be out so. If you
> really want to implement your own, TI DSP code is unlikely to
> be a good
> starting point. I assume that code is fixed point. You really need a
> floating point codec to get any decent speed on a PC. Pentiums and
> Athlons lack saturating arithmetic (MMX actually has partial
> saturating arithmetic, but it isn't much use for anything but image
> processing), so a fixed point implementation ends up very slow.
> 

Hello Steve,
from what you have written it seems very unlikely that we are gonna buy
the official G.729 codec for embedded hardware?
I don't know much about it but would a MIPS32 24kf be enough? Just
speculating from here
http://www.mips.com/products/processors/32-64-bit-cores/mips32-24k/

Thanks,
Claudio


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From brian at freeswitch.org  Thu Jan  7 08:56:55 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jan 2010 10:56:55 -0600
Subject: [Freeswitch-users] Codecs and things
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0F362B8@BESONE.corp.dom>
References: <A2167AE7633A5C4DAE56EEBD1428ECC0F362B8@BESONE.corp.dom>
Message-ID: <7A65B817-E490-4582-8D43-8531FFA61CC4@freeswitch.org>

You usually still have to pay a license even if you buy a DSP that is capable of doing it.  

/b

On Jan 7, 2010, at 10:46 AM, Cavalera Claudio Luigi wrote:

> Hello Steve,
> from what you have written it seems very unlikely that we are gonna buy
> the official G.729 codec for embedded hardware?
> I don't know much about it but would a MIPS32 24kf be enough? Just
> speculating from here
> http://www.mips.com/products/processors/32-64-bit-cores/mips32-24k/
> 
> Thanks,
> Claudio

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From msc at freeswitch.org  Thu Jan  7 09:20:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Jan 2010 09:20:07 -0800
Subject: [Freeswitch-users] Call through gateway without register >
	sends to gateway name?
In-Reply-To: <26979541.post@talk.nabble.com>
References: <26979541.post@talk.nabble.com>
Message-ID: <87f2f3b91001070920r12028425k266ba050d9cec7f8@mail.gmail.com>

On Thu, Jan 7, 2010 at 6:56 AM, peely <freeswitch at peely.com> wrote:

>
> Hi,
>
> I have a problem where I'm trying to send calls to a gateway that does not
> support registration.
>
> In my external sip profile directory I have a file containing:
>
> <include>
>    <gateway name="mygateway">
>
>
>
>
>
>    </gateway>
> </include>
>
> Then in my dialplan I have a dialplan with an action of:
>
> <action application="bridge" data="sofia/gateway/mygateway/1234567890"/>
>
> However, when I call out, the Sofia diag shows:
>
> sres_send_dns_query(0x7f19ac011150, 0x7f19a401cdf0) id=20898 NAPTR
> mygateway
> (to [172.16.1.1]:53)
>
> Could somebody please tell me how I get the gateway config to send INVITEs
> to a specific IP?
>
>
Is that your actual gateway definition? I see a bunch of blank lines and I
don't know if that's a typo or what. Be sure that you have this in your
gateway:
<param name="register" value="false"/>

Of course, you still need the username and password fields, even if they
just have dummy values, plus you should have the realm specified. If you're
still having issues then pastebin your whole gateway file (redacting private
info) and also a capture of a debug trace of a failed call from start to
finish.

-MC
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From msc at freeswitch.org  Thu Jan  7 09:26:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Jan 2010 09:26:31 -0800
Subject: [Freeswitch-users] Calls getting queued?
In-Reply-To: <1b46b4e81001070743v5389ef3ewb75dbeb06b402c9a@mail.gmail.com>
References: <1b46b4e81001070743v5389ef3ewb75dbeb06b402c9a@mail.gmail.com>
Message-ID: <87f2f3b91001070926j60139f2cxe0ae9687e25a302e@mail.gmail.com>

On Thu, Jan 7, 2010 at 7:43 AM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Hi, I'm having a strange problem with FS. I'm using a few JS scripts
> to generate calls and bridge them together. Usually everything works
> just fine, but them at some point it's like if FS choked, calls for
> the first leg of the bridges are apparently made, but the second leg
> is never called. The call is not hanged up for several minutes and the
> system keeps opening new channels but never connecting a call.
>
> For example, right now, doing 'show channels' on the console, I get a
> list of 72 open channels (it's adding up, it was 40 a couple minutes
> ago), but doing a 'show calls' gives me 0 active calls. The usual
> behavior, when everything's working fine, is to get twice as many
> channels as there are active calls and no channels at all when there
> are no calls, unless they haven't been bridged yet.
>
> The originate string is something like this:
>
> var stUsRing = "%(2000,4000,440,480)";
> var timeout = 45;
> originate_str1 = "{api_hangup_hook=jsapi::callback.js
> l1,execute_on_answer=lua answered.lua 1
>
> c2c_call,ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,ringback='"+stUsRing+"',medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]"+dialstr1;
>
> Where diasltr1 has the phonenumber and and gateway info. The
> callback.js has a curl request to update some call info on an external
> database and answered.lua calls a ruby script through the os.execute()
> function (I know, I should be doing all this through the event socket,
> I was doing that but had trouble and had to come up with a quick
> solution).
>
> The system is not loaded at all, at least not for what I think and
> read that FS can handle. We are having at most 10 concurrent calls (20
> channels), with maybe 5 to 10 calls per minute.
>
> What worries me is not only that I don't know where the problem is,
> but that I have no clue how to debug it or send you guys more
> "lowlevel" and detailed information to give you an insight about
> what's going on. Any help would be greatly appreciated!
>
> Thanks!
>
> Nico
>
> First off you'll want to get familiar with the resources mentioned here:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

It has good tips on how to collect and report information.

Second, I recommend that you pastebin your relevant portion of the dialplan
and the whole javascript program that you are using so that others can take
a look.

Last thing: if you restart FreeSWITCH does everything work fine for a while
but then eventually it breaks down and exhibits the behavior that you are
reporting?

-MC
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From neil.burgess at redmatter.com  Thu Jan  7 07:52:52 2010
From: neil.burgess at redmatter.com (Neil Burgess)
Date: Thu, 7 Jan 2010 15:52:52 +0000
Subject: [Freeswitch-users] UNIMRCP RECOGNIZE headers
Message-ID: <787302A89ACCE24DA8F56DA101E77C841392F8C2D8@THHS2E12BE1X.hostedservice2.net>

Hello,

Wondering if anyone can help with a unimrcp question.

We are using the UNIMRCP client in FreeSwitch to communicate with a commercial UNIMRCP server (Loquendo). We need to control such items as Speech Timeouts that are occurring on the server side, however we have been advised that the only way these can be affected is via  headers in the MRCP requests. So, we need to be able to set headers such as Speech-Complete-Timeout, Speech-Incomplete-Timeout, Recognition-Timeout, etc in the MRCP RECOGNIZE request.  Is there a mechanism in FreeSwitch which we can use to pass these down before, (or as) we issue the "detect_speech" command. We are happy to use a jscript interface, or whatever if such a capability is available.

Many thanks,
Neil
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From nicolas at medularis.com  Thu Jan  7 10:12:25 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Thu, 7 Jan 2010 15:12:25 -0300
Subject: [Freeswitch-users] Calls getting queued?
In-Reply-To: <87f2f3b91001070926j60139f2cxe0ae9687e25a302e@mail.gmail.com>
References: <1b46b4e81001070743v5389ef3ewb75dbeb06b402c9a@mail.gmail.com>
	<87f2f3b91001070926j60139f2cxe0ae9687e25a302e@mail.gmail.com>
Message-ID: <1b46b4e81001071012y237fb8c0jd68232137b7d8e96@mail.gmail.com>

Michael,

Thanks for your help. Yes, if I restart FS things go back to normal
for a while and then the same thing starts happening again.

The weird thing is, it started only 2 days ago, and happened only once
or twice. Before that I had no trouble, and I only made 1 change,
which I reverted, but it wasn't that. Today it's happening all the
time, if I restart FS things will work for maybe an hour and then it
will start doing the same thing.

I'm guessing it might be something external to FS, like curl calls not
finishing properly because of the url they are requesting or something
like that.

What kind of info should I collect? I don't think it has to do with
sofia or any sip-related problems. I'm also using the default
dialplan, no changes at all, I'm doing everything through JS, well and
one really small lua script.

This is the main JS file:
It originates 2 calls and bridges them.

- http://pastebin.freeswitch.org/11706


This is another JS script which gets called when each call is hanged up:
It gets some info and then requests a url using curl to update call
status on an external db.

- http://pastebin.freeswitch.org/11707


This lua script calls a ruby script to do some other stuff when a call
is answered:

- http://pastebin.freeswitch.org/11708


Thanks!


Nico



On Thu, Jan 7, 2010 at 2:26 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Thu, Jan 7, 2010 at 7:43 AM, Nicolas Brenner <nicolas at medularis.com>
> wrote:
>>
>> Hi, I'm having a strange problem with FS. I'm using a few JS scripts
>> to generate calls and bridge them together. Usually everything works
>> just fine, but them at some point it's like if FS choked, calls for
>> the first leg of the bridges are apparently made, but the second leg
>> is never called. The call is not hanged up for several minutes and the
>> system keeps opening new channels but never connecting a call.
>>
>> For example, right now, doing 'show channels' on the console, I get a
>> list of 72 open channels (it's adding up, it was 40 a couple minutes
>> ago), but doing a 'show calls' gives me 0 active calls. The usual
>> behavior, when everything's working fine, is to get twice as many
>> channels as there are active calls and no channels at all when there
>> are no calls, unless they haven't been bridged yet.
>>
>> The originate string is something like this:
>>
>> var stUsRing = "%(2000,4000,440,480)";
>> var timeout = 45;
>> originate_str1 = "{api_hangup_hook=jsapi::callback.js
>> l1,execute_on_answer=lua answered.lua 1
>>
>> c2c_call,ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,ringback='"+stUsRing+"',medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]"+dialstr1;
>>
>> Where diasltr1 has the phonenumber and and gateway info. The
>> callback.js has a curl request to update some call info on an external
>> database and answered.lua calls a ruby script through the os.execute()
>> function (I know, I should be doing all this through the event socket,
>> I was doing that but had trouble and had to come up with a quick
>> solution).
>>
>> The system is not loaded at all, at least not for what I think and
>> read that FS can handle. We are having at most 10 concurrent calls (20
>> channels), with maybe 5 to 10 calls per minute.
>>
>> What worries me is not only that I don't know where the problem is,
>> but that I have no clue how to debug it or send you guys more
>> "lowlevel" and detailed information to give you an insight about
>> what's going on. Any help would be greatly appreciated!
>>
>> Thanks!
>>
>> Nico
>>
> First off you'll want to get familiar with the resources mentioned here:
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>
> It has good tips on how to collect and report information.
>
> Second, I recommend that you pastebin your relevant portion of the dialplan
> and the whole javascript program that you are using so that others can take
> a look.
>
> Last thing: if you restart FreeSWITCH does everything work fine for a while
> but then eventually it breaks down and exhibits the behavior that you are
> reporting?
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From anthony.minessale at gmail.com  Thu Jan  7 10:22:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Jan 2010 12:22:03 -0600
Subject: [Freeswitch-users] Calls getting queued?
In-Reply-To: <1b46b4e81001071012y237fb8c0jd68232137b7d8e96@mail.gmail.com>
References: <1b46b4e81001070743v5389ef3ewb75dbeb06b402c9a@mail.gmail.com>
	<87f2f3b91001070926j60139f2cxe0ae9687e25a302e@mail.gmail.com>
	<1b46b4e81001071012y237fb8c0jd68232137b7d8e96@mail.gmail.com>
Message-ID: <191c3a031001071022j1e5796fdvf63900f5968bc01b@mail.gmail.com>

try setting the timeout in curl

conf/autoload_configs/xml_curl.conf.xml:
<param name="timeout" value="5"/>

On Thu, Jan 7, 2010 at 12:12 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Michael,
>
> Thanks for your help. Yes, if I restart FS things go back to normal
> for a while and then the same thing starts happening again.
>
> The weird thing is, it started only 2 days ago, and happened only once
> or twice. Before that I had no trouble, and I only made 1 change,
> which I reverted, but it wasn't that. Today it's happening all the
> time, if I restart FS things will work for maybe an hour and then it
> will start doing the same thing.
>
> I'm guessing it might be something external to FS, like curl calls not
> finishing properly because of the url they are requesting or something
> like that.
>
> What kind of info should I collect? I don't think it has to do with
> sofia or any sip-related problems. I'm also using the default
> dialplan, no changes at all, I'm doing everything through JS, well and
> one really small lua script.
>
> This is the main JS file:
> It originates 2 calls and bridges them.
>
> - http://pastebin.freeswitch.org/11706
>
>
> This is another JS script which gets called when each call is hanged up:
> It gets some info and then requests a url using curl to update call
> status on an external db.
>
> - http://pastebin.freeswitch.org/11707
>
>
> This lua script calls a ruby script to do some other stuff when a call
> is answered:
>
> - http://pastebin.freeswitch.org/11708
>
>
> Thanks!
>
>
> Nico
>
>
>
> On Thu, Jan 7, 2010 at 2:26 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >
> >
> > On Thu, Jan 7, 2010 at 7:43 AM, Nicolas Brenner <nicolas at medularis.com>
> > wrote:
> >>
> >> Hi, I'm having a strange problem with FS. I'm using a few JS scripts
> >> to generate calls and bridge them together. Usually everything works
> >> just fine, but them at some point it's like if FS choked, calls for
> >> the first leg of the bridges are apparently made, but the second leg
> >> is never called. The call is not hanged up for several minutes and the
> >> system keeps opening new channels but never connecting a call.
> >>
> >> For example, right now, doing 'show channels' on the console, I get a
> >> list of 72 open channels (it's adding up, it was 40 a couple minutes
> >> ago), but doing a 'show calls' gives me 0 active calls. The usual
> >> behavior, when everything's working fine, is to get twice as many
> >> channels as there are active calls and no channels at all when there
> >> are no calls, unless they haven't been bridged yet.
> >>
> >> The originate string is something like this:
> >>
> >> var stUsRing = "%(2000,4000,440,480)";
> >> var timeout = 45;
> >> originate_str1 = "{api_hangup_hook=jsapi::callback.js
> >> l1,execute_on_answer=lua answered.lua 1
> >>
> >>
> c2c_call,ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,ringback='"+stUsRing+"',medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]"+dialstr1;
> >>
> >> Where diasltr1 has the phonenumber and and gateway info. The
> >> callback.js has a curl request to update some call info on an external
> >> database and answered.lua calls a ruby script through the os.execute()
> >> function (I know, I should be doing all this through the event socket,
> >> I was doing that but had trouble and had to come up with a quick
> >> solution).
> >>
> >> The system is not loaded at all, at least not for what I think and
> >> read that FS can handle. We are having at most 10 concurrent calls (20
> >> channels), with maybe 5 to 10 calls per minute.
> >>
> >> What worries me is not only that I don't know where the problem is,
> >> but that I have no clue how to debug it or send you guys more
> >> "lowlevel" and detailed information to give you an insight about
> >> what's going on. Any help would be greatly appreciated!
> >>
> >> Thanks!
> >>
> >> Nico
> >>
> > First off you'll want to get familiar with the resources mentioned here:
> > http://wiki.freeswitch.org/wiki/Reporting_Bugs
> >
> > It has good tips on how to collect and report information.
> >
> > Second, I recommend that you pastebin your relevant portion of the
> dialplan
> > and the whole javascript program that you are using so that others can
> take
> > a look.
> >
> > Last thing: if you restart FreeSWITCH does everything work fine for a
> while
> > but then eventually it breaks down and exhibits the behavior that you are
> > reporting?
> >
> > -MC
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From linux4michelle at tamay-dogan.net  Thu Jan  7 10:27:31 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Thu, 7 Jan 2010 19:27:31 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <7b197bef1001060939rd1e08f3h2cb82cd89e50225e@mail.gmail.com>
References: <20100103184636.GW5547@tamay-dogan.net>
	<20100106135210.GG5547@tamay-dogan.net>
	<CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>
	<7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com>
	<20100106171413.GI5547@tamay-dogan.net>
	<7b197bef1001060939rd1e08f3h2cb82cd89e50225e@mail.gmail.com>
Message-ID: <20100107182731.GL5547@tamay-dogan.net>

Good evening Giovanni,

Am 2010-01-06 18:39:22, schrieb Giovanni Maruzzelli:
> Hello Michelle,
> 
> it supports all modules that accepts standard ETSI AT-GSM commands
> (so, let's say all of them).

Currently I havepayed over 20.000 Euro for ETSI and ANS specs  and  have
currently not the money to buy more...  Are the ETSI  AT-GSM  specs  are
free available?  --  If yes, I need them!

> Maybe I do not understand the second question, what do you means for
> Endpoints?
> 
> If you're talking usb endpoints, you'll need a modem endpoint (that
> can be seen as a serial port), and (if you need audio, eg not just
> SMSs but voice calls too) you need an audio endpoint (that can be seen
> as a soundcard).

I mean the USB-Endpoints...

If you have an USB-Microcontroller where the USB port is  a  device,  it
identify it self over the Endpoint 0 and is for us non usable.

And no it comes, where I haveproblems to understnd HOW the GSM Modem  is
working but I will assume some things:

The EP1 of the USB-pot is configured for bidirectional Data transmission
andwill controll out Device and is normaly  configured  as  /dev/ttyUSB0
and this is, where we use the AT commands to  get  infos  from  the  GSM
modem/cellphone and send/receive SMS.

Now We need EP2 and configure it as streaming output for the Audio port.

EP3 would be the streaming input for the Audio Port.

is this right up to here?

Then, EP4 would be the bidirectinal dataport  for  the  UMTS  and  HS*PA
Tranceiver, since it is entirely independant from the rest  of  the  GSM
modem/cellphone.

Is this right?

If yes, then it is easier as I was thinking...


> Many modules and cellphones can be seen as HDSPA or GPRS modems, just
> check their specs.

My "Nokia 6120 classic" has in total 13 endpoints...  Hell, where can  I
get an USB-Microcontroller which support this mass of USB Endpoints?

Most ARM9/11 support not more then 7 or 9.  :-(

> For audio, if the module/cellphone/modem does not offer an audio usb
> endpoint (eg cannot be seen as a soundcard) one trick is to connect
> the headset jack to an usb soundcard (you can find soundcard with for
> factor like a dongle based on cm-108 chipset for under $10). I'll
> publish the schema of the cable needed from hadset jack in the
> phone/module to the usb soundcard).
> 
> If I have not get what you asked, please explain more your question.

Most important things are the above desibed understanding  problem  with
the USB Endpoints

I have the HSPA and GS frontends (Maxim and Infineon chips) here and  my
selfemade simple GSM/GPRS cell-phone is already working,  but  has  less
functionality as the cell-phones from Year 2000  :-D

Hey, ist is my first experience of developing GSM stuff.

I an to develop a VERY simple GSM/UMTS/HSPA USB-Modem which do  its  job
without balast.

So if someone can help me with infos, I am very open...

I prefer FreeSwitch over Asterisk which froze in the  last  2  years  to
many times in situations where it should not freeze, exspecialy if  I  a
call a Chip-Manfacturer (Maxim/TI) Tech-Support.  --  It is not funny!

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From cmrienzo at gmail.com  Thu Jan  7 11:22:46 2010
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Thu, 7 Jan 2010 14:22:46 -0500
Subject: [Freeswitch-users] UNIMRCP RECOGNIZE headers
In-Reply-To: <E1D3989424711B43A00C5476DE5E1A7001E300D1@hqmail01.soundbite.com>
References: <E1D3989424711B43A00C5476DE5E1A7001E300D1@hqmail01.soundbite.com>
Message-ID: <7C265966-14D1-48E9-870F-2EE7B52A15BC@gmail.com>

Neil,

In general, you can set most MRCP params in mod_unimrcp like this:

detect_speech unimrcp {speech-complete-timeout=5000,speech-incomplete-timeout=5000}grammar grammar-name

These params will remain set on the speech handle until freeswitch closes it.  For ASR, that's when the call ends.




> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neil Burgess
> Sent: Thursday, January 07, 2010 10:53 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] UNIMRCP RECOGNIZE headers
>  
> Hello,
>  
> Wondering if anyone can help with a unimrcp question.
>  
> We are using the UNIMRCP client in FreeSwitch to communicate with a commercial UNIMRCP server (Loquendo). We need to control such items as Speech Timeouts that are occurring on the server side, however we have been advised that the only way these can be affected is via  headers in the MRCP requests. So, we need to be able to set headers such as Speech-Complete-Timeout, Speech-Incomplete-Timeout, Recognition-Timeout, etc in the MRCP RECOGNIZE request.  Is there a mechanism in FreeSwitch which we can use to pass these down before, (or as) we issue the ?detect_speech? command. We are happy to use a jscript interface, or whatever if such a capability is available.
>  
> Many thanks,
> Neil
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From achaloyan at yahoo.com  Thu Jan  7 11:51:48 2010
From: achaloyan at yahoo.com (Arsen Chaloyan)
Date: Thu, 7 Jan 2010 11:51:48 -0800 (PST)
Subject: [Freeswitch-users] UNIMRCP RECOGNIZE headers
In-Reply-To: <787302A89ACCE24DA8F56DA101E77C841392F8C2D8@THHS2E12BE1X.hostedservice2.net>
References: <787302A89ACCE24DA8F56DA101E77C841392F8C2D8@THHS2E12BE1X.hostedservice2.net>
Message-ID: <594856.45588.qm@web111302.mail.gq1.yahoo.com>

Hello Neil,


I guess you are using a commercial MRCP server such as Loquendo, at least I'm not aware of UniMRCP server based Loquendo ASR :)
Though someone asked me about such a solution a few months ago.

As of actual request, looking at the code of mod_unimrcp, I'd say such an option exists. See recog_channel_set_params().
Looking through mod_unimrcp wiki examples, I'd say the following should do what you need

switch_ivr_detect_speech(session, "unimrcp", "{recognition-timeout=15000}yesno", "yesno-name", "", ah);

-- 
Arsen Chaloyan
The author of UniMRCP 
http://www.unimrcp.org




________________________________
From: Neil Burgess <neil.burgess at redmatter.com>
To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
Sent: Thu, January 7, 2010 7:52:52 PM
Subject: [Freeswitch-users] UNIMRCP RECOGNIZE headers

 
Hello,
 
Wondering if anyone can help with a unimrcp question. 
 
We are using the UNIMRCP client in FreeSwitch to communicate
with a commercial UNIMRCP server (Loquendo). We need to control such items as Speech
Timeouts that are occurring on the server side, however we have been advised
that the only way these can be affected is via  headers in the MRCP
requests. So, we need to be able to set headers such as Speech-Complete-Timeout,
Speech-Incomplete-Timeout, Recognition-Timeout, etc in the MRCP RECOGNIZE request. 
Is there a mechanism in FreeSwitch which we can use to pass these down before,
(or as) we issue the ?detect_speech? command. We are happy to use a
jscript interface, or whatever if such a capability is available.
 
Many thanks,
Neil 
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From msc at freeswitch.org  Thu Jan  7 12:01:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Jan 2010 12:01:00 -0800
Subject: [Freeswitch-users] Announcement: FSComm - The FreeSWITCH-based
	softphone
Message-ID: <87f2f3b91001071201vb759713w3f37b192cd09020f@mail.gmail.com>

We are happy to announce a new project: FSComm, a FreeSWITCH-based
softphone. Read the story here <http://bit.ly/4HLF71>. We look forward to
watching this project grow and become a truly useful tool for VoIP users
everywhere.

-Michael
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From jmesquita at freeswitch.org  Thu Jan  7 12:27:01 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 7 Jan 2010 18:27:01 -0200
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<B732197E-1123-4738-A69F-F84206FCE38A@freeswitch.org>
	<0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org>
	<0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org>
	<23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com>
	<60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org>
	<23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com>
	<ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com>
	<23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com>
	<B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>
Message-ID: <ebbc2d551001071227y22d6eaa8y43742fb4328a67b2@mail.gmail.com>

There is a wiki page up now.

http://wiki.freeswitch.org/wiki/FSComm

It's a bit poor at the moment, but I will fill in more stuff when I feel
better (really sick now).

Regards,
Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927


On Thu, Jan 7, 2010 at 2:25 PM, Brian West <brian at freeswitch.org> wrote:

> Lets schedule FSComm on the weekly conference call... We need people to
> step up and take some roles in both FreeSWITCH and FSComm projects... Even
> if its just testing bugs and collecting info.
>
> Thanks,
> Brian
>
> On Jan 7, 2010, at 10:11 AM, Seven Du wrote:
>
> > :lol. I do like to involve into this. I saw you have done a lot of
> > works. I read some code and here are some questions:
> >
> > 1) what's your nick on IRC? I'm seven(or seven_ ?)
> > 2) Are you developing on Windows? How can I compile on Mac(I have no
> > experience on QT)?
> > 3) Does is needs fsGUI? I downloaded fsGUI dmg file and it complains
> > no mods on start.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Thu Jan  7 12:36:11 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jan 2010 14:36:11 -0600
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <ebbc2d551001071227y22d6eaa8y43742fb4328a67b2@mail.gmail.com>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<B732197E-1123-4738-A69F-F84206FCE38A@freeswitch.org>
	<0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org>
	<0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org>
	<23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com>
	<60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org>
	<23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com>
	<ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com>
	<23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com>
	<B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>
	<ebbc2d551001071227y22d6eaa8y43742fb4328a67b2@mail.gmail.com>
Message-ID: <D03B492C-74B9-4B38-B67B-017690FF372C@freeswitch.org>

Everyone should get JM's paypal and toss him some cash for all the good work he's doing... Without him this project wouldn't have become a reality.

/b

On Jan 7, 2010, at 2:27 PM, Jo?o Mesquita wrote:

> There is a wiki page up now.
> 
> http://wiki.freeswitch.org/wiki/FSComm
> 
> It's a bit poor at the moment, but I will fill in more stuff when I feel better (really sick now).
> 
> Regards,
> Jo?o Mesquita
> FreeSWITCH? Solutions
> t: +1 (646) 4959927

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From jerry.richards at teotech.com  Thu Jan  7 13:42:34 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 7 Jan 2010 13:42:34 -0800
Subject: [Freeswitch-users] DESTINATION_OUT_OF_ORDER
In-Reply-To: <407CE59D-5591-4838-B3D7-A0EC674844F7@freeswitch.org>
References: <B5012C4A3A0D4FA0956D48F368577C1D@greyhawk.tonecommander.com><87f2f3b91001051633xa410d57p18950e803d724634@mail.gmail.com><4BA552E277A945128169919160C9A57A@greyhawk.tonecommander.com>
	<407CE59D-5591-4838-B3D7-A0EC674844F7@freeswitch.org>
Message-ID: <5F093406C6D045CAB941A284E3A17178@greyhawk.tonecommander.com>

Brian,
 
Thank you.  The issue did have to do with SCTP package installation.
Apparently, after I installed the SCTP packages yesterday, I did not re-run
./confgure.  Anyway, now my PRI is working on the new server.
 
Thank You and Best Regards,
Jerry
 


  _____  

From: Brian West [mailto:brian at freeswitch.org] 
Sent: Wednesday, January 06, 2010 2:33 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] DESTINATION_OUT_OF_ORDER


Again if you did not have the SCTP libs and dev headers installed when you
did ./configure you'll never get this working SCTP is required to have boost
work.  Also don't use Pre9 get the latest from latest.freeswitch.org and run
with that please. 

/b

On Jan 6, 2010, at 4:22 PM, Jerry Richards wrote:



By the way, I posted another pastebin log
(http://pastebin.freeswitch.org/11694) that shows the output of the "make"
and "make install sounds-install moh-install..." commands.  Just prior to
these makes I executed "make clean" and "make distclean".  You will notice
in the makefile output that ozmod_sangoma_boost.c never appears to get
compiled.  Shouldn't everything be compiled in this case?  I also confirmed
that my working server does have the ozmod_sangoma_boost.so file located in
the right place, which is why it's working.
 
Could version 1.0.5pre9 introduced this bug?  Could that be why the older
server (which original ran with 1.0.4) works and the new one doesn't?
 
Thanks and Best Regards,
Jerry
 


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From larclap at yahoo.com  Thu Jan  7 14:15:14 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 7 Jan 2010 14:15:14 -0800
Subject: [Freeswitch-users] Compile error fscomm?
Message-ID: <012901ca8fe6$e36b71c0$aa425540$@com>

I just downloaded the fscomm project and loaded it into vs2008. I've never
programmed in C++ (or c), just C#, so I can't make anything of the following
two messages:

 

Error      1              error PRJ0019: A tool returned an error code from
"RCC resources.qrc"  FSComm              FSComm

 

Warning               2              The following environment variables
were not found: $(QTDIR)                Project FSComm

 

Any suggestions? I do see the file "resources.qrc" in the fscomm folder.

 

The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)

 

Lars

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From msc at freeswitch.org  Thu Jan  7 14:35:54 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Jan 2010 14:35:54 -0800
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <012901ca8fe6$e36b71c0$aa425540$@com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>
Message-ID: <87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:

>  I just downloaded the fscomm project and loaded it into vs2008. I?ve
> never programmed in C++ (or c), just C#, so I can?t make anything of the
> following two messages:
>
>
>
> Error      1              error PRJ0019: A tool returned an error code from
> "RCC resources.qrc"  FSComm              FSComm
>
>
>
> Warning               2              The following environment variables
> were not found: $(QTDIR)                Project FSComm
>
>
>
> Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.
>
>
>
> The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)
>
>
>
> Lars
>
Do you have Qt 4.6 installed?
-MC
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From larclap at yahoo.com  Thu Jan  7 15:11:15 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 7 Jan 2010 15:11:15 -0800
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>
Message-ID: <014801ca8fee$b75f8780$261e9680$@com>

No Qt installed. I just checked out from
http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into
VS2008. 

Do I need to get
http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

 

 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:

I just downloaded the fscomm project and loaded it into vs2008. I've never
programmed in C++ (or c), just C#, so I can't make anything of the following
two messages:

 

Error      1              error PRJ0019: A tool returned an error code from
"RCC resources.qrc"  FSComm              FSComm

 

Warning               2              The following environment variables
were not found: $(QTDIR)                Project FSComm

 

Any suggestions? I do see the file "resources.qrc" in the fscomm folder.

 

The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)

 

Lars

Do you have Qt 4.6 installed?
-MC

 

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From jcasale at activenetwerx.com  Thu Jan  7 15:24:15 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Thu, 7 Jan 2010 23:24:15 +0000
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9070B8B@prato.activenetwerx.local>

>the big rule for the time being is stick with 5.3 5.4 appears to have some bugs in the toolchain and libc

OMG, I have been messing with my broken fax and zap for like two weeks?
Someone shoot me...

Unless you install of a dvd and avoid using public repo's, that's kind of
hard? Do you guys have any idea when this could be resolved, I am going to
hold off on my migration then.

Thanks!
jlc



From brian at freeswitch.org  Thu Jan  7 15:30:59 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jan 2010 17:30:59 -0600
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4 (fixed)
In-Reply-To: <20100105065356.AEE0612F5@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20091228233838.75E611694@sinclaire.sibble.net>,
	<C2EB0262-CB8A-44F5-9978-1C5F8073832F@freeswitch.org>
	<20100105065356.AEE0612F5@sinclaire.sibble.net>
Message-ID: <8C20E0B1-0E56-4892-9D36-4D6DF3331244@freeswitch.org>

Harondel,
	Please update your FreeSWITCH source rev 16204 fixes the SAS passing in all cases from my testing.  You'll need the latest ZRTP Lib and zfone application to make this work... I'm not too sure Tiviphone does this yet as I don't have one to test with.  This also fixes the issue when both sides are enrolled.  Next we will fix the video portion so both video and audio will go thru zrtp.

Please try it and let me know.

Thanks,
Brian

On Jan 5, 2010, at 12:54 AM, Harondel J. Sibble wrote:

> Maybe that's what's affecting me now.....
> 
> I've both phones registered (confirmed by calling 9787) on both devices and 
> it says each device is already enrolled. (how does one un-enroll????). Both 
> phones are running the Tivi 2.0.7 beta. 
> 
> Now however, other than the first call I made between devices after 
> enrollment, the sas is not matching anymore.
> 
> I set both these options in the console
> 
> global_action application="set" data="zrtp_enrollment=true"
> global_setvar zrtp_secure_media=true
> 
> What should I be looking for in the console output
> 
> On 28 Dec 2009 at 17:49, Brian West wrote:
> 
>> I'm still not done with this I think we found a bug in the lib... Viktor
>> fixed it today and I'm going to retry after I get done testing G729 more
>> today!  ;)
>> 
>> /b
>> 
>> On Dec 28, 2009, at 5:38 PM, Harondel J. Sibble wrote:
>> 
>>> Hmm, okay, I went back to basics and did a full rebuild for 1.0.4 svn
>> trunk, 
>>> I am able to now enroll my nokia e61i running the beta 2.0.7 Tiviphone 
>>> client, however I am not seeing the enrollment option popup in zfone 0.92
>>> build 218 on windows in front of an x-lite client.
> 
> -- 
> Harondel J. Sibble 
> Sibble Computer Consulting
> Creating Solutions for the small and medium business computer user.
> help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
> (604) 739-3709 (voice)
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jmesquita at freeswitch.org  Thu Jan  7 15:45:56 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 7 Jan 2010 21:45:56 -0200
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <014801ca8fee$b75f8780$261e9680$@com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>
	<014801ca8fee$b75f8780$261e9680$@com>
Message-ID: <ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>

I don't have a Windows machine to test that. Maybe jlenk could give us a
hand since he is the one who has created the visual studio project?

Regards,
Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927


On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:

>  No Qt installed. I just checked out from
> http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into
> VS2008.
>
> Do I need to get
> http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, January 07, 2010 2:36 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Compile error fscomm?
>
>
>
>
>
> On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:
>
> I just downloaded the fscomm project and loaded it into vs2008. I?ve never
> programmed in C++ (or c), just C#, so I can?t make anything of the following
> two messages:
>
>
>
> Error      1              error PRJ0019: A tool returned an error code from
> "RCC resources.qrc"  FSComm              FSComm
>
>
>
> Warning               2              The following environment variables
> were not found: $(QTDIR)                Project FSComm
>
>
>
> Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.
>
>
>
> The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)
>
>
>
> Lars
>
> Do you have Qt 4.6 installed?
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Jan  7 16:01:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Jan 2010 18:01:56 -0600
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9070B8B@prato.activenetwerx.local>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E9070B8B@prato.activenetwerx.local>
Message-ID: <191c3a031001071601if302c1bsd8a9a2af7c166cf7@mail.gmail.com>

if you get centos5.3 or 5.2 it will be resolved because it's not centos5.4
which is the only bad one atm, newer is not always better.



On Thu, Jan 7, 2010 at 5:24 PM, Joseph L. Casale
<jcasale at activenetwerx.com>wrote:

> >the big rule for the time being is stick with 5.3 5.4 appears to have some
> bugs in the toolchain and libc
>
> OMG, I have been messing with my broken fax and zap for like two weeks?
> Someone shoot me...
>
> Unless you install of a dvd and avoid using public repo's, that's kind of
> hard? Do you guys have any idea when this could be resolved, I am going to
> hold off on my migration then.
>
> Thanks!
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Thu Jan  7 16:03:07 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jan 2010 18:03:07 -0600
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>
	<014801ca8fee$b75f8780$261e9680$@com>
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
Message-ID: <F31651BB-C5E4-429F-ADAF-512B641F2CE1@freeswitch.org>

Remember we have #fscomm on irc.freenode.net please join there if you wish to get involved... help out...   ;)

Thanks,
Brian

On Jan 7, 2010, at 5:45 PM, Jo?o Mesquita wrote:

> I don't have a Windows machine to test that. Maybe jlenk could give us a hand since he is the one who has created the visual studio project?
> 
> Regards,
> Jo?o Mesquita
> FreeSWITCH? Solutions
> t: +1 (646) 4959927




From freeswitch at aastral.net  Thu Jan  7 16:15:42 2010
From: freeswitch at aastral.net (Bill W.)
Date: Thu, 07 Jan 2010 19:15:42 -0500
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
Message-ID: <4B46792E.6090805@aastral.net>

Personally, I'm migrating to openSUSE, because I'm tired of RedHat's
non-standard kernel backports and outdated packages.

Others in the VoIP industry (ViciDial) recommend SuSE because of their
experience with with RedHat's long-standing perl bug, process preemption
set to desktop, version number mismatch on packages and non-standard
libraries.

But SuSE is not without it's problems.  The perl module for FreeSWITCH
will compile on SuSE, but FreeSWITCH coredumps when trying to use it.
Haven't had time to research that one.

Also, if you're going to use wanpipe with Sangoma cards on SuSE, check
with Sangoma to make sure wanpipe will compile on 11.2.   It wouldn't
compile for me and I reported the bug to Sangoma, but I haven't heard
anything back yet.  Wanpipe compiles just fine on 11.1.

Having said that, the Suse installation is very similar to
CentOS/RedHat.  I've run FreeSWITCH on both, and it works just fine on
both.

Hope this helps,
Bill



Anthony Minessale wrote:
> the big rule for the time being is stick with 5.3 5.4 appears to have
> some bugs in the toolchain and libc
> 
> On Thu, Jan 7, 2010 at 6:02 AM, Joseph L. Casale
> <jcasale at activenetwerx.com <mailto:jcasale at activenetwerx.com>> wrote:
> 
>     >Since CentOS is recommend for FS but i can't see a CentOS specific
>     installation guide on wiki as we have a separate >guide for Ubuntu. 
>     >
>     >Do we have similar guide like this
>     one http://wiki.freeswitch.org/wiki/SBC_Setup actually its for
>     debian but good thing >is that it also explains which extra
>     >services should be stopped or removed for better performance. 
>     >
>     >Do we have similar for CentOS?
> 
>     Check out the http://wiki.freeswitch.org/wiki/Installation_Guide
> 
> 
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Thu Jan  7 16:49:11 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Jan 2010 18:49:11 -0600
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <4B46792E.6090805@aastral.net>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
	<4B46792E.6090805@aastral.net>
Message-ID: <191c3a031001071649g26ad7257pb9ba6e7045224f5a@mail.gmail.com>

The last guy using SUSE had filesystem problems so bad he almost cried while
he was apologizing for how much time he wasted asking for help about it once
he tried a different one.  Not to say you can't use whatever you want but we
are going to tread carefully about which distros we support.


On Thu, Jan 7, 2010 at 6:15 PM, Bill W. <freeswitch at aastral.net> wrote:

> Personally, I'm migrating to openSUSE, because I'm tired of RedHat's
> non-standard kernel backports and outdated packages.
>
> Others in the VoIP industry (ViciDial) recommend SuSE because of their
> experience with with RedHat's long-standing perl bug, process preemption
> set to desktop, version number mismatch on packages and non-standard
> libraries.
>
> But SuSE is not without it's problems.  The perl module for FreeSWITCH
> will compile on SuSE, but FreeSWITCH coredumps when trying to use it.
> Haven't had time to research that one.
>
> Also, if you're going to use wanpipe with Sangoma cards on SuSE, check
> with Sangoma to make sure wanpipe will compile on 11.2.   It wouldn't
> compile for me and I reported the bug to Sangoma, but I haven't heard
> anything back yet.  Wanpipe compiles just fine on 11.1.
>
> Having said that, the Suse installation is very similar to
> CentOS/RedHat.  I've run FreeSWITCH on both, and it works just fine on
> both.
>
> Hope this helps,
> Bill
>
>
>
> Anthony Minessale wrote:
> > the big rule for the time being is stick with 5.3 5.4 appears to have
> > some bugs in the toolchain and libc
> >
> > On Thu, Jan 7, 2010 at 6:02 AM, Joseph L. Casale
> > <jcasale at activenetwerx.com <mailto:jcasale at activenetwerx.com>> wrote:
> >
> >     >Since CentOS is recommend for FS but i can't see a CentOS specific
> >     installation guide on wiki as we have a separate >guide for Ubuntu.
> >     >
> >     >Do we have similar guide like this
> >     one http://wiki.freeswitch.org/wiki/SBC_Setup actually its for
> >     debian but good thing >is that it also explains which extra
> >     >services should be stopped or removed for better performance.
> >     >
> >     >Do we have similar for CentOS?
> >
> >     Check out the http://wiki.freeswitch.org/wiki/Installation_Guide
> >
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:+19193869900
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Thu Jan  7 16:52:25 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jan 2010 18:52:25 -0600
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <4B46792E.6090805@aastral.net>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
	<4B46792E.6090805@aastral.net>
Message-ID: <E5EFC626-353A-4B54-9F22-59C0C48E3B95@freeswitch.org>

Good luck with that you'll have an ass load of problems.  The reason its stable is the backports and outdated packages.  Bleeding edge will only screw you over... just ask KJV... He was on OpenSuSE and had nothing but weird problems.

/b

On Jan 7, 2010, at 6:15 PM, Bill W. wrote:

> Personally, I'm migrating to openSUSE, because I'm tired of RedHat's
> non-standard kernel backports and outdated packages.
> 
> Others in the VoIP industry (ViciDial) recommend SuSE because of their
> experience with with RedHat's long-standing perl bug, process preemption
> set to desktop, version number mismatch on packages and non-standard
> libraries.
> 
> But SuSE is not without it's problems.  The perl module for FreeSWITCH
> will compile on SuSE, but FreeSWITCH coredumps when trying to use it.
> Haven't had time to research that one.
> 
> Also, if you're going to use wanpipe with Sangoma cards on SuSE, check
> with Sangoma to make sure wanpipe will compile on 11.2.   It wouldn't
> compile for me and I reported the bug to Sangoma, but I haven't heard
> anything back yet.  Wanpipe compiles just fine on 11.1.
> 
> Having said that, the Suse installation is very similar to
> CentOS/RedHat.  I've run FreeSWITCH on both, and it works just fine on
> both.
> 
> Hope this helps,
> Bill




From jcasale at activenetwerx.com  Thu Jan  7 17:02:46 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 8 Jan 2010 01:02:46 +0000
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <191c3a031001071649g26ad7257pb9ba6e7045224f5a@mail.gmail.com>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
	<4B46792E.6090805@aastral.net>
	<191c3a031001071649g26ad7257pb9ba6e7045224f5a@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9072253@prato.activenetwerx.local>

>The last guy using SUSE had filesystem problems so bad he almost cried while
>he was apologizing for how much time he wasted asking for help about it once
>he tried a different one.? Not to say you can't use whatever you want but we
>are going to tread carefully about which distros we support.

Which begs the question, which distro do _you_ run on and suggest behind the
scenes:)



From jason at jasonjgw.net  Thu Jan  7 17:04:57 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 8 Jan 2010 12:04:57 +1100
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <191c3a031001071649g26ad7257pb9ba6e7045224f5a@mail.gmail.com>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
	<4B46792E.6090805@aastral.net>
	<191c3a031001071649g26ad7257pb9ba6e7045224f5a@mail.gmail.com>
Message-ID: <20100108010457.GA23508@jdc.jasonjgw.net>

Anthony Minessale <anthony.minessale at gmail.com> wrote:
> The last guy using SUSE had filesystem problems so bad he almost cried while
> he was apologizing for how much time he wasted asking for help about it once
> he tried a different one.  

Are they still using reiserfs by default? If I were using Reiserfs (which I'm
not, and I'm not using Suse either), I would be moving to EXT3, EXT4, or XFS.




From a.alalousi at gmail.com  Thu Jan  7 17:05:23 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Fri, 8 Jan 2010 01:05:23 +0000
Subject: [Freeswitch-users] FAS detection with FS
Message-ID: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>

Hi everyone,

Was just wondering what/if anyone is doing any work on FAS detection and
spoofed ring tones. Be great to discuss some ideas.

Regards,

Ahmed.

-- 
Ahmed Naji
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From freeswitch at aastral.net  Thu Jan  7 17:05:55 2010
From: freeswitch at aastral.net (Bill W.)
Date: Thu, 07 Jan 2010 20:05:55 -0500
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <E5EFC626-353A-4B54-9F22-59C0C48E3B95@freeswitch.org>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>	<4B46792E.6090805@aastral.net>
	<E5EFC626-353A-4B54-9F22-59C0C48E3B95@freeswitch.org>
Message-ID: <4B4684F3.8030504@aastral.net>

Wow, I haven't heard of these issues.  Obviously this concerns me.  Are
these documented anywhere so I can research this?  How do I get in touch
with KJV?

Thanks!
Bill



Brian West wrote:
> Good luck with that you'll have an ass load of problems.  The reason its stable is the backports and outdated packages.  Bleeding edge will only screw you over... just ask KJV... He was on OpenSuSE and had nothing but weird problems.
> 
> 



From anthony.minessale at gmail.com  Thu Jan  7 17:11:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Jan 2010 19:11:34 -0600
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9072253@prato.activenetwerx.local>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
	<4B46792E.6090805@aastral.net>
	<191c3a031001071649g26ad7257pb9ba6e7045224f5a@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E9072253@prato.activenetwerx.local>
Message-ID: <191c3a031001071711g776488e7t474806fb3527f194@mail.gmail.com>

We use CentOS 5.3 64 bit and have very few issues,
The problem is since it's an older kernel there may be some real time
improvements we could benefit from so we will probably play with some newer
bleeding kernels to compare and stay ahead of the curve but still rely on
CentOS to keep the users happy.


On Thu, Jan 7, 2010 at 7:02 PM, Joseph L. Casale
<jcasale at activenetwerx.com>wrote:

> >The last guy using SUSE had filesystem problems so bad he almost cried
> while
> >he was apologizing for how much time he wasted asking for help about it
> once
> >he tried a different one.  Not to say you can't use whatever you want but
> we
> >are going to tread carefully about which distros we support.
>
> Which begs the question, which distro do _you_ run on and suggest behind
> the
> scenes:)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From dujinfang at gmail.com  Thu Jan  7 17:21:30 2010
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 8 Jan 2010 09:21:30 +0800
Subject: [Freeswitch-users] Failed to connect to a SKYPE API
In-Reply-To: <27062783.post@talk.nabble.com>
References: <27062783.post@talk.nabble.com>
Message-ID: <23f91031001071721g3f220722o5a07e8e7c74ae339@mail.gmail.com>

To make others help you easier, you'd better include more informations

1) what's your FS version/OS?
2) did you followed all steps as the wiki page said to generate
.Skype/ conf files ?
3) there is a client.c in source code, can you compile and make sure
it works like this?

./client :101   #where 101 is your display no.
4) what your skypiax.conf looks like?

2010/1/8 Oscav <oscav at hotmail.fr>:
>
> Hi,
>
> I'm trying to use to SkypeIAX. When I load the mod_skypiax, I got the
> following error :
>
> ?Failed to connect to a SKYPE API for interface_id=1, no SKYPE client
> running, please (re)start Skype client. Skypiax exiting
>
> Skype is running with the correct account and skypiax.conf use the same
> account. I was expecting a permission request from the Skype user but
> nothing happens.
>
> Somebody knows how I can solve this ??
>
> Many thanks.
> --
> View this message in context: http://old.nabble.com/Failed-to-connect-to-a-SKYPE-API-tp27062783p27062783.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Thu Jan  7 17:23:57 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jan 2010 19:23:57 -0600
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <4B4684F3.8030504@aastral.net>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>	<4B46792E.6090805@aastral.net>
	<E5EFC626-353A-4B54-9F22-59C0C48E3B95@freeswitch.org>
	<4B4684F3.8030504@aastral.net>
Message-ID: <74C18F18-89EB-49DF-B5CE-C8D56856ABCD@freeswitch.org>

He's on the list Karl J. Vesterling

/b

On Jan 7, 2010, at 7:05 PM, Bill W. wrote:

> Wow, I haven't heard of these issues.  Obviously this concerns me.  Are
> these documented anywhere so I can research this?  How do I get in touch
> with KJV?
> 
> Thanks!
> Bill
> 
> 
> 
> Brian West wrote:
>> Good luck with that you'll have an ass load of problems.  The reason its stable is the backports and outdated packages.  Bleeding edge will only screw you over... just ask KJV... He was on OpenSuSE and had nothing but weird problems.
>> 
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From max.bridgewater at gmail.com  Thu Jan  7 18:36:37 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Thu, 7 Jan 2010 21:36:37 -0500
Subject: [Freeswitch-users] Skypiax on CentOS
Message-ID: <c55d46951001071836i55516863r359bb56695744054@mail.gmail.com>

Hi,

Has anybody installed Skypiax on CentOS 5 lately? The documentation is based
on Skype 2.0.0 (skype-2.0.0.72-centos.i586.rpm) which apparently is not
available online anymore. I am hoping that you guys can help me either 1)
get Skypiax to run with the latest Skype or 2) share this old version so i
can follow the existing documentation. My preference would be the second
option though ;)

Thanks,
Max.
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From brian at freeswitch.org  Thu Jan  7 18:49:57 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Jan 2010 20:49:57 -0600
Subject: [Freeswitch-users] Skypiax on CentOS
In-Reply-To: <c55d46951001071836i55516863r359bb56695744054@mail.gmail.com>
References: <c55d46951001071836i55516863r359bb56695744054@mail.gmail.com>
Message-ID: <581E0571-986A-450D-98E3-D3AAF34E21C6@freeswitch.org>

Just download the static binary build for linux... problem solved.

/b

On Jan 7, 2010, at 8:36 PM, Max Bridgewater wrote:

> Hi,
> 
> Has anybody installed Skypiax on CentOS 5 lately? The documentation is based on Skype 2.0.0 (skype-2.0.0.72-centos.i586.rpm) which apparently is not available online anymore. I am hoping that you guys can help me either   1) get Skypiax to run with the latest  Skype or 2) share this old version so i can follow the existing documentation. My preference would be the second option though ;)
> 
> Thanks,
> Max.
> _________

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From william.suffill at gmail.com  Thu Jan  7 18:52:49 2010
From: william.suffill at gmail.com (William Suffill)
Date: Thu, 7 Jan 2010 21:52:49 -0500
Subject: [Freeswitch-users] Skypiax on CentOS
In-Reply-To: <c55d46951001071836i55516863r359bb56695744054@mail.gmail.com>
References: <c55d46951001071836i55516863r359bb56695744054@mail.gmail.com>
Message-ID: <6b65470d1001071852i752b8306r51a48576090b4be@mail.gmail.com>

Skype appears to be pushing a new version on the linux side but the old
packages are still available but not linked anywhere.

http://download.skype.com/linux/skype-2.0.0.72-centos.i586.rpm
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From max.bridgewater at gmail.com  Thu Jan  7 19:03:24 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Thu, 7 Jan 2010 22:03:24 -0500
Subject: [Freeswitch-users] Skypiax on CentOS
In-Reply-To: <6b65470d1001071852i752b8306r51a48576090b4be@mail.gmail.com>
References: <c55d46951001071836i55516863r359bb56695744054@mail.gmail.com>
	<6b65470d1001071852i752b8306r51a48576090b4be@mail.gmail.com>
Message-ID: <c55d46951001071903q5a23639ehee09da7640351a74@mail.gmail.com>

Thanks you very much guys.

Ad Brian: I tried the static binary build but was blocked by a qt4-x11
missing and I couldn't find an obvious solution to that. Hence my  inquiry
in the group before spending time on things that other people potentially
already solved.

max.

On Thu, Jan 7, 2010 at 9:52 PM, William Suffill
<william.suffill at gmail.com>wrote:

> Skype appears to be pushing a new version on the linux side but the old
> packages are still available but not linked anywhere.
>
> http://download.skype.com/linux/skype-2.0.0.72-centos.i586.rpm
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jeff at jefflenk.com  Thu Jan  7 19:16:40 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 7 Jan 2010 21:16:40 -0600
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>,
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>,
	<014801ca8fee$b75f8780$261e9680$@com>,
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
Message-ID: <COL117-W486989B811B372C35935AAB9700@phx.gbl>


The windows support is very experimental at this time!

 

You must manually install http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe

 

Then set the environment variable QTDIR in the environment variables. This can be set from the Computer/Properties/Advanced system settings/Environment Variables/User Variables settings screen.

 

QTDIR=c:\qt\4.6.0 - or wherever you installed it

 

then restart VS

 


 


Date: Thu, 7 Jan 2010 21:45:56 -0200
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

I don't have a Windows machine to test that. Maybe jlenk could give us a hand since he is the one who has created the visual studio project?


Regards,Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927



On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:




No Qt installed. I just checked out from http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into VS2008. 
Do I need to get http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?
 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?



 
 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:


I just downloaded the fscomm project and loaded it into vs2008. I?ve never programmed in C++ (or c), just C#, so I can?t make anything of the following two messages:
 
Error      1              error PRJ0019: A tool returned an error code from "RCC resources.qrc"  FSComm              FSComm
 
Warning               2              The following environment variables were not found: $(QTDIR)                Project FSComm
 
Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.
 
The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)
 
Lars

Do you have Qt 4.6 installed?
-MC
 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


 		 	   		  
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From pekkis50 at gmail.com  Thu Jan  7 19:25:56 2010
From: pekkis50 at gmail.com (Pekka Kurki)
Date: Fri, 08 Jan 2010 04:25:56 +0100
Subject: [Freeswitch-users] really no installer for w2k anywhere?
Message-ID: <4B46A5C4.9040809@gmail.com>

all installer versions fail with  missing  getnameinfo/getaddressinfo 
support in w2k.




From jmesquita at freeswitch.org  Thu Jan  7 19:28:31 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 8 Jan 2010 01:28:31 -0200
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <COL117-W486989B811B372C35935AAB9700@phx.gbl>
References: <012901ca8fe6$e36b71c0$aa425540$@com>
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>
	<014801ca8fee$b75f8780$261e9680$@com>
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>
Message-ID: <ebbc2d551001071928u422b1884r6ba556b8b7a06e80@mail.gmail.com>

Jeff, any chance we can get this on the wiki?

We have created a page here: http://wiki.freeswitch.org/wiki/FSComm

I am looking for a Windows machine to do testing on it as well.

Regards,
Jo?o Mesquita


On Fri, Jan 8, 2010 at 1:16 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

>  The windows support is very experimental at this time!
>
> You must manually install
> http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
>
> Then set the environment variable QTDIR in the environment variables. This
> can be set from the Computer/Properties/Advanced system settings/Environment
> Variables/User Variables settings screen.
>
> QTDIR=c:\qt\4.6.0 - or wherever you installed it
>
> then restart VS
>
>
>
> ------------------------------
> Date: Thu, 7 Jan 2010 21:45:56 -0200
> From: jmesquita at freeswitch.org
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Compile error fscomm?
>
> I don't have a Windows machine to test that. Maybe jlenk could give us a
> hand since he is the one who has created the visual studio project?
>
> Regards,
> Jo?o Mesquita
> FreeSWITCH? Solutions
> t: +1 (646) 4959927
>
>
> On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:
>
>  No Qt installed. I just checked out from
> http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into
> VS2008.
>
> Do I need to get
> http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, January 07, 2010 2:36 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Compile error fscomm?
>
>
>
>
>
> On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:
>
> I just downloaded the fscomm project and loaded it into vs2008. I?ve never
> programmed in C++ (or c), just C#, so I can?t make anything of the following
> two messages:
>
>
>
> Error      1              error PRJ0019: A tool returned an error code from
> "RCC resources.qrc"  FSComm              FSComm
>
>
>
> Warning               2              The following environment variables
> were not found: $(QTDIR)                Project FSComm
>
>
>
> Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.
>
>
>
> The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)
>
>
>
> Lars
>
> Do you have Qt 4.6 installed?
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up
> now. <http://clk.atdmt.com/GBL/go/196390709/direct/01/>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jeff at jefflenk.com  Thu Jan  7 19:36:52 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 7 Jan 2010 21:36:52 -0600
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <COL117-W486989B811B372C35935AAB9700@phx.gbl>
References: <012901ca8fe6$e36b71c0$aa425540$@com>, ,
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>, ,
	<014801ca8fee$b75f8780$261e9680$@com>, ,
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>,
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>
Message-ID: <COL117-W45E166F06B72FC096E4ACB9700@phx.gbl>


I will update the project files tommorow to account for the new project files/locations in the last 24 hours too.
 


From: jeff at jefflenk.com
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 7 Jan 2010 21:16:40 -0600
Subject: Re: [Freeswitch-users] Compile error fscomm?



The windows support is very experimental at this time!
 
You must manually install http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
 
Then set the environment variable QTDIR in the environment variables. This can be set from the Computer/Properties/Advanced system settings/Environment Variables/User Variables settings screen.
 
QTDIR=c:\qt\4.6.0 - or wherever you installed it
 
then restart VS
 

 


Date: Thu, 7 Jan 2010 21:45:56 -0200
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

I don't have a Windows machine to test that. Maybe jlenk could give us a hand since he is the one who has created the visual studio project? 


Regards,Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927



On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:




No Qt installed. I just checked out from http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into VS2008. 
Do I need to get http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?
 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?



 
 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:


I just downloaded the fscomm project and loaded it into vs2008. I?ve never programmed in C++ (or c), just C#, so I can?t make anything of the following two messages:
 
Error      1              error PRJ0019: A tool returned an error code from "RCC resources.qrc"  FSComm              FSComm
 
Warning               2              The following environment variables were not found: $(QTDIR)                Project FSComm
 
Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.
 
The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)
 
Lars

Do you have Qt 4.6 installed?
-MC
 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. 		 	   		  
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From jeff at jefflenk.com  Thu Jan  7 19:39:41 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 7 Jan 2010 21:39:41 -0600
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <ebbc2d551001071928u422b1884r6ba556b8b7a06e80@mail.gmail.com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>,
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>,
	<014801ca8fee$b75f8780$261e9680$@com>,
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>,
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>,
	<ebbc2d551001071928u422b1884r6ba556b8b7a06e80@mail.gmail.com>
Message-ID: <COL117-W15F7382FBF45603F4000E1B9700@phx.gbl>


yep sure thing I will update/add some of this information tom. after updating the project files.
 


Date: Fri, 8 Jan 2010 01:28:31 -0200
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

Jeff, any chance we can get this on the wiki?


We have created a page here: http://wiki.freeswitch.org/wiki/FSComm


I am looking for a Windows machine to do testing on it as well.


Regards,Jo?o Mesquita



On Fri, Jan 8, 2010 at 1:16 AM, Jeff Lenk <jeff at jefflenk.com> wrote:


The windows support is very experimental at this time!
 
You must manually install http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
 
Then set the environment variable QTDIR in the environment variables. This can be set from the Computer/Properties/Advanced system settings/Environment Variables/User Variables settings screen.
 
QTDIR=c:\qt\4.6.0 - or wherever you installed it
 
then restart VS
 

 


Date: Thu, 7 Jan 2010 21:45:56 -0200
From: jmesquita at freeswitch.org



To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

I don't have a Windows machine to test that. Maybe jlenk could give us a hand since he is the one who has created the visual studio project? 


Regards,Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927



On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:




No Qt installed. I just checked out from http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into VS2008. 

Do I need to get http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?

 


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?




 

 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:



I just downloaded the fscomm project and loaded it into vs2008. I?ve never programmed in C++ (or c), just C#, so I can?t make anything of the following two messages:

 

Error      1              error PRJ0019: A tool returned an error code from "RCC resources.qrc"  FSComm              FSComm

 

Warning               2              The following environment variables were not found: $(QTDIR)                Project FSComm

 

Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.

 

The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)

 

Lars


Do you have Qt 4.6 installed?
-MC

 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now.
_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From mike at jerris.com  Thu Jan  7 19:43:44 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 7 Jan 2010 22:43:44 -0500
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
Message-ID: <9AC8A302-DAC3-42FE-B1BB-91EB8A717248@jerris.com>

what is FAS ?

Mike

On Jan 7, 2010, at 8:05 PM, Ahmed Naji wrote:

> Was just wondering what/if anyone is doing any work on FAS detection and spoofed ring tones. Be great to discuss some ideas.




From max.bridgewater at gmail.com  Thu Jan  7 19:48:45 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Thu, 7 Jan 2010 22:48:45 -0500
Subject: [Freeswitch-users] hw:dummy not visible
Message-ID: <c55d46951001071948s558060fera2b08fecd667c544@mail.gmail.com>

Hi,

I got a few more Skypiax questions. Please bear with me.

1) Skype should use hw:dummy as audio device. But where do I set this on
Skype? In Options>Sound Devices, the only devices I see are; "Default
device" and "hdmi". My guess was that alsa-utils or some other ALSA related
lib would install this. But it seems this is not happening. Am i missing
something?

2) To create the configuration directory, I connect to the FreeNX XServer
running on my remote machine. I use NX to connect to the remote Xserver. I
can start all sort of GUI applications this way. But when I try to run
skypiax_auth, I get the following error message: "Cannot open X Display
':0.0', exiting". Any idea?

Thanks again,

Max.
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From nicolas at medularis.com  Thu Jan  7 19:58:26 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 8 Jan 2010 00:58:26 -0300
Subject: [Freeswitch-users] Ruby ESL missing pthread
Message-ID: <1b46b4e81001071958i17ccb6bey6eeecd9d94eb8438@mail.gmail.com>

I followed the instructions on the wiki to compile the Ruby version of
ESL, but then when I tried to run the examples, I kept getting a
"undefined symbol: pthread_mutexattr_init" error. I ran ldd on the
ESL.so file in the libs/esl/ruby folder and found it wasn't linked
against pthread, so I manually added -lpthread to the Makefile,
recompiled and got it to work. The only other version I tried was the
Perl one and it compiled and worked without needing anything.



From mrene_lists at avgs.ca  Thu Jan  7 20:38:30 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 8 Jan 2010 05:38:30 +0100
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <9AC8A302-DAC3-42FE-B1BB-91EB8A717248@jerris.com>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
	<9AC8A302-DAC3-42FE-B1BB-91EB8A717248@jerris.com>
Message-ID: <065B7BF9-3D3E-4FB5-84DC-ABB7BC24D16E@avgs.ca>

False Answer Supervision.. 200 ok while it's still ringing

Sent from my iPhone

On 2010-01-08, at 4:43 AM, Michael Jerris <mike at jerris.com> wrote:

> what is FAS ?
>
> Mike
>
> On Jan 7, 2010, at 8:05 PM, Ahmed Naji wrote:
>
>> Was just wondering what/if anyone is doing any work on FAS  
>> detection and spoofed ring tones. Be great to discuss some ideas.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From jcasale at activenetwerx.com  Thu Jan  7 20:57:10 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 8 Jan 2010 04:57:10 +0000
Subject: [Freeswitch-users] SPA3102 Help
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9072618@prato.activenetwerx.local>

Reading the wiki, I have Line 1 (the fxs port) configured as ext 1001
and PSTN User (the fxo port) configured as ext 1000. I am a bit unsure
of the dial plan part of the wiki's config?

Under "VoIP-To-PSTN Gateway Setup" I set "Line 1 VoIP Caller DP:" to a
dial plan with (xx.) which dials whatever is sent to it?

Under "PSTN-To-VoIP Gateway Setup" I set "PSTN Caller Default DP:" to a
dial plan with (<:ABCD>S0) where ABCD could be a group dial for example
like 2000 as in the default config?

Do I understand this correctly? I would try this out, but I don't have a
pstn here and want to pass this off configured for delivery tomorrow and
be prepared for any adjustments via ssh if need be...

Thanks!
jlc



From jmesquita at freeswitch.org  Thu Jan  7 21:17:53 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 8 Jan 2010 02:17:53 -0300
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <COL117-W15F7382FBF45603F4000E1B9700@phx.gbl>
References: <012901ca8fe6$e36b71c0$aa425540$@com>
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>
	<014801ca8fee$b75f8780$261e9680$@com>
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>
	<ebbc2d551001071928u422b1884r6ba556b8b7a06e80@mail.gmail.com>
	<COL117-W15F7382FBF45603F4000E1B9700@phx.gbl>
Message-ID: <ebbc2d551001072117i5759a8efn680024657964c964@mail.gmail.com>

You are tha man! Thank you!

On Friday, January 8, 2010, Jeff Lenk <jeff at jefflenk.com> wrote:
>
>
>
>
>
> yep sure thing I will?update/add some of this information?tom. after updating the project files.
>
>
> Date: Fri, 8 Jan 2010 01:28:31 -0200
> From: jmesquita at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Compile error fscomm?
>
> Jeff, any chance we can get this on the wiki?
>
>
> We have created a page here: http://wiki.freeswitch.org/wiki/FSComm
>
>
> I am looking for a Windows machine to do testing on it as well.
>
>
> Regards,Jo?o Mesquita
>
>
> On Fri, Jan 8, 2010 at 1:16 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>
> The windows support is very experimental at this time!
>
> You must manually install http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
>
> Then set the environment variable QTDIR in the environment variables. This can be set from the Computer/Properties/Advanced system settings/Environment Variables/User Variables settings screen.
>
> QTDIR=c:\qt\4.6.0 - or wherever you installed it
>
> then restart VS
>
>
>
>
> Date: Thu, 7 Jan 2010 21:45:56 -0200
> From: jmesquita at freeswitch.org
>
>
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Compile error fscomm?
>
> I don't have a Windows machine to test that. Maybe jlenk could give us a hand since he is the one who has created the visual studio project?
>
>
> Regards,Jo?o Mesquita
> FreeSWITCH? Solutions
> t: +1 (646) 4959927
>
>
> On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:
>
>
>
> No Qt installed. I just checked out from http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into VS2008.
> Do I need to get http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
> Sent: Thursday, January 07, 2010 2:36 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Compile error fscomm?
>
>
>
>
>
>
>
> On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:
>
>
> I just downloaded the fscomm project and loaded it into vs2008. I?ve never programmed in C++ (or c), just C#, so I can?t make anything of the following two messages:
>
> Error????? 1????????????? error PRJ0019: A tool returned an error code from "RCC resources.qrc"? FSComm????????????? FSComm
>
> Warning?????????????? 2????????????? The following environment variables were not found: $(QTDIR)??????????????? Project FSComm
>
> Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.
>
> The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)
>
> Lars
>
>
> Do you have Qt 4.6 installed?
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now.?<http://clk.atdmt.com/GBL/go/196390709/direct/01/>
> _______________________________________________
> FreeSWITCH-users mailing list
> Hotmail: Trusted email with powerful SPAM protection. Sign up now.?<http://clk.atdmt.com/GBL/go/196390707/direct/01/>
>

-- 
Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927



From dujinfang at gmail.com  Thu Jan  7 21:29:14 2010
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 8 Jan 2010 13:29:14 +0800
Subject: [Freeswitch-users] hw:dummy not visible
In-Reply-To: <c55d46951001071948s558060fera2b08fecd667c544@mail.gmail.com>
References: <c55d46951001071948s558060fera2b08fecd667c544@mail.gmail.com>
Message-ID: <23f91031001072129v6a229573t79b7a6c17c89486b@mail.gmail.com>

2010/1/8 Max Bridgewater <max.bridgewater at gmail.com>:
> Hi,
> I got a few more Skypiax questions. Please bear with me.
> 1) Skype should use?hw:dummy?as audio device. But where do I set this on
> Skype? In Options>Sound Devices, the only devices I see are; "Default
> device" and "hdmi". My guess was that?alsa-utils or some other ALSA related
> lib would install this. But it seems this is not happening. Am i missing
> something?

lsmod ?
modprobe snd_dummy ?


> 2) To create the configuration directory, I connect to the FreeNX XServer
> running on my remote machine. I use NX to connect to the remote Xserver. I
> can start all sort of GUI applications this way. But when I try to run
> skypiax_auth, I get the following error message: "Cannot open X Display
> ':0.0', exiting". Any idea?

skypiax_auth :101  #or other display number skype running on

> Thanks again,
> Max.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From nicolas at medularis.com  Thu Jan  7 21:33:30 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 8 Jan 2010 02:33:30 -0300
Subject: [Freeswitch-users] Need to fake ringback
Message-ID: <1b46b4e81001072133h3f4c9bf1g68e25d27fd71edc1@mail.gmail.com>

I'm trying to fake a ringback for leg1 of a two-legged call without success.

I'm doing this with a JS script, originating one leg first, then the
second and executing the bridge application on both afterwards. I
would like to fake a ringback on the first call while they wait for
the second call to connect.

I tried originating the call and then setting the ringback (like the
example here: http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones),
and I also tried setting the ringback in the originate command, but
none of those worked. I even tried using the variable
instant_ringback, but it didn't work either.


This is the code for the first case:

ostr = "{ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]sofia/gateway/mygw/123456789";
session1 = new Session(ostr);
if (session1.ready())
{
session1.execute("set","instant_ringback=%(2000,4000,440.0,480.0)");
}


And this is the originate string for the second case:

var stUsRing = "%(2000,4000,440,480)";
ostr = "{ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,ringback='"+stUsRing+"',medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]sofia/gateway/mygw/123456789";


Thanks for your help!


Nico



From scott.torr.fs at letterboxes.org  Thu Jan  7 22:24:56 2010
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Fri, 08 Jan 2010 17:24:56 +1100
Subject: [Freeswitch-users] Segmentation fault (core dumped) on "shutdown"
	if mod_skypiax loaded
Message-ID: <1262931896.12045.1353566301@webmail.messagingengine.com>

Hi,

I'm getting a Segmentation fault on "shutdown" if mod_skypiax is loaded
on the current build.
The audio on skypiax calls is also extremely choppy.

This only occurred after a "make current" from 15787.

Has anyone else experienced this problem?

VMware Server Version 2.0.1
ubuntu-8.04.3-server-amd64.iso (update/upgrade)
FreeSWITCH Version 1.0.trunk (16172)
skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb
mod_skypiax 

http://jira.freeswitch.org/browse/MODSKYPIAX-68

Regards,
Scott

 



From pmhshz at gmail.com  Thu Jan  7 22:46:41 2010
From: pmhshz at gmail.com (shehzad p)
Date: Thu, 7 Jan 2010 22:46:41 -0800 (PST)
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<B98B6921358D0C49BD21CD023F1F52C302623BE0@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
Message-ID: <27071973.post@talk.nabble.com>




Brian West-3 wrote:
> 
> You could but I think you want to stream RTP to a multicast it would  
> be better off building an rtp format mod so you can record rtp:// 
> x.x.x.x:5000 and play from rtp://y.y.y.y:5000
> 
> /b
> 
> 

I was looking for such functionality, but unfortunately it seems not present
right now, I am willing to build the rtp format mod as described by Brian.,
and will provide back to trunk. 

Although I have modified mod_skel application for use in dialplan in custom
application, I need to have a basic understanding regarding format mod. 
Will anybody please guide me from where to starts?
-- 
View this message in context: http://old.nabble.com/stream-a-file-multicast-with-mod_esf-tp21976696p27071973.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From pmhshz at gmail.com  Thu Jan  7 22:51:42 2010
From: pmhshz at gmail.com (shehzad p)
Date: Thu, 7 Jan 2010 22:51:42 -0800 (PST)
Subject: [Freeswitch-users] Re cording call into existing file
Message-ID: <26975973.post@talk.nabble.com>


Hi,

while recording a file using session_record, can i continue the existing
recorded file?  So that the existing record will remain as it is and new
recording will be added into that file?

Thanks
msp
-- 
View this message in context: http://old.nabble.com/Recording-call-into-existing-file-tp26975973p26975973.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From pmhshz at gmail.com  Thu Jan  7 22:57:30 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Fri, 8 Jan 2010 12:27:30 +0530
Subject: [Freeswitch-users] Re cording call into existing file
In-Reply-To: <191c3a030912310704g5c57d296pbbe553aa46c63c4e@mail.gmail.com>
References: <26975973.post@talk.nabble.com>
	<191c3a030912310704g5c57d296pbbe553aa46c63c4e@mail.gmail.com>
Message-ID: <a792304b1001072257r7c29a43fo4a772c15a092616f@mail.gmail.com>

On Thu, Dec 31, 2009 at 8:34 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> set RECORD_APPEND=true on the channel and all recordings will behave this
> way to formats which support it
> (curently mod_sndfile for WAV etc)
>
>
> On Thu, Dec 31, 2009 at 12:49 AM, shehzad p <pmhshz at gmail.com> wrote:
>
>>
>> Hi,
>>
>> while recording a file using session_record, can i continue the existing
>> recorded file?  So that the existing record will remain as it is and new
>> recording will be added into that file?
>>
>> Thanks
>> msp
>> --
>> View this message in context:
>> http://old.nabble.com/Recording-call-into-existing-file-tp26975973p26975973.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
Great, Thanks Anthony.
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From jingwei.yang at gmail.com  Fri Jan  8 00:42:20 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Fri, 8 Jan 2010 16:42:20 +0800
Subject: [Freeswitch-users]  IVR and TTS
Message-ID: <13529f9d1001080042w420c9f0vdbc5b7a57edd1e6c@mail.gmail.com>

Hi Guys,

I need to implement a function using IVR and TTS. Here's the scenario.

1. User A calls in
2. FS plays a welcome message and directs A to press '1' to continue
3. FS detects A's number and finds A's address from the database and plays
another piece of voice message including the address info just found

I understand this logic can be implemented using javascript. However, in
this scenario, the database is a remote one and the local js has no access
to it. What I'm planning to do is write a Java program, talking to FS via
ESL. Could someone please tell me what event FS will trigger after user A
selects a certain option and how to inform the FS to continue the rest of
IVR menu after finding the address?

Thanks,
-Jingwei
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From mattdfong at gmail.com  Fri Jan  8 01:29:32 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Fri, 8 Jan 2010 16:29:32 +0700
Subject: [Freeswitch-users] Connecting GSM Cards to FreeSWITCH
Message-ID: <4256bf831001080129m69c6b5d1s20557da27f040bca@mail.gmail.com>

Does anyone have any hardware recommendations for setting up 4+ GSM cell
phone lines to work with FreeSWITCH?
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From jonas.gauffin at gmail.com  Fri Jan  8 01:39:46 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Fri, 8 Jan 2010 10:39:46 +0100
Subject: [Freeswitch-users] hangup cause: NORMAL_UNSPECIFIED
Message-ID: <d021d3be1001080139m5b4cd7c6ge87b19e4ad6a62fe@mail.gmail.com>

Hello,

Is it possible to get a more detailed reason (in the log) to
why NORMAL_UNSPECIFIED was returned as hang up cause?

2010-01-08 09:30:50.987200 [DEBUG] sofia.c:3831 Channel
sofia/external/070738xxxx entering state [terminated][904]
2010-01-08 09:30:50.987200 [NOTICE] sofia.c:4461 Hangup
sofia/external/070738xxxx [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED]

Regards,
  Jonas
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From darklion11 at yahoo.com  Fri Jan  8 02:25:10 2010
From: darklion11 at yahoo.com (Edmar Cruz)
Date: Fri, 8 Jan 2010 02:25:10 -0800 (PST)
Subject: [Freeswitch-users] Personal Greeting
In-Reply-To: <1262066443847-4226681.post@n2.nabble.com>
References: <1262066443847-4226681.post@n2.nabble.com>
Message-ID: <27073953.post@talk.nabble.com>


You can set it in the dialplan

<action application="playback" data="yourgreeting.wav" />

For some cases softphones has its own greeting :working:

Hope it can help you..


sharad-5 wrote:
> 
> 
> 
> Hi
> 
> I am new to Freeswitch so my question may be a stupid question.
> 
> I just want to know how to disable the personal greeting to the default
> one. 
> One user has recorded his personal greeting now how can he make this
> default.
> 
> I could not find any option for the same.
> 
> Plz advice.
> 
> Thanks & regards
> Sharad garg
> -- 
> View this message in context:
> http://n2.nabble.com/Personal-Greeting-tp4226681p4226681.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://old.nabble.com/Personal-Greeting-tp26951471p27073953.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From gavin.henry at gmail.com  Fri Jan  8 03:31:57 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 8 Jan 2010 11:31:57 +0000
Subject: [Freeswitch-users] FSComm builds OK on Fedora F-12 with QT4 4.5.3
	but doesn't save SIP account
Message-ID: <13ca621c1001080331l363122a5l3b093adb5782538e@mail.gmail.com>

Hi,

Just a quick one to say this builds ok with:

Compiled FSComm version: 1.0.trunk (16209M)
FreeSWITCH Version 1.0.trunk (16209M)

But it doesn't want to save my SIP account details. How to debug?

Thanks.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From vinuth.madinur at gmail.com  Fri Jan  8 04:16:20 2010
From: vinuth.madinur at gmail.com (Vinuth Madinur)
Date: Fri, 8 Jan 2010 17:46:20 +0530
Subject: [Freeswitch-users] IVR and TTS
In-Reply-To: <13529f9d1001080042w420c9f0vdbc5b7a57edd1e6c@mail.gmail.com>
References: <13529f9d1001080042w420c9f0vdbc5b7a57edd1e6c@mail.gmail.com>
Message-ID: <910309031001080416r6064670bic4e6285494f8d25b@mail.gmail.com>

Hi,

"DTMF" event will be raised in ESL, when "A" presses a key. It'll be raised
for each key pressed.

Alternatively you can use play_and_get_digits.

To continue FS execution after you fetch the address, you just need to
invoke the "speak" command on that socket.

Since FS is handling inbound calls, you can use the outbound event socket,
where a new connection will be opened per call from FS to your java program.

One way to know what events are raised in ESL, you can telnet to 8021 port,
authenticate and send "events plain all" command. Configure FS dialplan for
an extension which will just answer a call when it comes in. Then call this
extension from a softphone, press a key and you'll see the corresponding
event in the telnet console.


Thanks,
Vinuth.



On Fri, Jan 8, 2010 at 2:12 PM, Jingwei Yang <jingwei.yang at gmail.com> wrote:

> Hi Guys,
>
> I need to implement a function using IVR and TTS. Here's the scenario.
>
> 1. User A calls in
> 2. FS plays a welcome message and directs A to press '1' to continue
> 3. FS detects A's number and finds A's address from the database and plays
> another piece of voice message including the address info just found
>
> I understand this logic can be implemented using javascript. However, in
> this scenario, the database is a remote one and the local js has no access
> to it. What I'm planning to do is write a Java program, talking to FS via
> ESL. Could someone please tell me what event FS will trigger after user A
> selects a certain option and how to inform the FS to continue the rest of
> IVR menu after finding the address?
>
> Thanks,
> -Jingwei
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vhatz at kinetix.gr  Fri Jan  8 04:44:21 2010
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Fri, 08 Jan 2010 14:44:21 +0200
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
Message-ID: <4B4728A5.4040805@kinetix.gr>

Hello Ahmed,

I don't think there is a reliable way to detect FAS on a per call basis. 
Even audio detection software can be confused by strange ringtones, 
answering machines, etc.

The most reliable way we have found is to use statistics from CDRs and 
see if the INVITE-to-200(OK) delay averaged over a number of calls 
appears to be too small. If it is, then it is possible that you got FAS 
for those calls.

But this can only tell you if you have been experiencing FAS in past 
calls, ie you will not know you are getting FAS in real time.

It is still useful however: after you detect a possible FAS case via 
statistics you can place a few test calls yourself to verify that there 
actually exists FAS (and this is the only information that a carrier 
will accept in a trouble ticket to prove to them that they actually give 
you FAS).

I hope this helps.

Best regards,
Vlasis Hatzistavrou.


On 8/1/10 3:05 ??, Ahmed Naji wrote:
> Hi everyone,
>
> Was just wondering what/if anyone is doing any work on FAS detection 
> and spoofed ring tones. Be great to discuss some ideas.
>
> Regards,
>
> Ahmed.
>
> -- 
> Ahmed Naji
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From codecomplete at free.fr  Fri Jan  8 05:03:00 2010
From: codecomplete at free.fr (Fred-145)
Date: Fri, 8 Jan 2010 05:03:00 -0800 (PST)
Subject: [Freeswitch-users]  IP PBX and NAT firewalls
Message-ID: <27075600.post@talk.nabble.com>


Hello,

    I read a couple of thorough articles on SIP, and I'd like to make sure I
got things right when it comes to using SIP with NAT routers.
   
I know that, ideally, the IP PBX should be located in the DMZ to void
NAT-related issues in SIP, but SOHO routers don't necessarily support this,
so I'll assume that the SIP caller "Alice" and the IP PBX (eg. Freeswitch or
Asterisk) server are located in a non-routable, private LAN, while the
remote callee "Bob" is located on the Internet (either behind their own NAT
router, or connected with a public, routable address). The SIP phone of
Alice and Bob are both logged on to the Freeswitch server:

http://img46.imageshack.us/img46/5120/sipnatrouters.jpg

1. When Alice wants to call Bob, her SIP phone sends an SIP packet to the
Freeswitch server with her private IP address and a UDP port that it opened
to let incoming RTP packets from Bob

2. Freeswitch rings Bob's phone through the UDP port is used to register
with Freeswitch (usually, UDP5060). Bob's phone replies to Freeswitch with
his public IP address and the RTP port it chose to receive voice packets
from Alice

3. Once Bob picks up the phone, RTP voice packets flow directly between
Alice and Bob, while Freeswitch remains in the loop to handle call signaling
such as closing the connection when someone hangs up the call.

Provided this is how things work... there are three issues when one or all
SIP end-points are located in a (different) private LAN:
1. End-points use their private IP and a private UDP port for RTP. A server
has to translate this into a routable IP address, and...
2. it must negotiate with the NAT firewall to make sure this RTP port is
available, and if not, open some other port, and...
3. the server must rewrite the SDP packet to use this public port

I have a couple of questions:

1. Can Freeswitch/Asterisk handle this rewriting/negotiation?

2. Provided the NAT firewall doesn't support UPnP/NAT-PMP, does it mean I
must a) enable STUN in Freeswitch, b) set SIP end-points so that they use a
fixed port for RTP, and c) configure the NAT firewall to map this UDP port
to point to the SIP end-point?

3. Should SIP end-points be configured to use STUN/NAT, or should I let the
server handle the IP/port rewriting itself?

Thank you for any help.
-- 
View this message in context: http://old.nabble.com/IP-PBX-and-NAT-firewalls-tp27075600p27075600.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From steve at justfone.com  Fri Jan  8 05:17:53 2010
From: steve at justfone.com (Steven Brown)
Date: Fri, 8 Jan 2010 13:17:53 +0000
Subject: [Freeswitch-users] Connecting GSM Cards to FreeSWITCH
Message-ID: <3e6d7b0c1001080517i7af2a275oda7baac9429a9c29@mail.gmail.com>

Hi,

I have used Portech gateways with good results, you can get cards and
external gateways with various numbers of GSM channels, see

http://www.portech.com.tw/p3-product1_1.asp?Pid=13

Regards

Steve



From gmaruzz at celliax.org  Fri Jan  8 05:23:46 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 8 Jan 2010 14:23:46 +0100
Subject: [Freeswitch-users] Connecting GSM Cards to FreeSWITCH
In-Reply-To: <3e6d7b0c1001080517i7af2a275oda7baac9429a9c29@mail.gmail.com>
References: <3e6d7b0c1001080517i7af2a275oda7baac9429a9c29@mail.gmail.com>
Message-ID: <7b197bef1001080523g31c28424i3a1ba13049421642@mail.gmail.com>

Hi Mattew,

in a short while (before monday) will be available mod_gsmopen that
maybe can fit your needs.

http://wiki.freeswitch.org/wiki/GSMopen

-giovanni

On Fri, Jan 8, 2010 at 2:17 PM, Steven Brown <steve at justfone.com> wrote:
> Hi,
>
> I have used Portech gateways with good results, you can get cards and
> external gateways with various numbers of GSM channels, see
>
> http://www.portech.com.tw/p3-product1_1.asp?Pid=13
>
> Regards
>
> Steve
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From gmaruzz at celliax.org  Fri Jan  8 05:38:19 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 8 Jan 2010 14:38:19 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <20100107182731.GL5547@tamay-dogan.net>
References: <20100103184636.GW5547@tamay-dogan.net>
	<20100106135210.GG5547@tamay-dogan.net> 
	<CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>
	<7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com> 
	<20100106171413.GI5547@tamay-dogan.net>
	<7b197bef1001060939rd1e08f3h2cb82cd89e50225e@mail.gmail.com> 
	<20100107182731.GL5547@tamay-dogan.net>
Message-ID: <7b197bef1001080538u5ed67082r6df84d287c0c06c1@mail.gmail.com>

On Thu, Jan 7, 2010 at 7:27 PM, Michelle Konzack
<linux4michelle at tamay-dogan.net> wrote:
>> it supports all modules that accepts standard ETSI AT-GSM commands
>> (so, let's say all of them).
>
> Currently I havepayed over 20.000 Euro for ETSI and ANS specs ?and ?have
> currently not the money to buy more... ?Are the ETSI ?AT-GSM ?specs ?are
> free available? ?-- ?If yes, I need them!

yes is part of the GSM specifications that are freely available. Just
google for them. Anyway, any GSM modem or module (but not all
cellphones) supports at least a fair share of those specs.

>
>> Maybe I do not understand the second question, what do you means for
>> Endpoints?
>>
>> If you're talking usb endpoints, you'll need a modem endpoint (that
>> can be seen as a serial port), and (if you need audio, eg not just
>> SMSs but voice calls too) you need an audio endpoint (that can be seen
>> as a soundcard).
>
> I mean the USB-Endpoints...
>
> If you have an USB-Microcontroller where the USB port is ?a ?device, ?it
> identify it self over the Endpoint 0 and is for us non usable.
>
> And no it comes, where I haveproblems to understnd HOW the GSM Modem ?is
> working but I will assume some things:
>
> The EP1 of the USB-pot is configured for bidirectional Data transmission
> andwill controll out Device and is normaly ?configured ?as ?/dev/ttyUSB0
> and this is, where we use the AT commands to ?get ?infos ?from ?the ?GSM
> modem/cellphone and send/receive SMS.
>
> Now We need EP2 and configure it as streaming output for the Audio port.
>
> EP3 would be the streaming input for the Audio Port.
>
> is this right up to here?
>
> Then, EP4 would be the bidirectinal dataport ?for ?the ?UMTS ?and ?HS*PA
> Tranceiver, since it is entirely independant from the rest ?of ?the ?GSM
> modem/cellphone.
>
> Is this right?

Is completely up to the implementation.

I suggest you use lsusb with full debug/verbosity turned on, it will
tell you (almost) all.


>
> If yes, then it is easier as I was thinking...
>
>
>> Many modules and cellphones can be seen as HDSPA or GPRS modems, just
>> check their specs.
>
> My "Nokia 6120 classic" has in total 13 endpoints... ?Hell, where can ?I
> get an USB-Microcontroller which support this mass of USB Endpoints?
>
> Most ARM9/11 support not more then 7 or 9. ?:-(

I think you just needs to interface the modem endpoint and the audio
endpoint. The others are probably Human Interfaces for volume, color,
keyboard, whatever. You control the full phone features through AT
commands exchanged through the modem endpoint.


>
>> For audio, if the module/cellphone/modem does not offer an audio usb
>> endpoint (eg cannot be seen as a soundcard) one trick is to connect
>> the headset jack to an usb soundcard (you can find soundcard with for
>> factor like a dongle based on cm-108 chipset for under $10). I'll
>> publish the schema of the cable needed from hadset jack in the
>> phone/module to the usb soundcard).
>>
>> If I have not get what you asked, please explain more your question.
>
> Most important things are the above desibed understanding ?problem ?with
> the USB Endpoints
>
> I have the HSPA and GS frontends (Maxim and Infineon chips) here and ?my
> selfemade simple GSM/GPRS cell-phone is already working, ?but ?has ?less
> functionality as the cell-phones from Year 2000 ?:-D

you don't need them all (at least if you don't want to make dirty pics
with a kludge full of wires... hey, that can be arousing! :))

I suggest you use a ready-made GSM/GPRS/HDSPA/whatever module, that
contains all that you need and is available from Chinese suppliers for
very low prices. You can find some module that allows you to directly
tap in the GSM pcm audio stream, that would means you will not need to
sample and convert from analog to digital (so, no cpu power at all, no
dsp, no nothing).


If you would like to keep me in the loop, I would like to know how you progress.

Happy hacking

-giovanni

>
> Hey, ist is my first experience of developing GSM stuff.
>
> I an to develop a VERY simple GSM/UMTS/HSPA USB-Modem which do ?its ?job
> without balast.
>
> So if someone can help me with infos, I am very open...
>
> I prefer FreeSwitch over Asterisk which froze in the ?last ?2 ?years ?to
> many times in situations where it should not freeze, exspecialy if ?I ?a
> call a Chip-Manfacturer (Maxim/TI) Tech-Support. ?-- ?It is not funny!
>
> Thanks, Greetings and nice Day/Evening
> ? ?Michelle Konzack
> ? ?Systemadministrator
> ? ?Electronic Engineer
> ? ?Tamay Dogan Network
> ? ?Debian GNU/Linux Consultant
>
> --
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
> ##################### Debian GNU/Linux Consultant #####################
> <http://www.tamay-dogan.net/> ? ? ? ? ? ? ? ? Michelle Konzack
> <http://www.can4linux.org/> ? ? ? ? ? ? ? ? ? Apt. 917
> <http://www.flexray4linux.org/> ? ? ? ? ? ? ? 50, rue de Soultz
> Jabber linux4michelle at jabber.ccc.de ? ? ? ? ? 67100 Strabourg/France
> IRC ? ?#Debian (irc.icq.com) ? ? ? ? ? ? ? ? ?Tel. DE: +49 177 9351947
> ICQ ? ?#328449886 ? ? ? ? ? ? ? ? ? ? ? ? ? ? Tel. FR: +33 ?6 ?61925193
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From kdjakovic at hotmail.com  Fri Jan  8 06:03:55 2010
From: kdjakovic at hotmail.com (katarina djakovic)
Date: Fri, 8 Jan 2010 15:03:55 +0100
Subject: [Freeswitch-users] How to suspend certain users from the
	system	using flush_inbound_reg
In-Reply-To: <59CAAB6B-5259-4B78-9C9C-676ACF04D6B1@jerris.com>
References: <SNT137-w54C41FF82F0E62B26415E9B3750@phx.gbl>,
	<59CAAB6B-5259-4B78-9C9C-676ACF04D6B1@jerris.com>
Message-ID: <SNT137-w3221FCED6EE7F190EDCC65B3700@phx.gbl>


Dear all,

 

thanks for your suggestions, they helped us to understand what was happening. 

 

The mistake was ours, we had auth-calls parameter in the profile set to false, and that was the cause of the problem, since the calls went through regardless of the directory settings.

 

Thanks again,

Katarina
 


From: mike at jerris.com
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 3 Jan 2010 20:24:15 -0500
Subject: Re: [Freeswitch-users] How to suspend certain users from the system using flush_inbound_reg


First a note.  Registration and authentication are completely different. Removing the registration has to do with the switch knowin where to send the calls and nothing to do with auth for receiving calls.  There is one caveat to this.  We do support nonce count, and it could be using the auth from the previous registration that is still valid.  Double check the nc from the registrations and the call and see if that rings true.  We may want to add something to explicitly expire the nonce when youflush reg but I need some confirmation on that first.  Otherwise the other responces seem to cover the possibilities.  Crank up the debug and check sip trace for more details on what is allowing the call through and report back.


Mike

On Jan 3, 2010, at 11:50 AM, katarina djakovic <kdjakovic at hotmail.com> wrote:




Hi,
 
we are trying to figure out how to suspend certain subscribers from our system and we have some problems with removing thier 
registrations. The UAs are ATAs.
 
This is what we do:
 
1) We remove the subscriber extension from the conf\directory .xml files
2) We do reloadxml
3) We flush user's registration with flush_inbound_reg 
 
but, the users are still able to make calls as if they were still registered. To make it clearer, their registrations are removed from 
the registration list (checked with sofia status), but they system still accepts the calls from them. From this, it seems that if ATA is never rebooted - we are not able to ban these users from the system.
 
Only after the ATA is rebooted user is not able to make calls any more, as the ATA can not register any more - since they users are 
removed from the directory. But before we reboot ATA everything works as nothing had been done.
 
Does anyone have an idea what are we doing wrong? We expect that after the registration is removed from the FS the UA should not be 
able to make a call but this is not what happnes.
 
Can anybody help please?
Thanks,
Katarina



Keep your friends updated? even when you?re not signed in. 

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_3:092010
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From max.bridgewater at gmail.com  Fri Jan  8 06:14:39 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 8 Jan 2010 09:14:39 -0500
Subject: [Freeswitch-users] hw:dummy not visible
In-Reply-To: <23f91031001072129v6a229573t79b7a6c17c89486b@mail.gmail.com>
References: <c55d46951001071948s558060fera2b08fecd667c544@mail.gmail.com>
	<23f91031001072129v6a229573t79b7a6c17c89486b@mail.gmail.com>
Message-ID: <c55d46951001080614u3024b2a3t71f8d8e066ebe13a@mail.gmail.com>

Wow, it works like a charm. And of course, your two suggestions were spot
on.

Gracie Mille.

Max.


On Fri, Jan 8, 2010 at 12:29 AM, Seven Du <dujinfang at gmail.com> wrote:

> 2010/1/8 Max Bridgewater <max.bridgewater at gmail.com>:
> > Hi,
> > I got a few more Skypiax questions. Please bear with me.
> > 1) Skype should use hw:dummy as audio device. But where do I set this on
> > Skype? In Options>Sound Devices, the only devices I see are; "Default
> > device" and "hdmi". My guess was that alsa-utils or some other ALSA
> related
> > lib would install this. But it seems this is not happening. Am i missing
> > something?
>
> lsmod ?
> modprobe snd_dummy ?
>
>
> > 2) To create the configuration directory, I connect to the FreeNX XServer
> > running on my remote machine. I use NX to connect to the remote Xserver.
> I
> > can start all sort of GUI applications this way. But when I try to run
> > skypiax_auth, I get the following error message: "Cannot open X Display
> > ':0.0', exiting". Any idea?
>
> skypiax_auth :101  #or other display number skype running on
>
> > Thanks again,
> > Max.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at celliax.org  Fri Jan  8 06:26:14 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 8 Jan 2010 15:26:14 +0100
Subject: [Freeswitch-users] hw:dummy not visible
In-Reply-To: <c55d46951001080614u3024b2a3t71f8d8e066ebe13a@mail.gmail.com>
References: <c55d46951001071948s558060fera2b08fecd667c544@mail.gmail.com> 
	<23f91031001072129v6a229573t79b7a6c17c89486b@mail.gmail.com> 
	<c55d46951001080614u3024b2a3t71f8d8e066ebe13a@mail.gmail.com>
Message-ID: <7b197bef1001080626y4ca13f2br6d5aceed9c2db9a8@mail.gmail.com>

On Fri, Jan 8, 2010 at 3:14 PM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Wow, it works like a charm. And of course, your two suggestions were spot
> on.

Seven is an authority on using mod_skypiax!

-giovanni


>
> Gracie Mille.
>
> Max.
>
>
> On Fri, Jan 8, 2010 at 12:29 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> 2010/1/8 Max Bridgewater <max.bridgewater at gmail.com>:
>> > Hi,
>> > I got a few more Skypiax questions. Please bear with me.
>> > 1) Skype should use?hw:dummy?as audio device. But where do I set this on
>> > Skype? In Options>Sound Devices, the only devices I see are; "Default
>> > device" and "hdmi". My guess was that?alsa-utils or some other ALSA
>> > related
>> > lib would install this. But it seems this is not happening. Am i missing
>> > something?
>>
>> lsmod ?
>> modprobe snd_dummy ?
>>
>>
>> > 2) To create the configuration directory, I connect to the FreeNX
>> > XServer
>> > running on my remote machine. I use NX to connect to the remote Xserver.
>> > I
>> > can start all sort of GUI applications this way. But when I try to run
>> > skypiax_auth, I get the following error message: "Cannot open X Display
>> > ':0.0', exiting". Any idea?
>>
>> skypiax_auth :101 ?#or other display number skype running on
>>
>> > Thanks again,
>> > Max.
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From gmaruzz at celliax.org  Fri Jan  8 06:29:51 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 8 Jan 2010 15:29:51 +0100
Subject: [Freeswitch-users] Skypiax on CentOS
In-Reply-To: <c55d46951001071903q5a23639ehee09da7640351a74@mail.gmail.com>
References: <c55d46951001071836i55516863r359bb56695744054@mail.gmail.com> 
	<6b65470d1001071852i752b8306r51a48576090b4be@mail.gmail.com> 
	<c55d46951001071903q5a23639ehee09da7640351a74@mail.gmail.com>
Message-ID: <7b197bef1001080629h6b812e0dt67d66dcb740d21a7@mail.gmail.com>

I would use the version signaled by William (
http://download.skype.com/linux/skype-2.0.0.72-centos.i586.rpm ) or,
better yet, if you can find the static version of 2.0.0.72.

If you, or others, find that static version, please post the link here.

Thanks,

-giovanni


On Fri, Jan 8, 2010 at 4:03 AM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Thanks you very much guys.
> Ad Brian: I tried the static binary build but was blocked by a qt4-x11
> missing and I couldn't find an obvious solution to that. Hence my ?inquiry
> in the group before spending time on things that other people potentially
> already solved.
> max.
>
> On Thu, Jan 7, 2010 at 9:52 PM, William Suffill <william.suffill at gmail.com>
> wrote:
>>
>> Skype appears to be pushing a new version on the linux side but the old
>> packages are still available but not linked anywhere.
>>
>> http://download.skype.com/linux/skype-2.0.0.72-centos.i586.rpm
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From Claudio.Cavalera at italtel.it  Fri Jan  8 06:30:05 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Fri, 8 Jan 2010 15:30:05 +0100
Subject: [Freeswitch-users] Codecs and things
In-Reply-To: <7A65B817-E490-4582-8D43-8531FFA61CC4@freeswitch.org>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F362C3@BESONE.corp.dom>

Hi Bkw,
my doubt is not the price but about the availability of the official
mod_G729 to be used on embedded hardware.
Thanks,
Claudio


________________________________

	From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West
	Sent: Thursday, January 07, 2010 5:57 PM
	To: freeswitch-users at lists.freeswitch.org
	Subject: Re: [Freeswitch-users] Codecs and things
	
	
	You usually still have to pay a license even if you buy a DSP
that is capable of doing it.   

	/b

	On Jan 7, 2010, at 10:46 AM, Cavalera Claudio Luigi wrote:


		Hello Steve,
		from what you have written it seems very unlikely that
we are gonna buy
		the official G.729 codec for embedded hardware?
		I don't know much about it but would a MIPS32 24kf be
enough? Just
		speculating from here
	
http://www.mips.com/products/processors/32-64-bit-cores/mips32-24k/
		
		Thanks,
		Claudio



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From brian at freeswitch.org  Fri Jan  8 06:41:55 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 08:41:55 -0600
Subject: [Freeswitch-users] Need to fake ringback
In-Reply-To: <1b46b4e81001072133h3f4c9bf1g68e25d27fd71edc1@mail.gmail.com>
References: <1b46b4e81001072133h3f4c9bf1g68e25d27fd71edc1@mail.gmail.com>
Message-ID: <91E856D5-B499-4C5F-A6A3-E2BB36C926C6@freeswitch.org>

Are you using proxy media?

/b

On Jan 7, 2010, at 11:33 PM, Nicolas Brenner wrote:

> I'm trying to fake a ringback for leg1 of a two-legged call without success.




From brian at freeswitch.org  Fri Jan  8 06:43:07 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 08:43:07 -0600
Subject: [Freeswitch-users] hangup cause: NORMAL_UNSPECIFIED
In-Reply-To: <d021d3be1001080139m5b4cd7c6ge87b19e4ad6a62fe@mail.gmail.com>
References: <d021d3be1001080139m5b4cd7c6ge87b19e4ad6a62fe@mail.gmail.com>
Message-ID: <583FB3F4-9CFC-48A8-ADE0-30FD43F8F1F1@freeswitch.org>

Not without the siptrace and sofia loglevel all 9

/b

On Jan 8, 2010, at 3:39 AM, Jonas Gauffin wrote:

> Hello,
> 
> Is it possible to get a more detailed reason (in the log) to why NORMAL_UNSPECIFIED was returned as hang up cause?
> 
> 2010-01-08 09:30:50.987200 [DEBUG] sofia.c:3831 Channel sofia/external/070738xxxx entering state [terminated][904]
> 2010-01-08 09:30:50.987200 [NOTICE] sofia.c:4461 Hangup sofia/external/070738xxxx [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED]




From brian at freeswitch.org  Fri Jan  8 06:51:44 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 08:51:44 -0600
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <27071973.post@talk.nabble.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<B98B6921358D0C49BD21CD023F1F52C302623BE0@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
Message-ID: <378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>

www.bkw.org/mod_rtp_stream.tgz is a skel I started on.. You can take that and finish it up if you like.

/b

On Jan 8, 2010, at 12:46 AM, shehzad p wrote:

> I was looking for such functionality, but unfortunately it seems not present
> right now, I am willing to build the rtp format mod as described by Brian.,
> and will provide back to trunk. 
> 
> Although I have modified mod_skel application for use in dialplan in custom
> application, I need to have a basic understanding regarding format mod. 
> Will anybody please guide me from where to starts?




From max.bridgewater at gmail.com  Fri Jan  8 07:15:00 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 8 Jan 2010 10:15:00 -0500
Subject: [Freeswitch-users] Skypiax on CentOS
In-Reply-To: <7b197bef1001080629h6b812e0dt67d66dcb740d21a7@mail.gmail.com>
References: <c55d46951001071836i55516863r359bb56695744054@mail.gmail.com>
	<6b65470d1001071852i752b8306r51a48576090b4be@mail.gmail.com>
	<c55d46951001071903q5a23639ehee09da7640351a74@mail.gmail.com>
	<7b197bef1001080629h6b812e0dt67d66dcb740d21a7@mail.gmail.com>
Message-ID: <c55d46951001080715x1401b6f5tdde6c99b3c60655c@mail.gmail.com>

That's indeed what I used. Thanks everybody.

On Fri, Jan 8, 2010 at 9:29 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> I would use the version signaled by William (
> http://download.skype.com/linux/skype-2.0.0.72-centos.i586.rpm ) or,
> better yet, if you can find the static version of 2.0.0.72.
>
> If you, or others, find that static version, please post the link here.
>
> Thanks,
>
> -giovanni
>
>
> On Fri, Jan 8, 2010 at 4:03 AM, Max Bridgewater
> <max.bridgewater at gmail.com> wrote:
> > Thanks you very much guys.
> > Ad Brian: I tried the static binary build but was blocked by a qt4-x11
> > missing and I couldn't find an obvious solution to that. Hence my
>  inquiry
> > in the group before spending time on things that other people potentially
> > already solved.
> > max.
> >
> > On Thu, Jan 7, 2010 at 9:52 PM, William Suffill <
> william.suffill at gmail.com>
> > wrote:
> >>
> >> Skype appears to be pushing a new version on the linux side but the old
> >> packages are still available but not linked anywhere.
> >>
> >> http://download.skype.com/linux/skype-2.0.0.72-centos.i586.rpm
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.bridgewater at gmail.com  Fri Jan  8 07:19:52 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 8 Jan 2010 10:19:52 -0500
Subject: [Freeswitch-users] Skypiax and Socket API
Message-ID: <c55d46951001080719p743df4faufe6d3964060392cb@mail.gmail.com>

Hey,

I see that it's possible to send chat messages to Skype users using sk on
top of socket api and skypiax. But how do I received chat messages via the
socket api? Are there events generated by Skypiax? If so, do we have a list
somewhere?

Thanks,
Max.
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From pmhshz at gmail.com  Fri Jan  8 07:25:55 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Fri, 8 Jan 2010 20:55:55 +0530
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<B98B6921358D0C49BD21CD023F1F52C302623BE0@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
Message-ID: <a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>

Thanks Brian,

Great... Let me start the developing it.
Although I am analyzing the mod_local_stream, to understand exact working of
format module, Will anybody please let me know any similar thing, I can take
reference?


On Fri, Jan 8, 2010 at 8:21 PM, Brian West <brian at freeswitch.org> wrote:

> www.bkw.org/mod_rtp_stream.tgz is a skel I started on.. You can take that
> and finish it up if you like.
>
> /b
>
> On Jan 8, 2010, at 12:46 AM, shehzad p wrote:
>
> > I was looking for such functionality, but unfortunately it seems not
> present
> > right now, I am willing to build the rtp format mod as described by
> Brian.,
> > and will provide back to trunk.
> >
> > Although I have modified mod_skel application for use in dialplan in
> custom
> > application, I need to have a basic understanding regarding format mod.
> > Will anybody please guide me from where to starts?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Fri Jan  8 07:33:48 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 8 Jan 2010 10:33:48 -0500
Subject: [Freeswitch-users] Ruby ESL missing pthread
In-Reply-To: <1b46b4e81001071958i17ccb6bey6eeecd9d94eb8438@mail.gmail.com>
References: <1b46b4e81001071958i17ccb6bey6eeecd9d94eb8438@mail.gmail.com>
Message-ID: <8A3B5711-727D-4BDB-97AE-7AD20C4528F7@jerris.com>

Please open up a bug on jira for me for this issue.

Mike

On Jan 7, 2010, at 10:58 PM, Nicolas Brenner <nicolas at medularis.com>  
wrote:

> I followed the instructions on the wiki to compile the Ruby version of
> ESL, but then when I tried to run the examples, I kept getting a
> "undefined symbol: pthread_mutexattr_init" error. I ran ldd on the
> ESL.so file in the libs/esl/ruby folder and found it wasn't linked
> against pthread, so I manually added -lpthread to the Makefile,
> recompiled and got it to work. The only other version I tried was the
> Perl one and it compiled and worked without needing anything.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From mike at jerris.com  Fri Jan  8 07:47:34 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 8 Jan 2010 10:47:34 -0500
Subject: [Freeswitch-users] Personal Greeting
In-Reply-To: <27073953.post@talk.nabble.com>
References: <1262066443847-4226681.post@n2.nabble.com>
	<27073953.post@talk.nabble.com>
Message-ID: <53FBFF55-D94D-4BCA-81EC-230F00DDC344@jerris.com>

Huh?  What does this have to do with his question?

On Jan 8, 2010, at 5:25 AM, Edmar Cruz <darklion11 at yahoo.com> wrote:

>
> You can set it in the dialplan
>
> <action application="playback" data="yourgreeting.wav" />
>
> For some cases softphones has its own greeting :working:
>
> Hope it can help you..
>
>
> sharad-5 wrote:
>>
>>
>>
>> Hi
>>
>> I am new to Freeswitch so my question may be a stupid question.
>>
>> I just want to know how to disable the personal greeting to the  
>> default
>> one.
>> One user has recorded his personal greeting now how can he make this
>> default.
>>
>> I could not find any option for the same.
>>
>> Plz advice.
>>
>> Thanks & regards
>> Sharad garg
>> -- 
>> View this message in context:
>> http://n2.nabble.com/Personal-Greeting-tp4226681p4226681.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>>
>
> -- 
> View this message in context: http://old.nabble.com/Personal-Greeting-tp26951471p27073953.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From gmaruzz at celliax.org  Fri Jan  8 07:47:43 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 8 Jan 2010 16:47:43 +0100
Subject: [Freeswitch-users] Skypiax and Socket API
In-Reply-To: <c55d46951001080719p743df4faufe6d3964060392cb@mail.gmail.com>
References: <c55d46951001080719p743df4faufe6d3964060392cb@mail.gmail.com>
Message-ID: <7b197bef1001080747u7ea7852l4cf776e02ec07f16@mail.gmail.com>

yes, it uses standard MESSAGE messages (chat api)

to test it:

telnet localhost 8021

-> auth ClueCon

-> events plain message

or, if you prefere xml:

-> events xml message

this will subscribe to the events of type message.

You can send messages using the standard chat API of FS (like with
JINGLE and sofia/SIP/SIMPLE)

-giovanni


On Fri, Jan 8, 2010 at 4:19 PM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Hey,
>
> I see that it's possible to send chat messages to Skype users using sk on
> top of socket api and skypiax. But how do I received chat messages via the
> socket api? Are there events generated by Skypiax? If so, do we have a list
> somewhere?
>
> Thanks,
> Max.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From a.alalousi at gmail.com  Fri Jan  8 08:07:52 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Fri, 8 Jan 2010 16:07:52 +0000
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <4B4728A5.4040805@kinetix.gr>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
	<4B4728A5.4040805@kinetix.gr>
Message-ID: <cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>

Hi Valsis,

Thanks for this.

My line of thought is to tone-detect secondary ringing tones post 200(OK) to
detect FAS (False Answer Supervision). This should eliminate at least a good
proportion of calls, and it can be done real time through a
script/modules/...etc.

Working along this thought, at least you are minimising the hit cost-wise to
a few seconds at most.

As to answering machines and fake conferences, fake network messages ...etc,
one can possibly use voice detection, perhaps with heuristic and statistical
training.

Just a thought ..

Regards,

Ahmed.


2010/1/8 Vlasis Hatzistavrou (KTI) <vhatz at kinetix.gr>

>  Hello Ahmed,
>
> I don't think there is a reliable way to detect FAS on a per call basis.
> Even audio detection software can be confused by strange ringtones,
> answering machines, etc.
>
> The most reliable way we have found is to use statistics from CDRs and see
> if the INVITE-to-200(OK) delay averaged over a number of calls appears to be
> too small. If it is, then it is possible that you got FAS for those calls.
>
> But this can only tell you if you have been experiencing FAS in past calls,
> ie you will not know you are getting FAS in real time.
>
> It is still useful however: after you detect a possible FAS case via
> statistics you can place a few test calls yourself to verify that there
> actually exists FAS (and this is the only information that a carrier will
> accept in a trouble ticket to prove to them that they actually give you
> FAS).
>
> I hope this helps.
>
> Best regards,
> Vlasis Hatzistavrou.
>
>
> On 8/1/10 3:05 ??, Ahmed Naji wrote:
>
> Hi everyone,
>
> Was just wondering what/if anyone is doing any work on FAS detection and
> spoofed ring tones. Be great to discuss some ideas.
>
> Regards,
>
> Ahmed.
>
> --
> Ahmed Naji
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From oscav at hotmail.fr  Fri Jan  8 08:14:26 2010
From: oscav at hotmail.fr (Oscav)
Date: Fri, 8 Jan 2010 08:14:26 -0800 (PST)
Subject: [Freeswitch-users] Failed to connect to a SKYPE API
In-Reply-To: <27062783.post@talk.nabble.com>
References: <27062783.post@talk.nabble.com>
Message-ID: <27078464.post@talk.nabble.com>


Im' running FS on windows server 2003 64bits


Oscav wrote:
> 
> Hi,
> 
> I'm trying to use to SkypeIAX. When I load the mod_skypiax, I got the
> following error :
> 
>  Failed to connect to a SKYPE API for interface_id=1, no SKYPE client
> running, please (re)start Skype client. Skypiax exiting
> 
> Skype is running with the correct account and skypiax.conf use the same
> account. I was expecting a permission request from the Skype user but
> nothing happens.
> 
> Somebody knows how I can solve this ??
> 
> Many thanks.
> 

-- 
View this message in context: http://old.nabble.com/Failed-to-connect-to-a-SKYPE-API-tp27062783p27078464.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From max.bridgewater at gmail.com  Fri Jan  8 08:15:56 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 8 Jan 2010 11:15:56 -0500
Subject: [Freeswitch-users] Skypiax and Socket API
In-Reply-To: <7b197bef1001080747u7ea7852l4cf776e02ec07f16@mail.gmail.com>
References: <c55d46951001080719p743df4faufe6d3964060392cb@mail.gmail.com>
	<7b197bef1001080747u7ea7852l4cf776e02ec07f16@mail.gmail.com>
Message-ID: <c55d46951001080815j22388bcai471b9986747e631b@mail.gmail.com>

Awesome; I will try it and report next week on my experience.

On Fri, Jan 8, 2010 at 10:47 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> yes, it uses standard MESSAGE messages (chat api)
>
> to test it:
>
> telnet localhost 8021
>
> -> auth ClueCon
>
> -> events plain message
>
> or, if you prefere xml:
>
> -> events xml message
>
> this will subscribe to the events of type message.
>
> You can send messages using the standard chat API of FS (like with
> JINGLE and sofia/SIP/SIMPLE)
>
> -giovanni
>
>
> On Fri, Jan 8, 2010 at 4:19 PM, Max Bridgewater
> <max.bridgewater at gmail.com> wrote:
> > Hey,
> >
> > I see that it's possible to send chat messages to Skype users using sk on
> > top of socket api and skypiax. But how do I received chat messages via
> the
> > socket api? Are there events generated by Skypiax? If so, do we have a
> list
> > somewhere?
> >
> > Thanks,
> > Max.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Fri Jan  8 08:20:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Jan 2010 10:20:14 -0600
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
	<4B4728A5.4040805@kinetix.gr>
	<cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>
Message-ID: <191c3a031001080820i426194cw58b8a2cf8b123493@mail.gmail.com>

you could use the existing tone_detect app after the false answer to xfer to
an extension of your choice if ring tone was detected.


2010/1/8 Ahmed Naji <a.alalousi at gmail.com>

> Hi Valsis,
>
> Thanks for this.
>
> My line of thought is to tone-detect secondary ringing tones post 200(OK)
> to detect FAS (False Answer Supervision). This should eliminate at least a
> good proportion of calls, and it can be done real time through a
> script/modules/...etc.
>
> Working along this thought, at least you are minimising the hit cost-wise
> to a few seconds at most.
>
> As to answering machines and fake conferences, fake network messages
> ...etc, one can possibly use voice detection, perhaps with heuristic and
> statistical training.
>
> Just a thought ..
>
> Regards,
>
> Ahmed.
>
>
> 2010/1/8 Vlasis Hatzistavrou (KTI) <vhatz at kinetix.gr>
>
>  Hello Ahmed,
>>
>> I don't think there is a reliable way to detect FAS on a per call basis.
>> Even audio detection software can be confused by strange ringtones,
>> answering machines, etc.
>>
>> The most reliable way we have found is to use statistics from CDRs and see
>> if the INVITE-to-200(OK) delay averaged over a number of calls appears to be
>> too small. If it is, then it is possible that you got FAS for those calls.
>>
>> But this can only tell you if you have been experiencing FAS in past
>> calls, ie you will not know you are getting FAS in real time.
>>
>> It is still useful however: after you detect a possible FAS case via
>> statistics you can place a few test calls yourself to verify that there
>> actually exists FAS (and this is the only information that a carrier will
>> accept in a trouble ticket to prove to them that they actually give you
>> FAS).
>>
>> I hope this helps.
>>
>> Best regards,
>> Vlasis Hatzistavrou.
>>
>>
>> On 8/1/10 3:05 ??, Ahmed Naji wrote:
>>
>> Hi everyone,
>>
>> Was just wondering what/if anyone is doing any work on FAS detection and
>> spoofed ring tones. Be great to discuss some ideas.
>>
>> Regards,
>>
>> Ahmed.
>>
>> --
>> Ahmed Naji
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From gmaruzz at celliax.org  Fri Jan  8 08:20:47 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 8 Jan 2010 17:20:47 +0100
Subject: [Freeswitch-users] Failed to connect to a SKYPE API
In-Reply-To: <27078464.post@talk.nabble.com>
References: <27062783.post@talk.nabble.com> <27078464.post@talk.nabble.com>
Message-ID: <7b197bef1001080820m371dd494v4c8fd55a07e0c6a0@mail.gmail.com>

you are probably running skype and FS as different windows users
(maybe skype as yourself and fs as administrator, or local_account, or
whatever).

both skype client AND freeSWITCH need to be running as the same
windows user, so FS can find the Skype client

-giovanni


On Fri, Jan 8, 2010 at 5:14 PM, Oscav <oscav at hotmail.fr> wrote:
>
> Im' running FS on windows server 2003 64bits
>
>
> Oscav wrote:
>>
>> Hi,
>>
>> I'm trying to use to SkypeIAX. When I load the mod_skypiax, I got the
>> following error :
>>
>> ?Failed to connect to a SKYPE API for interface_id=1, no SKYPE client
>> running, please (re)start Skype client. Skypiax exiting
>>
>> Skype is running with the correct account and skypiax.conf use the same
>> account. I was expecting a permission request from the Skype user but
>> nothing happens.
>>
>> Somebody knows how I can solve this ??
>>
>> Many thanks.
>>
>
> --
> View this message in context: http://old.nabble.com/Failed-to-connect-to-a-SKYPE-API-tp27062783p27078464.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From jonas.gauffin at gmail.com  Fri Jan  8 08:25:14 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Fri, 8 Jan 2010 17:25:14 +0100
Subject: [Freeswitch-users] hangup cause: NORMAL_UNSPECIFIED
In-Reply-To: <583FB3F4-9CFC-48A8-ADE0-30FD43F8F1F1@freeswitch.org>
References: <d021d3be1001080139m5b4cd7c6ge87b19e4ad6a62fe@mail.gmail.com>
	<583FB3F4-9CFC-48A8-ADE0-30FD43F8F1F1@freeswitch.org>
Message-ID: <d021d3be1001080825m1948cf9drbfb3a1f3ebd99b90@mail.gmail.com>

Consider it a feature request then :) A one liner would be enough, just a
bit more about why the call failed.

On Fri, Jan 8, 2010 at 3:43 PM, Brian West <brian at freeswitch.org> wrote:

> Not without the siptrace and sofia loglevel all 9
>
> /b
>
> On Jan 8, 2010, at 3:39 AM, Jonas Gauffin wrote:
>
> > Hello,
> >
> > Is it possible to get a more detailed reason (in the log) to why
> NORMAL_UNSPECIFIED was returned as hang up cause?
> >
> > 2010-01-08 09:30:50.987200 [DEBUG] sofia.c:3831 Channel
> sofia/external/070738xxxx entering state [terminated][904]
> > 2010-01-08 09:30:50.987200 [NOTICE] sofia.c:4461 Hangup
> sofia/external/070738xxxx [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED]
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Fri Jan  8 08:35:36 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 10:35:36 -0600
Subject: [Freeswitch-users] hangup cause: NORMAL_UNSPECIFIED
In-Reply-To: <d021d3be1001080825m1948cf9drbfb3a1f3ebd99b90@mail.gmail.com>
References: <d021d3be1001080139m5b4cd7c6ge87b19e4ad6a62fe@mail.gmail.com>
	<583FB3F4-9CFC-48A8-ADE0-30FD43F8F1F1@freeswitch.org>
	<d021d3be1001080825m1948cf9drbfb3a1f3ebd99b90@mail.gmail.com>
Message-ID: <3872821A-B842-40C8-ADD7-1079C35EBC6C@freeswitch.org>

A one liner can't solve this... their are so many things interacting... the 90X usually means you have given something to sofia it didn't like.

/b

On Jan 8, 2010, at 10:25 AM, Jonas Gauffin wrote:

> Consider it a feature request then :) A one liner would be enough, just a bit more about why the call failed.
> 
> On Fri, Jan 8, 2010 at 3:43 PM, Brian West <brian at freeswitch.org> wrote:
> Not without the siptrace and sofia loglevel all 9
> 
> /b

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From vhatz at kinetix.gr  Fri Jan  8 08:48:21 2010
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Fri, 08 Jan 2010 18:48:21 +0200
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>	<4B4728A5.4040805@kinetix.gr>
	<cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>
Message-ID: <4B4761D5.5010907@kinetix.gr>

Hello Ahmed,

On 8/1/10 6:07 ??, Ahmed Naji wrote:
> My line of thought is to tone-detect secondary ringing tones post 
> 200(OK) to detect FAS (False Answer Supervision). This should 
> eliminate at least a good proportion of calls, and it can be done real 
> time through a script/modules/...etc.
>

Tone detection can work only in cases where actual tones are sent over 
the audio. There are some routes (especially mobile routes) where 
different ringtones or music can be used instead of ringback tones, 
which render the whole tone detection effort difficult. There are also 
cases where no audio at all is sent until the called party answers the 
phone. Not to mention that the "FASed" route's behavior can change over 
time, rendering an effort for permanent FAS detection futile.

> Working along this thought, at least you are minimising the hit 
> cost-wise to a few seconds at most.

Well, even if you manage to detect FAS given to you by your termination 
provider, you cannot really minimize loss, as typically calls are 
charged by the time difference between 200(OK) and BYE. You can't really 
dispute anything even if you manage to detect FAS in real time, because 
you cannot prove this in a court or to anyone else by presenting CDRs 
only. You can't really pinpoint in a list of calls which ones had FAS. 
And of course, the terminating partner who gives FAS will deny it most 
of the times.

>
> As to answering machines and fake conferences, fake network messages 
> ...etc, one can possibly use voice detection, perhaps with heuristic 
> and statistical training.

You will need to combine tones detection, music detection, answering 
machine detection, which in the end makes the whole effort pointless. 
You will need to use DSP in software or in hardware to even try 
accomplish this (which means additional cost), and for what? For trying 
to detect FAS on a route which is not suitable for production use in the 
first place? And for today's wholesale margins?

IMHO the best solution is policy:

1) gather statistics from your CDRs (FreeSWITCH has lots of variables 
that you can use in your CDRs)
2) place test calls in cases where the stats show a possible problem
3) send a trouble ticket to the offending carrier in cases of FAS
and
4) stop using the route until the "problem" is fixed.

There are "carriers" who give FAS issues all the time and you can't 
really deal with them... It's best to just avoid them completely after 
repeated FAS infractions.

Best regards,
Vlasis Hatzistavrou.



From larclap at yahoo.com  Fri Jan  8 08:53:25 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 8 Jan 2010 08:53:25 -0800
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <COL117-W486989B811B372C35935AAB9700@phx.gbl>
References: <012901ca8fe6$e36b71c0$aa425540$@com>,
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>,
	<014801ca8fee$b75f8780$261e9680$@com>,
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>
Message-ID: <013a01ca9083$19a10920$4ce31b60$@com>

Thanks for the instructions Jeff.

 

After installing QT and setting the environmental variable, I get the
following error on build:

 

Error      1              error PRJ0019: A tool returned an error code from
"MOC prefportaudio.h"          FSComm              FSComm

 

Do I need to have checked-out the entire FreeSWITCH trunk in order to build
fscomm? I did not, just
http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm.

 

Lars

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Thursday, January 07, 2010 7:17 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

 

The windows support is very experimental at this time!
 
You must manually install
http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
 
Then set the environment variable QTDIR in the environment variables. This
can be set from the Computer/Properties/Advanced system settings/Environment
Variables/User Variables settings screen.
 
QTDIR=c:\qt\4.6.0 - or wherever you installed it
 
then restart VS
 

 

  _____  

Date: Thu, 7 Jan 2010 21:45:56 -0200
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

I don't have a Windows machine to test that. Maybe jlenk could give us a
hand since he is the one who has created the visual studio project? 

 

Regards,
Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927



On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:

No Qt installed. I just checked out from
http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into
VS2008. 

Do I need to get
http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

 

 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:

I just downloaded the fscomm project and loaded it into vs2008. I?ve never
programmed in C++ (or c), just C#, so I can?t make anything of the following
two messages:

 

Error      1              error PRJ0019: A tool returned an error code from
"RCC resources.qrc"  FSComm              FSComm

 

Warning               2              The following environment variables
were not found: $(QTDIR)                Project FSComm

 

Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.

 

The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)

 

Lars

Do you have Qt 4.6 installed?
-MC

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

 

  _____  

Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up
<http://clk.atdmt.com/GBL/go/196390709/direct/01/>  now.

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From msc at freeswitch.org  Fri Jan  8 08:57:19 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 8 Jan 2010 08:57:19 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly!
Message-ID: <87f2f3b91001080857l2f61903cw7f92a850f6c27718@mail.gmail.com>

Please call in! We'll mingle for a bit and then get started. Agenda is here:
http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_08

-Michael
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From lists at redbonez.net  Fri Jan  8 08:59:09 2010
From: lists at redbonez.net (Adam Ford)
Date: Fri, 8 Jan 2010 09:59:09 -0700
Subject: [Freeswitch-users] Delay in connecting inbound calls
Message-ID: <016e01ca9083$e6de14f0$b49a3ed0$@net>

I have setup a FreeSWITCH system using OpenZAP and Redfone>foneBridge2 for
my connection to the PSTN and there seems to be a 3-5 second delay between
when the incoming call is answered and the parties are able to hear each
other.  This is only with inbound calls, outbound audio connection is
instant.  

 

Any suggestions on where to start troubleshooting this issue?

 

-AF

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From brian at freeswitch.org  Fri Jan  8 09:05:19 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 11:05:19 -0600
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <013a01ca9083$19a10920$4ce31b60$@com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>,
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>,
	<014801ca8fee$b75f8780$261e9680$@com>,
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>
	<013a01ca9083$19a10920$4ce31b60$@com>
Message-ID: <68D59A25-D24F-4D1D-A548-67CE33D1EEEB@freeswitch.org>

You have to have freeswitch built and installed... and you need to do it all right now.

/b

On Jan 8, 2010, at 10:53 AM, Lars Zeb wrote:

> Thanks for the instructions Jeff.
>  
> After installing QT and setting the environmental variable, I get the following error on build:
>  
> Error      1              error PRJ0019: A tool returned an error code from "MOC prefportaudio.h"          FSComm              FSComm
>  
> Do I need to have checked-out the entire FreeSWITCH trunk in order to build fscomm? I did not, just http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm.
>  
> Lars

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From dujinfang at gmail.com  Fri Jan  8 10:00:58 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 9 Jan 2010 02:00:58 +0800
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<B732197E-1123-4738-A69F-F84206FCE38A@freeswitch.org>
	<0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org>
	<0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org>
	<23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com>
	<60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org>
	<23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com>
	<ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com>
	<23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com>
	<B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>
Message-ID: <23f91031001081000t26bfebfcy222a1ffc13b95a1e@mail.gmail.com>

Bad luck, I had trouble to join the conference due to my bad internet.
Good thing is that I compiled and run on Mac and dialed into the
conference successfully.

Definitely I can help test and reporting bugs, and might be more
helpful on small bug fixes and implement new features though I need to
learn QT first. But first of all I think it would be better if we have
a feature list for the new phone.

2010/1/8, Brian West <brian at freeswitch.org>:
> Lets schedule FSComm on the weekly conference call... We need people to step
> up and take some roles in both FreeSWITCH and FSComm projects... Even if its
> just testing bugs and collecting info.
>
> Thanks,
> Brian
>
> On Jan 7, 2010, at 10:11 AM, Seven Du wrote:
>
>> :lol. I do like to involve into this. I saw you have done a lot of
>> works. I read some code and here are some questions:
>>
>> 1) what's your nick on IRC? I'm seven(or seven_ ?)
>> 2) Are you developing on Windows? How can I compile on Mac(I have no
>> experience on QT)?
>> 3) Does is needs fsGUI? I downloaded fsGUI dmg file and it complains
>> no mods on start.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From jcasale at activenetwerx.com  Fri Jan  8 10:29:48 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 8 Jan 2010 18:29:48 +0000
Subject: [Freeswitch-users] Purge Stale Registrations
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90734F9@prato.activenetwerx.local>

Looking through the cli, I thought this might be attainable with
sofia status profile internal flush_inbound_reg|rescan for example
but it's not removing stake registrations from UA's I am not expecting
to see. Anyway to do this?

Thanks!
jlc



From brian at freeswitch.org  Fri Jan  8 10:40:32 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 12:40:32 -0600
Subject: [Freeswitch-users] Purge Stale Registrations
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E90734F9@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E90734F9@prato.activenetwerx.local>
Message-ID: <72E53884-A014-43AF-93C7-A33EE240A221@freeswitch.org>

how do we know they are stale?  They remove when they expire or your endpoint unregisters on shutdown.

/b

On Jan 8, 2010, at 12:29 PM, Joseph L. Casale wrote:

> Looking through the cli, I thought this might be attainable with
> sofia status profile internal flush_inbound_reg|rescan for example
> but it's not removing stake registrations from UA's I am not expecting
> to see. Anyway to do this?
> 
> Thanks!
> jlc




From jcasale at activenetwerx.com  Fri Jan  8 10:53:52 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 8 Jan 2010 18:53:52 +0000
Subject: [Freeswitch-users] Purge Stale Registrations
In-Reply-To: <72E53884-A014-43AF-93C7-A33EE240A221@freeswitch.org>
References: <CA5A491E9DEFBE4CB777DE97E21575E90734F9@prato.activenetwerx.local>
	<72E53884-A014-43AF-93C7-A33EE240A221@freeswitch.org>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90735AE@prato.activenetwerx.local>

>how do we know they are stale?  They remove when they expire or your endpoint unregisters on shutdown.

Right, should have been more specific: I made changes to a few UA's and
I see the new UA's but wanted to remove the old reg's forcibly to clean
up the status so it's more readable. Only wanted to use this during testing
of course...



From nicolas at medularis.com  Fri Jan  8 11:02:08 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 8 Jan 2010 16:02:08 -0300
Subject: [Freeswitch-users] Need to fake ringback
In-Reply-To: <91E856D5-B499-4C5F-A6A3-E2BB36C926C6@freeswitch.org>
References: <1b46b4e81001072133h3f4c9bf1g68e25d27fd71edc1@mail.gmail.com>
	<91E856D5-B499-4C5F-A6A3-E2BB36C926C6@freeswitch.org>
Message-ID: <1b46b4e81001081102k36c76792n5619426d05cca8cd@mail.gmail.com>

Not that I'm aware.


On Fri, Jan 8, 2010 at 11:41 AM, Brian West <brian at freeswitch.org> wrote:
> Are you using proxy media?
>
> /b
>
> On Jan 7, 2010, at 11:33 PM, Nicolas Brenner wrote:
>
>> I'm trying to fake a ringback for leg1 of a two-legged call without success.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From nicolas at medularis.com  Fri Jan  8 11:02:46 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 8 Jan 2010 16:02:46 -0300
Subject: [Freeswitch-users] Need to fake ringback
In-Reply-To: <1b46b4e81001081102k36c76792n5619426d05cca8cd@mail.gmail.com>
References: <1b46b4e81001072133h3f4c9bf1g68e25d27fd71edc1@mail.gmail.com>
	<91E856D5-B499-4C5F-A6A3-E2BB36C926C6@freeswitch.org>
	<1b46b4e81001081102k36c76792n5619426d05cca8cd@mail.gmail.com>
Message-ID: <1b46b4e81001081102h2f644d28r86e14d7ba151e5c4@mail.gmail.com>

I'm using the default dialplan.


On Fri, Jan 8, 2010 at 4:02 PM, Nicolas Brenner <nicolas at medularis.com> wrote:
> Not that I'm aware.
>
>
> On Fri, Jan 8, 2010 at 11:41 AM, Brian West <brian at freeswitch.org> wrote:
>> Are you using proxy media?
>>
>> /b
>>
>> On Jan 7, 2010, at 11:33 PM, Nicolas Brenner wrote:
>>
>>> I'm trying to fake a ringback for leg1 of a two-legged call without success.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>



From kristian.kielhofner at gmail.com  Fri Jan  8 11:03:35 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 8 Jan 2010 14:03:35 -0500
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <D03B492C-74B9-4B38-B67B-017690FF372C@freeswitch.org>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org>
	<23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com>
	<60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org>
	<23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com>
	<ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com>
	<23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com>
	<B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>
	<ebbc2d551001071227y22d6eaa8y43742fb4328a67b2@mail.gmail.com>
	<D03B492C-74B9-4B38-B67B-017690FF372C@freeswitch.org>
Message-ID: <2d9149cd1001081103m4d2b6852l9848ce4c82005fd3@mail.gmail.com>

What is his PayPal?  I've been looking through FSComm and it looks
sick...  I love the idea of being able to tweak my SOFTPHONE params
just like I would tweak FS.  Sick, just sick.

Speaking of sick, I hope JM gets better soon :).

On Thu, Jan 7, 2010 at 3:36 PM, Brian West <brian at freeswitch.org> wrote:
> Everyone should get JM's paypal and toss him some cash for all the good work
> he's doing... Without him this project wouldn't have become a reality.
> /b


-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From nicolas at medularis.com  Fri Jan  8 11:13:52 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 8 Jan 2010 16:13:52 -0300
Subject: [Freeswitch-users] Ruby ESL missing pthread
In-Reply-To: <8A3B5711-727D-4BDB-97AE-7AD20C4528F7@jerris.com>
References: <1b46b4e81001071958i17ccb6bey6eeecd9d94eb8438@mail.gmail.com>
	<8A3B5711-727D-4BDB-97AE-7AD20C4528F7@jerris.com>
Message-ID: <1b46b4e81001081113j40ed02fehd34eb2b99d1a5ef6@mail.gmail.com>

Thanks!

http://jira.freeswitch.org/browse/ESL-29

On Fri, Jan 8, 2010 at 12:33 PM, Michael Jerris <mike at jerris.com> wrote:
> Please open up a bug on jira for me for this issue.
>
> Mike



From linux4michelle at tamay-dogan.net  Fri Jan  8 11:32:45 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Fri, 8 Jan 2010 20:32:45 +0100
Subject: [Freeswitch-users] FreeSwitch and GSM/HSPA Modems
In-Reply-To: <7b197bef1001080538u5ed67082r6df84d287c0c06c1@mail.gmail.com>
References: <20100103184636.GW5547@tamay-dogan.net>
	<20100106135210.GG5547@tamay-dogan.net>
	<CC40A355-43DA-4EFE-A77B-D607FE650E96@jerris.com>
	<7b197bef1001060759w1b179fapf42aa47e3d7574b6@mail.gmail.com>
	<20100106171413.GI5547@tamay-dogan.net>
	<7b197bef1001060939rd1e08f3h2cb82cd89e50225e@mail.gmail.com>
	<20100107182731.GL5547@tamay-dogan.net>
	<7b197bef1001080538u5ed67082r6df84d287c0c06c1@mail.gmail.com>
Message-ID: <20100108193244.GR5547@tamay-dogan.net>

Hello Giovanni,

thankyou forthe answer...

> you don't need them all (at least if you don't want to make dirty pics
> with a kludge full of wires... hey, that can be arousing! :))

My Longsun L580 can be put into Web-Cam mode...

But it is cheal chinese Dual-SIM/Standby Cell-Phone which  can  not  use
anything in parallel.  :-/

> I suggest you use a ready-made GSM/GPRS/HDSPA/whatever module, that
> contains all that you need and is available from Chinese suppliers for
> very low prices. You can find some module that allows you to directly
> tap in the GSM pcm audio stream, that would means you will not need to
> sample and convert from analog to digital (so, no cpu power at all, no
> dsp, no nothing).

I will see, they cost arround 110 to 230 US$...

> If you would like to keep me in the loop, I would like to know how you
> progress.

If my ISP would repair my ADSL I had my website back online...

    <http://www.electronica.tamay-dogan.net/>

Fully "Open Hardware Development" in the sense of "Open Source" with some
small limitations...  or advantages.

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From jerry.richards at teotech.com  Fri Jan  8 11:38:06 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 8 Jan 2010 11:38:06 -0800
Subject: [Freeswitch-users] http://jira.freeswitch.org/browse/FSCORE-262
Message-ID: <39ED1D5111554D33AF502370BC34D288@greyhawk.tonecommander.com>

Is there a plan to fix this JIRA issue:
http://jira.freeswitch.org/browse/FSCORE-262

This is causing a problem in sharing presence data between FS and another
gateway.

Thanks,
Jerry




From brian at freeswitch.org  Fri Jan  8 11:43:40 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 13:43:40 -0600
Subject: [Freeswitch-users] http://jira.freeswitch.org/browse/FSCORE-262
In-Reply-To: <39ED1D5111554D33AF502370BC34D288@greyhawk.tonecommander.com>
References: <39ED1D5111554D33AF502370BC34D288@greyhawk.tonecommander.com>
Message-ID: <5817BC3F-C091-42E3-AAAE-2CF4A21A9531@freeswitch.org>

Do you happen to have a patch for that?

/b

On Jan 8, 2010, at 1:38 PM, Jerry Richards wrote:

> Is there a plan to fix this JIRA issue:
> http://jira.freeswitch.org/browse/FSCORE-262
> 
> This is causing a problem in sharing presence data between FS and another
> gateway.
> 
> Thanks,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From william.suffill at gmail.com  Fri Jan  8 12:09:04 2010
From: william.suffill at gmail.com (William Suffill)
Date: Fri, 8 Jan 2010 15:09:04 -0500
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <2d9149cd1001081103m4d2b6852l9848ce4c82005fd3@mail.gmail.com>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> 
	<23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com> 
	<60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org>
	<23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com> 
	<ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com> 
	<23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com> 
	<B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>
	<ebbc2d551001071227y22d6eaa8y43742fb4328a67b2@mail.gmail.com> 
	<D03B492C-74B9-4B38-B67B-017690FF372C@freeswitch.org>
	<2d9149cd1001081103m4d2b6852l9848ce4c82005fd3@mail.gmail.com>
Message-ID: <6b65470d1001081209v64d20f9ehdd7a7b4c6516c540@mail.gmail.com>

jmesquita at gmail.com is his Paypal account. Ya hope he gets better as well.

-- W
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From a.alalousi at gmail.com  Fri Jan  8 13:43:59 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Fri, 8 Jan 2010 21:43:59 +0000
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <191c3a031001080820i426194cw58b8a2cf8b123493@mail.gmail.com>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
	<4B4728A5.4040805@kinetix.gr>
	<cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>
	<191c3a031001080820i426194cw58b8a2cf8b123493@mail.gmail.com>
Message-ID: <cd93b3701001081343o35b6752wdd44f4f97155bae6@mail.gmail.com>

That's what I had in mind + some other techniques to address things to help
with answering machines and recorded messages.

Vasilis rightly states some of the challenges in doing this, and he is also
right on the legal and effort/benefit ratio.

My aim behind all of this is not to really prosecute or dispute issues with
carriers - we have an active policy of barring FAS routes after the second
offence, so that's covered. The aim is to be able to detect such routes as
an when they arise, and be proactive rather than reactive in dealing with
this horrendous and disgusting practice that plagues the telecom industry.

Best regards,

Ahmed.


2010/1/8 Anthony Minessale <anthony.minessale at gmail.com>

> you could use the existing tone_detect app after the false answer to xfer
> to an extension of your choice if ring tone was detected.
>
>
> 2010/1/8 Ahmed Naji <a.alalousi at gmail.com>
>
> Hi Valsis,
>>
>> Thanks for this.
>>
>> My line of thought is to tone-detect secondary ringing tones post 200(OK)
>> to detect FAS (False Answer Supervision). This should eliminate at least a
>> good proportion of calls, and it can be done real time through a
>> script/modules/...etc.
>>
>> Working along this thought, at least you are minimising the hit cost-wise
>> to a few seconds at most.
>>
>> As to answering machines and fake conferences, fake network messages
>> ...etc, one can possibly use voice detection, perhaps with heuristic and
>> statistical training.
>>
>> Just a thought ..
>>
>> Regards,
>>
>> Ahmed.
>>
>>
>> 2010/1/8 Vlasis Hatzistavrou (KTI) <vhatz at kinetix.gr>
>>
>>  Hello Ahmed,
>>>
>>> I don't think there is a reliable way to detect FAS on a per call basis.
>>> Even audio detection software can be confused by strange ringtones,
>>> answering machines, etc.
>>>
>>> The most reliable way we have found is to use statistics from CDRs and
>>> see if the INVITE-to-200(OK) delay averaged over a number of calls appears
>>> to be too small. If it is, then it is possible that you got FAS for those
>>> calls.
>>>
>>> But this can only tell you if you have been experiencing FAS in past
>>> calls, ie you will not know you are getting FAS in real time.
>>>
>>> It is still useful however: after you detect a possible FAS case via
>>> statistics you can place a few test calls yourself to verify that there
>>> actually exists FAS (and this is the only information that a carrier will
>>> accept in a trouble ticket to prove to them that they actually give you
>>> FAS).
>>>
>>> I hope this helps.
>>>
>>> Best regards,
>>> Vlasis Hatzistavrou.
>>>
>>>
>>> On 8/1/10 3:05 ??, Ahmed Naji wrote:
>>>
>>> Hi everyone,
>>>
>>> Was just wondering what/if anyone is doing any work on FAS detection and
>>> spoofed ring tones. Be great to discuss some ideas.
>>>
>>> Regards,
>>>
>>> Ahmed.
>>>
>>> --
>>> Ahmed Naji
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Ahmed Naji
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From a.alalousi at gmail.com  Fri Jan  8 13:57:16 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Fri, 8 Jan 2010 21:57:16 +0000
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <4B4761D5.5010907@kinetix.gr>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
	<4B4728A5.4040805@kinetix.gr>
	<cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>
	<4B4761D5.5010907@kinetix.gr>
Message-ID: <cd93b3701001081357i56c39e15j262ed05e5351a31a@mail.gmail.com>

Hi Vasilis,

Good points .. my comments inline.

Regards,

Ahmed.

2010/1/8 Vlasis Hatzistavrou (KTI) <vhatz at kinetix.gr>

> Hello Ahmed,
>
> On 8/1/10 6:07 ??, Ahmed Naji wrote:
> > My line of thought is to tone-detect secondary ringing tones post
> > 200(OK) to detect FAS (False Answer Supervision). This should
> > eliminate at least a good proportion of calls, and it can be done real
> > time through a script/modules/...etc.
> >
>
> Tone detection can work only in cases where actual tones are sent over
> the audio. There are some routes (especially mobile routes) where
> different ringtones or music can be used instead of ringback tones,
> which render the whole tone detection effort difficult. There are also
> cases where no audio at all is sent until the called party answers the
> phone. Not to mention that the "FASed" route's behavior can change over
> time, rendering an effort for permanent FAS detection futile.
>
> True. One way of dealing with coloured ringing tones is quiet straight
forward with acoustics libraries for C++ and such like. No audio is also
possible to deal with in this fashion.

> Working along this thought, at least you are minimising the hit
> > cost-wise to a few seconds at most.
>
> Well, even if you manage to detect FAS given to you by your termination
> provider, you cannot really minimize loss, as typically calls are
> charged by the time difference between 200(OK) and BYE.

True, but the idea is to minimise this duration, and hangup the call ASAP.

You can't really
> dispute anything even if you manage to detect FAS in real time, because
> you cannot prove this in a court or to anyone else by presenting CDRs
> only. You can't really pinpoint in a list of calls which ones had FAS.
> And of course, the terminating partner who gives FAS will deny it most
> of the times.
>
Absolutely right on all accounts. At least in doing so, you are minimising
the financial hit, particularly on busy routes that carry a high premium.


>
> >
> > As to answering machines and fake conferences, fake network messages
> > ...etc, one can possibly use voice detection, perhaps with heuristic
> > and statistical training.
>
> You will need to combine tones detection, music detection, answering
> machine detection, which in the end makes the whole effort pointless.
>
Debatable. Hard, yes, but pointless ? I don't really agree.


> You will need to use DSP in software or in hardware to even try
> accomplish this (which means additional cost),

DSP techniques - to some extent, if you really want to go that far. The idea
is to do it all in FS, and not really employ expensive software/hardware
techniques as you so rightly state.


> and for what? For trying
> to detect FAS on a route which is not suitable for production use in the
> first place? And for today's wholesale margins?
>
Here's the thing: even on those terms, it is worth it. For so many
destinations (e.g. Central/Sub-Saharan Africa and such like), there really
isn't much of a distinction between "reliable" wholesale routes and retail
routes. You use the best you could to carry your traffic and my concern is
for my retail customers. People like Vodafone, Orange and BT don't take too
kindly to being FASed, and if you carry their traffic, then you by default
are under contractual obligation to cushion the hit where they can prove a
case. One case, for one of those routes that carry 250K minutes/day will
bankrupt a small/medium carrier, even if you end up winning the legal
argument.

What I'm trying to do is not make an unusable route usable. Rather, to
detect an unusable route ASAP, place the terminating carrier on a very short
leash, and disconnect them as early as possible should they re-offend.


> IMHO the best solution is policy:
>
> 1) gather statistics from your CDRs (FreeSWITCH has lots of variables
> that you can use in your CDRs)
> 2) place test calls in cases where the stats show a possible problem
> 3) send a trouble ticket to the offending carrier in cases of FAS
> and
> 4) stop using the route until the "problem" is fixed.
>
> I hear what you're saying. And how much is the time and man effort involved
in manually dealing with this ?


> There are "carriers" who give FAS issues all the time and you can't
> really deal with them... It's best to just avoid them completely after
> repeated FAS infractions.
>
> Best regards,
> Vlasis Hatzistavrou.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From freeswitch at aastral.net  Fri Jan  8 15:39:54 2010
From: freeswitch at aastral.net (Bill W)
Date: Fri, 08 Jan 2010 18:39:54 -0500
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <74C18F18-89EB-49DF-B5CE-C8D56856ABCD@freeswitch.org>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>	<4B46792E.6090805@aastral.net>	<E5EFC626-353A-4B54-9F22-59C0C48E3B95@freeswitch.org>	<4B4684F3.8030504@aastral.net>
	<74C18F18-89EB-49DF-B5CE-C8D56856ABCD@freeswitch.org>
Message-ID: <4B47C24A.3080305@aastral.net>

Hey Everyone,

I spoke with Karl, and for the sake of completing this thread I'm 
posting the results.

Ultimately the problem Karl found with OpenSUSE was the SQLite libraries 
leading to database corruption, and freeswitch misbehaving because of 
that.  He also was trying to do traffic shaping for his application and 
ran into problems with shaping not working right on multi-core x86_64 
kernels in SuSE.

I asked him about using odbc in the core to get around the sqlite bug 
but he didn't bother because he needed traffic shaping that worked.

So it's not so much that SuSE was a bad distro, but rather that it 
didn't work well with freeswitch and traffic shaping on multi-core 
x86_64.  He did mention that SuSE 11.1 is nice in general but to stay 
away from 11.2 because gcc segfaults on any significant build.

Hope this helps the community!
Bill



His response:
====================
The big thing you should learn from the investment of my time in the lab 
here is simply this, "listen to Brian".  From now on, I'm considering 
him the EF Hutton on #freeswitch.

(Editors Note: For you youngsters out there, EF Hutton's tagline was 
"When EF Hutton talks, people listen.")

When I added about 60 phones to the system, it essentially "blew up" 
whereby phones wouldn't register, the database would get corrupted, etc...

We worked around the problem by making the freeswitch "db" directory a 
ramdisk, but that only mitigated the problem, and didn't entirely fix it.

Oh yeah, and although mod_perl will compile in freeswitch, it will bomb 
out and segfault when attempting to run any mod_perl scripts in 
freeswitch.  The fix is to recompile perl from source.  Even then, I 
still had problems.

Other problems with the remaining Suse installations still are:

1.) When you answer, it will hang up.  No rhyme, reason, or otherwise.
2.) Occasionally (rarely) we have issues with audio not passing through 
correctly.  One way audio, or none at all.
3.) When you dial, it will take 10 seconds to go through enum lookups 
and the like before finally hitting the PSTN gateway.

Oh yeah, about RedHat...  Don't bother with it's sister CentOS either. 
I tested it here in the lab, and it was like getting in a time machine 
and going back to 2007.   Also, Centos is busted.  You'll find that 
Linux kernel will re-transmit IP packets from processes long since dead. 
  Suffice it to say, when it does this to RTP traffic it drives things 
bugnutz. (Ed. His test was to run 20mbit/sec through centos with a 
gigabit card overnight and it dropped 150 packets.  He did try different 
cards.)

Mandrake - recommend this to people you dislike.  If they're ignorant 
enough you'll find them thanking you for it.

Debian - almost awesome, but if failed miserably in the lab with packet 
shaping.  I mean, it thought it was working, but the overall quality was 
hit & miss.  Other than that, no complaints.  What was really attractive 
was it's got Cyrus 2.3 out of the box, so if you're using Cyrus and not 
using it for packet shaping you might consider Debian an option.  Be 
advised, since it flunked packet shaping we never bothered to 
compile/test freeswitch on it, so do your own research.

Ubuntu, make no mistake...
We tested Ubuntu pretty heavily here in the lab.
Even the packet shaping works (HTB & SFQ).
With Suse I has to custom-compile the kernel, and packet shaping ONLY 
worked with 2.6.28.8 - 2.6.28.10, the rest were buggy and the problems 
manifested themselves in ways that you'd think would be totally 
unrelated.  I should be more specific and state x86_64 multi-core. 
x86_64 single core seemed to do packet shaping (somewhat nicely) in 
2.6.18 on up.  Previous to that it was rather "interesting"...

Save yourself the headache and go with Ubuntu.

==================================




Brian West wrote:
> He's on the list Karl J. Vesterling
> 
> /b
> 
> On Jan 7, 2010, at 7:05 PM, Bill W. wrote:
> 
>> Wow, I haven't heard of these issues.  Obviously this concerns me.  Are
>> these documented anywhere so I can research this?  How do I get in touch
>> with KJV?
>>
>> Thanks!
>> Bill
>>
>>
>>
>> Brian West wrote:
>>> Good luck with that you'll have an ass load of problems.  The reason its stable is the backports and outdated packages.  Bleeding edge will only screw you over... just ask KJV... He was on OpenSuSE and had nothing but weird problems.
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From vhatz at kinetix.gr  Fri Jan  8 16:02:05 2010
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Sat, 09 Jan 2010 02:02:05 +0200
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <cd93b3701001081357i56c39e15j262ed05e5351a31a@mail.gmail.com>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>	<4B4728A5.4040805@kinetix.gr>	<cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>	<4B4761D5.5010907@kinetix.gr>
	<cd93b3701001081357i56c39e15j262ed05e5351a31a@mail.gmail.com>
Message-ID: <4B47C77D.4080502@kinetix.gr>

Hello Ahmed,

>
>     IMHO the best solution is policy:
>
>     1) gather statistics from your CDRs (FreeSWITCH has lots of variables
>     that you can use in your CDRs)
>     2) place test calls in cases where the stats show a possible problem
>     3) send a trouble ticket to the offending carrier in cases of FAS
>     and
>     4) stop using the route until the "problem" is fixed.
>
> I hear what you're saying. And how much is the time and man effort 
> involved in manually dealing with this ?
>  
>
Unfortunately, it is a lot of work. In fact it is a 24x7 manual work... 
What makes things a bit easier is that a FAS route will have a short 
setup-connect delay, and statistics will definitely show a heavily FASed 
route, although a route with a small percentage of FAS will still go 
under the radar. And of course, using statistical methods doesn't give 
you results in real time...

We have experimented with tone_detect app as Anthony Minesalle suggested 
for detecting tones to apply to FAS problems and it works very well. But 
after a while we dropped the whole FAS detection idea because of the so 
many different FAS scenarios out there... Plus, quite a few methods that 
monitor audio involve decompressing the incoming audio before processing 
it. This means that you need to have audio transcoding on FS, which a 
bad things in terms of perceived audio quality and CPU power/scaling. In 
that case you would almost certainly need quite a few G729 codec 
licenses, as most carriers use this codec to send you their traffic.

Nonetheless if you want to pursue this and you need help I'd be happy to 
assist, as FAS is a huge problem for carriers.

Best regards,
Vlasis.
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From msc at freeswitch.org  Fri Jan  8 16:17:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 8 Jan 2010 16:17:26 -0800
Subject: [Freeswitch-users] Delay in connecting inbound calls
In-Reply-To: <016e01ca9083$e6de14f0$b49a3ed0$@net>
References: <016e01ca9083$e6de14f0$b49a3ed0$@net>
Message-ID: <87f2f3b91001081617t1844119bp1ece7ae951c68836@mail.gmail.com>

On Fri, Jan 8, 2010 at 8:59 AM, Adam Ford <lists at redbonez.net> wrote:

>  I have setup a FreeSWITCH system using OpenZAP and Redfone>foneBridge2
> for my connection to the PSTN and there seems to be a 3-5 second delay
> between when the incoming call is answered and the parties are able to hear
> each other.  This is only with inbound calls, outbound audio connection is
> instant.
>
>
>
> Any suggestions on where to start troubleshooting this issue?
>
Start by getting a debug log of an incoming call and putting it on pastebin
so others can take a look.
-MC
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From jmesquita at freeswitch.org  Fri Jan  8 17:13:32 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 8 Jan 2010 22:13:32 -0300
Subject: [Freeswitch-users] FSComm builds OK on Fedora F-12 with QT4
	4.5.3 but doesn't save SIP account
In-Reply-To: <13ca621c1001080331l363122a5l3b093adb5782538e@mail.gmail.com>
References: <13ca621c1001080331l363122a5l3b093adb5782538e@mail.gmail.com>
Message-ID: <ebbc2d551001081713r70da95e3i115769116a17390a@mail.gmail.com>

Gavin,

http://wiki.freeswitch.org/wiki/FSComm#Configuration

Sofia configuration is not yet persisted. I have it kinda working on my
local copy and I hope to have it commited by the end of tomorrow. Till then,
you will have to do the configuration manually like described on the wiki.

Regards,

Jo?o Mesquita


On Fri, Jan 8, 2010 at 8:31 AM, Gavin Henry <gavin.henry at gmail.com> wrote:

> Hi,
>
> Just a quick one to say this builds ok with:
>
> Compiled FSComm version: 1.0.trunk (16209M)
> FreeSWITCH Version 1.0.trunk (16209M)
>
> But it doesn't want to save my SIP account details. How to debug?
>
> Thanks.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dave at 3c.co.uk  Fri Jan  8 17:17:24 2010
From: dave at 3c.co.uk (David Knell)
Date: Sat, 09 Jan 2010 01:17:24 +0000
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <4B47C77D.4080502@kinetix.gr>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
	<4B4728A5.4040805@kinetix.gr>
	<cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>
	<4B4761D5.5010907@kinetix.gr>
	<cd93b3701001081357i56c39e15j262ed05e5351a31a@mail.gmail.com>
	<4B47C77D.4080502@kinetix.gr>
Message-ID: <1262999844.21753.52.camel@local.freepabx.com>

> Nonetheless if you want to pursue this and you need help I'd be happy
> to assist, as FAS is a huge problem for carriers.

FYI - here's some guys advertising "multilingual FAS" - presumably for
those crooked enough to want to inflate their margins in this manner and
too thick to do it for themselves:
http://www.calltermination.com/forums/thread270514.html

--Dave







From jcasale at activenetwerx.com  Fri Jan  8 17:39:56 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 9 Jan 2010 01:39:56 +0000
Subject: [Freeswitch-users] rxfax ending dialplan
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9074638@prato.activenetwerx.local>

Hey Guys,
In my initial testing with an old version of fs, my dialplan with a system
call after the rxfax app would execute, but now the dialplan ends at rxfax
after it writes the file out.

Anyone know what var is different by default now so I can allow this to
continue on to hit the script and then end after?

Thanks,
jlc



From brian at freeswitch.org  Fri Jan  8 17:46:51 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 19:46:51 -0600
Subject: [Freeswitch-users] rxfax ending dialplan
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9074638@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E9074638@prato.activenetwerx.local>
Message-ID: <CCB7B570-7845-4BA2-985B-DC40922E5E6E@freeswitch.org>

use api_hangup_hook variable.

/b

On Jan 8, 2010, at 7:39 PM, Joseph L. Casale wrote:

> Anyone know what var is different by default now so I can allow this to
> continue on to hit the script and then end after?




From jingwei.yang at gmail.com  Fri Jan  8 17:58:07 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Sat, 9 Jan 2010 09:58:07 +0800
Subject: [Freeswitch-users] IVR and TTS
In-Reply-To: <910309031001080416r6064670bic4e6285494f8d25b@mail.gmail.com>
References: <13529f9d1001080042w420c9f0vdbc5b7a57edd1e6c@mail.gmail.com>
	<910309031001080416r6064670bic4e6285494f8d25b@mail.gmail.com>
Message-ID: <13529f9d1001081758h4a32fae7q83401def09f2f34e@mail.gmail.com>

Hi Vinuth,

Thanks a lot for your great response!

Regards,
-Jingwei

On Fri, Jan 8, 2010 at 8:16 PM, Vinuth Madinur <vinuth.madinur at gmail.com>wrote:

> Hi,
>
> "DTMF" event will be raised in ESL, when "A" presses a key. It'll be raised
> for each key pressed.
>
> Alternatively you can use play_and_get_digits.
>
> To continue FS execution after you fetch the address, you just need to
> invoke the "speak" command on that socket.
>
> Since FS is handling inbound calls, you can use the outbound event socket,
> where a new connection will be opened per call from FS to your java program.
>
> One way to know what events are raised in ESL, you can telnet to 8021 port,
> authenticate and send "events plain all" command. Configure FS dialplan for
> an extension which will just answer a call when it comes in. Then call this
> extension from a softphone, press a key and you'll see the corresponding
> event in the telnet console.
>
>
> Thanks,
> Vinuth.
>
>
>
> On Fri, Jan 8, 2010 at 2:12 PM, Jingwei Yang <jingwei.yang at gmail.com>
>  wrote:
>
>> Hi Guys,
>>
>> I need to implement a function using IVR and TTS. Here's the scenario.
>>
>> 1. User A calls in
>> 2. FS plays a welcome message and directs A to press '1' to continue
>> 3. FS detects A's number and finds A's address from the database and plays
>> another piece of voice message including the address info just found
>>
>> I understand this logic can be implemented using javascript. However, in
>> this scenario, the database is a remote one and the local js has no access
>> to it. What I'm planning to do is write a Java program, talking to FS via
>> ESL. Could someone please tell me what event FS will trigger after user A
>> selects a certain option and how to inform the FS to continue the rest of
>> IVR menu after finding the address?
>>
>> Thanks,
>> -Jingwei
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jcasale at activenetwerx.com  Fri Jan  8 18:23:47 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 9 Jan 2010 02:23:47 +0000
Subject: [Freeswitch-users] rxfax ending dialplan
In-Reply-To: <CCB7B570-7845-4BA2-985B-DC40922E5E6E@freeswitch.org>
References: <CA5A491E9DEFBE4CB777DE97E21575E9074638@prato.activenetwerx.local>
	<CCB7B570-7845-4BA2-985B-DC40922E5E6E@freeswitch.org>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90746C8@prato.activenetwerx.local>

>use api_hangup_hook variable.

Thanks Brian!
I'll update the wiki for the fax related stuff once I am positive
I have it nailed...
jlc



From anthony.minessale at gmail.com  Fri Jan  8 18:46:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Jan 2010 20:46:48 -0600
Subject: [Freeswitch-users] Need to fake ringback
In-Reply-To: <1b46b4e81001081102h2f644d28r86e14d7ba151e5c4@mail.gmail.com>
References: <1b46b4e81001072133h3f4c9bf1g68e25d27fd71edc1@mail.gmail.com>
	<91E856D5-B499-4C5F-A6A3-E2BB36C926C6@freeswitch.org>
	<1b46b4e81001081102k36c76792n5619426d05cca8cd@mail.gmail.com>
	<1b46b4e81001081102h2f644d28r86e14d7ba151e5c4@mail.gmail.com>
Message-ID: <191c3a031001081846i2c31b73bp714813d71a21b611@mail.gmail.com>

ringback variable must be set on the originating leg (the A leg aka inbound
leg of the call)
then that channel must be provided as the 2nd arg to the session constructor
after the dial string or used with bridge.

session.setVariable("ringback", "%(2000,4000,440,480)");
session.execute("bridge", "sofia/internal/foo at bar.com");

or
// passing session as the 2nd arg allows the ringback to play on session
while session2 is established.
session2 = new Session("sofia/internal/foo at bar.com", session);
bridge(session, session2); //or whatever


On Fri, Jan 8, 2010 at 1:02 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> I'm using the default dialplan.
>
>
> On Fri, Jan 8, 2010 at 4:02 PM, Nicolas Brenner <nicolas at medularis.com>
> wrote:
> > Not that I'm aware.
> >
> >
> > On Fri, Jan 8, 2010 at 11:41 AM, Brian West <brian at freeswitch.org>
> wrote:
> >> Are you using proxy media?
> >>
> >> /b
> >>
> >> On Jan 7, 2010, at 11:33 PM, Nicolas Brenner wrote:
> >>
> >>> I'm trying to fake a ringback for leg1 of a two-legged call without
> success.
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Jan  8 18:48:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Jan 2010 20:48:30 -0600
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<B98B6921358D0C49BD21CD023F1F52C302623BE0@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
Message-ID: <191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>

look at mod_tone_stream its fairly basic
but really look at any mod in formats dir


On Fri, Jan 8, 2010 at 9:25 AM, MohammedShehzad <pmhshz at gmail.com> wrote:

> Thanks Brian,
>
> Great... Let me start the developing it.
> Although I am analyzing the mod_local_stream, to understand exact working
> of format module, Will anybody please let me know any similar thing, I can
> take reference?
>
>
>
> On Fri, Jan 8, 2010 at 8:21 PM, Brian West <brian at freeswitch.org> wrote:
>
>> www.bkw.org/mod_rtp_stream.tgz is a skel I started on.. You can take that
>> and finish it up if you like.
>>
>> /b
>>
>> On Jan 8, 2010, at 12:46 AM, shehzad p wrote:
>>
>> > I was looking for such functionality, but unfortunately it seems not
>> present
>> > right now, I am willing to build the rtp format mod as described by
>> Brian.,
>> > and will provide back to trunk.
>> >
>> > Although I have modified mod_skel application for use in dialplan in
>> custom
>> > application, I need to have a basic understanding regarding format mod.
>> > Will anybody please guide me from where to starts?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Fri Jan  8 19:31:18 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Jan 2010 21:31:18 -0600
Subject: [Freeswitch-users] stream a file multicast with mod_esf
In-Reply-To: <191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
References: <B98B6921358D0C49BD21CD023F1F52C302623B4B@hbghs04a.ww300.siemens.net>
	<B98B6921358D0C49BD21CD023F1F52C302623BE0@hbghs04a.ww300.siemens.net>
	<7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org>
	<27071973.post@talk.nabble.com>
	<378CDEC3-5A41-4EF9-927A-311F2531E6AB@freeswitch.org>
	<a792304b1001080725y19ff28camd37425927550f311@mail.gmail.com>
	<191c3a031001081848w9036207yf6a4b473ec233243@mail.gmail.com>
Message-ID: <85C74329-0D7C-48B2-B7A8-D4AA67C6E3F2@freeswitch.org>

Yah the skel of mod_rtp_stream was basted on mod_tone_stream so its a great jumping off point.

/b

On Jan 8, 2010, at 8:48 PM, Anthony Minessale wrote:

> look at mod_tone_stream its fairly basic
> but really look at any mod in formats dir
> 




From red.rain.seven at gmail.com  Sat Jan  9 00:47:09 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 9 Jan 2010 16:47:09 +0800
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <4B47C24A.3080305@aastral.net>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>
	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>
	<4B46792E.6090805@aastral.net>
	<E5EFC626-353A-4B54-9F22-59C0C48E3B95@freeswitch.org>
	<4B4684F3.8030504@aastral.net>
	<74C18F18-89EB-49DF-B5CE-C8D56856ABCD@freeswitch.org>
	<4B47C24A.3080305@aastral.net>
Message-ID: <59ad9ca11001090047u7e3b150bpa27ff1936f41993@mail.gmail.com>

Which version of Ubuntu was tested against?
It's surprising to find such testing result about CentOS.




On Sat, Jan 9, 2010 at 7:39 AM, Bill W <freeswitch at aastral.net> wrote:

> Hey Everyone,
>
> I spoke with Karl, and for the sake of completing this thread I'm
> posting the results.
>
> Ultimately the problem Karl found with OpenSUSE was the SQLite libraries
> leading to database corruption, and freeswitch misbehaving because of
> that.  He also was trying to do traffic shaping for his application and
> ran into problems with shaping not working right on multi-core x86_64
> kernels in SuSE.
>
> I asked him about using odbc in the core to get around the sqlite bug
> but he didn't bother because he needed traffic shaping that worked.
>
> So it's not so much that SuSE was a bad distro, but rather that it
> didn't work well with freeswitch and traffic shaping on multi-core
> x86_64.  He did mention that SuSE 11.1 is nice in general but to stay
> away from 11.2 because gcc segfaults on any significant build.
>
> Hope this helps the community!
> Bill
>
>
>
> His response:
> ====================
> The big thing you should learn from the investment of my time in the lab
> here is simply this, "listen to Brian".  From now on, I'm considering
> him the EF Hutton on #freeswitch.
>
> (Editors Note: For you youngsters out there, EF Hutton's tagline was
> "When EF Hutton talks, people listen.")
>
> When I added about 60 phones to the system, it essentially "blew up"
> whereby phones wouldn't register, the database would get corrupted, etc...
>
> We worked around the problem by making the freeswitch "db" directory a
> ramdisk, but that only mitigated the problem, and didn't entirely fix it.
>
> Oh yeah, and although mod_perl will compile in freeswitch, it will bomb
> out and segfault when attempting to run any mod_perl scripts in
> freeswitch.  The fix is to recompile perl from source.  Even then, I
> still had problems.
>
> Other problems with the remaining Suse installations still are:
>
> 1.) When you answer, it will hang up.  No rhyme, reason, or otherwise.
> 2.) Occasionally (rarely) we have issues with audio not passing through
> correctly.  One way audio, or none at all.
> 3.) When you dial, it will take 10 seconds to go through enum lookups
> and the like before finally hitting the PSTN gateway.
>
> Oh yeah, about RedHat...  Don't bother with it's sister CentOS either.
> I tested it here in the lab, and it was like getting in a time machine
> and going back to 2007.   Also, Centos is busted.  You'll find that
> Linux kernel will re-transmit IP packets from processes long since dead.
>  Suffice it to say, when it does this to RTP traffic it drives things
> bugnutz. (Ed. His test was to run 20mbit/sec through centos with a
> gigabit card overnight and it dropped 150 packets.  He did try different
> cards.)
>
> Mandrake - recommend this to people you dislike.  If they're ignorant
> enough you'll find them thanking you for it.
>
> Debian - almost awesome, but if failed miserably in the lab with packet
> shaping.  I mean, it thought it was working, but the overall quality was
> hit & miss.  Other than that, no complaints.  What was really attractive
> was it's got Cyrus 2.3 out of the box, so if you're using Cyrus and not
> using it for packet shaping you might consider Debian an option.  Be
> advised, since it flunked packet shaping we never bothered to
> compile/test freeswitch on it, so do your own research.
>
> Ubuntu, make no mistake...
> We tested Ubuntu pretty heavily here in the lab.
> Even the packet shaping works (HTB & SFQ).
> With Suse I has to custom-compile the kernel, and packet shaping ONLY
> worked with 2.6.28.8 - 2.6.28.10, the rest were buggy and the problems
> manifested themselves in ways that you'd think would be totally
> unrelated.  I should be more specific and state x86_64 multi-core.
> x86_64 single core seemed to do packet shaping (somewhat nicely) in
> 2.6.18 on up.  Previous to that it was rather "interesting"...
>
> Save yourself the headache and go with Ubuntu.
>
> ==================================
>
>
>
>
> Brian West wrote:
> > He's on the list Karl J. Vesterling
> >
> > /b
> >
> > On Jan 7, 2010, at 7:05 PM, Bill W. wrote:
> >
> >> Wow, I haven't heard of these issues.  Obviously this concerns me.  Are
> >> these documented anywhere so I can research this?  How do I get in touch
> >> with KJV?
> >>
> >> Thanks!
> >> Bill
> >>
> >>
> >>
> >> Brian West wrote:
> >>> Good luck with that you'll have an ass load of problems.  The reason
> its stable is the backports and outdated packages.  Bleeding edge will only
> screw you over... just ask KJV... He was on OpenSuSE and had nothing but
> weird problems.
> >>>
> >>>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From a.afzali2003 at gmail.com  Sat Jan  9 02:29:28 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sat, 9 Jan 2010 13:59:28 +0330
Subject: [Freeswitch-users] mod_xml_ldap compile error
Message-ID: <debff5a81001090229p40d0ad7yc642cb84f2174fde@mail.gmail.com>

Hi,

I'm using CentOS 5.4 x86-64 and just mod_xml_ldap causes this compile error.
I've found an exactly same issue in the list by Keith Laaks that responded
by Patrick. Patrick says :

I had the same issue and MikeJ (one of the core developers) looked at
it. Conclusion was that it is an openldap issue and iirc the solution is
to libtoolize libraries/liblutil/Makefile.in so that when running
configure a Makefile with proper compiler flags is generated in
libraries/liblutil/

My point is :
1) As I found in openldap 2.4.19, the package already is libtool support. if
I try libtoolize on openldap package , then :

You should update your `aclocal.m4' by running aclocal.
Putting files in AC_CONFIG_AUX_DIR, `build'.
libtoolize: `config.guess' exists: use `--force' to overwrite
libtoolize: `config.sub' exists: use `--force' to overwrite
libtoolize: `ltmain.sh' exists: use `--force' to overwrite

2) Anthony says : "CentOS 5.4 appears to have some bugs in the toolchain and
libc"

Is this issue because of those bugs? How I can workaround that?

Regards,
-- afshin

making all mod_xml_ldap
Creating mod_xml_ldap.so...
/usr/bin/ld:
/root/freeswitch-1.0.5-20100106-0400/libs/openldap-2.4.19/libraries/liblutil/liblutil.a(sasl.o):
relocation R_X86_64_32S against `a local symbol' can not be used when making
a shared object; recompile with -fPIC
/root/freeswitch-1.0.5-20100106-0400/libs/openldap-2.4.19/libraries/liblutil/liblutil.a:
could not read symbols: Bad value
collect2: ld returned 1 exit status
gcc -DWITH_OPENLDAP -DLDAP_DEPRECATED
-I/root/freeswitch-1.0.5-20100106-0400/libs/openldap-2.4.19/include
-I/root/freeswitch-1.0.5-20100106-0400/src/include
-I/root/freeswitch-1.0.5-20100106-0400/src/include
-I/root/freeswitch-1.0.5-20100106-0400/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
-g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE
-shared -o .libs/mod_xml_ldap.so -shared -Wl,-x .libs/mod_xml_ldap.o
/root/freeswitch-1.0.5-20100106-0400/.libs/libfreeswitch.so
-L/root/freeswitch-1.0.5-20100106-0400/libs/apr-util/xml/expat/lib
/root/freeswitch-1.0.5-20100106-0400/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/root/freeswitch-1.0.5-20100106-0400/libs/apr/.libs/libapr-1.a -luuid -lrt
-lcrypt -lpthread -L/root/freeswitch-1.0.5-20100106-0400/libs/srtp
-L/usr/kerberos/lib64 -ldl -lz -lncurses
/root/freeswitch-1.0.5-20100106-0400/libs/openldap-2.4.19/libraries/libldap_r/.libs/libldap_r.a
-lsasl2 -lssl -lcrypto -pthread
/root/freeswitch-1.0.5-20100106-0400/libs/openldap-2.4.19/libraries/liblber/.libs/liblber.a
-lm -lresolv
/root/freeswitch-1.0.5-20100106-0400/libs/openldap-2.4.19/libraries/liblutil/liblutil.a
-Wl,--rpath -Wl,/usr/local/freeswitch/lib -Wl,--rpath
-Wl,/usr/local/freeswitch/mod
make[5]: *** [mod_xml_ldap.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_xml_ldap-all] Error 1
make[2]: *** [all-recursive] Error 1
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From nicolas at medularis.com  Sat Jan  9 04:46:45 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Sat, 9 Jan 2010 09:46:45 -0300
Subject: [Freeswitch-users] Need to fake ringback
In-Reply-To: <191c3a031001081846i2c31b73bp714813d71a21b611@mail.gmail.com>
References: <1b46b4e81001072133h3f4c9bf1g68e25d27fd71edc1@mail.gmail.com>
	<91E856D5-B499-4C5F-A6A3-E2BB36C926C6@freeswitch.org>
	<1b46b4e81001081102k36c76792n5619426d05cca8cd@mail.gmail.com>
	<1b46b4e81001081102h2f644d28r86e14d7ba151e5c4@mail.gmail.com>
	<191c3a031001081846i2c31b73bp714813d71a21b611@mail.gmail.com>
Message-ID: <1b46b4e81001090446kc5cdd0fi21816dcc3c4d8f5a@mail.gmail.com>

Thanks!

I thought I was doing exactly the same, but I was creating the 2nd
session like this:

session2 = new Session(ostr2); // ostr is the corresponding originate command

So it was missing the "link" to the first leg. Doing it like this solved it:

session2 = new Session(ostr2, session1);

Now it works great!



On Fri, Jan 8, 2010 at 11:46 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> ringback variable must be set on the originating leg (the A leg aka inbound
> leg of the call)
> then that channel must be provided as the 2nd arg to the session constructor
> after the dial string or used with bridge.
>
> session.setVariable("ringback", "%(2000,4000,440,480)");
> session.execute("bridge", "sofia/internal/foo at bar.com");
>
> or
> // passing session as the 2nd arg allows the ringback to play on session
> while session2 is established.
> session2 = new Session("sofia/internal/foo at bar.com", session);
> bridge(session, session2); //or whatever
>
>
> On Fri, Jan 8, 2010 at 1:02 PM, Nicolas Brenner <nicolas at medularis.com>
> wrote:
>>
>> I'm using the default dialplan.
>>
>>
>> On Fri, Jan 8, 2010 at 4:02 PM, Nicolas Brenner <nicolas at medularis.com>
>> wrote:
>> > Not that I'm aware.
>> >
>> >
>> > On Fri, Jan 8, 2010 at 11:41 AM, Brian West <brian at freeswitch.org>
>> > wrote:
>> >> Are you using proxy media?
>> >>
>> >> /b
>> >>
>> >> On Jan 7, 2010, at 11:33 PM, Nicolas Brenner wrote:
>> >>
>> >>> I'm trying to fake a ringback for leg1 of a two-legged call without
>> >>> success.
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From craigesmith at gmail.com  Sat Jan  9 03:59:14 2010
From: craigesmith at gmail.com (Craig Smith)
Date: Sat, 9 Jan 2010 06:59:14 -0500
Subject: [Freeswitch-users] Voicemail - Is it possible?
Message-ID: <14dd859f1001090359y3761d773of56c0118d1da124@mail.gmail.com>

Hi!

I have a legacy PBX, Mitel to be specific, and an ailing AVST voice
mail system that I'm looking to replace with Freeswitch.

I know I can toggle the MWI on or off for a particular extension on
the Mitel by dialing a feature code plus extension.

Is there a way to use voicemail so that I can turn on/off MWI based on
the status of the mailbox.

Turning the light on seems like it would be the easy part, shutting it
off when all messages have been read has me perplexed.

If I used MySQL, could I scan flags on messages in mailboxes to toggle my MWI?

Thanks,

Craig



From a.alalousi at gmail.com  Sat Jan  9 07:30:28 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Sat, 9 Jan 2010 15:30:28 +0000
Subject: [Freeswitch-users] FAS detection with FS
In-Reply-To: <1262999844.21753.52.camel@local.freepabx.com>
References: <cd93b3701001071705w5bae2e88pe22308a799907a98@mail.gmail.com>
	<4B4728A5.4040805@kinetix.gr>
	<cd93b3701001080807p7d2f3480lc0ac3b1d9cf24a25@mail.gmail.com>
	<4B4761D5.5010907@kinetix.gr>
	<cd93b3701001081357i56c39e15j262ed05e5351a31a@mail.gmail.com>
	<4B47C77D.4080502@kinetix.gr>
	<1262999844.21753.52.camel@local.freepabx.com>
Message-ID: <cd93b3701001090730p48c9e98vf14b6ee65225d1ea@mail.gmail.com>

Hi David,

Oh God, yes, them.

They are Russians, based in Moscow. Personally, I think such flagrant abuse
of technology for fraud should be pursued to the maximum possible legal
extent, but alas .. Russian telcos, like the profitable mineral and carbon
industries are Mafia territory.

Regards,

Ahmed.

2010/1/9 David Knell <dave at 3c.co.uk>

> > Nonetheless if you want to pursue this and you need help I'd be happy
> > to assist, as FAS is a huge problem for carriers.
>
> FYI - here's some guys advertising "multilingual FAS" - presumably for
> those crooked enough to want to inflate their margins in this manner and
> too thick to do it for themselves:
> http://www.calltermination.com/forums/thread270514.html
>
> --Dave
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From a.alalousi at gmail.com  Sat Jan  9 07:49:43 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Sat, 9 Jan 2010 15:49:43 +0000
Subject: [Freeswitch-users] Rewriting hangup causes
Message-ID: <cd93b3701001090749t36c750c9k55b8ab36e72dacfe@mail.gmail.com>

Dear All,

I have a need to remap hangup causes returned to customers on one of our
clusters. Is there an easy way of achieving this in FS ?

Regards,

Ahmed.
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From anthony.minessale at gmail.com  Sat Jan  9 08:47:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 9 Jan 2010 10:47:52 -0600
Subject: [Freeswitch-users] Voicemail - Is it possible?
In-Reply-To: <14dd859f1001090359y3761d773of56c0118d1da124@mail.gmail.com>
References: <14dd859f1001090359y3761d773of56c0118d1da124@mail.gmail.com>
Message-ID: <191c3a031001090847s15d79443ibdce45b33e3a3017@mail.gmail.com>

Use esl to listen for mwi events on fs then make the call to turn on the
light manually with originate

On Jan 9, 2010 7:36 AM, "Craig Smith" <craigesmith at gmail.com> wrote:

Hi!

I have a legacy PBX, Mitel to be specific, and an ailing AVST voice
mail system that I'm looking to replace with Freeswitch.

I know I can toggle the MWI on or off for a particular extension on
the Mitel by dialing a feature code plus extension.

Is there a way to use voicemail so that I can turn on/off MWI based on
the status of the mailbox.

Turning the light on seems like it would be the easy part, shutting it
off when all messages have been read has me perplexed.

If I used MySQL, could I scan flags on messages in mailboxes to toggle my
MWI?

Thanks,

Craig

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From freeswitch at aastral.net  Sat Jan  9 08:48:20 2010
From: freeswitch at aastral.net (Bill W.)
Date: Sat, 09 Jan 2010 11:48:20 -0500
Subject: [Freeswitch-users] Installing freeswitch on CentOS
In-Reply-To: <59ad9ca11001090047u7e3b150bpa27ff1936f41993@mail.gmail.com>
References: <c96b95651001070300o13ad87bbr49f8e9b0f771e000@mail.gmail.com>	<CA5A491E9DEFBE4CB777DE97E21575E906FD48@prato.activenetwerx.local>	<191c3a031001070714v21ad0f46v47f300b1dce1429a@mail.gmail.com>	<4B46792E.6090805@aastral.net>	<E5EFC626-353A-4B54-9F22-59C0C48E3B95@freeswitch.org>	<4B4684F3.8030504@aastral.net>	<74C18F18-89EB-49DF-B5CE-C8D56856ABCD@freeswitch.org>	<4B47C24A.3080305@aastral.net>
	<59ad9ca11001090047u7e3b150bpa27ff1936f41993@mail.gmail.com>
Message-ID: <4B48B354.7000907@aastral.net>

Hey Henry,

I'm not sure what version of Ubuntu he tested against.  The other thing
to note is that he tested the different distros in VMware virtual
machines first.  And only the ones that passed his test there, went on
to testing on real hardware (Debian and Ubuntu).  So there may have been
an an interaction with VMWare that wasn't accounted for.  Again his
focus was fixing the sqlite bug, and reliable traffic shaping.


So here are the results of my initial testing of Ubuntu in MY environment.

I installed Ubuntu server 9.10 i386 on real hardware.

First off, there was an install bug where grub wouldn't install itself
into the MBR on my disks that were already partitioned.  (Raid 1 /boot
partition).  I had to blow away all partitions and re-create.  Then it
installed okay.  It did, however, have a nice feature where you could
specify a hot-spare for RAID 1 during setup.

Secondly, balance-alb mode appears broken for bonded interfaces with my
switch (HP Procurve).  Ubuntu just sat there drooling and not
transmitting packets.  As soon as I changed over to balance-tlb, things
worked.

Thirdly, the clustering tools (openais/pacemaker) are in their infancy.
Openais doesn't even come with an init script to get it started.

And as a general distro thing, Ubuntu doesn't have any tool as elegant
as chkconfig for managing the init scripts.  The administrative features
of the OS don't feel nearly as refined as CentOS or SuSE.

Since I'm trying to create a highly-available clustered freeswitch
environment I need a cluster-ready distro to handle that.  Not having
OpenAIS/pacemaker work out-of-the-box is a show-stopper for me.

So it looks like my choices are CentOS or SuSE.  Right now, I'm leaning
towards using Suse 11.1 and just using ODBC for the freeswitch core
database to get around the SQLite bug.  That way, at least I can use a
current kernel and current software.  True, I can't use the freeswitch
perl module.  But RedHat has had a long-standing perl bug as well.
(Google redhat perl bug).

Also, Karl reported GCC segfaulting during big compiles on SuSE 11.2 but
I haven't run into any.  True, I haven't compiled any kernels, but
freeswitch compiles just fine.

Hope this helps,
Bill



Henry Huang wrote:
> Which version of Ubuntu was tested against?
> It's surprising to find such testing result about CentOS.
> 
> 
> 



From max.bridgewater at gmail.com  Sat Jan  9 09:00:54 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sat, 9 Jan 2010 12:00:54 -0500
Subject: [Freeswitch-users] Help with Portech <-> Freeswitch
Message-ID: <c55d46951001090900p36df42e3ga91e61812d5b8d03@mail.gmail.com>

Hi Guys,

It appears quite a few people in the list are using Portech. Can you please
help me connect Freeswitch to  it for termination puposes?

Here is what I've done so far but without success.

In Freeswitch I created a profile and stored it in under
/usr/local/freeswitch/conf/sip_profiles/external/. Here is the content:

<include>
  <gateway name="portech">
    <param name="username" value="noauthentication" />
    <param name="password" value="noauthentication" />
    <param name="proxy" value="74.24.22.59" />
    <param name="realm" value="74.24.22.59" />
    <param name="register" value="false" />
  </gateway>
</include>

Then, in the /usr/local/freeswitch/confi/dialplan/default.xml, I added the
following extension:

   <extension name="portech">
      <condition field="destination_number" expression="^2801$">
        <action application="bridge"
data="sofia/gateway/portech/5147237479"/>
      </condition>
    </extension>

In Portech MV374, what I did is simply adding one entry in the Mobile/Lan to
mobile table  that consists of  URL: 74.24.22.59 and call Num: #.

Now, when I connect to Freeswitch with Xlite and try to dial extension 2801
it simply says User Busy. i don't even see that attempts are being made to
connect to the Portech gateway.

Any idea?
Thanks in advance.

Max.
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From jcasale at activenetwerx.com  Sat Jan  9 09:40:21 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 9 Jan 2010 17:40:21 +0000
Subject: [Freeswitch-users] Gateway Configuration
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9074E14@prato.activenetwerx.local>

It seems there are two ways to configure an spa3102's fxo port w/
pbx's, you can set the dial plan to <phone_#_of_pstn>@<freeswitch_ip>
or <ext_to_dial>@<freeswitch_ip>.

>From fs's perspective, what exactly is the difference here?
Are there any significant differences between the two methods?
Are there any best practices that should be considered?

Incoming sip did's and a zap line I had all were configured so that
they entered the public context filtered by <phone_#>.

Most of the examples I see for setting up the spa don't function like this
but a couple do?

Thanks!
jlc



From lortas at freenet.de  Sun Jan 10 12:11:16 2010
From: lortas at freenet.de (Holger von Rhein)
Date: Sun, 10 Jan 2010 21:11:16 +0100
Subject: [Freeswitch-users] looking for supported hardware
Message-ID: <4B4A3464.7070503@freenet.de>

Hi,

I plan to set up a telephone system at home. At the moment I have just 
one analogue hardware phone connected to my DSL-Splitter. To build a 
system capable to be my telephone system, I have looked for a list of 
supported hardware by freeswitch, but I could not find one. :(

Especial, I want to know which PCI(e)-Cards to connect an analogue 
hardware phone are well supported / recommended by freeswitch.

Does http://www.asterisk.org/astdocs/node12.html also apply for freeswitch?


On the other side, are there any features of an analogue modem, I have 
to pay attention for? Or will any Linux supported modem be okay to dial out?


Thanks for any hint.


Holger



From larclap at yahoo.com  Sun Jan 10 13:45:59 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Sun, 10 Jan 2010 13:45:59 -0800
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <COL117-W486989B811B372C35935AAB9700@phx.gbl>
References: <012901ca8fe6$e36b71c0$aa425540$@com>,
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>,
	<014801ca8fee$b75f8780$261e9680$@com>,
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>
Message-ID: <010001ca923e$4cd49bb0$e67dd310$@com>

Jeff,

 

I downloaded and install QT as you suggested, and created an environmental
variable QTDIR as C:\Qt\4.6.0 (the installation directory).

 

I then downloaded FreeSWITCH source via svn (v16230) and built it
successfully. I then opened the FSCOMM project within the FreeSWITCH
directory and attempted to build it. It failed with the messages below. Any
ideas on what I might have done wrong? The environment is Windows 7/64.

 

Error  1      error LNK2019: unresolved external symbol
__imp__switch_core_set_globals at 0 referenced in function "public: __thiscall
FSHost::FSHost(class QObject *)" (??0FSHost@@QAE at PAVQObject@@@Z)
fshost.obj       FSComm

Error  2      error LNK2019: unresolved external symbol
__imp__switch_core_setrlimits at 0 referenced in function "public: __thiscall
FSHost::FSHost(class QObject *)" (??0FSHost@@QAE at PAVQObject@@@Z)
fshost.obj       FSComm

Error  3      error LNK2019: unresolved external symbol
__imp__switch_core_destroy at 0 referenced in function "protected: virtual void
__thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj
FSComm

Error  4      error LNK2019: unresolved external symbol
__imp__switch_event_unbind_callback at 4 referenced in function "protected:
virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)
fshost.obj       FSComm

Error  5      error LNK2019: unresolved external symbol
__imp__switch_core_runtime_loop at 4 referenced in function "protected: virtual
void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)    fshost.obj
FSComm

Error  6      error LNK2019: unresolved external symbol
__imp__switch_core_init_and_modload at 12 referenced in function "protected:
virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ) fshost.obj
FSComm

Error  7      error LNK2019: unresolved external symbol
__imp__switch_log_printf referenced in function "protected: virtual void
__thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj
FSComm

Error  8      error LNK2001: unresolved external symbol
__imp__switch_log_printf   mod_qsettings.obj       FSComm

Error  9      error LNK2001: unresolved external symbol
__imp__switch_log_printf   prefportaudio.obj       FSComm

Error  10     error LNK2019: unresolved external symbol
__imp__switch_event_bind at 20 referenced in function "protected: virtual void
__thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj
FSComm

Error  11     error LNK2019: unresolved external symbol
__imp__switch_core_init at 12 referenced in function "protected: virtual void
__thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj
FSComm

Error  12     error LNK2001: unresolved external symbol
__imp__SWITCH_GLOBAL_dirs  fshost.obj    FSComm

Error  13     error LNK2019: unresolved external symbol
__imp__switch_event_dup at 8 referenced in function "void __cdecl
eventHandlerCallback(struct switch_event *)"
(?eventHandlerCallback@@YAXPAUswitch_event@@@Z)       fshost.obj    FSComm

Error  14     error LNK2019: unresolved external symbol
__imp__switch_event_name at 4 referenced in function "private: enum
switch_status_t __thiscall FSHost::processAlegEvent(struct switch_event
*,class QString)"
(?processAlegEvent at FSHost@@AAE?AW4switch_status_t@@PAUswitch_event@@VQString
@@@Z) fshost.obj    FSComm

Error  15     error LNK2019: unresolved external symbol
__imp__switch_event_get_header at 8 referenced in function "private: enum
switch_status_t __thiscall FSHost::processAlegEvent(struct switch_event
*,class QString)"
(?processAlegEvent at FSHost@@AAE?AW4switch_status_t@@PAUswitch_event@@VQString
@@@Z)     fshost.obj       FSComm

Error  16     error LNK2019: unresolved external symbol
__imp__switch_api_execute at 16 referenced in function "public: enum
switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class
QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)
fshost.obj    FSComm

Error  17     error LNK2019: unresolved external symbol
__imp__switch_console_stream_raw_write referenced in function "public: enum
switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class
QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)
fshost.obj    FSComm

Error  18     error LNK2019: unresolved external symbol
__imp__switch_console_stream_write referenced in function "public: enum
switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class
QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)
fshost.obj    FSComm

Error  19     error LNK2019: unresolved external symbol
__imp__switch_xml_parse_str at 8 referenced in function "public: struct
switch_xml * __thiscall XMLBinding::getConfigXML(class QString)"
(?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)
mod_qsettings.obj    FSComm

Error  20     error LNK2019: unresolved external symbol
__imp__switch_event_expand_headers at 8 referenced in function "public: struct
switch_xml * __thiscall XMLBinding::getConfigXML(class QString)"
(?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)
mod_qsettings.obj    FSComm

Error  21     error LNK2019: unresolved external symbol
__imp__switch_event_add_header_string at 16 referenced in function "public:
struct switch_xml * __thiscall XMLBinding::getConfigXML(class QString)"
(?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)
mod_qsettings.obj    FSComm

Error  22     error LNK2019: unresolved external symbol
__imp__switch_event_create_subclass_detailed at 24 referenced in function
_switch_event_create_plain      mod_qsettings.obj    FSComm

Error  23     error LNK2019: unresolved external symbol
__imp__switch_xml_bind_search_function_ret at 16 referenced in function "enum
switch_status_t __cdecl do_config(void)"
(?do_config@@YA?AW4switch_status_t@@XZ)    mod_qsettings.obj    FSComm

Error  24     error LNK2019: unresolved external symbol
__imp__switch_xml_parse_section_string at 4 referenced in function "enum
switch_status_t __cdecl do_config(void)"
(?do_config@@YA?AW4switch_status_t@@XZ)       mod_qsettings.obj    FSComm

Error  25     error LNK2019: unresolved external symbol
__imp__switch_xml_attr_soft at 8 referenced in function "enum switch_status_t
__cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)
mod_qsettings.obj    FSComm

Error  26     error LNK2019: unresolved external symbol
__imp__switch_xml_free at 4 referenced in function "enum switch_status_t
__cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)
mod_qsettings.obj       FSComm

Error  27     error LNK2019: unresolved external symbol
__imp__switch_xml_child at 8 referenced in function "enum switch_status_t
__cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)
mod_qsettings.obj       FSComm

Error  28     error LNK2019: unresolved external symbol
__imp__switch_xml_open_cfg at 12 referenced in function "enum switch_status_t
__cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)
mod_qsettings.obj    FSComm

Error  29     error LNK2019: unresolved external symbol
__imp__switch_find_local_ip at 16 referenced in function "public: void
__thiscall PrefSofia::readConfig(void)" (?readConfig at PrefSofia@@QAEXXZ)
prefsofia.obj FSComm

Error  30     fatal error LNK1120: 27 unresolved externals
debug\FSComm.exe     FSComm

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Thursday, January 07, 2010 7:17 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

 

The windows support is very experimental at this time!
 
You must manually install
http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
 
Then set the environment variable QTDIR in the environment variables. This
can be set from the Computer/Properties/Advanced system settings/Environment
Variables/User Variables settings screen.
 
QTDIR=c:\qt\4.6.0 - or wherever you installed it
 
then restart VS
 

 

  _____  

Date: Thu, 7 Jan 2010 21:45:56 -0200
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

I don't have a Windows machine to test that. Maybe jlenk could give us a
hand since he is the one who has created the visual studio project? 

 

Regards,
Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927



On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:

No Qt installed. I just checked out from
http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into
VS2008. 

Do I need to get
http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

 

 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:

I just downloaded the fscomm project and loaded it into vs2008. I?ve never
programmed in C++ (or c), just C#, so I can?t make anything of the following
two messages:

 

Error      1              error PRJ0019: A tool returned an error code from
"RCC resources.qrc"  FSComm              FSComm

 

Warning               2              The following environment variables
were not found: $(QTDIR)                Project FSComm

 

Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.

 

The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)

 

Lars

Do you have Qt 4.6 installed?
-MC

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

 

  _____  

Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up
<http://clk.atdmt.com/GBL/go/196390709/direct/01/>  now.

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From jeff at jefflenk.com  Sun Jan 10 14:16:01 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sun, 10 Jan 2010 16:16:01 -0600
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <010001ca923e$4cd49bb0$e67dd310$@com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>, ,
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>, ,
	<014801ca8fee$b75f8780$261e9680$@com>, ,
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>,
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>,
	<010001ca923e$4cd49bb0$e67dd310$@com>
Message-ID: <COL117-W6092C7A9A05EF45B305C54B96E0@phx.gbl>


you must also add FreeSwitchCoreLib as a dependency of FSComm

 

please update http://wiki.freeswitch.org/wiki/FSComm with any additional information you found helpful


 


From: larclap at yahoo.com
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 10 Jan 2010 13:45:59 -0800
Subject: Re: [Freeswitch-users] Compile error fscomm?







Jeff,
 
I downloaded and install QT as you suggested, and created an environmental variable QTDIR as C:\Qt\4.6.0 (the installation directory).
 
I then downloaded FreeSWITCH source via svn (v16230) and built it successfully. I then opened the FSCOMM project within the FreeSWITCH directory and attempted to build it. It failed with the messages below. Any ideas on what I might have done wrong? The environment is Windows 7/64.
 
Error  1      error LNK2019: unresolved external symbol __imp__switch_core_set_globals at 0 referenced in function "public: __thiscall FSHost::FSHost(class QObject *)" (??0FSHost@@QAE at PAVQObject@@@Z)   fshost.obj       FSComm
Error  2      error LNK2019: unresolved external symbol __imp__switch_core_setrlimits at 0 referenced in function "public: __thiscall FSHost::FSHost(class QObject *)" (??0FSHost@@QAE at PAVQObject@@@Z)   fshost.obj       FSComm
Error  3      error LNK2019: unresolved external symbol __imp__switch_core_destroy at 0 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj    FSComm
Error  4      error LNK2019: unresolved external symbol __imp__switch_event_unbind_callback at 4 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)    fshost.obj       FSComm
Error  5      error LNK2019: unresolved external symbol __imp__switch_core_runtime_loop at 4 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)    fshost.obj       FSComm
Error  6      error LNK2019: unresolved external symbol __imp__switch_core_init_and_modload at 12 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ) fshost.obj       FSComm
Error  7      error LNK2019: unresolved external symbol __imp__switch_log_printf referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj    FSComm
Error  8      error LNK2001: unresolved external symbol __imp__switch_log_printf   mod_qsettings.obj       FSComm
Error  9      error LNK2001: unresolved external symbol __imp__switch_log_printf   prefportaudio.obj       FSComm
Error  10     error LNK2019: unresolved external symbol __imp__switch_event_bind at 20 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj    FSComm
Error  11     error LNK2019: unresolved external symbol __imp__switch_core_init at 12 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj    FSComm
Error  12     error LNK2001: unresolved external symbol __imp__SWITCH_GLOBAL_dirs  fshost.obj    FSComm
Error  13     error LNK2019: unresolved external symbol __imp__switch_event_dup at 8 referenced in function "void __cdecl eventHandlerCallback(struct switch_event *)" (?eventHandlerCallback@@YAXPAUswitch_event@@@Z)       fshost.obj    FSComm
Error  14     error LNK2019: unresolved external symbol __imp__switch_event_name at 4 referenced in function "private: enum switch_status_t __thiscall FSHost::processAlegEvent(struct switch_event *,class QString)" (?processAlegEvent at FSHost@@AAE?AW4switch_status_t@@PAUswitch_event@@VQString@@@Z) fshost.obj    FSComm
Error  15     error LNK2019: unresolved external symbol __imp__switch_event_get_header at 8 referenced in function "private: enum switch_status_t __thiscall FSHost::processAlegEvent(struct switch_event *,class QString)" (?processAlegEvent at FSHost@@AAE?AW4switch_status_t@@PAUswitch_event@@VQString@@@Z)     fshost.obj       FSComm
Error  16     error LNK2019: unresolved external symbol __imp__switch_api_execute at 16 referenced in function "public: enum switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)   fshost.obj    FSComm
Error  17     error LNK2019: unresolved external symbol __imp__switch_console_stream_raw_write referenced in function "public: enum switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)  fshost.obj    FSComm
Error  18     error LNK2019: unresolved external symbol __imp__switch_console_stream_write referenced in function "public: enum switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)    fshost.obj    FSComm
Error  19     error LNK2019: unresolved external symbol __imp__switch_xml_parse_str at 8 referenced in function "public: struct switch_xml * __thiscall XMLBinding::getConfigXML(class QString)" (?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)      mod_qsettings.obj    FSComm
Error  20     error LNK2019: unresolved external symbol __imp__switch_event_expand_headers at 8 referenced in function "public: struct switch_xml * __thiscall XMLBinding::getConfigXML(class QString)" (?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)      mod_qsettings.obj    FSComm
Error  21     error LNK2019: unresolved external symbol __imp__switch_event_add_header_string at 16 referenced in function "public: struct switch_xml * __thiscall XMLBinding::getConfigXML(class QString)" (?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)      mod_qsettings.obj    FSComm
Error  22     error LNK2019: unresolved external symbol __imp__switch_event_create_subclass_detailed at 24 referenced in function _switch_event_create_plain      mod_qsettings.obj    FSComm
Error  23     error LNK2019: unresolved external symbol __imp__switch_xml_bind_search_function_ret at 16 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)    mod_qsettings.obj    FSComm
Error  24     error LNK2019: unresolved external symbol __imp__switch_xml_parse_section_string at 4 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)       mod_qsettings.obj    FSComm
Error  25     error LNK2019: unresolved external symbol __imp__switch_xml_attr_soft at 8 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)       mod_qsettings.obj    FSComm
Error  26     error LNK2019: unresolved external symbol __imp__switch_xml_free at 4 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)   mod_qsettings.obj       FSComm
Error  27     error LNK2019: unresolved external symbol __imp__switch_xml_child at 8 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)   mod_qsettings.obj       FSComm
Error  28     error LNK2019: unresolved external symbol __imp__switch_xml_open_cfg at 12 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)       mod_qsettings.obj    FSComm
Error  29     error LNK2019: unresolved external symbol __imp__switch_find_local_ip at 16 referenced in function "public: void __thiscall PrefSofia::readConfig(void)" (?readConfig at PrefSofia@@QAEXXZ)       prefsofia.obj FSComm
Error  30     fatal error LNK1120: 27 unresolved externals    debug\FSComm.exe     FSComm
 


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk
Sent: Thursday, January 07, 2010 7:17 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?
 
The windows support is very experimental at this time!
 
You must manually install http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
 
Then set the environment variable QTDIR in the environment variables. This can be set from the Computer/Properties/Advanced system settings/Environment Variables/User Variables settings screen.
 
QTDIR=c:\qt\4.6.0 - or wherever you installed it
 
then restart VS
 

 



Date: Thu, 7 Jan 2010 21:45:56 -0200
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

I don't have a Windows machine to test that. Maybe jlenk could give us a hand since he is the one who has created the visual studio project? 

 

Regards,Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927



On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:


No Qt installed. I just checked out from http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into VS2008. 
Do I need to get http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?
 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?


 
 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:


I just downloaded the fscomm project and loaded it into vs2008. I?ve never programmed in C++ (or c), just C#, so I can?t make anything of the following two messages:
 
Error      1              error PRJ0019: A tool returned an error code from "RCC resources.qrc"  FSComm              FSComm
 
Warning               2              The following environment variables were not found: $(QTDIR)                Project FSComm
 
Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.
 
The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)
 
Lars

Do you have Qt 4.6 installed?
-MC
 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 
 



Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. 		 	   		  
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From jcasale at activenetwerx.com  Sun Jan 10 15:08:25 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sun, 10 Jan 2010 23:08:25 +0000
Subject: [Freeswitch-users] looking for supported hardware
In-Reply-To: <4B4A3464.7070503@freenet.de>
References: <4B4A3464.7070503@freenet.de>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9076377@prato.activenetwerx.local>

>Especial, I want to know which PCI(e)-Cards to connect an analogue 
>hardware phone are well supported / recommended by freeswitch.
>
>Does http://www.asterisk.org/astdocs/node12.html also apply for freeswitch?

Those are expensive, and a IMHO the OpenZAP implementation is still in its
infancy and I can't see it getting a strong movement behind it to make it better.

External IP based gateway's are much more easier to deal with. I have heard
good things about Patton's and AudioCodes and both make some cheap models
that are way less money than the digium cards. I set up an SPA2102 for a pair
of FXS's I needed for faxing its working flawless, I am just setting up an
SPA3102 as I needed an FXO port and that's 3/4 working. Those Linksys devices
were very cheap, ~80 CDN I paid for each.

>On the other side, are there any features of an analogue modem, I have 
>to pay attention for? Or will any Linux supported modem be okay to dial out?

Linux will support them fine, but that's not the only concern, IMHO you want
to avoid those, there is nothing that makes them appealing anymore from
my perspective...

jlc



From darklion11 at yahoo.com  Sun Jan 10 17:26:02 2010
From: darklion11 at yahoo.com (Edmar Cruz)
Date: Sun, 10 Jan 2010 17:26:02 -0800 (PST)
Subject: [Freeswitch-users]  Change Domain Freeswitch
Message-ID: <27104680.post@talk.nabble.com>


Dear All,

   How can i change the domain of my freeswitch 52.236.125.12 to
sip.grandminister.com to be able to detect the presence of the user whos
online or not...

Thanks
Edmar
-- 
View this message in context: http://old.nabble.com/Change-Domain-Freeswitch-tp27104680p27104680.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From larclap at yahoo.com  Sun Jan 10 18:25:00 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Sun, 10 Jan 2010 18:25:00 -0800
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <COL117-W6092C7A9A05EF45B305C54B96E0@phx.gbl>
References: <012901ca8fe6$e36b71c0$aa425540$@com>, ,
	<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>, ,
	<014801ca8fee$b75f8780$261e9680$@com>, ,
	<ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>,
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>,
	<010001ca923e$4cd49bb0$e67dd310$@com>
	<COL117-W6092C7A9A05EF45B305C54B96E0@phx.gbl>
Message-ID: <014b01ca9265$46fa6e50$d4ef4af0$@com>

After realizing I need to build FreeSWITCH, which builds FreeSwitchCore.lib,
I then open fscomm and build it after setting the dependency. Now I get
these messages, which is really strange since the configuration for each
FreeSWITCH and fscomm is set to Debug. The missing files are in .\debug.
What am I missing?

 

Error  1      fatal error C1083: Cannot open source file:
'.\release\qrc_resources.cpp': No such file or directory     c1xx   FSComm

Error  2      fatal error C1083: Cannot open source file:
'.\release\moc_prefsofia.cpp': No such file or directory     c1xx   FSComm

Error  3      fatal error C1083: Cannot open source file:
'.\release\moc_prefportaudio.cpp': No such file or directory  c1xx   FSComm

Error  4      fatal error C1083: Cannot open source file:
'.\release\moc_prefdialog.cpp': No such file or directory     c1xx   FSComm

Error  5      fatal error C1083: Cannot open source file:
'.\release\moc_mainwindow.cpp': No such file or directory     c1xx   FSComm

Error  6      fatal error C1083: Cannot open source file:
'.\release\moc_fshost.cpp': No such file or directory     c1xx   FSComm

Error  7      fatal error C1083: Cannot open source file:
'.\release\moc_accountdialog.cpp': No such file or directory  c1xx   FSComm

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Sunday, January 10, 2010 2:16 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

 

you must also add FreeSwitchCoreLib as a dependency of FSComm
 
please update http://wiki.freeswitch.org/wiki/FSComm with any additional
information you found helpful

 

  _____  

From: larclap at yahoo.com
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 10 Jan 2010 13:45:59 -0800
Subject: Re: [Freeswitch-users] Compile error fscomm?

Jeff,

 

I downloaded and install QT as you suggested, and created an environmental
variable QTDIR as C:\Qt\4.6.0 (the installation directory).

 

I then downloaded FreeSWITCH source via svn (v16230) and built it
successfully. I then opened the FSCOMM project within the FreeSWITCH
directory and attempted to build it. It failed with the messages below. Any
ideas on what I might have done wrong? The environment is Windows 7/64.

 

Error  1      error LNK2019: unresolved external symbol
__imp__switch_core_set_globals at 0 referenced in function "public: __thiscall
FSHost::FSHost(class QObject *)" (??0FSHost@@QAE at PAVQObject@@@Z)
fshost.obj       FSComm

Error  2      error LNK2019: unresolved external symbol
__imp__switch_core_setrlimits at 0 referenced in function "public: __thiscall
FSHost::FSHost(class QObject *)" (??0FSHost@@QAE at PAVQObject@@@Z)
fshost.obj       FSComm

Error  3      error LNK2019: unresolved external symbol
__imp__switch_core_destroy at 0 referenced in function "protected: virtual void
__thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj
FSComm

Error  4      error LNK2019: unresolved external symbol
__imp__switch_event_unbind_callback at 4 referenced in function "protected:
virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)
fshost.obj       FSComm

Error  5      error LNK2019: unresolved external symbol
__imp__switch_core_runtime_loop at 4 referenced in function "protected: virtual
void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)    fshost.obj
FSComm

Error  6      error LNK2019: unresolved external symbol
__imp__switch_core_init_and_modload at 12 referenced in function "protected:
virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ) fshost.obj
FSComm

Error  7      error LNK2019: unresolved external symbol
__imp__switch_log_printf referenced in function "protected: virtual void
__thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj
FSComm

Error  8      error LNK2001: unresolved external symbol
__imp__switch_log_printf   mod_qsettings.obj       FSComm

Error  9      error LNK2001: unresolved external symbol
__imp__switch_log_printf   prefportaudio.obj       FSComm

Error  10     error LNK2019: unresolved external symbol
__imp__switch_event_bind at 20 referenced in function "protected: virtual void
__thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj
FSComm

Error  11     error LNK2019: unresolved external symbol
__imp__switch_core_init at 12 referenced in function "protected: virtual void
__thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj
FSComm

Error  12     error LNK2001: unresolved external symbol
__imp__SWITCH_GLOBAL_dirs  fshost.obj    FSComm

Error  13     error LNK2019: unresolved external symbol
__imp__switch_event_dup at 8 referenced in function "void __cdecl
eventHandlerCallback(struct switch_event *)"
(?eventHandlerCallback@@YAXPAUswitch_event@@@Z)       fshost.obj    FSComm

Error  14     error LNK2019: unresolved external symbol
__imp__switch_event_name at 4 referenced in function "private: enum
switch_status_t __thiscall FSHost::processAlegEvent(struct switch_event
*,class QString)"
(?processAlegEvent at FSHost@@AAE?AW4switch_status_t@@PAUswitch_event@@VQString
@@@Z) fshost.obj    FSComm

Error  15     error LNK2019: unresolved external symbol
__imp__switch_event_get_header at 8 referenced in function "private: enum
switch_status_t __thiscall FSHost::processAlegEvent(struct switch_event
*,class QString)"
(?processAlegEvent at FSHost@@AAE?AW4switch_status_t@@PAUswitch_event@@VQString
@@@Z)     fshost.obj       FSComm

Error  16     error LNK2019: unresolved external symbol
__imp__switch_api_execute at 16 referenced in function "public: enum
switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class
QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)
fshost.obj    FSComm

Error  17     error LNK2019: unresolved external symbol
__imp__switch_console_stream_raw_write referenced in function "public: enum
switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class
QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)
fshost.obj    FSComm

Error  18     error LNK2019: unresolved external symbol
__imp__switch_console_stream_write referenced in function "public: enum
switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class
QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)
fshost.obj    FSComm

Error  19     error LNK2019: unresolved external symbol
__imp__switch_xml_parse_str at 8 referenced in function "public: struct
switch_xml * __thiscall XMLBinding::getConfigXML(class QString)"
(?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)
mod_qsettings.obj    FSComm

Error  20     error LNK2019: unresolved external symbol
__imp__switch_event_expand_headers at 8 referenced in function "public: struct
switch_xml * __thiscall XMLBinding::getConfigXML(class QString)"
(?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)
mod_qsettings.obj    FSComm

Error  21     error LNK2019: unresolved external symbol
__imp__switch_event_add_header_string at 16 referenced in function "public:
struct switch_xml * __thiscall XMLBinding::getConfigXML(class QString)"
(?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)
mod_qsettings.obj    FSComm

Error  22     error LNK2019: unresolved external symbol
__imp__switch_event_create_subclass_detailed at 24 referenced in function
_switch_event_create_plain      mod_qsettings.obj    FSComm

Error  23     error LNK2019: unresolved external symbol
__imp__switch_xml_bind_search_function_ret at 16 referenced in function "enum
switch_status_t __cdecl do_config(void)"
(?do_config@@YA?AW4switch_status_t@@XZ)    mod_qsettings.obj    FSComm

Error  24     error LNK2019: unresolved external symbol
__imp__switch_xml_parse_section_string at 4 referenced in function "enum
switch_status_t __cdecl do_config(void)"
(?do_config@@YA?AW4switch_status_t@@XZ)       mod_qsettings.obj    FSComm

Error  25     error LNK2019: unresolved external symbol
__imp__switch_xml_attr_soft at 8 referenced in function "enum switch_status_t
__cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)
mod_qsettings.obj    FSComm

Error  26     error LNK2019: unresolved external symbol
__imp__switch_xml_free at 4 referenced in function "enum switch_status_t
__cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)
mod_qsettings.obj       FSComm

Error  27     error LNK2019: unresolved external symbol
__imp__switch_xml_child at 8 referenced in function "enum switch_status_t
__cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)
mod_qsettings.obj       FSComm

Error  28     error LNK2019: unresolved external symbol
__imp__switch_xml_open_cfg at 12 referenced in function "enum switch_status_t
__cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)
mod_qsettings.obj    FSComm

Error  29     error LNK2019: unresolved external symbol
__imp__switch_find_local_ip at 16 referenced in function "public: void
__thiscall PrefSofia::readConfig(void)" (?readConfig at PrefSofia@@QAEXXZ)
prefsofia.obj FSComm

Error  30     fatal error LNK1120: 27 unresolved externals
debug\FSComm.exe     FSComm

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Thursday, January 07, 2010 7:17 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

 

The windows support is very experimental at this time!
 
You must manually install
http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
 
Then set the environment variable QTDIR in the environment variables. This
can be set from the Computer/Properties/Advanced system settings/Environment
Variables/User Variables settings screen.
 
QTDIR=c:\qt\4.6.0 - or wherever you installed it
 
then restart VS
 

 

  _____  

Date: Thu, 7 Jan 2010 21:45:56 -0200
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

I don't have a Windows machine to test that. Maybe jlenk could give us a
hand since he is the one who has created the visual studio project? 

 

Regards,
Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927

On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:

No Qt installed. I just checked out from
http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into
VS2008. 

Do I need to get
http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

 

 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:

I just downloaded the fscomm project and loaded it into vs2008. I?ve never
programmed in C++ (or c), just C#, so I can?t make anything of the following
two messages:

 

Error      1              error PRJ0019: A tool returned an error code from
"RCC resources.qrc"  FSComm              FSComm

 

Warning               2              The following environment variables
were not found: $(QTDIR)                Project FSComm

 

Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.

 

The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)

 

Lars

Do you have Qt 4.6 installed?
-MC

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

 

  _____  

Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now.
<http://clk.atdmt.com/GBL/go/196390709/direct/01/> 

 

  _____  

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From andrew at hijacked.us  Sun Jan 10 18:52:48 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Sun, 10 Jan 2010 21:52:48 -0500
Subject: [Freeswitch-users] really no installer for w2k anywhere?
In-Reply-To: <4B46A5C4.9040809@gmail.com>
References: <4B46A5C4.9040809@gmail.com>
Message-ID: <20100111025248.GA10774@hijacked.us>

On Fri, Jan 08, 2010 at 04:25:56AM +0100, Pekka Kurki wrote:
> all installer versions fail with  missing  getnameinfo/getaddressinfo 
> support in w2k.
>

Sofia and some other FS bits use some XP and higher APIs, I never had
the time to properly try to backport FS to win2k.

Really you might want to consider upgrading from an OS that is now over
a decade old if you actually want to use it for a production purpose.
Alternately post a bounty for doing the backport (its a little more
complicated than I'd hoped, IIRC).

Andrew



From jeff at jefflenk.com  Sun Jan 10 20:18:18 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sun, 10 Jan 2010 22:18:18 -0600
Subject: [Freeswitch-users] Compile error fscomm?
In-Reply-To: <014b01ca9265$46fa6e50$d4ef4af0$@com>
References: <012901ca8fe6$e36b71c0$aa425540$@com>,
	,,<87f2f3b91001071435s226f1286qaab3ef8a7e234e97@mail.gmail.com>,
	,,<014801ca8fee$b75f8780$261e9680$@com>,
	, , <ebbc2d551001071545n3f430f43y577440b30378a971@mail.gmail.com>, ,
	<COL117-W486989B811B372C35935AAB9700@phx.gbl>, ,
	<010001ca923e$4cd49bb0$e67dd310$@com>,
	<COL117-W6092C7A9A05EF45B305C54B96E0@phx.gbl>,
	<014b01ca9265$46fa6e50$d4ef4af0$@com>
Message-ID: <COL117-W27A9215BD93159A12F8BEEB96D0@phx.gbl>


Please make sure you are on svn16231 or later
 


From: larclap at yahoo.com
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 10 Jan 2010 18:25:00 -0800
Subject: Re: [Freeswitch-users] Compile error fscomm?







After realizing I need to build FreeSWITCH, which builds FreeSwitchCore.lib, I then open fscomm and build it after setting the dependency. Now I get these messages, which is really strange since the configuration for each FreeSWITCH and fscomm is set to Debug. The missing files are in .\debug. What am I missing?
 
Error  1      fatal error C1083: Cannot open source file: '.\release\qrc_resources.cpp': No such file or directory     c1xx   FSComm
Error  2      fatal error C1083: Cannot open source file: '.\release\moc_prefsofia.cpp': No such file or directory     c1xx   FSComm
Error  3      fatal error C1083: Cannot open source file: '.\release\moc_prefportaudio.cpp': No such file or directory  c1xx   FSComm
Error  4      fatal error C1083: Cannot open source file: '.\release\moc_prefdialog.cpp': No such file or directory     c1xx   FSComm
Error  5      fatal error C1083: Cannot open source file: '.\release\moc_mainwindow.cpp': No such file or directory     c1xx   FSComm
Error  6      fatal error C1083: Cannot open source file: '.\release\moc_fshost.cpp': No such file or directory     c1xx   FSComm
Error  7      fatal error C1083: Cannot open source file: '.\release\moc_accountdialog.cpp': No such file or directory  c1xx   FSComm
 


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk
Sent: Sunday, January 10, 2010 2:16 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?
 
you must also add FreeSwitchCoreLib as a dependency of FSComm
 
please update http://wiki.freeswitch.org/wiki/FSComm with any additional information you found helpful

 



From: larclap at yahoo.com
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 10 Jan 2010 13:45:59 -0800
Subject: Re: [Freeswitch-users] Compile error fscomm?

Jeff,
 
I downloaded and install QT as you suggested, and created an environmental variable QTDIR as C:\Qt\4.6.0 (the installation directory).
 
I then downloaded FreeSWITCH source via svn (v16230) and built it successfully. I then opened the FSCOMM project within the FreeSWITCH directory and attempted to build it. It failed with the messages below. Any ideas on what I might have done wrong? The environment is Windows 7/64.
 
Error  1      error LNK2019: unresolved external symbol __imp__switch_core_set_globals at 0 referenced in function "public: __thiscall FSHost::FSHost(class QObject *)" (??0FSHost@@QAE at PAVQObject@@@Z)   fshost.obj       FSComm
Error  2      error LNK2019: unresolved external symbol __imp__switch_core_setrlimits at 0 referenced in function "public: __thiscall FSHost::FSHost(class QObject *)" (??0FSHost@@QAE at PAVQObject@@@Z)   fshost.obj       FSComm
Error  3      error LNK2019: unresolved external symbol __imp__switch_core_destroy at 0 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj    FSComm
Error  4      error LNK2019: unresolved external symbol __imp__switch_event_unbind_callback at 4 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)    fshost.obj       FSComm
Error  5      error LNK2019: unresolved external symbol __imp__switch_core_runtime_loop at 4 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)    fshost.obj       FSComm
Error  6      error LNK2019: unresolved external symbol __imp__switch_core_init_and_modload at 12 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ) fshost.obj       FSComm
Error  7      error LNK2019: unresolved external symbol __imp__switch_log_printf referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj    FSComm
Error  8      error LNK2001: unresolved external symbol __imp__switch_log_printf   mod_qsettings.obj       FSComm
Error  9      error LNK2001: unresolved external symbol __imp__switch_log_printf   prefportaudio.obj       FSComm
Error  10     error LNK2019: unresolved external symbol __imp__switch_event_bind at 20 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj    FSComm
Error  11     error LNK2019: unresolved external symbol __imp__switch_core_init at 12 referenced in function "protected: virtual void __thiscall FSHost::run(void)" (?run at FSHost@@MAEXXZ)      fshost.obj    FSComm
Error  12     error LNK2001: unresolved external symbol __imp__SWITCH_GLOBAL_dirs  fshost.obj    FSComm
Error  13     error LNK2019: unresolved external symbol __imp__switch_event_dup at 8 referenced in function "void __cdecl eventHandlerCallback(struct switch_event *)" (?eventHandlerCallback@@YAXPAUswitch_event@@@Z)       fshost.obj    FSComm
Error  14     error LNK2019: unresolved external symbol __imp__switch_event_name at 4 referenced in function "private: enum switch_status_t __thiscall FSHost::processAlegEvent(struct switch_event *,class QString)" (?processAlegEvent at FSHost@@AAE?AW4switch_status_t@@PAUswitch_event@@VQString@@@Z) fshost.obj    FSComm
Error  15     error LNK2019: unresolved external symbol __imp__switch_event_get_header at 8 referenced in function "private: enum switch_status_t __thiscall FSHost::processAlegEvent(struct switch_event *,class QString)" (?processAlegEvent at FSHost@@AAE?AW4switch_status_t@@PAUswitch_event@@VQString@@@Z)     fshost.obj       FSComm
Error  16     error LNK2019: unresolved external symbol __imp__switch_api_execute at 16 referenced in function "public: enum switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)   fshost.obj    FSComm
Error  17     error LNK2019: unresolved external symbol __imp__switch_console_stream_raw_write referenced in function "public: enum switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)  fshost.obj    FSComm
Error  18     error LNK2019: unresolved external symbol __imp__switch_console_stream_write referenced in function "public: enum switch_status_t __thiscall FSHost::sendCmd(char const *,char const *,class QString *)" (?sendCmd at FSHost@@QAE?AW4switch_status_t@@PBD0PAVQString@@@Z)    fshost.obj    FSComm
Error  19     error LNK2019: unresolved external symbol __imp__switch_xml_parse_str at 8 referenced in function "public: struct switch_xml * __thiscall XMLBinding::getConfigXML(class QString)" (?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)      mod_qsettings.obj    FSComm
Error  20     error LNK2019: unresolved external symbol __imp__switch_event_expand_headers at 8 referenced in function "public: struct switch_xml * __thiscall XMLBinding::getConfigXML(class QString)" (?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)      mod_qsettings.obj    FSComm
Error  21     error LNK2019: unresolved external symbol __imp__switch_event_add_header_string at 16 referenced in function "public: struct switch_xml * __thiscall XMLBinding::getConfigXML(class QString)" (?getConfigXML at XMLBinding@@QAEPAUswitch_xml@@VQString@@@Z)      mod_qsettings.obj    FSComm
Error  22     error LNK2019: unresolved external symbol __imp__switch_event_create_subclass_detailed at 24 referenced in function _switch_event_create_plain      mod_qsettings.obj    FSComm
Error  23     error LNK2019: unresolved external symbol __imp__switch_xml_bind_search_function_ret at 16 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)    mod_qsettings.obj    FSComm
Error  24     error LNK2019: unresolved external symbol __imp__switch_xml_parse_section_string at 4 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)       mod_qsettings.obj    FSComm
Error  25     error LNK2019: unresolved external symbol __imp__switch_xml_attr_soft at 8 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)       mod_qsettings.obj    FSComm
Error  26     error LNK2019: unresolved external symbol __imp__switch_xml_free at 4 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)   mod_qsettings.obj       FSComm
Error  27     error LNK2019: unresolved external symbol __imp__switch_xml_child at 8 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)   mod_qsettings.obj       FSComm
Error  28     error LNK2019: unresolved external symbol __imp__switch_xml_open_cfg at 12 referenced in function "enum switch_status_t __cdecl do_config(void)" (?do_config@@YA?AW4switch_status_t@@XZ)       mod_qsettings.obj    FSComm
Error  29     error LNK2019: unresolved external symbol __imp__switch_find_local_ip at 16 referenced in function "public: void __thiscall PrefSofia::readConfig(void)" (?readConfig at PrefSofia@@QAEXXZ)       prefsofia.obj FSComm
Error  30     fatal error LNK1120: 27 unresolved externals    debug\FSComm.exe     FSComm
 


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk
Sent: Thursday, January 07, 2010 7:17 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?
 
The windows support is very experimental at this time!
 
You must manually install http://get.qt.nokia.com/qt/source/qt-win-opensource-4.6.0-vs2008.exe
 
Then set the environment variable QTDIR in the environment variables. This can be set from the Computer/Properties/Advanced system settings/Environment Variables/User Variables settings screen.
 
QTDIR=c:\qt\4.6.0 - or wherever you installed it
 
then restart VS
 

 



Date: Thu, 7 Jan 2010 21:45:56 -0200
From: jmesquita at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?

I don't have a Windows machine to test that. Maybe jlenk could give us a hand since he is the one who has created the visual studio project? 

 

Regards,Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927

On Thu, Jan 7, 2010 at 9:11 PM, Lars Zeb <larclap at yahoo.com> wrote:


No Qt installed. I just checked out from http://svn.freeswitch.org/svn/freeswitch/trunk/fscomm and loaded it into VS2008. 
Do I need to get http://get.qt.nokia.com/qtsdk/qt-sdk-win-opensource-2009.05.exe ?
 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, January 07, 2010 2:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Compile error fscomm?


 
 

On Thu, Jan 7, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:


I just downloaded the fscomm project and loaded it into vs2008. I?ve never programmed in C++ (or c), just C#, so I can?t make anything of the following two messages:
 
Error      1              error PRJ0019: A tool returned an error code from "RCC resources.qrc"  FSComm              FSComm
 
Warning               2              The following environment variables were not found: $(QTDIR)                Project FSComm
 
Any suggestions? I do see the file ?resources.qrc? in the fscomm folder.
 
The environment is Windows 7 64bit & VS2008 (Version 9.0.30729.1 SP)
 
Lars

Do you have Qt 4.6 installed?
-MC
 

_______________________________________________
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From jmesquita at freeswitch.org  Sun Jan 10 20:46:18 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 11 Jan 2010 01:46:18 -0300
Subject: [Freeswitch-users] MacOSX
In-Reply-To: <6b65470d1001081209v64d20f9ehdd7a7b4c6516c540@mail.gmail.com>
References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net>
	<60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org>
	<23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com>
	<ebbc2d550912300627t7d164f9cnb7c772ac7ea67e91@mail.gmail.com>
	<23f91031001070811m1bfcea40j5bedd81b8eeb3c46@mail.gmail.com>
	<B6423326-77F5-43A4-9A3D-5BFCC451089F@freeswitch.org>
	<ebbc2d551001071227y22d6eaa8y43742fb4328a67b2@mail.gmail.com>
	<D03B492C-74B9-4B38-B67B-017690FF372C@freeswitch.org>
	<2d9149cd1001081103m4d2b6852l9848ce4c82005fd3@mail.gmail.com>
	<6b65470d1001081209v64d20f9ehdd7a7b4c6516c540@mail.gmail.com>
Message-ID: <ebbc2d551001102046o1dcf14cbkd9a639ccd16ef713@mail.gmail.com>

Thank you both for the support. This week I will make even more changes to
FSComm. Stay tuned and I hope you like it.

As for the sponsorship, Kristian, me and the project rely on that! ;-) Thank
you for even asking!

Regards,
Jo?o Mesquita
Paypal: jmesquita at gmail.com


On Fri, Jan 8, 2010 at 5:09 PM, William Suffill
<william.suffill at gmail.com>wrote:

> jmesquita at gmail.com is his Paypal account. Ya hope he gets better as well.
>
> -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mayamatakeshi at gmail.com  Sun Jan 10 21:00:57 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Mon, 11 Jan 2010 14:00:57 +0900
Subject: [Freeswitch-users] Avoiding restart of transfer_ringback file
	during successive bridge actions.
Message-ID: <15b9404e1001102100u157db096t636f06c0fa774291@mail.gmail.com>

Hi,
I'm implementing ACD functionality using a dialplan.
Basically, I prepare a list of members in a group and try each one of them
in successive bridge actions.
I cannot use a single bridge action with all members separated with pipes
because I am required to check some conditions like if the member is already
in another call. So what I do is to add a transfer action after the bridge
so that the call reenters the dialplan with the remaining of the list.
Up to this point things are fine.
However, if the call is answered and transferred to the group, I set
transfer_ringback so that I can play a file while the bridge is happening.
In case transfer_ringback is set to something like "local_stream://moh",
then we hear a small glitch every time a bridge action happens as the bridge
doesn't take into account that the ringback was already set up by a previous
operation and sets up the playfile operation again. But this is also fine.
The problem is in case transfer_ringback is set to an actual file and not
local_stream. In this case, every bridge action will cause restart of the
playback from the beginning.
So, is it possible to ask bridge to do not setup playback if it is already
running referencing the same file?

regards,
takeshi
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From gabe at gundy.org  Sun Jan 10 21:36:15 2010
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sun, 10 Jan 2010 22:36:15 -0700
Subject: [Freeswitch-users] fifo funk?
Message-ID: <903da5681001102136y6d51f960hd8c10dde4e74ac26@mail.gmail.com>

Anyone else notice that adding and deleting members of a queue works
better then reparsing with the exact same member info?

It seems to keep better track of where to send the next call.  Not a
big deal, just wondering if there is something going on that I don't
know about.

All in all we've really like fifos :) Thanks FS devs!


Gabe



From dujinfang at gmail.com  Sun Jan 10 21:49:52 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 11 Jan 2010 13:49:52 +0800
Subject: [Freeswitch-users] fifo funk?
In-Reply-To: <903da5681001102136y6d51f960hd8c10dde4e74ac26@mail.gmail.com>
References: <903da5681001102136y6d51f960hd8c10dde4e74ac26@mail.gmail.com>
Message-ID: <23f91031001102149x1f1913amefa97d02981261d@mail.gmail.com>

see fifo_member add/delete api

2010/1/11 Gabriel Gunderson <gabe at gundy.org>:
> Anyone else notice that adding and deleting members of a queue works
> better then reparsing with the exact same member info?
>
> It seems to keep better track of where to send the next call. ?Not a
> big deal, just wondering if there is something going on that I don't
> know about.
>
> All in all we've really like fifos :) Thanks FS devs!
>
>
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From sharad at coraltele.com  Sun Jan 10 23:30:55 2010
From: sharad at coraltele.com (Sharad)
Date: Sun, 10 Jan 2010 23:30:55 -0800 (PST)
Subject: [Freeswitch-users] Personal Greeting
In-Reply-To: <53FBFF55-D94D-4BCA-81EC-230F00DDC344@jerris.com>
References: <1262066443847-4226681.post@n2.nabble.com>
	<27073953.post@talk.nabble.com>
	<53FBFF55-D94D-4BCA-81EC-230F00DDC344@jerris.com>
Message-ID: <1263195055974-4284196.post@n2.nabble.com>



Michael Jerris wrote:
> 
> Huh?  What does this have to do with his question?
> 
> On Jan 8, 2010, at 5:25 AM, Edmar Cruz <darklion11 at yahoo.com> wrote:
> 
>>
>> You can set it in the dialplan
>>
>> <action application="playback" data="yourgreeting.wav" />
>>
>> For some cases softphones has its own greeting :working:
>>
>> Hope it can help you..
>>
>>
>> sharad-5 wrote:
>>>
>>>
>>>
>>> Hi
>>>
>>> I am new to Freeswitch so my question may be a stupid question.
>>>
>>> I just want to know how to disable the personal greeting to the  
>>> default
>>> one.
>>> One user has recorded his personal greeting now how can he make this
>>> default.
>>>
>>> I could not find any option for the same.
>>>
>>> Plz advice.
>>>
>>> Thanks & regards
>>> Sharad garg
>>> -- 
>>> View this message in context:
>>> http://n2.nabble.com/Personal-Greeting-tp4226681p4226681.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> -- 
>> View this message in context:
>> http://old.nabble.com/Personal-Greeting-tp26951471p27073953.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

Correct..I also think, dialplan wont help me.

sharad

-- 
View this message in context: http://n2.nabble.com/Personal-Greeting-tp4226681p4284196.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From steve at justfone.com  Mon Jan 11 01:10:21 2010
From: steve at justfone.com (Steven Brown)
Date: Mon, 11 Jan 2010 09:10:21 +0000
Subject: [Freeswitch-users] RE Help with Portech <-> Freeswitch (Max
	Bridgewater)
Message-ID: <3e6d7b0c1001110110k474ba234lfc91210735e3a63d@mail.gmail.com>

Hi,

Two ways I've done it (not to say these are the 'correct' ways but
they do work for me after a lot of trial and error) are either to get
the Portech to register on FS as a regular endpoint then  bridge to it
as below

<extension name="gsm">
      <condition field="destination_number" expression="^9(.*)$">
	<action application="set" data="hangup_after_bridge=true"/>
        <action application="bridge" data="sofia/internal/1005^$1@${domain}">

where the Portech registers as 1005 and I prefix the number I wish to
dial with 9

or alternatively just bridge directly to it

<action application="bridge" data="sofia/default/$1 at 192.168.1.3:5060"/>

Where 192.168.1.3 is my Portech

In both cases you need to configure the Portech Lan to Mobile Table as below

Item  URL                 Call Num
0 	192.168.1.2 	#

where 192.168.1.2 is my FS box

This should get things going for one GSM channel, to get multiple
channels running you will have to tweak it differently depending on
the age of the Portech as I've discovered different firmware versions
seem to behave very differently in this respect.

Hope this helps

Steve




> Message: 1
> Date: Sat, 9 Jan 2010 12:00:54 -0500
> From: Max Bridgewater <max.bridgewater at gmail.com>
> Subject: [Freeswitch-users] Help with Portech <-> Freeswitch
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
> ? ? ? ?<c55d46951001090900p36df42e3ga91e61812d5b8d03 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Guys,
>
> It appears quite a few people in the list are using Portech. Can you please
> help me connect Freeswitch to ?it for termination puposes?
>
> Here is what I've done so far but without success.
>
> In Freeswitch I created a profile and stored it in under
> /usr/local/freeswitch/conf/sip_profiles/external/. Here is the content:
>
> <include>
> ?<gateway name="portech">
> ? ?<param name="username" value="noauthentication" />
> ? ?<param name="password" value="noauthentication" />
> ? ?<param name="proxy" value="74.24.22.59" />
> ? ?<param name="realm" value="74.24.22.59" />
> ? ?<param name="register" value="false" />
> ?</gateway>
> </include>
>
> Then, in the /usr/local/freeswitch/confi/dialplan/default.xml, I added the
> following extension:
>
> ? <extension name="portech">
> ? ? ?<condition field="destination_number" expression="^2801$">
> ? ? ? ?<action application="bridge"
> data="sofia/gateway/portech/5147237479"/>
> ? ? ?</condition>
> ? ?</extension>
>
> In Portech MV374, what I did is simply adding one entry in the Mobile/Lan to
> mobile table ?that consists of ?URL: 74.24.22.59 and call Num: #.
>
> Now, when I connect to Freeswitch with Xlite and try to dial extension 2801
> it simply says User Busy. i don't even see that attempts are being made to
> connect to the Portech gateway.
>
> Any idea?
> Thanks in advance.
>
> Max.



From oscav at hotmail.fr  Mon Jan 11 01:43:05 2010
From: oscav at hotmail.fr (Oscav)
Date: Mon, 11 Jan 2010 01:43:05 -0800 (PST)
Subject: [Freeswitch-users]  URGENT : DTMF during bridge
Message-ID: <27107895.post@talk.nabble.com>


Hi,

I need to handle DTMF during bridge in order to hangup the called party on
caller request. The DTMF sequence should be ##. Any idea on how to do that??

Thanks.
-- 
View this message in context: http://old.nabble.com/URGENT-%3A-DTMF-during-bridge-tp27107895p27107895.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From mcampbellsmith at gmail.com  Mon Jan 11 03:16:33 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 11 Jan 2010 22:16:33 +1100
Subject: [Freeswitch-users] Bypass_media mode
Message-ID: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>

Hi!
Hi!

I am calling from extension 2010 to extension 1000.  Both have ip
addesses 192.168.1.x.

In the 2000 series dialplan (a separate context) I have the following
to try to enable bypass_media.

            <extension name="Local_1000_9xxx">
                  <condition field="destination_number"
expression="^(10[01][0-9]|9\d{3})$">
                        <action application="set" data="dialed_extension=$1"/>
                        <action application="set" data="proxy_media=false"/>
                        <action application="set" data="bypass_media=true"/>
                        <action application="transfer"
data="${dialed_extension} XML default"/>
                  </condition>
            </extension>

Is this how bypass media should be enabled?

This fails fo me (the calls hang up and no audio).  The debug trace is
in http://pastebin.freeswitch.org/11737

What have I done wrong?

Thanks



From r.mokhtarpour at yahoo.com  Mon Jan 11 01:57:43 2010
From: r.mokhtarpour at yahoo.com (reza mokhtarpour)
Date: Mon, 11 Jan 2010 01:57:43 -0800 (PST)
Subject: [Freeswitch-users] gtalk and g723 codec
Message-ID: <769338.99062.qm@web33205.mail.mud.yahoo.com>

Hi there

I am begginer with FS , I use FS plus Gtalk?? everything is OK with PCMU codec but whenever? i replace it whit G723 codec i got "This codec is only usable in passthrough mode!" error.

SIP GW? >? FreeSwitch? >? Gtalk

these are my configuration files :

------- public.xml ----------------

<extension name="Gtalk">
??<condition field="source" expression="mod_sofia"/>
??<condition field="destination_number" expression="^1234$">
?<!-- <action application="set" data="proxy_media=true"/> -->
? <action application="set" data="bypass_media=true"/>
????<action application="bridge" data="dingaling/gmail.com/xxxxx at gmail.com"/>
??</condition>
</extension>

------------? dingaling.conf.xml ------------

<configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
? <settings>
??? <param name="debug" value="1"/>
??? <param name="codec-prefs" value="G723"/>
? </settings>

? <X-PRE-PROCESS cmd="include" data="../jingle_profiles/*.xml"/>

</configuration>

---------- sip_profiles/external.xml ---------------

<!-- 
?<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
?<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
-->
<param name="inbound-codec-prefs" value="G723"/>
? <param name="outbound-codec-prefs" value="G723"/>


I was googling for a while but I got nothing.
Thanks in advance.





      
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From a.alalousi at gmail.com  Mon Jan 11 04:18:11 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Mon, 11 Jan 2010 12:18:11 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
Message-ID: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>

Dear All,

I posted a thread re the subject but didn't get any joy, so perhaps second
time lucky.

I need to rewrite a couple of hangup causes to mean NORMAL_CONGESTION and no
matter which variables I set, this isn't happening. The idea is to control
what codes are returned to an end point after a successful bridge, as well
as deal with what codes are returned if the bridge is unsuccessful (e.g.
user_busy, originator_cancel ...etc).

I've had limited success by setting hangup_after_bridge=false then bridging
to error/<required_code>. This, however only works when the B-leg terminates
the call after a successful answer. Any other codes are not rewritten.

I've also tried playing with the bridge_hangup_code and hangup_code
variables prior and after bridging, still no joy. I have also set
sip_ignore_remote_cause=true prior to entering the bridge, as well
explicitly in vars.xml.

By the way, I'm running in proxy-media mode, but I did try it with
bypass-media as well. Same symptoms, same behaviour.

Any help with this would be highly appreciated.

Regards,

Ahmed.
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From john_re at fastmail.us  Mon Jan 11 04:21:57 2010
From: john_re at fastmail.us (giovanni_re)
Date: Mon, 11 Jan 2010 04:21:57 -0800
Subject: [Freeswitch-users] TONIGHT Join 5-6P Mon 11th - 1st Evening Meeting
 test IRC & VOIP online freeswitch at BerkeleyTIP-Global - for forwarding
Message-ID: <1263212517.5989.1354001243@webmail.messagingengine.com>

You're invited to the first test of the Global freeswitch bimonthly
evening meetings at BerkeleyTIP-Global.  :)

Join in tonight, Monday Jan 11, 5-6P Pacific, 8-9P Eastern, = Tues Jan
12 1A-2A UTC.
http://sites.google.com/site/berkeleytip/schedule

On #berkeleytip on irc.freenode.net,
& on voip - whatever is working - try btip server first.
http://sites.google.com/site/berkeleytip/remote-attendance

This will be an online only meeting - no in person meeting at UCB.

Hot topics:  Community Leadership Summit review of interesting sessions,
Spring 2010 efforts for UCB & all UC's & all college activities,
Upcoming KDE conference end of next week, for 1 week, in Los Angeles.

What do _you_ want to discuss?

==
Some people have asked for an evening meeting, because: a) they can't
make weekend meetings, b) they want more BTIP-Global. ;)

So, this will be a test, everyone invited, to see if we can make this
work.

==  BerkeleyTIP-Global is the Global All Free SW HW & Culture meeting
online via VOIP.
http://sites.google.com/site/berkeleytip/

Join the global mailing list, say "hi", & what you're interested in. :)
http://groups.google.com/group/BerkTIPGlobal

For Forwarding: You are invited to forward this announcement wherever it
might be appreciated.



From devel at thom.fr.eu.org  Mon Jan 11 04:53:45 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Mon, 11 Jan 2010 13:53:45 +0100
Subject: [Freeswitch-users] Sending call notification tone to a busy FXS port
Message-ID: <08abf4fa1e11a07bda4381fda8f93879@thom.fr.eu.org>

Hello,

I was just wondering if it is possible (and how) to send a call
notification tone to a phone connected to an FXS port and which is already
in communication.

Thanks

Fran?ois



From a.alalousi at gmail.com  Mon Jan 11 05:15:01 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Mon, 11 Jan 2010 13:15:01 +0000
Subject: [Freeswitch-users] Codecs and things
In-Reply-To: <b39273530912221645w8897121lf2938f59f541df8f@mail.gmail.com>
References: <cd93b3700912221255ve466667qb6e40961dbd05d99@mail.gmail.com>
	<b39273530912221645w8897121lf2938f59f541df8f@mail.gmail.com>
Message-ID: <cd93b3701001110515o47da76ffgc695222f3c4c6b97@mail.gmail.com>

Hi Rupa,

Thanks for your feedback.

One thing that caught my eye in your reply is you mentioning that proxy
media is a special hack for T38. The reason I'm using proxy mode is to fully
mask the identity of end points from each other. If there is another of
achieving this in FS then I am definitely very interested in hearing about
it and implementing it.

You also mention regular mode, and I couldn't find mention of this anywhere
in the Wiki. So far, my understanding was that you could run FS in either
proxy or bypass media modes. Can you elaborate a little bit and discuss this
a bit more ?

Thanks a lot in advance.

Regards,

Ahmed.


2009/12/23 Rupa Schomaker <rupa at rupa.com>

> On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:
> > Hello people,
> >
> > Can someone please clear the following ambiguities with codecs:
> >
> > Are we definitively able to run pass-through codecs (e.g. G.729) in Proxy
> > Media mode, or does FS need to be running in bypass-media ? the Wiki is
> not
> > clear in this regard
>
> Yes, you can use proxy media, bypass media, or even regular mode if
> you don't transcode (special for g729).  Proxy media is really a
> special hack that should only be used for T38 passthrough.  If you are
> using it for other purposes, think about it some more....
>
> > When an A-leg has negotiated a pass-through media codec, can the B-leg be
> > transcoded into a non-pass-through codec, and vice-versa ? think A-leg
> > incoming with a G.729 codec, and target for B-leg needs to be setup with
> a
> > GSM-codec, say
>
> That would require transcoding - which can't be done if the codec is
> pass-through.
>
> > Where in the developer's set of documentation are codecs discussed ? I
> would
> > like to start porting some code of mine for G.729a/b/ab form a ti DSP
> > platform to FS. FS lacking full G.729 support is proving quite a
> hindrance,
> > and there is no clear direction from the dev community as to when the
> same
> > will be available. Incidentally, any news on this effort ? where are we
> with
> > code, and what's an ETA for a Beta ?
>
> I'd say look at the broadvoice or other simple self-contained codecs
> are done.  Currently the only supported g729 solution is to use a
> digium board with mod_dahdi_codec.
>
> I don't have any info on a software based g729 solution.
>
> > On the same lines as (3) above, there is a codec dev template in the
> source
> > tree. Again, where can I find documentation relating to this ? the
> template
> > has hardly any docs at all.
> >
> > Best regards and warm wishes for a Merry Christmas and a great New Year
> to
> > one and all.
> >
> > Ahmed.
> >
> >
> > --
> > Ahmed A. Ibrahim-Naji Al-Alousi
> > Ph.D., MIEE, MBCS
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From rupa at rupa.com  Mon Jan 11 06:18:40 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 11 Jan 2010 08:18:40 -0600
Subject: [Freeswitch-users] Avoiding restart of transfer_ringback file
	during successive bridge actions.
In-Reply-To: <15b9404e1001102100u157db096t636f06c0fa774291@mail.gmail.com>
References: <15b9404e1001102100u157db096t636f06c0fa774291@mail.gmail.com>
Message-ID: <b39273531001110618r7d3df336i1cd9fb5d13a9ee7d@mail.gmail.com>

Maybe set a var that, when set, causes the dp to skip restarting the
ringback?

On Sun, Jan 10, 2010 at 11:00 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

> Hi,
> I'm implementing ACD functionality using a dialplan.
> Basically, I prepare a list of members in a group and try each one of them
> in successive bridge actions.
> I cannot use a single bridge action with all members separated with pipes
> because I am required to check some conditions like if the member is already
> in another call. So what I do is to add a transfer action after the bridge
> so that the call reenters the dialplan with the remaining of the list.
> Up to this point things are fine.
> However, if the call is answered and transferred to the group, I set
> transfer_ringback so that I can play a file while the bridge is happening.
> In case transfer_ringback is set to something like "local_stream://moh",
> then we hear a small glitch every time a bridge action happens as the bridge
> doesn't take into account that the ringback was already set up by a previous
> operation and sets up the playfile operation again. But this is also fine.
> The problem is in case transfer_ringback is set to an actual file and not
> local_stream. In this case, every bridge action will cause restart of the
> playback from the beginning.
> So, is it possible to ask bridge to do not setup playback if it is already
> running referencing the same file?
>
> regards,
> takeshi
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From rupa at rupa.com  Mon Jan 11 06:23:14 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 11 Jan 2010 08:23:14 -0600
Subject: [Freeswitch-users] Codecs and things
In-Reply-To: <cd93b3701001110515o47da76ffgc695222f3c4c6b97@mail.gmail.com>
References: <cd93b3700912221255ve466667qb6e40961dbd05d99@mail.gmail.com>
	<b39273530912221645w8897121lf2938f59f541df8f@mail.gmail.com>
	<cd93b3701001110515o47da76ffgc695222f3c4c6b97@mail.gmail.com>
Message-ID: <b39273531001110623q4795d6a9i2bc8bfa2760ba99a@mail.gmail.com>

On Mon, Jan 11, 2010 at 7:15 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:

> Hi Rupa,
>
> Thanks for your feedback.
>
> One thing that caught my eye in your reply is you mentioning that proxy
> media is a special hack for T38. The reason I'm using proxy mode is to fully
> mask the identity of end points from each other. If there is another of
> achieving this in FS then I am definitely very interested in hearing about
> it and implementing it.
>
>
The stock answer is that FS is not a proxy.  If you want a proxy use proxy
software (opensips/kamilio/whatever).  Proxy media mode is generally not the
right answer.


> You also mention regular mode, and I couldn't find mention of this anywhere
> in the Wiki. So far, my understanding was that you could run FS in either
> proxy or bypass media modes. Can you elaborate a little bit and discuss this
> a bit more ?
>
>
Regular mode is just the default mode FS runs in where all media passes
through it.


> Thanks a lot in advance.
>
> Regards,
>
> Ahmed.
>
>
> 2009/12/23 Rupa Schomaker <rupa at rupa.com>
>
> On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>> > Hello people,
>> >
>> > Can someone please clear the following ambiguities with codecs:
>> >
>> > Are we definitively able to run pass-through codecs (e.g. G.729) in
>> Proxy
>> > Media mode, or does FS need to be running in bypass-media ? the Wiki is
>> not
>> > clear in this regard
>>
>> Yes, you can use proxy media, bypass media, or even regular mode if
>> you don't transcode (special for g729).  Proxy media is really a
>> special hack that should only be used for T38 passthrough.  If you are
>> using it for other purposes, think about it some more....
>>
>> > When an A-leg has negotiated a pass-through media codec, can the B-leg
>> be
>> > transcoded into a non-pass-through codec, and vice-versa ? think A-leg
>> > incoming with a G.729 codec, and target for B-leg needs to be setup with
>> a
>> > GSM-codec, say
>>
>> That would require transcoding - which can't be done if the codec is
>> pass-through.
>>
>> > Where in the developer's set of documentation are codecs discussed ? I
>> would
>> > like to start porting some code of mine for G.729a/b/ab form a ti DSP
>> > platform to FS. FS lacking full G.729 support is proving quite a
>> hindrance,
>> > and there is no clear direction from the dev community as to when the
>> same
>> > will be available. Incidentally, any news on this effort ? where are we
>> with
>> > code, and what's an ETA for a Beta ?
>>
>> I'd say look at the broadvoice or other simple self-contained codecs
>> are done.  Currently the only supported g729 solution is to use a
>> digium board with mod_dahdi_codec.
>>
>> I don't have any info on a software based g729 solution.
>>
>> > On the same lines as (3) above, there is a codec dev template in the
>> source
>> > tree. Again, where can I find documentation relating to this ? the
>> template
>> > has hardly any docs at all.
>> >
>> > Best regards and warm wishes for a Merry Christmas and a great New Year
>> to
>> > one and all.
>> >
>> > Ahmed.
>> >
>> >
>> > --
>> > Ahmed A. Ibrahim-Naji Al-Alousi
>> > Ph.D., MIEE, MBCS
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From a.alalousi at gmail.com  Mon Jan 11 06:39:20 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Mon, 11 Jan 2010 14:39:20 +0000
Subject: [Freeswitch-users] Codecs and things
In-Reply-To: <b39273531001110623q4795d6a9i2bc8bfa2760ba99a@mail.gmail.com>
References: <cd93b3700912221255ve466667qb6e40961dbd05d99@mail.gmail.com>
	<b39273530912221645w8897121lf2938f59f541df8f@mail.gmail.com>
	<cd93b3701001110515o47da76ffgc695222f3c4c6b97@mail.gmail.com>
	<b39273531001110623q4795d6a9i2bc8bfa2760ba99a@mail.gmail.com>
Message-ID: <cd93b3701001110639t8b222d1i344ae445fb86f501@mail.gmail.com>

Hi Rupa,

Thanks for this.

Re: regular mode, are we saying to set both bypass-media and proxy-media to
false, and this would put it into regular mode ?

I'll look into alternatives re: proxy per your feedback.

Regards,

Ahmed.


2010/1/11 Rupa Schomaker <rupa at rupa.com>

>
>
> On Mon, Jan 11, 2010 at 7:15 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>
>> Hi Rupa,
>>
>> Thanks for your feedback.
>>
>> One thing that caught my eye in your reply is you mentioning that proxy
>> media is a special hack for T38. The reason I'm using proxy mode is to fully
>> mask the identity of end points from each other. If there is another of
>> achieving this in FS then I am definitely very interested in hearing about
>> it and implementing it.
>>
>>
> The stock answer is that FS is not a proxy.  If you want a proxy use proxy
> software (opensips/kamilio/whatever).  Proxy media mode is generally not the
> right answer.
>
>
>> You also mention regular mode, and I couldn't find mention of this
>> anywhere in the Wiki. So far, my understanding was that you could run FS in
>> either proxy or bypass media modes. Can you elaborate a little bit and
>> discuss this a bit more ?
>>
>>
> Regular mode is just the default mode FS runs in where all media passes
> through it.
>
>
>> Thanks a lot in advance.
>>
>> Regards,
>>
>> Ahmed.
>>
>>
>> 2009/12/23 Rupa Schomaker <rupa at rupa.com>
>>
>> On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>>> > Hello people,
>>> >
>>> > Can someone please clear the following ambiguities with codecs:
>>> >
>>> > Are we definitively able to run pass-through codecs (e.g. G.729) in
>>> Proxy
>>> > Media mode, or does FS need to be running in bypass-media ? the Wiki is
>>> not
>>> > clear in this regard
>>>
>>> Yes, you can use proxy media, bypass media, or even regular mode if
>>> you don't transcode (special for g729).  Proxy media is really a
>>> special hack that should only be used for T38 passthrough.  If you are
>>> using it for other purposes, think about it some more....
>>>
>>> > When an A-leg has negotiated a pass-through media codec, can the B-leg
>>> be
>>> > transcoded into a non-pass-through codec, and vice-versa ? think A-leg
>>> > incoming with a G.729 codec, and target for B-leg needs to be setup
>>> with a
>>> > GSM-codec, say
>>>
>>> That would require transcoding - which can't be done if the codec is
>>> pass-through.
>>>
>>> > Where in the developer's set of documentation are codecs discussed ? I
>>> would
>>> > like to start porting some code of mine for G.729a/b/ab form a ti DSP
>>> > platform to FS. FS lacking full G.729 support is proving quite a
>>> hindrance,
>>> > and there is no clear direction from the dev community as to when the
>>> same
>>> > will be available. Incidentally, any news on this effort ? where are we
>>> with
>>> > code, and what's an ETA for a Beta ?
>>>
>>> I'd say look at the broadvoice or other simple self-contained codecs
>>> are done.  Currently the only supported g729 solution is to use a
>>> digium board with mod_dahdi_codec.
>>>
>>> I don't have any info on a software based g729 solution.
>>>
>>> > On the same lines as (3) above, there is a codec dev template in the
>>> source
>>> > tree. Again, where can I find documentation relating to this ? the
>>> template
>>> > has hardly any docs at all.
>>> >
>>> > Best regards and warm wishes for a Merry Christmas and a great New Year
>>> to
>>> > one and all.
>>> >
>>> > Ahmed.
>>> >
>>> >
>>> > --
>>> > Ahmed A. Ibrahim-Naji Al-Alousi
>>> > Ph.D., MIEE, MBCS
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Ahmed Naji
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From sos at sokhapkin.dyndns.org  Mon Jan 11 06:48:49 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 11 Jan 2010 09:48:49 -0500
Subject: [Freeswitch-users] Codecs and things
In-Reply-To: <cd93b3701001110639t8b222d1i344ae445fb86f501@mail.gmail.com>
References: <cd93b3700912221255ve466667qb6e40961dbd05d99@mail.gmail.com>
	<b39273531001110623q4795d6a9i2bc8bfa2760ba99a@mail.gmail.com>
	<cd93b3701001110639t8b222d1i344ae445fb86f501@mail.gmail.com>
Message-ID: <201001110948.49710.sos@sokhapkin.dyndns.org>

Regular mode is when neither variable is set or set to false.

On Monday 11 January 2010, Ahmed Naji wrote:
> Hi Rupa,
>
> Thanks for this.
>
> Re: regular mode, are we saying to set both bypass-media and proxy-media to
> false, and this would put it into regular mode ?
>
> I'll look into alternatives re: proxy per your feedback.
>
> Regards,
>
> Ahmed.
>
>
> 2010/1/11 Rupa Schomaker <rupa at rupa.com>
>
> > On Mon, Jan 11, 2010 at 7:15 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:
> >> Hi Rupa,
> >>
> >> Thanks for your feedback.
> >>
> >> One thing that caught my eye in your reply is you mentioning that proxy
> >> media is a special hack for T38. The reason I'm using proxy mode is to
> >> fully mask the identity of end points from each other. If there is
> >> another of achieving this in FS then I am definitely very interested in
> >> hearing about it and implementing it.
> >
> > The stock answer is that FS is not a proxy.  If you want a proxy use
> > proxy software (opensips/kamilio/whatever).  Proxy media mode is
> > generally not the right answer.
> >
> >> You also mention regular mode, and I couldn't find mention of this
> >> anywhere in the Wiki. So far, my understanding was that you could run FS
> >> in either proxy or bypass media modes. Can you elaborate a little bit
> >> and discuss this a bit more ?
> >
> > Regular mode is just the default mode FS runs in where all media passes
> > through it.
> >
> >> Thanks a lot in advance.
> >>
> >> Regards,
> >>
> >> Ahmed.
> >>
> >>
> >> 2009/12/23 Rupa Schomaker <rupa at rupa.com>
> >>
> >> On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:
> >>> > Hello people,
> >>> >
> >>> > Can someone please clear the following ambiguities with codecs:
> >>> >
> >>> > Are we definitively able to run pass-through codecs (e.g. G.729) in
> >>>
> >>> Proxy
> >>>
> >>> > Media mode, or does FS need to be running in bypass-media ? the Wiki
> >>> > is
> >>>
> >>> not
> >>>
> >>> > clear in this regard
> >>>
> >>> Yes, you can use proxy media, bypass media, or even regular mode if
> >>> you don't transcode (special for g729).  Proxy media is really a
> >>> special hack that should only be used for T38 passthrough.  If you are
> >>> using it for other purposes, think about it some more....
> >>>
> >>> > When an A-leg has negotiated a pass-through media codec, can the
> >>> > B-leg
> >>>
> >>> be
> >>>
> >>> > transcoded into a non-pass-through codec, and vice-versa ? think
> >>> > A-leg incoming with a G.729 codec, and target for B-leg needs to be
> >>> > setup
> >>>
> >>> with a
> >>>
> >>> > GSM-codec, say
> >>>
> >>> That would require transcoding - which can't be done if the codec is
> >>> pass-through.
> >>>
> >>> > Where in the developer's set of documentation are codecs discussed ?
> >>> > I
> >>>
> >>> would
> >>>
> >>> > like to start porting some code of mine for G.729a/b/ab form a ti DSP
> >>> > platform to FS. FS lacking full G.729 support is proving quite a
> >>>
> >>> hindrance,
> >>>
> >>> > and there is no clear direction from the dev community as to when the
> >>>
> >>> same
> >>>
> >>> > will be available. Incidentally, any news on this effort ? where are
> >>> > we
> >>>
> >>> with
> >>>
> >>> > code, and what's an ETA for a Beta ?
> >>>
> >>> I'd say look at the broadvoice or other simple self-contained codecs
> >>> are done.  Currently the only supported g729 solution is to use a
> >>> digium board with mod_dahdi_codec.
> >>>
> >>> I don't have any info on a software based g729 solution.
> >>>
> >>> > On the same lines as (3) above, there is a codec dev template in the
> >>>
> >>> source
> >>>
> >>> > tree. Again, where can I find documentation relating to this ? the
> >>>
> >>> template
> >>>
> >>> > has hardly any docs at all.
> >>> >
> >>> > Best regards and warm wishes for a Merry Christmas and a great New
> >>> > Year
> >>>
> >>> to
> >>>
> >>> > one and all.
> >>> >
> >>> > Ahmed.
> >>> >
> >>> >
> >>> > --
> >>> > Ahmed A. Ibrahim-Naji Al-Alousi
> >>> > Ph.D., MIEE, MBCS
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> >>>
> >>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>
> >>> > http://www.freeswitch.org
> >>>
> >>> --
> >>> -Rupa
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> >>>s http://www.freeswitch.org
> >>
> >> --
> >> Ahmed Naji
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > -Rupa
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From rupa at rupa.com  Mon Jan 11 06:52:22 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 11 Jan 2010 08:52:22 -0600
Subject: [Freeswitch-users] Codecs and things
In-Reply-To: <cd93b3701001110639t8b222d1i344ae445fb86f501@mail.gmail.com>
References: <cd93b3700912221255ve466667qb6e40961dbd05d99@mail.gmail.com>
	<b39273530912221645w8897121lf2938f59f541df8f@mail.gmail.com>
	<cd93b3701001110515o47da76ffgc695222f3c4c6b97@mail.gmail.com>
	<b39273531001110623q4795d6a9i2bc8bfa2760ba99a@mail.gmail.com>
	<cd93b3701001110639t8b222d1i344ae445fb86f501@mail.gmail.com>
Message-ID: <b39273531001110652k6920cdb4k48f9729f5fa5f13e@mail.gmail.com>

On Mon, Jan 11, 2010 at 8:39 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:

> Hi Rupa,
>
> Thanks for this.
>
> Re: regular mode, are we saying to set both bypass-media and proxy-media to
> false, and this would put it into regular mode ?
>
>
Or just don't set either.  Default is regular mode (why I called it regular
mode).  I'm not sure what the proper term is, "media mode" maybe ?


> I'll look into alternatives re: proxy per your feedback.
>
> Regards,
>
> Ahmed.
>
>
> 2010/1/11 Rupa Schomaker <rupa at rupa.com>
>
>
>>
>> On Mon, Jan 11, 2010 at 7:15 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>>
>>> Hi Rupa,
>>>
>>> Thanks for your feedback.
>>>
>>> One thing that caught my eye in your reply is you mentioning that proxy
>>> media is a special hack for T38. The reason I'm using proxy mode is to fully
>>> mask the identity of end points from each other. If there is another of
>>> achieving this in FS then I am definitely very interested in hearing about
>>> it and implementing it.
>>>
>>>
>> The stock answer is that FS is not a proxy.  If you want a proxy use proxy
>> software (opensips/kamilio/whatever).  Proxy media mode is generally not the
>> right answer.
>>
>>
>>> You also mention regular mode, and I couldn't find mention of this
>>> anywhere in the Wiki. So far, my understanding was that you could run FS in
>>> either proxy or bypass media modes. Can you elaborate a little bit and
>>> discuss this a bit more ?
>>>
>>>
>> Regular mode is just the default mode FS runs in where all media passes
>> through it.
>>
>>
>>> Thanks a lot in advance.
>>>
>>> Regards,
>>>
>>> Ahmed.
>>>
>>>
>>> 2009/12/23 Rupa Schomaker <rupa at rupa.com>
>>>
>>> On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji <a.alalousi at gmail.com>
>>>> wrote:
>>>> > Hello people,
>>>> >
>>>> > Can someone please clear the following ambiguities with codecs:
>>>> >
>>>> > Are we definitively able to run pass-through codecs (e.g. G.729) in
>>>> Proxy
>>>> > Media mode, or does FS need to be running in bypass-media ? the Wiki
>>>> is not
>>>> > clear in this regard
>>>>
>>>> Yes, you can use proxy media, bypass media, or even regular mode if
>>>> you don't transcode (special for g729).  Proxy media is really a
>>>> special hack that should only be used for T38 passthrough.  If you are
>>>> using it for other purposes, think about it some more....
>>>>
>>>> > When an A-leg has negotiated a pass-through media codec, can the B-leg
>>>> be
>>>> > transcoded into a non-pass-through codec, and vice-versa ? think A-leg
>>>> > incoming with a G.729 codec, and target for B-leg needs to be setup
>>>> with a
>>>> > GSM-codec, say
>>>>
>>>> That would require transcoding - which can't be done if the codec is
>>>> pass-through.
>>>>
>>>> > Where in the developer's set of documentation are codecs discussed ? I
>>>> would
>>>> > like to start porting some code of mine for G.729a/b/ab form a ti DSP
>>>> > platform to FS. FS lacking full G.729 support is proving quite a
>>>> hindrance,
>>>> > and there is no clear direction from the dev community as to when the
>>>> same
>>>> > will be available. Incidentally, any news on this effort ? where are
>>>> we with
>>>> > code, and what's an ETA for a Beta ?
>>>>
>>>> I'd say look at the broadvoice or other simple self-contained codecs
>>>> are done.  Currently the only supported g729 solution is to use a
>>>> digium board with mod_dahdi_codec.
>>>>
>>>> I don't have any info on a software based g729 solution.
>>>>
>>>> > On the same lines as (3) above, there is a codec dev template in the
>>>> source
>>>> > tree. Again, where can I find documentation relating to this ? the
>>>> template
>>>> > has hardly any docs at all.
>>>> >
>>>> > Best regards and warm wishes for a Merry Christmas and a great New
>>>> Year to
>>>> > one and all.
>>>> >
>>>> > Ahmed.
>>>> >
>>>> >
>>>> > --
>>>> > Ahmed A. Ibrahim-Naji Al-Alousi
>>>> > Ph.D., MIEE, MBCS
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Ahmed Naji
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From brian at freeswitch.org  Mon Jan 11 07:01:18 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Jan 2010 09:01:18 -0600
Subject: [Freeswitch-users] gtalk and g723 codec
In-Reply-To: <769338.99062.qm@web33205.mail.mud.yahoo.com>
References: <769338.99062.qm@web33205.mail.mud.yahoo.com>
Message-ID: <A8CA30A1-555D-4145-AC23-D468E21CE54A@freeswitch.org>

You can't use G723, Its only a passthru codec.

/b

On Jan 11, 2010, at 3:57 AM, reza mokhtarpour wrote:

> Hi there
> 
> I am begginer with FS , I use FS plus Gtalk   everything is OK with PCMU codec but whenever  i replace it whit G723 codec i got "This codec is only usable in passthrough mode!" error.
> 
> SIP GW  >  FreeSwitch  >  Gtalk
> 
> these are my configuration files :

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From ibrahimtunali at gmail.com  Mon Jan 11 07:43:59 2010
From: ibrahimtunali at gmail.com (itunali)
Date: Mon, 11 Jan 2010 07:43:59 -0800 (PST)
Subject: [Freeswitch-users]  Performance issue on 1.0.4 vs 1.0.5
Message-ID: <27112490.post@talk.nabble.com>


Hi,

I did performance tests to measure that freeswitch limits. The test just
dial echo extension 9996 at default context and wait 6 sec then hangup.

I used sipp test tool and set all variables/environment as described on
(http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations)

I reached 80 cps on 1.0.4 (install from freeswitch-drivers ppa) but not get
similar rate on SVN trunk build (1.5 cps) and get many retransmission SIP
packets.

I build SVN trunk code with default ./bootstrap.sh && ./configure && make &&
make install process.

My server specs;
Ubuntu 9.10 Karmic 
Linux 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 17:01:44 UTC 2009 x86_64
GNU/Linux

Is there any .deb packets for 1.0.5?

Regards,
Ibrahim
-- 
View this message in context: http://old.nabble.com/Performance-issue-on-1.0.4-vs-1.0.5-tp27112490p27112490.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From jcasale at activenetwerx.com  Mon Jan 11 08:15:48 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Mon, 11 Jan 2010 16:15:48 +0000
Subject: [Freeswitch-users] Outbound call problem
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9076F42@prato.activenetwerx.local>

Likely an issue with my SPA3102, but when I route a call
to its FXO port, I can almost faintly hear the operator
if its misdialed, but otherwise the connection is loaded
with feedback and static.

Anyone have a suggestion on where to start looking? Inbound
from that FXO port is flawless.

Thanks,
jlc



From jerry.richards at teotech.com  Mon Jan 11 08:48:48 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 11 Jan 2010 08:48:48 -0800
Subject: [Freeswitch-users] http://jira.freeswitch.org/browse/FSCORE-262
In-Reply-To: <5817BC3F-C091-42E3-AAAE-2CF4A21A9531@freeswitch.org>
References: <39ED1D5111554D33AF502370BC34D288@greyhawk.tonecommander.com>
	<5817BC3F-C091-42E3-AAAE-2CF4A21A9531@freeswitch.org>
Message-ID: <D266691DAA854ABD9B266AC8187802AE@greyhawk.tonecommander.com>

No I don't have a patch, but I suspect it might be a sofia SIP stack issue.

Jerry
 

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Friday, January 08, 2010 11:44 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] http://jira.freeswitch.org/browse/FSCORE-262

Do you happen to have a patch for that?

/b

On Jan 8, 2010, at 1:38 PM, Jerry Richards wrote:

> Is there a plan to fix this JIRA issue:
> http://jira.freeswitch.org/browse/FSCORE-262
> 
> This is causing a problem in sharing presence data between FS and 
> another gateway.
> 
> Thanks,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org







From mike at jerris.com  Mon Jan 11 09:03:56 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 11 Jan 2010 12:03:56 -0500
Subject: [Freeswitch-users] http://jira.freeswitch.org/browse/FSCORE-262
In-Reply-To: <D266691DAA854ABD9B266AC8187802AE@greyhawk.tonecommander.com>
References: <39ED1D5111554D33AF502370BC34D288@greyhawk.tonecommander.com>
	<5817BC3F-C091-42E3-AAAE-2CF4A21A9531@freeswitch.org>
	<D266691DAA854ABD9B266AC8187802AE@greyhawk.tonecommander.com>
Message-ID: <4B646557-1ED8-4767-A87F-846EC071D64E@jerris.com>

as noted on the bug, please try r16218 

Mike

On Jan 11, 2010, at 11:48 AM, Jerry Richards wrote:

> No I don't have a patch, but I suspect it might be a sofia SIP stack issue.
> 
> Jerry
> 
> 
> -----Original Message-----
> From: Brian West [mailto:brian at freeswitch.org] 
> Sent: Friday, January 08, 2010 11:44 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] http://jira.freeswitch.org/browse/FSCORE-262
> 
> Do you happen to have a patch for that?
> 
> /b
> 
> On Jan 8, 2010, at 1:38 PM, Jerry Richards wrote:
> 
>> Is there a plan to fix this JIRA issue:
>> http://jira.freeswitch.org/browse/FSCORE-262
>> 
>> This is causing a problem in sharing presence data between FS and 
>> another gateway.
>> 
>> Thanks,
>> Jerry
>> 



From talk2ram at gmail.com  Mon Jan 11 09:17:10 2010
From: talk2ram at gmail.com (ram)
Date: Mon, 11 Jan 2010 22:47:10 +0530
Subject: [Freeswitch-users] Outbound call problem
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9076F42@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E9076F42@prato.activenetwerx.local>
Message-ID: <b74751491001110917n6709a93cua5dcdce67232ce7a@mail.gmail.com>

post the logs

ram

On Mon, Jan 11, 2010 at 9:45 PM, Joseph L. Casale <jcasale at activenetwerx.com
> wrote:

> Likely an issue with my SPA3102, but when I route a call
> to its FXO port, I can almost faintly hear the operator
> if its misdialed, but otherwise the connection is loaded
> with feedback and static.
>
> Anyone have a suggestion on where to start looking? Inbound
> from that FXO port is flawless.
>
> Thanks,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tayeb.meftah at gmail.com  Mon Jan 11 09:04:44 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Mon, 11 Jan 2010 18:04:44 +0100
Subject: [Freeswitch-users] Performance issue on 1.0.4 vs 1.0.5
In-Reply-To: <27112490.post@talk.nabble.com>
References: <27112490.post@talk.nabble.com>
Message-ID: <E712CC50028F4DA0B5A53289032A839A@PCdeuser>

    hi ibrahim,
please try  to apt-get update
and apt-get upgrade and return fidback
thanks
----- Original Message ----- 
From: "itunali" <ibrahimtunali at gmail.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, January 11, 2010 4:43 PM
Subject: [Freeswitch-users] Performance issue on 1.0.4 vs 1.0.5


>
> Hi,
>
> I did performance tests to measure that freeswitch limits. The test just
> dial echo extension 9996 at default context and wait 6 sec then hangup.
>
> I used sipp test tool and set all variables/environment as described on
> (http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations)
>
> I reached 80 cps on 1.0.4 (install from freeswitch-drivers ppa) but not 
> get
> similar rate on SVN trunk build (1.5 cps) and get many retransmission SIP
> packets.
>
> I build SVN trunk code with default ./bootstrap.sh && ./configure && make 
> &&
> make install process.
>
> My server specs;
> Ubuntu 9.10 Karmic
> Linux 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 17:01:44 UTC 2009 x86_64
> GNU/Linux
>
> Is there any .deb packets for 1.0.5?
>
> Regards,
> Ibrahim
> -- 
> View this message in context: 
> http://old.nabble.com/Performance-issue-on-1.0.4-vs-1.0.5-tp27112490p27112490.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From anatoliy at kounitskiy.com  Mon Jan 11 09:18:49 2010
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Mon, 11 Jan 2010 19:18:49 +0200
Subject: [Freeswitch-users] Problem between revision 16223 and 16242 on
	Debian (stable) 64bit
Message-ID: <1263230329.2504.33.camel@lenovor400-laptop>

Hello,
I just made a checkout of the svn and tried to configure it, but there
is an error in the arp-util lib, after the ./bootstrap.sh


Freeswitch revision: 16242
OS: Debian 64b (stable)
Command used: ./configure --prefix=/usr/local/freeswitch
--enable-optimization --enable-64
Error:
checking for Expat in xml/expat... yes
configuring package in xml/expat now
configure: error: expected an absolute directory name for --bindir:
NONE/bin
configure failed for xml/expat
configure: error: ./configure.gnu failed for libs/apr-util

If I execute "sh configure.gnu"/"sh configure.gnu --enable-64" in
libs/apr-util it goes without an error.

On the same server with Freeswitch revision: 16223 (with the same
configure command), it goes as planned - without errors.

Regards,

-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540




From brian at freeswitch.org  Mon Jan 11 09:20:48 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Jan 2010 11:20:48 -0600
Subject: [Freeswitch-users] Performance issue on 1.0.4 vs 1.0.5
In-Reply-To: <E712CC50028F4DA0B5A53289032A839A@PCdeuser>
References: <27112490.post@talk.nabble.com>
	<E712CC50028F4DA0B5A53289032A839A@PCdeuser>
Message-ID: <B6E8FC8D-EB6F-4D70-B606-439CC8DFB8B9@freeswitch.org>

How about we compile from src?  

/b

On Jan 11, 2010, at 11:04 AM, Meftah Tayeb wrote:

>    hi ibrahim,
> please try  to apt-get update
> and apt-get upgrade and return fidback
> thanks




From andrew at hijacked.us  Mon Jan 11 10:12:03 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Mon, 11 Jan 2010 13:12:03 -0500
Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS
	based callcenter)
In-Reply-To: <855e4dcf1001041842u46c6d36q9c2e4ece2ced7380@mail.gmail.com>
References: <20091219014359.GA21798@hijacked.us>
	<855e4dcf1001041842u46c6d36q9c2e4ece2ced7380@mail.gmail.com>
Message-ID: <20100111181203.GD10774@hijacked.us>

On Tue, Jan 05, 2010 at 03:42:57PM +1300, Tim Uckun wrote:
> > http://opencsm.org/downloads/spice-telephony-0.9.6.tar.gz
> >
> 
> There seems to be something wrong with both opencsm.org and wiki.opencsm.org.
> 
> Just thought I'd let you know.
>

Yeah, my company wanted me to move it, so I re-hosted it as a github
project (with a new name):

http://github.com/Vagabond/OpenACD

I hadn't announced the change yet because I've been away for the last
week and didn't have time. The wiki contents haven't been moved over
yet, but they needed some cleanup anyway.

On the other hand attended transfer support finally materialized.

Andrew



From anatoliy at kounitskiy.com  Mon Jan 11 10:30:49 2010
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Mon, 11 Jan 2010 20:30:49 +0200
Subject: [Freeswitch-users] Problem between revision 16223 and 16242 on
 Debian (stable) 64bit
In-Reply-To: <1263230329.2504.33.camel@lenovor400-laptop>
References: <1263230329.2504.33.camel@lenovor400-laptop>
Message-ID: <1263234649.2504.35.camel@lenovor400-laptop>

Ok, i found in which revision it is broken.

Until revision 16237 - it works as charm
In Revision 16238 - it doesn't work

svn log --revision 16237:16238
------------------------------------------------------------------------
r16238 | mikej | 2010-01-11 16:36:29 +0200 (Mon, 11 Jan 2010) | 1 line

wip move towards adding directory layout control to configure
------------------------------------------------------------------------

Regards,

On Mon, 2010-01-11 at 19:18 +0200, Anatoliy Kounitskiy wrote:
> Hello,
> I just made a checkout of the svn and tried to configure it, but there
> is an error in the arp-util lib, after the ./bootstrap.sh
> 
> 
> Freeswitch revision: 16242
> OS: Debian 64b (stable)
> Command used: ./configure --prefix=/usr/local/freeswitch
> --enable-optimization --enable-64
> Error:
> checking for Expat in xml/expat... yes
> configuring package in xml/expat now
> configure: error: expected an absolute directory name for --bindir:
> NONE/bin
> configure failed for xml/expat
> configure: error: ./configure.gnu failed for libs/apr-util
> 
> If I execute "sh configure.gnu"/"sh configure.gnu --enable-64" in
> libs/apr-util it goes without an error.
> 
> On the same server with Freeswitch revision: 16223 (with the same
> configure command), it goes as planned - without errors.
> 
> Regards,
> 


-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540




From gabe at gundy.org  Mon Jan 11 11:05:29 2010
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 11 Jan 2010 12:05:29 -0700
Subject: [Freeswitch-users] fifo funk?
In-Reply-To: <23f91031001102149x1f1913amefa97d02981261d@mail.gmail.com>
References: <903da5681001102136y6d51f960hd8c10dde4e74ac26@mail.gmail.com>
	<23f91031001102149x1f1913amefa97d02981261d@mail.gmail.com>
Message-ID: <903da5681001111105j25e46e98n8e6cf095651c010f@mail.gmail.com>

On Sun, Jan 10, 2010 at 10:49 PM, Seven Du <dujinfang at gmail.com> wrote:
> see fifo_member add/delete api

I must not have been clear.  We *do* use the add and del api and find
that it works well (better than reparsing).

Gabe



From anthony.minessale at gmail.com  Mon Jan 11 11:13:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Jan 2010 13:13:07 -0600
Subject: [Freeswitch-users] fifo funk?
In-Reply-To: <903da5681001111105j25e46e98n8e6cf095651c010f@mail.gmail.com>
References: <903da5681001102136y6d51f960hd8c10dde4e74ac26@mail.gmail.com>
	<23f91031001102149x1f1913amefa97d02981261d@mail.gmail.com>
	<903da5681001111105j25e46e98n8e6cf095651c010f@mail.gmail.com>
Message-ID: <191c3a031001111113t4f61ff8t21c6c80bce35f53f@mail.gmail.com>

that is the best way.

The re-parse is a much more harsh operation designed for configuration
changes.


On Mon, Jan 11, 2010 at 1:05 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Sun, Jan 10, 2010 at 10:49 PM, Seven Du <dujinfang at gmail.com> wrote:
> > see fifo_member add/delete api
>
> I must not have been clear.  We *do* use the add and del api and find
> that it works well (better than reparsing).
>
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From r.mokhtarpour at yahoo.com  Mon Jan 11 12:04:02 2010
From: r.mokhtarpour at yahoo.com (reza mokhtarpour)
Date: Mon, 11 Jan 2010 12:04:02 -0800 (PST)
Subject: [Freeswitch-users] gtalk and g723 codec
In-Reply-To: <A8CA30A1-555D-4145-AC23-D468E21CE54A@freeswitch.org>
Message-ID: <702240.99335.qm@web33207.mail.mud.yahoo.com>

Sorry ,? I don't understand.

i want use it in this mode .

my SIP gateway originate call with g723 codec and gtalk support this codec? so these endpoints should can communicate with each other in pass through mode.

if it is not true please correct me.

--- On Mon, 1/11/10, Brian West <brian at freeswitch.org> wrote:

From: Brian West <brian at freeswitch.org>
Subject: Re: [Freeswitch-users] gtalk and g723 codec
To: freeswitch-users at lists.freeswitch.org
Date: Monday, January 11, 2010, 7:01 AM

You can't use G723, Its only a passthru codec.
/b
On Jan 11, 2010, at 3:57 AM, reza mokhtarpour wrote:
Hi there

I am begginer with FS , I use FS plus Gtalk?? everything is OK with PCMU codec but whenever? i replace it whit G723 codec i got "This codec is only usable in passthrough mode!" error.

SIP GW? >? FreeSwitch? >? Gtalk

these are my configuration files :

-----Inline Attachment Follows-----

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      
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From brian at freeswitch.org  Mon Jan 11 12:12:21 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Jan 2010 14:12:21 -0600
Subject: [Freeswitch-users] gtalk and g723 codec
In-Reply-To: <702240.99335.qm@web33207.mail.mud.yahoo.com>
References: <702240.99335.qm@web33207.mail.mud.yahoo.com>
Message-ID: <C76820EA-8DCD-4034-9CFC-D3FD4D6D07E8@freeswitch.org>

It might not work correctly in that configuration.

/b

On Jan 11, 2010, at 2:04 PM, reza mokhtarpour wrote:

> Sorry ,  I don't understand.
> 
> i want use it in this mode .
> 
> my SIP gateway originate call with g723 codec and gtalk support this codec  so these endpoints should can communicate with each other in pass through mode.
> 
> if it is not true please correct me.

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From anthony.minessale at gmail.com  Mon Jan 11 12:18:08 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Jan 2010 14:18:08 -0600
Subject: [Freeswitch-users] Performance issue on 1.0.4 vs 1.0.5
In-Reply-To: <27112490.post@talk.nabble.com>
References: <27112490.post@talk.nabble.com>
Message-ID: <191c3a031001111218l645342d8k5ba8b1304e303025@mail.gmail.com>

We have a policy against getting involved in load testing.
Many people are getting well beyond 1.5 cps with trunk so you are probably
doing something wrong.
That's about all I have to offer on the topic.



On Mon, Jan 11, 2010 at 9:43 AM, itunali <ibrahimtunali at gmail.com> wrote:

>
> Hi,
>
> I did performance tests to measure that freeswitch limits. The test just
> dial echo extension 9996 at default context and wait 6 sec then hangup.
>
> I used sipp test tool and set all variables/environment as described on
> (http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations)
>
> I reached 80 cps on 1.0.4 (install from freeswitch-drivers ppa) but not get
> similar rate on SVN trunk build (1.5 cps) and get many retransmission SIP
> packets.
>
> I build SVN trunk code with default ./bootstrap.sh && ./configure && make
> &&
> make install process.
>
> My server specs;
> Ubuntu 9.10 Karmic
> Linux 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 17:01:44 UTC 2009 x86_64
> GNU/Linux
>
> Is there any .deb packets for 1.0.5?
>
> Regards,
> Ibrahim
> --
> View this message in context:
> http://old.nabble.com/Performance-issue-on-1.0.4-vs-1.0.5-tp27112490p27112490.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at van.lammeren.net  Mon Jan 11 12:49:35 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Mon, 11 Jan 2010 15:49:35 -0500
Subject: [Freeswitch-users] lua+sqlite example?
In-Reply-To: <06ca01ca777a$a04125e0$e0c371a0$@com>
References: <04a201ca7623$2c0b2020$84216060$@com>
	<191c3a030912070935u183ff728j8b2c99576da1f5b8@mail.gmail.com>
	<06ca01ca777a$a04125e0$e0c371a0$@com>
Message-ID: <5d2828f1001111249t29331786t6dc9705897b77442@mail.gmail.com>

LuaSQL supports both sqlite and sqlite3 natively.
You can find more info here:
http://www.keplerproject.org/luasql/index.html

And you can download the source from here:
http://luaforge.net/frs/?group_id=12

Mike van Lammeren
<http://luaforge.net/frs/?group_id=12>

On Mon, Dec 7, 2009 at 3:19 PM, Steve Klein <sklein at singular.com> wrote:

>  Thanks. We?ll look at that.
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, December 07, 2009 9:35 AM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] lua+sqlite example?
>
>
>
> yes if you use the lua odbc sql plugin you should be able to use that for
> sqlite, they may also have a native one.
>
>  On Sat, Dec 5, 2009 at 9:21 PM, Steve Klein <sklein at singular.com> wrote:
>
> Greetings. We are attempting to add sqlite access to an IVR application we
> are prototyping. We are using lua for the scripts. Is there an example
> anywhere of a lua + sqlite script? Do we need to install luasql? Any
> help/pointers greatly appreciated.
>
>
>
> --Steve Klein
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> No virus found in this incoming message.
> Checked by AVG - www.avg.com
> Version: 8.5.426 / Virus Database: 270.14.83/2529 - Release Date: 12/07/09
> 07:33:00
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jcasale at activenetwerx.com  Mon Jan 11 13:34:02 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Mon, 11 Jan 2010 21:34:02 +0000
Subject: [Freeswitch-users] Outbound call problem
In-Reply-To: <b74751491001110917n6709a93cua5dcdce67232ce7a@mail.gmail.com>
References: <CA5A491E9DEFBE4CB777DE97E21575E9076F42@prato.activenetwerx.local>
	<b74751491001110917n6709a93cua5dcdce67232ce7a@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9077570@prato.activenetwerx.local>

>post the logs
?
Ram,
I don't know what to say:) I spent two days on this without results.
Someone onsite finally called me back with a chance to dial out and
it worked? I had to review the log three times because I didn't believe
it...

Can't say I mind, but I wish I knew why, heh.

Thanks!
jlc




From a.alalousi at gmail.com  Mon Jan 11 14:10:39 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Mon, 11 Jan 2010 22:10:39 +0000
Subject: [Freeswitch-users] Revision 16238,
	16237 compiling and call handling issue
Message-ID: <cd93b3701001111410labccf1eme3e9c680cb13c249@mail.gmail.com>

People,

It seems there are  a few things that are broken in this release. So far,
I've come across two issues: first of, configure fails, per what others
reported, complaining about NONE/bin not being an absolute directory path;
1637 has no problems with that.

There is also what appears to be a serious call handling issue that was not
present in 1.0.4 trunks which is the following:

Calls are initiated correctly, Leg-B is set-up and remote end rings,
ring-back is reported to Leg-A, remote end picks up, but Leg-A still gets
the ringing tone. This is pretty much the case with both internal and
external profiles, irrespective of whether or not a gateway is used to route
the call, irrespective of which media mode FS is running in and irrespective
of codec.

I know dev and support groups don't like to get involved in performance and
load testing loops, but for he who cares, performance on 1.0.5 trunks is
miserable compared to 1.0.4. I've bombarded 1.0.4 with over 150 cps with
hardly a glitch on quad-xeon machines with 6Gb RAM and GigE controllers.
1.0.5 falls far, far short of that and the figure is a fraction at just
under 20 for the same spec. machine. I run CentOS 5.0.4, and have had the
exact same experience with Debian stable and Ubuntu Karmic on the same
hardware.

I'm tracing the call handling issue now, and will report back if I find
anything useful. Meantime, it would be great if someone from support can
open a Jira for this or let me know how to do that - sorry, my own ignorance
of how the dev/support system works .. only converted to FS 6 weeks ago :).
Happy to post traces and findings.

Regards,

Ahmed.
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From anthony.minessale at gmail.com  Mon Jan 11 14:32:22 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Jan 2010 16:32:22 -0600
Subject: [Freeswitch-users] Revision 16238,
	16237 compiling and call 	handling issue
In-Reply-To: <cd93b3701001111410labccf1eme3e9c680cb13c249@mail.gmail.com>
References: <cd93b3701001111410labccf1eme3e9c680cb13c249@mail.gmail.com>
Message-ID: <191c3a031001111432j5734c6begc52d5398b4844c0e@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Reporting_Bugs


On Mon, Jan 11, 2010 at 4:10 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:

> People,
>
> It seems there are  a few things that are broken in this release. So far,
> I've come across two issues: first of, configure fails, per what others
> reported, complaining about NONE/bin not being an absolute directory path;
> 1637 has no problems with that.
>
> There is also what appears to be a serious call handling issue that was not
> present in 1.0.4 trunks which is the following:
>
> Calls are initiated correctly, Leg-B is set-up and remote end rings,
> ring-back is reported to Leg-A, remote end picks up, but Leg-A still gets
> the ringing tone. This is pretty much the case with both internal and
> external profiles, irrespective of whether or not a gateway is used to route
> the call, irrespective of which media mode FS is running in and irrespective
> of codec.
>
> I know dev and support groups don't like to get involved in performance and
> load testing loops, but for he who cares, performance on 1.0.5 trunks is
> miserable compared to 1.0.4. I've bombarded 1.0.4 with over 150 cps with
> hardly a glitch on quad-xeon machines with 6Gb RAM and GigE controllers.
> 1.0.5 falls far, far short of that and the figure is a fraction at just
> under 20 for the same spec. machine. I run CentOS 5.0.4, and have had the
> exact same experience with Debian stable and Ubuntu Karmic on the same
> hardware.
>
> I'm tracing the call handling issue now, and will report back if I find
> anything useful. Meantime, it would be great if someone from support can
> open a Jira for this or let me know how to do that - sorry, my own ignorance
> of how the dev/support system works .. only converted to FS 6 weeks ago :).
> Happy to post traces and findings.
>
> Regards,
>
> Ahmed.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From djbinter at yahoo.com  Mon Jan 11 14:41:14 2010
From: djbinter at yahoo.com (DJB)
Date: Mon, 11 Jan 2010 14:41:14 -0800 (PST)
Subject: [Freeswitch-users] Bypass Media mode seems to be broken
Message-ID: <78283.656.qm@web37506.mail.mud.yahoo.com>

I wonder whether anyone experienced this problem.

SVN Version: 16249

Trace log:  http://pastebin.freeswitch.org/11754

Dialplan:
   <extension name="tollfree">
     <condition field="destination_number" expression="^(\+1|1)?(8(00|88|77|66)[2-9]\d{6})$">
       <action application="set" data="bypass_media=true"/>
       <action application="bridge" data="sofia/external/1$2 at sip.tollfreegateway.com"/>
     </condition>
   </extension>

Problem:
FS is not passing 200 OK to inbound leg when it received from outbound leg


      
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From mike at van.lammeren.net  Mon Jan 11 14:53:24 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Mon, 11 Jan 2010 17:53:24 -0500
Subject: [Freeswitch-users] How to provide dynamic directory information?
Message-ID: <5d2828f1001111453t8044774w3cc882988e18ced@mail.gmail.com>

Hello!

I'd like to be able to have FreeSWITCH check a database for authorization,
every time a user registers. There are some great examples on the wiki,
which use either MOD_XML_CURL or Lua to dynamically provide a dialplan, but
I cannot find an example for providing a directory.

I am developing an application that will have thousands of users, and will
run on multiple FreeSWITCH servers behind a load balancer. Ideally,
FreeSWITCH would only look-up directory information, specifically, username
and password, whenever a user attempts to connect. The directory information
will be changing regularly, as users are added or removed from the system.

Is this possible with FreeSWITCH? Or can only dialplan information be
provided dynamically?

I've written a script in Lua that provides the XML data, such as that found
in the example /freeswitch/conf/directory/default/ folders, and I try to
call it with this bit of XML in /freeswitch/conf/directory/default.xml:

<groups>
<group name="default">
<users>
<param name="xml-handler-script" value="/directory.lua"/>
<param name="xml-handler-bindings" value="users"/>
</users>
</groups>

Is this the right approach? Am I going about this the right way?

I would appreciate any tips that anyone can provide!

Thanks!

Mike van Lammeren
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From mcampbellsmith at gmail.com  Mon Jan 11 14:58:05 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 12 Jan 2010 09:58:05 +1100
Subject: [Freeswitch-users] Bypass Media mode seems to be broken
In-Reply-To: <78283.656.qm@web37506.mail.mud.yahoo.com>
References: <78283.656.qm@web37506.mail.mud.yahoo.com>
Message-ID: <33c87fa31001111458s595c0164l7bd64c9031b38ed7@mail.gmail.com>

I sent this yesterday (my FS version is FreeSWITCH Version 1.0.trunk (16131) )

Not sure if this is related ...

Hi!

I am calling from extension 2010 to extension 1000.  Both have ip
addesses 192.168.1.x.

In the 2000 series dialplan (a separate context) I have the following
to try to enable bypass_media.

           <extension name="Local_1000_9xxx">
                 <condition field="destination_number"
expression="^(10[01][0-9]|9\d{3})$">
                       <action application="set" data="dialed_extension=$1"/>
                       <action application="set" data="proxy_media=false"/>
                       <action application="set" data="bypass_media=true"/>
                       <action application="transfer"
data="${dialed_extension} XML default"/>
                 </condition>
           </extension>

Is this how bypass media should be enabled?

This fails fo me (the calls hang up and no audio).  The debug trace is
in http://pastebin.freeswitch.org/11737

What have I done wrong?

Thanks


On Tue, Jan 12, 2010 at 9:41 AM, DJB <djbinter at yahoo.com> wrote:
> I wonder whether anyone experienced this problem.
> SVN Version: 16249
> Trace log: ?http://pastebin.freeswitch.org/11754
> Dialplan:
> ?? <extension name="tollfree">
> ?? ? <condition field="destination_number"
> expression="^(\+1|1)?(8(00|88|77|66)[2-9]\d{6})$">
> ?? ? ? <action application="set" data="bypass_media=true"/>
> ?? ? ? <action application="bridge"
> data="sofia/external/1$2 at sip.tollfreegateway.com"/>
> ?? ? </condition>
> ?? </extension>
> Problem:
> FS is not passing 200 OK to inbound leg when it received from outbound leg
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From egable+freeswitch at gmail.com  Mon Jan 11 15:02:37 2010
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Mon, 11 Jan 2010 18:02:37 -0500
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
Message-ID: <bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>

Assuming you're using SVN trunk, there is a bug that is likely causing
your issue. Anthony is fixing it right now.

On Mon, Jan 11, 2010 at 6:16 AM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Hi!
> Hi!
>
> I am calling from extension 2010 to extension 1000. ?Both have ip
> addesses 192.168.1.x.
>
> In the 2000 series dialplan (a separate context) I have the following
> to try to enable bypass_media.
>
> ? ? ? ? ? ?<extension name="Local_1000_9xxx">
> ? ? ? ? ? ? ? ? ?<condition field="destination_number"
> expression="^(10[01][0-9]|9\d{3})$">
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="dialed_extension=$1"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="proxy_media=false"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="bypass_media=true"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="transfer"
> data="${dialed_extension} XML default"/>
> ? ? ? ? ? ? ? ? ?</condition>
> ? ? ? ? ? ?</extension>
>
> Is this how bypass media should be enabled?
>
> This fails fo me (the calls hang up and no audio). ?The debug trace is
> in http://pastebin.freeswitch.org/11737
>
> What have I done wrong?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero



From egable+freeswitch at gmail.com  Mon Jan 11 15:06:30 2010
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Mon, 11 Jan 2010 18:06:30 -0500
Subject: [Freeswitch-users] Bypass Media mode seems to be broken
In-Reply-To: <33c87fa31001111458s595c0164l7bd64c9031b38ed7@mail.gmail.com>
References: <78283.656.qm@web37506.mail.mud.yahoo.com>
	<33c87fa31001111458s595c0164l7bd64c9031b38ed7@mail.gmail.com>
Message-ID: <bf6923ed1001111506q471ac349md187f4df91ae9280@mail.gmail.com>

Anthony reproduced it and is fixing it right now.


On Mon, Jan 11, 2010 at 5:58 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> I sent this yesterday (my FS version is FreeSWITCH Version 1.0.trunk (16131) )
>
> Not sure if this is related ...
>
> Hi!
>
> I am calling from extension 2010 to extension 1000. ?Both have ip
> addesses 192.168.1.x.
>
> In the 2000 series dialplan (a separate context) I have the following
> to try to enable bypass_media.
>
> ? ? ? ? ? <extension name="Local_1000_9xxx">
> ? ? ? ? ? ? ? ? <condition field="destination_number"
> expression="^(10[01][0-9]|9\d{3})$">
> ? ? ? ? ? ? ? ? ? ? ? <action application="set" data="dialed_extension=$1"/>
> ? ? ? ? ? ? ? ? ? ? ? <action application="set" data="proxy_media=false"/>
> ? ? ? ? ? ? ? ? ? ? ? <action application="set" data="bypass_media=true"/>
> ? ? ? ? ? ? ? ? ? ? ? <action application="transfer"
> data="${dialed_extension} XML default"/>
> ? ? ? ? ? ? ? ? </condition>
> ? ? ? ? ? </extension>
>
> Is this how bypass media should be enabled?
>
> This fails fo me (the calls hang up and no audio). ?The debug trace is
> in http://pastebin.freeswitch.org/11737
>
> What have I done wrong?
>
> Thanks
>
>
> On Tue, Jan 12, 2010 at 9:41 AM, DJB <djbinter at yahoo.com> wrote:
>> I wonder whether anyone experienced this problem.
>> SVN Version: 16249
>> Trace log: ?http://pastebin.freeswitch.org/11754
>> Dialplan:
>> ?? <extension name="tollfree">
>> ?? ? <condition field="destination_number"
>> expression="^(\+1|1)?(8(00|88|77|66)[2-9]\d{6})$">
>> ?? ? ? <action application="set" data="bypass_media=true"/>
>> ?? ? ? <action application="bridge"
>> data="sofia/external/1$2 at sip.tollfreegateway.com"/>
>> ?? ? </condition>
>> ?? </extension>
>> Problem:
>> FS is not passing 200 OK to inbound leg when it received from outbound leg
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero



From mcampbellsmith at gmail.com  Mon Jan 11 15:09:40 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 12 Jan 2010 10:09:40 +1100
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
Message-ID: <33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>

Thanks Eliot.  I'm using FreeSWITCH Version 1.0.trunk (16131).   Do
you kno wif the fault was present in 16131?

Cheers

On Tue, Jan 12, 2010 at 10:02 AM, Eliot Gable
<egable+freeswitch at gmail.com> wrote:
> Assuming you're using SVN trunk, there is a bug that is likely causing
> your issue. Anthony is fixing it right now.
>
> On Mon, Jan 11, 2010 at 6:16 AM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>> Hi!
>> Hi!
>>
>> I am calling from extension 2010 to extension 1000. ?Both have ip
>> addesses 192.168.1.x.
>>
>> In the 2000 series dialplan (a separate context) I have the following
>> to try to enable bypass_media.
>>
>> ? ? ? ? ? ?<extension name="Local_1000_9xxx">
>> ? ? ? ? ? ? ? ? ?<condition field="destination_number"
>> expression="^(10[01][0-9]|9\d{3})$">
>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="dialed_extension=$1"/>
>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="proxy_media=false"/>
>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="bypass_media=true"/>
>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="transfer"
>> data="${dialed_extension} XML default"/>
>> ? ? ? ? ? ? ? ? ?</condition>
>> ? ? ? ? ? ?</extension>
>>
>> Is this how bypass media should be enabled?
>>
>> This fails fo me (the calls hang up and no audio). ?The debug trace is
>> in http://pastebin.freeswitch.org/11737
>>
>> What have I done wrong?
>>
>> Thanks
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Mon Jan 11 15:10:09 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Jan 2010 17:10:09 -0600
Subject: [Freeswitch-users] Bypass Media mode seems to be broken
In-Reply-To: <bf6923ed1001111506q471ac349md187f4df91ae9280@mail.gmail.com>
References: <78283.656.qm@web37506.mail.mud.yahoo.com>
	<33c87fa31001111458s595c0164l7bd64c9031b38ed7@mail.gmail.com>
	<bf6923ed1001111506q471ac349md187f4df91ae9280@mail.gmail.com>
Message-ID: <EBB7B2A4-295A-4E3D-B26B-855601858D27@freeswitch.org>

YEP YEP!

/b

On Jan 11, 2010, at 5:06 PM, Eliot Gable wrote:

> Anthony reproduced it and is fixing it right now.
> 




From mike at jerris.com  Mon Jan 11 15:18:12 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 11 Jan 2010 18:18:12 -0500
Subject: [Freeswitch-users] Revision 16238,
	16237 compiling and call handling issue
In-Reply-To: <cd93b3701001111410labccf1eme3e9c680cb13c249@mail.gmail.com>
References: <cd93b3701001111410labccf1eme3e9c680cb13c249@mail.gmail.com>
Message-ID: <FF3124B2-775D-4308-B405-DC9D14DC3396@jerris.com>


On Jan 11, 2010, at 5:10 PM, Ahmed Naji wrote:

> People,
> 
> It seems there are  a few things that are broken in this release. So far, I've come across two issues: first of, configure fails, per what others reported, complaining about NONE/bin not being an absolute directory path; 1637 has no problems with that.

fixed in tree already.




From egable+freeswitch at gmail.com  Mon Jan 11 15:22:07 2010
From: egable+freeswitch at gmail.com (Eliot Gable)
Date: Mon, 11 Jan 2010 18:22:07 -0500
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
Message-ID: <bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>

I can't say for certain. I know it's not in 16016. Wait for Anthony to
fix the issue in the current version, then update and try again.

On Mon, Jan 11, 2010 at 6:09 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Thanks Eliot. ?I'm using FreeSWITCH Version 1.0.trunk (16131). ? Do
> you kno wif the fault was present in 16131?
>
> Cheers
>
> On Tue, Jan 12, 2010 at 10:02 AM, Eliot Gable
> <egable+freeswitch at gmail.com> wrote:
>> Assuming you're using SVN trunk, there is a bug that is likely causing
>> your issue. Anthony is fixing it right now.
>>
>> On Mon, Jan 11, 2010 at 6:16 AM, Mark Campbell-Smith
>> <mcampbellsmith at gmail.com> wrote:
>>> Hi!
>>> Hi!
>>>
>>> I am calling from extension 2010 to extension 1000. ?Both have ip
>>> addesses 192.168.1.x.
>>>
>>> In the 2000 series dialplan (a separate context) I have the following
>>> to try to enable bypass_media.
>>>
>>> ? ? ? ? ? ?<extension name="Local_1000_9xxx">
>>> ? ? ? ? ? ? ? ? ?<condition field="destination_number"
>>> expression="^(10[01][0-9]|9\d{3})$">
>>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="dialed_extension=$1"/>
>>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="proxy_media=false"/>
>>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="bypass_media=true"/>
>>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="transfer"
>>> data="${dialed_extension} XML default"/>
>>> ? ? ? ? ? ? ? ? ?</condition>
>>> ? ? ? ? ? ?</extension>
>>>
>>> Is this how bypass media should be enabled?
>>>
>>> This fails fo me (the calls hang up and no audio). ?The debug trace is
>>> in http://pastebin.freeswitch.org/11737
>>>
>>> What have I done wrong?
>>>
>>> Thanks
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Eliot Gable
>>
>> "We do not inherit the Earth from our ancestors: we borrow it from our
>> children." ~David Brower
>>
>> "I decided the words were too conservative for me. We're not borrowing
>> from our children, we're stealing from them--and it's not even
>> considered to be a crime." ~David Brower
>>
>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>> live; not live to eat.) ~Marcus Tullius Cicero
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero



From a.alalousi at gmail.com  Mon Jan 11 15:32:33 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Mon, 11 Jan 2010 23:32:33 +0000
Subject: [Freeswitch-users] Revision 16238,
	16237 compiling and call 	handling issue
In-Reply-To: <191c3a031001111432j5734c6begc52d5398b4844c0e@mail.gmail.com>
References: <cd93b3701001111410labccf1eme3e9c680cb13c249@mail.gmail.com>
	<191c3a031001111432j5734c6begc52d5398b4844c0e@mail.gmail.com>
Message-ID: <cd93b3701001111532y6a9f6f74j18c85fc9b02553ba@mail.gmail.com>

Thanks for this.

2010/1/11 Anthony Minessale <anthony.minessale at gmail.com>

>
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>
>
> On Mon, Jan 11, 2010 at 4:10 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>
>> People,
>>
>> It seems there are  a few things that are broken in this release. So far,
>> I've come across two issues: first of, configure fails, per what others
>> reported, complaining about NONE/bin not being an absolute directory path;
>> 1637 has no problems with that.
>>
>> There is also what appears to be a serious call handling issue that was
>> not present in 1.0.4 trunks which is the following:
>>
>> Calls are initiated correctly, Leg-B is set-up and remote end rings,
>> ring-back is reported to Leg-A, remote end picks up, but Leg-A still gets
>> the ringing tone. This is pretty much the case with both internal and
>> external profiles, irrespective of whether or not a gateway is used to route
>> the call, irrespective of which media mode FS is running in and irrespective
>> of codec.
>>
>> I know dev and support groups don't like to get involved in performance
>> and load testing loops, but for he who cares, performance on 1.0.5 trunks is
>> miserable compared to 1.0.4. I've bombarded 1.0.4 with over 150 cps with
>> hardly a glitch on quad-xeon machines with 6Gb RAM and GigE controllers.
>> 1.0.5 falls far, far short of that and the figure is a fraction at just
>> under 20 for the same spec. machine. I run CentOS 5.0.4, and have had the
>> exact same experience with Debian stable and Ubuntu Karmic on the same
>> hardware.
>>
>> I'm tracing the call handling issue now, and will report back if I find
>> anything useful. Meantime, it would be great if someone from support can
>> open a Jira for this or let me know how to do that - sorry, my own ignorance
>> of how the dev/support system works .. only converted to FS 6 weeks ago :).
>> Happy to post traces and findings.
>>
>> Regards,
>>
>> Ahmed.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From nicolas at medularis.com  Mon Jan 11 15:42:58 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 11 Jan 2010 20:42:58 -0300
Subject: [Freeswitch-users] Calls getting queued?
In-Reply-To: <191c3a031001071022j1e5796fdvf63900f5968bc01b@mail.gmail.com>
References: <1b46b4e81001070743v5389ef3ewb75dbeb06b402c9a@mail.gmail.com>
	<87f2f3b91001070926j60139f2cxe0ae9687e25a302e@mail.gmail.com>
	<1b46b4e81001071012y237fb8c0jd68232137b7d8e96@mail.gmail.com>
	<191c3a031001071022j1e5796fdvf63900f5968bc01b@mail.gmail.com>
Message-ID: <1b46b4e81001111542t51629dd0ic22f5cc908283778@mail.gmail.com>

Thanks. I actually got rid of all the JS callbacks and left only the main JS
script which originates 2 calls and then bridges them together. I moved all
event detection to an Event Sockets daemon. I thought I was off the hook,
but today the issue started happening again, and there was no curl involved.

Without looking at sip traces, what do you think could create a situation
like this?

I have no idea how to reproduce this issue, except wait for a few hours or
maybe even a few days, so I'm not sure recording all sip traffic would be
such a good idea. How would I go about getting traces for this?

Thanks.


On Thu, Jan 7, 2010 at 3:22 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try setting the timeout in curl
>
> conf/autoload_configs/xml_curl.conf.xml:
> <param name="timeout" value="5"/>
>
>
> On Thu, Jan 7, 2010 at 12:12 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> Michael,
>>
>> Thanks for your help. Yes, if I restart FS things go back to normal
>> for a while and then the same thing starts happening again.
>>
>> The weird thing is, it started only 2 days ago, and happened only once
>> or twice. Before that I had no trouble, and I only made 1 change,
>> which I reverted, but it wasn't that. Today it's happening all the
>> time, if I restart FS things will work for maybe an hour and then it
>> will start doing the same thing.
>>
>> I'm guessing it might be something external to FS, like curl calls not
>> finishing properly because of the url they are requesting or something
>> like that.
>>
>> What kind of info should I collect? I don't think it has to do with
>> sofia or any sip-related problems. I'm also using the default
>> dialplan, no changes at all, I'm doing everything through JS, well and
>> one really small lua script.
>>
>> This is the main JS file:
>> It originates 2 calls and bridges them.
>>
>> - http://pastebin.freeswitch.org/11706
>>
>>
>> This is another JS script which gets called when each call is hanged up:
>> It gets some info and then requests a url using curl to update call
>> status on an external db.
>>
>> - http://pastebin.freeswitch.org/11707
>>
>>
>> This lua script calls a ruby script to do some other stuff when a call
>> is answered:
>>
>> - http://pastebin.freeswitch.org/11708
>>
>>
>> Thanks!
>>
>>
>> Nico
>>
>>
>>
>> On Thu, Jan 7, 2010 at 2:26 PM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> >
>> >
>> > On Thu, Jan 7, 2010 at 7:43 AM, Nicolas Brenner <nicolas at medularis.com>
>> > wrote:
>> >>
>> >> Hi, I'm having a strange problem with FS. I'm using a few JS scripts
>> >> to generate calls and bridge them together. Usually everything works
>> >> just fine, but them at some point it's like if FS choked, calls for
>> >> the first leg of the bridges are apparently made, but the second leg
>> >> is never called. The call is not hanged up for several minutes and the
>> >> system keeps opening new channels but never connecting a call.
>> >>
>> >> For example, right now, doing 'show channels' on the console, I get a
>> >> list of 72 open channels (it's adding up, it was 40 a couple minutes
>> >> ago), but doing a 'show calls' gives me 0 active calls. The usual
>> >> behavior, when everything's working fine, is to get twice as many
>> >> channels as there are active calls and no channels at all when there
>> >> are no calls, unless they haven't been bridged yet.
>> >>
>> >> The originate string is something like this:
>> >>
>> >> var stUsRing = "%(2000,4000,440,480)";
>> >> var timeout = 45;
>> >> originate_str1 = "{api_hangup_hook=jsapi::callback.js
>> >> l1,execute_on_answer=lua answered.lua 1
>> >>
>> >>
>> c2c_call,ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,ringback='"+stUsRing+"',medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]"+dialstr1;
>> >>
>> >> Where diasltr1 has the phonenumber and and gateway info. The
>> >> callback.js has a curl request to update some call info on an external
>> >> database and answered.lua calls a ruby script through the os.execute()
>> >> function (I know, I should be doing all this through the event socket,
>> >> I was doing that but had trouble and had to come up with a quick
>> >> solution).
>> >>
>> >> The system is not loaded at all, at least not for what I think and
>> >> read that FS can handle. We are having at most 10 concurrent calls (20
>> >> channels), with maybe 5 to 10 calls per minute.
>> >>
>> >> What worries me is not only that I don't know where the problem is,
>> >> but that I have no clue how to debug it or send you guys more
>> >> "lowlevel" and detailed information to give you an insight about
>> >> what's going on. Any help would be greatly appreciated!
>> >>
>> >> Thanks!
>> >>
>> >> Nico
>> >>
>> > First off you'll want to get familiar with the resources mentioned here:
>> > http://wiki.freeswitch.org/wiki/Reporting_Bugs
>> >
>> > It has good tips on how to collect and report information.
>> >
>> > Second, I recommend that you pastebin your relevant portion of the
>> dialplan
>> > and the whole javascript program that you are using so that others can
>> take
>> > a look.
>> >
>> > Last thing: if you restart FreeSWITCH does everything work fine for a
>> while
>> > but then eventually it breaks down and exhibits the behavior that you
>> are
>> > reporting?
>> >
>> > -MC
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Jan 11 16:07:59 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Jan 2010 18:07:59 -0600
Subject: [Freeswitch-users] Revision 16238,
	16237 compiling and call 	handling issue
In-Reply-To: <cd93b3701001111532y6a9f6f74j18c85fc9b02553ba@mail.gmail.com>
References: <cd93b3701001111410labccf1eme3e9c680cb13c249@mail.gmail.com>
	<191c3a031001111432j5734c6begc52d5398b4844c0e@mail.gmail.com>
	<cd93b3701001111532y6a9f6f74j18c85fc9b02553ba@mail.gmail.com>
Message-ID: <191c3a031001111607j63a72fdft6f47084820b81bf4@mail.gmail.com>

too late,
its fixed in rev 16250


On Mon, Jan 11, 2010 at 5:32 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:

> Thanks for this.
>
> 2010/1/11 Anthony Minessale <anthony.minessale at gmail.com>
>
>
>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>
>>
>> On Mon, Jan 11, 2010 at 4:10 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>>
>>> People,
>>>
>>> It seems there are  a few things that are broken in this release. So far,
>>> I've come across two issues: first of, configure fails, per what others
>>> reported, complaining about NONE/bin not being an absolute directory path;
>>> 1637 has no problems with that.
>>>
>>> There is also what appears to be a serious call handling issue that was
>>> not present in 1.0.4 trunks which is the following:
>>>
>>> Calls are initiated correctly, Leg-B is set-up and remote end rings,
>>> ring-back is reported to Leg-A, remote end picks up, but Leg-A still gets
>>> the ringing tone. This is pretty much the case with both internal and
>>> external profiles, irrespective of whether or not a gateway is used to route
>>> the call, irrespective of which media mode FS is running in and irrespective
>>> of codec.
>>>
>>> I know dev and support groups don't like to get involved in performance
>>> and load testing loops, but for he who cares, performance on 1.0.5 trunks is
>>> miserable compared to 1.0.4. I've bombarded 1.0.4 with over 150 cps with
>>> hardly a glitch on quad-xeon machines with 6Gb RAM and GigE controllers.
>>> 1.0.5 falls far, far short of that and the figure is a fraction at just
>>> under 20 for the same spec. machine. I run CentOS 5.0.4, and have had the
>>> exact same experience with Debian stable and Ubuntu Karmic on the same
>>> hardware.
>>>
>>> I'm tracing the call handling issue now, and will report back if I find
>>> anything useful. Meantime, it would be great if someone from support can
>>> open a Jira for this or let me know how to do that - sorry, my own ignorance
>>> of how the dev/support system works .. only converted to FS 6 weeks ago :).
>>> Happy to post traces and findings.
>>>
>>> Regards,
>>>
>>> Ahmed.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Mon Jan 11 16:08:40 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Jan 2010 18:08:40 -0600
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
Message-ID: <191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>

fixed in 16250

On Mon, Jan 11, 2010 at 5:22 PM, Eliot Gable
<egable+freeswitch at gmail.com<egable%2Bfreeswitch at gmail.com>
> wrote:

> I can't say for certain. I know it's not in 16016. Wait for Anthony to
> fix the issue in the current version, then update and try again.
>
> On Mon, Jan 11, 2010 at 6:09 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
> > Thanks Eliot.  I'm using FreeSWITCH Version 1.0.trunk (16131).   Do
> > you kno wif the fault was present in 16131?
> >
> > Cheers
> >
> > On Tue, Jan 12, 2010 at 10:02 AM, Eliot Gable
> > <egable+freeswitch at gmail.com <egable%2Bfreeswitch at gmail.com>> wrote:
> >> Assuming you're using SVN trunk, there is a bug that is likely causing
> >> your issue. Anthony is fixing it right now.
> >>
> >> On Mon, Jan 11, 2010 at 6:16 AM, Mark Campbell-Smith
> >> <mcampbellsmith at gmail.com> wrote:
> >>> Hi!
> >>> Hi!
> >>>
> >>> I am calling from extension 2010 to extension 1000.  Both have ip
> >>> addesses 192.168.1.x.
> >>>
> >>> In the 2000 series dialplan (a separate context) I have the following
> >>> to try to enable bypass_media.
> >>>
> >>>            <extension name="Local_1000_9xxx">
> >>>                  <condition field="destination_number"
> >>> expression="^(10[01][0-9]|9\d{3})$">
> >>>                        <action application="set"
> data="dialed_extension=$1"/>
> >>>                        <action application="set"
> data="proxy_media=false"/>
> >>>                        <action application="set"
> data="bypass_media=true"/>
> >>>                        <action application="transfer"
> >>> data="${dialed_extension} XML default"/>
> >>>                  </condition>
> >>>            </extension>
> >>>
> >>> Is this how bypass media should be enabled?
> >>>
> >>> This fails fo me (the calls hang up and no audio).  The debug trace is
> >>> in http://pastebin.freeswitch.org/11737
> >>>
> >>> What have I done wrong?
> >>>
> >>> Thanks
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> >> --
> >> Eliot Gable
> >>
> >> "We do not inherit the Earth from our ancestors: we borrow it from our
> >> children." ~David Brower
> >>
> >> "I decided the words were too conservative for me. We're not borrowing
> >> from our children, we're stealing from them--and it's not even
> >> considered to be a crime." ~David Brower
> >>
> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> >> live; not live to eat.) ~Marcus Tullius Cicero
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Mon Jan 11 16:08:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Jan 2010 18:08:24 -0600
Subject: [Freeswitch-users] Bypass Media mode seems to be broken
In-Reply-To: <EBB7B2A4-295A-4E3D-B26B-855601858D27@freeswitch.org>
References: <78283.656.qm@web37506.mail.mud.yahoo.com>
	<33c87fa31001111458s595c0164l7bd64c9031b38ed7@mail.gmail.com>
	<bf6923ed1001111506q471ac349md187f4df91ae9280@mail.gmail.com>
	<EBB7B2A4-295A-4E3D-B26B-855601858D27@freeswitch.org>
Message-ID: <191c3a031001111608o37eef419i7fcc556979b3e2ff@mail.gmail.com>

fixed in 16250


On Mon, Jan 11, 2010 at 5:10 PM, Brian West <brian at freeswitch.org> wrote:

> YEP YEP!
>
> /b
>
> On Jan 11, 2010, at 5:06 PM, Eliot Gable wrote:
>
> > Anthony reproduced it and is fixing it right now.
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Mon Jan 11 16:10:01 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Jan 2010 18:10:01 -0600
Subject: [Freeswitch-users] Calls getting queued?
In-Reply-To: <1b46b4e81001111542t51629dd0ic22f5cc908283778@mail.gmail.com>
References: <1b46b4e81001070743v5389ef3ewb75dbeb06b402c9a@mail.gmail.com>
	<87f2f3b91001070926j60139f2cxe0ae9687e25a302e@mail.gmail.com>
	<1b46b4e81001071012y237fb8c0jd68232137b7d8e96@mail.gmail.com>
	<191c3a031001071022j1e5796fdvf63900f5968bc01b@mail.gmail.com>
	<1b46b4e81001111542t51629dd0ic22f5cc908283778@mail.gmail.com>
Message-ID: <191c3a031001111610o66bbc21fxbceabd037dddcf76@mail.gmail.com>

js is notorious for garbage collection issues.
you would be wise to just build a dial string and use the bridge application
to bridge them rather than bridge them manually in JS

On Mon, Jan 11, 2010 at 5:42 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Thanks. I actually got rid of all the JS callbacks and left only the main
> JS script which originates 2 calls and then bridges them together. I moved
> all event detection to an Event Sockets daemon. I thought I was off the
> hook, but today the issue started happening again, and there was no curl
> involved.
>
> Without looking at sip traces, what do you think could create a situation
> like this?
>
> I have no idea how to reproduce this issue, except wait for a few hours or
> maybe even a few days, so I'm not sure recording all sip traffic would be
> such a good idea. How would I go about getting traces for this?
>
> Thanks.
>
>
> On Thu, Jan 7, 2010 at 3:22 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> try setting the timeout in curl
>>
>> conf/autoload_configs/xml_curl.conf.xml:
>> <param name="timeout" value="5"/>
>>
>>
>> On Thu, Jan 7, 2010 at 12:12 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>
>>> Michael,
>>>
>>> Thanks for your help. Yes, if I restart FS things go back to normal
>>> for a while and then the same thing starts happening again.
>>>
>>> The weird thing is, it started only 2 days ago, and happened only once
>>> or twice. Before that I had no trouble, and I only made 1 change,
>>> which I reverted, but it wasn't that. Today it's happening all the
>>> time, if I restart FS things will work for maybe an hour and then it
>>> will start doing the same thing.
>>>
>>> I'm guessing it might be something external to FS, like curl calls not
>>> finishing properly because of the url they are requesting or something
>>> like that.
>>>
>>> What kind of info should I collect? I don't think it has to do with
>>> sofia or any sip-related problems. I'm also using the default
>>> dialplan, no changes at all, I'm doing everything through JS, well and
>>> one really small lua script.
>>>
>>> This is the main JS file:
>>> It originates 2 calls and bridges them.
>>>
>>> - http://pastebin.freeswitch.org/11706
>>>
>>>
>>> This is another JS script which gets called when each call is hanged up:
>>> It gets some info and then requests a url using curl to update call
>>> status on an external db.
>>>
>>> - http://pastebin.freeswitch.org/11707
>>>
>>>
>>> This lua script calls a ruby script to do some other stuff when a call
>>> is answered:
>>>
>>> - http://pastebin.freeswitch.org/11708
>>>
>>>
>>> Thanks!
>>>
>>>
>>> Nico
>>>
>>>
>>>
>>> On Thu, Jan 7, 2010 at 2:26 PM, Michael Collins <msc at freeswitch.org>
>>> wrote:
>>> >
>>> >
>>> > On Thu, Jan 7, 2010 at 7:43 AM, Nicolas Brenner <nicolas at medularis.com
>>> >
>>> > wrote:
>>> >>
>>> >> Hi, I'm having a strange problem with FS. I'm using a few JS scripts
>>> >> to generate calls and bridge them together. Usually everything works
>>> >> just fine, but them at some point it's like if FS choked, calls for
>>> >> the first leg of the bridges are apparently made, but the second leg
>>> >> is never called. The call is not hanged up for several minutes and the
>>> >> system keeps opening new channels but never connecting a call.
>>> >>
>>> >> For example, right now, doing 'show channels' on the console, I get a
>>> >> list of 72 open channels (it's adding up, it was 40 a couple minutes
>>> >> ago), but doing a 'show calls' gives me 0 active calls. The usual
>>> >> behavior, when everything's working fine, is to get twice as many
>>> >> channels as there are active calls and no channels at all when there
>>> >> are no calls, unless they haven't been bridged yet.
>>> >>
>>> >> The originate string is something like this:
>>> >>
>>> >> var stUsRing = "%(2000,4000,440,480)";
>>> >> var timeout = 45;
>>> >> originate_str1 = "{api_hangup_hook=jsapi::callback.js
>>> >> l1,execute_on_answer=lua answered.lua 1
>>> >>
>>> >>
>>> c2c_call,ignore_early_media=true,originate_timeout=90,hangup_after_bridge=false,ringback='"+stUsRing+"',medularis_uuid="+uuid+",c2c_call=true,api_call=true,leg=1}[leg_timeout="+timeout+"]"+dialstr1;
>>> >>
>>> >> Where diasltr1 has the phonenumber and and gateway info. The
>>> >> callback.js has a curl request to update some call info on an external
>>> >> database and answered.lua calls a ruby script through the os.execute()
>>> >> function (I know, I should be doing all this through the event socket,
>>> >> I was doing that but had trouble and had to come up with a quick
>>> >> solution).
>>> >>
>>> >> The system is not loaded at all, at least not for what I think and
>>> >> read that FS can handle. We are having at most 10 concurrent calls (20
>>> >> channels), with maybe 5 to 10 calls per minute.
>>> >>
>>> >> What worries me is not only that I don't know where the problem is,
>>> >> but that I have no clue how to debug it or send you guys more
>>> >> "lowlevel" and detailed information to give you an insight about
>>> >> what's going on. Any help would be greatly appreciated!
>>> >>
>>> >> Thanks!
>>> >>
>>> >> Nico
>>> >>
>>> > First off you'll want to get familiar with the resources mentioned
>>> here:
>>> > http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>> >
>>> > It has good tips on how to collect and report information.
>>> >
>>> > Second, I recommend that you pastebin your relevant portion of the
>>> dialplan
>>> > and the whole javascript program that you are using so that others can
>>> take
>>> > a look.
>>> >
>>> > Last thing: if you restart FreeSWITCH does everything work fine for a
>>> while
>>> > but then eventually it breaks down and exhibits the behavior that you
>>> are
>>> > reporting?
>>> >
>>> > -MC
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mcampbellsmith at gmail.com  Mon Jan 11 16:14:47 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 12 Jan 2010 11:14:47 +1100
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
Message-ID: <33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>

Beauty!  Thanks.. will update now.

On Tue, Jan 12, 2010 at 11:08 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> fixed in 16250
>
> On Mon, Jan 11, 2010 at 5:22 PM, Eliot Gable <egable+freeswitch at gmail.com>
> wrote:
>>
>> I can't say for certain. I know it's not in 16016. Wait for Anthony to
>> fix the issue in the current version, then update and try again.
>>
>> On Mon, Jan 11, 2010 at 6:09 PM, Mark Campbell-Smith
>> <mcampbellsmith at gmail.com> wrote:
>> > Thanks Eliot. ?I'm using FreeSWITCH Version 1.0.trunk (16131). ? Do
>> > you kno wif the fault was present in 16131?
>> >
>> > Cheers
>> >
>> > On Tue, Jan 12, 2010 at 10:02 AM, Eliot Gable
>> > <egable+freeswitch at gmail.com> wrote:
>> >> Assuming you're using SVN trunk, there is a bug that is likely causing
>> >> your issue. Anthony is fixing it right now.
>> >>
>> >> On Mon, Jan 11, 2010 at 6:16 AM, Mark Campbell-Smith
>> >> <mcampbellsmith at gmail.com> wrote:
>> >>> Hi!
>> >>> Hi!
>> >>>
>> >>> I am calling from extension 2010 to extension 1000. ?Both have ip
>> >>> addesses 192.168.1.x.
>> >>>
>> >>> In the 2000 series dialplan (a separate context) I have the following
>> >>> to try to enable bypass_media.
>> >>>
>> >>> ? ? ? ? ? ?<extension name="Local_1000_9xxx">
>> >>> ? ? ? ? ? ? ? ? ?<condition field="destination_number"
>> >>> expression="^(10[01][0-9]|9\d{3})$">
>> >>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set"
>> >>> data="dialed_extension=$1"/>
>> >>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set"
>> >>> data="proxy_media=false"/>
>> >>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set"
>> >>> data="bypass_media=true"/>
>> >>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="transfer"
>> >>> data="${dialed_extension} XML default"/>
>> >>> ? ? ? ? ? ? ? ? ?</condition>
>> >>> ? ? ? ? ? ?</extension>
>> >>>
>> >>> Is this how bypass media should be enabled?
>> >>>
>> >>> This fails fo me (the calls hang up and no audio). ?The debug trace is
>> >>> in http://pastebin.freeswitch.org/11737
>> >>>
>> >>> What have I done wrong?
>> >>>
>> >>> Thanks
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >>
>> >> --
>> >> Eliot Gable
>> >>
>> >> "We do not inherit the Earth from our ancestors: we borrow it from our
>> >> children." ~David Brower
>> >>
>> >> "I decided the words were too conservative for me. We're not borrowing
>> >> from our children, we're stealing from them--and it's not even
>> >> considered to be a crime." ~David Brower
>> >>
>> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>> >> live; not live to eat.) ~Marcus Tullius Cicero
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Eliot Gable
>>
>> "We do not inherit the Earth from our ancestors: we borrow it from our
>> children." ~David Brower
>>
>> "I decided the words were too conservative for me. We're not borrowing
>> from our children, we're stealing from them--and it's not even
>> considered to be a crime." ~David Brower
>>
>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>> live; not live to eat.) ~Marcus Tullius Cicero
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From a.alalousi at gmail.com  Mon Jan 11 16:21:26 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Tue, 12 Jan 2010 00:21:26 +0000
Subject: [Freeswitch-users] Revision 16238,
	16237 compiling and call 	handling issue
In-Reply-To: <191c3a031001111607j63a72fdft6f47084820b81bf4@mail.gmail.com>
References: <cd93b3701001111410labccf1eme3e9c680cb13c249@mail.gmail.com>
	<191c3a031001111432j5734c6begc52d5398b4844c0e@mail.gmail.com>
	<cd93b3701001111532y6a9f6f74j18c85fc9b02553ba@mail.gmail.com>
	<191c3a031001111607j63a72fdft6f47084820b81bf4@mail.gmail.com>
Message-ID: <cd93b3701001111621l771c2333r56dd1a097fbf7a58@mail.gmail.com>

we are not worthy :)

well done.

2010/1/12 Anthony Minessale <anthony.minessale at gmail.com>

> too late,
> its fixed in rev 16250
>
>
>
> On Mon, Jan 11, 2010 at 5:32 PM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>
>> Thanks for this.
>>
>> 2010/1/11 Anthony Minessale <anthony.minessale at gmail.com>
>>
>>
>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>
>>>
>>> On Mon, Jan 11, 2010 at 4:10 PM, Ahmed Naji <a.alalousi at gmail.com>wrote:
>>>
>>>> People,
>>>>
>>>> It seems there are  a few things that are broken in this release. So
>>>> far, I've come across two issues: first of, configure fails, per what others
>>>> reported, complaining about NONE/bin not being an absolute directory path;
>>>> 1637 has no problems with that.
>>>>
>>>> There is also what appears to be a serious call handling issue that was
>>>> not present in 1.0.4 trunks which is the following:
>>>>
>>>> Calls are initiated correctly, Leg-B is set-up and remote end rings,
>>>> ring-back is reported to Leg-A, remote end picks up, but Leg-A still gets
>>>> the ringing tone. This is pretty much the case with both internal and
>>>> external profiles, irrespective of whether or not a gateway is used to route
>>>> the call, irrespective of which media mode FS is running in and irrespective
>>>> of codec.
>>>>
>>>> I know dev and support groups don't like to get involved in performance
>>>> and load testing loops, but for he who cares, performance on 1.0.5 trunks is
>>>> miserable compared to 1.0.4. I've bombarded 1.0.4 with over 150 cps with
>>>> hardly a glitch on quad-xeon machines with 6Gb RAM and GigE controllers.
>>>> 1.0.5 falls far, far short of that and the figure is a fraction at just
>>>> under 20 for the same spec. machine. I run CentOS 5.0.4, and have had the
>>>> exact same experience with Debian stable and Ubuntu Karmic on the same
>>>> hardware.
>>>>
>>>> I'm tracing the call handling issue now, and will report back if I find
>>>> anything useful. Meantime, it would be great if someone from support can
>>>> open a Jira for this or let me know how to do that - sorry, my own ignorance
>>>> of how the dev/support system works .. only converted to FS 6 weeks ago :).
>>>> Happy to post traces and findings.
>>>>
>>>> Regards,
>>>>
>>>> Ahmed.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Ahmed Naji
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From msc at freeswitch.org  Mon Jan 11 16:25:42 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 16:25:42 -0800
Subject: [Freeswitch-users] URGENT : DTMF during bridge
In-Reply-To: <27107895.post@talk.nabble.com>
References: <27107895.post@talk.nabble.com>
Message-ID: <87f2f3b91001111625t2e959dc0g6fd2fb1f58aa0da3@mail.gmail.com>

On Mon, Jan 11, 2010 at 1:43 AM, Oscav <oscav at hotmail.fr> wrote:

>
> Hi,
>
> I need to handle DTMF during bridge in order to hangup the called party on
> caller request. The DTMF sequence should be ##. Any idea on how to do
> that??
>
> I think you need bind_meta_app. Look at the wiki as well as the
Local_Extension example to see how bind_meta_app works. The catch for you,
though, is that you can't use ## as your digit sequence, instead you'll need
to use *# or ** or something like that.

-MC
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From msc at freeswitch.org  Mon Jan 11 16:36:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 16:36:03 -0800
Subject: [Freeswitch-users] How to provide dynamic directory information?
In-Reply-To: <5d2828f1001111453t8044774w3cc882988e18ced@mail.gmail.com>
References: <5d2828f1001111453t8044774w3cc882988e18ced@mail.gmail.com>
Message-ID: <87f2f3b91001111636j1613b6ak5b5de86804f2125e@mail.gmail.com>

On Mon, Jan 11, 2010 at 2:53 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Hello!
>
> I'd like to be able to have FreeSWITCH check a database for authorization,
> every time a user registers. There are some great examples on the wiki,
> which use either MOD_XML_CURL or Lua to dynamically provide a dialplan, but
> I cannot find an example for providing a directory.
>
> I am developing an application that will have thousands of users, and will
> run on multiple FreeSWITCH servers behind a load balancer. Ideally,
> FreeSWITCH would only look-up directory information, specifically, username
> and password, whenever a user attempts to connect. The directory information
> will be changing regularly, as users are added or removed from the system.
>
> Is this possible with FreeSWITCH? Or can only dialplan information be
> provided dynamically?
>
> I've written a script in Lua that provides the XML data, such as that found
> in the example /freeswitch/conf/directory/default/ folders, and I try to
> call it with this bit of XML in /freeswitch/conf/directory/default.xml:
>
> <groups>
> <group name="default">
> <users>
>  <param name="xml-handler-script" value="/directory.lua"/>
> <param name="xml-handler-bindings" value="users"/>
>  </users>
> </groups>
>
> Is this the right approach? Am I going about this the right way?
>

You can bind "directory" as well as "dialplan" and a few others. I
personally don't use xml_curl in production but for kicks I tried to learn
it and I documented some of my journey on my personal blog. (
http://telecommusings.blogspot.com/)

xml_curl was designed to scale and be applied in your type of scenario.
Raymond (intralanman on IRC) has played with it quite a bit as have a number
of others.
-MC
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From msc at freeswitch.org  Mon Jan 11 16:44:11 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 16:44:11 -0800
Subject: [Freeswitch-users] Gateway Configuration
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9074E14@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E9074E14@prato.activenetwerx.local>
Message-ID: <87f2f3b91001111644o766fccdap5ccf4fcde3a7536b@mail.gmail.com>

On Sat, Jan 9, 2010 at 9:40 AM, Joseph L. Casale
<jcasale at activenetwerx.com>wrote:

> It seems there are two ways to configure an spa3102's fxo port w/
> pbx's, you can set the dial plan to <phone_#_of_pstn>@<freeswitch_ip>
> or <ext_to_dial>@<freeswitch_ip>.
>
> >From fs's perspective, what exactly is the difference here?
> Are there any significant differences between the two methods?
> Are there any best practices that should be considered?
>

I've only got a PAP2T (2 FXS) but from what you describe I'd say that it
makes sense for the FXO port to be "phone_#_of_pstn" and FXS to be
"ext_to_dial." I believe FS tries to be endpoint-type agnostic in this
scenario. It's a SIP call in and gets routed in the dialplan.

>
> Incoming sip did's and a zap line I had all were configured so that
> they entered the public context filtered by <phone_#>.
>
> If a call is coming in from an actual PSTN line then hitting the public
context makes a lot of sense.

> Most of the examples I see for setting up the spa don't function like this
> but a couple do?
>
As usual "it depends." However, the simplest rule of thumb would be that FXS
ports are analogous to SIP users and FXO ports are analogous to SIP
gateways.

Out of curiosity, what is your application?
-MC
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From anatoliy at kounitskiy.com  Mon Jan 11 16:44:48 2010
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Tue, 12 Jan 2010 02:44:48 +0200
Subject: [Freeswitch-users] Problem between revision 16223 and 16242 on
 Debian (stable) 64bit
In-Reply-To: <1263234649.2504.35.camel@lenovor400-laptop>
References: <1263230329.2504.33.camel@lenovor400-laptop>
	<1263234649.2504.35.camel@lenovor400-laptop>
Message-ID: <1263257088.5515.6.camel@lenovor400-laptop>

Thank you for fixing it so fast :) (tested with 16250)

Regards,
Anatoliy


On Mon, 2010-01-11 at 20:30 +0200, Anatoliy Kounitskiy wrote:
> Ok, i found in which revision it is broken.
> 
> Until revision 16237 - it works as charm
> In Revision 16238 - it doesn't work
> 
> svn log --revision 16237:16238
> ------------------------------------------------------------------------
> r16238 | mikej | 2010-01-11 16:36:29 +0200 (Mon, 11 Jan 2010) | 1 line
> 
> wip move towards adding directory layout control to configure
> ------------------------------------------------------------------------
> 
> Regards,
> 
> On Mon, 2010-01-11 at 19:18 +0200, Anatoliy Kounitskiy wrote:
> > Hello,
> > I just made a checkout of the svn and tried to configure it, but there
> > is an error in the arp-util lib, after the ./bootstrap.sh
> > 
> > 
> > Freeswitch revision: 16242
> > OS: Debian 64b (stable)
> > Command used: ./configure --prefix=/usr/local/freeswitch
> > --enable-optimization --enable-64
> > Error:
> > checking for Expat in xml/expat... yes
> > configuring package in xml/expat now
> > configure: error: expected an absolute directory name for --bindir:
> > NONE/bin
> > configure failed for xml/expat
> > configure: error: ./configure.gnu failed for libs/apr-util
> > 
> > If I execute "sh configure.gnu"/"sh configure.gnu --enable-64" in
> > libs/apr-util it goes without an error.
> > 
> > On the same server with Freeswitch revision: 16223 (with the same
> > configure command), it goes as planned - without errors.
> > 
> > Regards,
> > 
> 
> 


-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540




From rupa at rupa.com  Mon Jan 11 16:47:48 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 11 Jan 2010 18:47:48 -0600
Subject: [Freeswitch-users] URGENT : DTMF during bridge
In-Reply-To: <27107895.post@talk.nabble.com>
References: <27107895.post@talk.nabble.com>
Message-ID: <b39273531001111647n2867f859rfa08cd506a5964cb@mail.gmail.com>

Is it really necessary to say "URGENT"?  I doubt anyone will respond any
faster/sooner.

Anyway:

bind_meta_app will do * + a char to do an action.  So *# could be hangup.

You could listen to DTMF events over ESL and do whatever action you want.
 Keep in mind you should have a maximum time duration between the first a
second char.  That way, if I hit # 1s into the call and then again 5min into
the call you don't hangup on me.

On Mon, Jan 11, 2010 at 3:43 AM, Oscav <oscav at hotmail.fr> wrote:

>
> Hi,
>
> I need to handle DTMF during bridge in order to hangup the called party on
> caller request. The DTMF sequence should be ##. Any idea on how to do
> that??
>
> Thanks.
> --
> View this message in context:
> http://old.nabble.com/URGENT-%3A-DTMF-during-bridge-tp27107895p27107895.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From msc at freeswitch.org  Mon Jan 11 16:50:56 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 16:50:56 -0800
Subject: [Freeswitch-users] Help with Portech <-> Freeswitch
In-Reply-To: <c55d46951001090900p36df42e3ga91e61812d5b8d03@mail.gmail.com>
References: <c55d46951001090900p36df42e3ga91e61812d5b8d03@mail.gmail.com>
Message-ID: <87f2f3b91001111650n6d635e8dk9de30a4249a6dd73@mail.gmail.com>

On Sat, Jan 9, 2010 at 9:00 AM, Max Bridgewater
<max.bridgewater at gmail.com>wrote:

> Hi Guys,
>
> It appears quite a few people in the list are using Portech. Can you please
> help me connect Freeswitch to  it for termination puposes?
>
> Here is what I've done so far but without success.
>
> In Freeswitch I created a profile and stored it in under
> /usr/local/freeswitch/conf/sip_profiles/external/. Here is the content:
>
> <include>
>   <gateway name="portech">
>     <param name="username" value="noauthentication" />
>     <param name="password" value="noauthentication" />
>     <param name="proxy" value="74.24.22.59" />
>     <param name="realm" value="74.24.22.59" />
>     <param name="register" value="false" />
>   </gateway>
> </include>
>
> Then, in the /usr/local/freeswitch/confi/dialplan/default.xml, I added the
> following extension:
>
>    <extension name="portech">
>       <condition field="destination_number" expression="^2801$">
>         <action application="bridge"
> data="sofia/gateway/portech/5147237479"/>
>       </condition>
>     </extension>
>
> In Portech MV374, what I did is simply adding one entry in the Mobile/Lan
> to mobile table  that consists of  URL: 74.24.22.59 and call Num: #.
>
> Now, when I connect to Freeswitch with Xlite and try to dial extension 2801
> it simply says User Busy. i don't even see that attempts are being made to
> connect to the Portech gateway.
>
> Any idea?
> Thanks in advance.
>

Turn on debug level console output and turn on SIP trace and make a test
call capturing the output. Put into a pastebin and reply to this thread with
the link. Hopefully the log will tell you (and us) what is going on.

FYI, this page has lots have handy tips for how to gather information and
report it:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

That page is your friend because it gives you lots of tools and skills for
gathering information. :)
-MC
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From msc at freeswitch.org  Mon Jan 11 16:53:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 16:53:03 -0800
Subject: [Freeswitch-users] URGENT : DTMF during bridge
In-Reply-To: <b39273531001111647n2867f859rfa08cd506a5964cb@mail.gmail.com>
References: <27107895.post@talk.nabble.com>
	<b39273531001111647n2867f859rfa08cd506a5964cb@mail.gmail.com>
Message-ID: <87f2f3b91001111653y5c78bca3r47170eaa064ee9d1@mail.gmail.com>

On Mon, Jan 11, 2010 at 4:47 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Is it really necessary to say "URGENT"?  I doubt anyone will respond any
> faster/sooner.
>
> Anyway:
>
> bind_meta_app will do * + a char to do an action.  So *# could be hangup.
>
> You could listen to DTMF events over ESL and do whatever action you want.
>  Keep in mind you should have a maximum time duration between the first a
> second char.  That way, if I hit # 1s into the call and then again 5min into
> the call you don't hangup on me.
>

Rupa is wise... :)
-MC
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From brian at freeswitch.org  Mon Jan 11 16:53:52 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Jan 2010 18:53:52 -0600
Subject: [Freeswitch-users] Problem between revision 16223 and 16242 on
	Debian (stable) 64bit
In-Reply-To: <1263257088.5515.6.camel@lenovor400-laptop>
References: <1263230329.2504.33.camel@lenovor400-laptop>
	<1263234649.2504.35.camel@lenovor400-laptop>
	<1263257088.5515.6.camel@lenovor400-laptop>
Message-ID: <5246A6FF-3A35-4153-B3E6-3BEB501A947D@freeswitch.org>

Tony has a wish list on the FAQ ;)  First question.

/b

On Jan 11, 2010, at 6:44 PM, Anatoliy Kounitskiy wrote:

> Thank you for fixing it so fast :) (tested with 16250)
> 
> Regards,
> Anatoliy




From msc at freeswitch.org  Mon Jan 11 16:55:08 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 16:55:08 -0800
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
Message-ID: <87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>

On Mon, Jan 11, 2010 at 4:18 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:

> Dear All,
>
> I posted a thread re the subject but didn't get any joy, so perhaps second
> time lucky.
>
> I need to rewrite a couple of hangup causes to mean NORMAL_CONGESTION and
> no matter which variables I set, this isn't happening. The idea is to
> control what codes are returned to an end point after a successful bridge,
> as well as deal with what codes are returned if the bridge is unsuccessful
> (e.g. user_busy, originator_cancel ...etc).
>
> I've had limited success by setting hangup_after_bridge=false then bridging
> to error/<required_code>. This, however only works when the B-leg terminates
> the call after a successful answer. Any other codes are not rewritten.
>
> I've also tried playing with the bridge_hangup_code and hangup_code
> variables prior and after bridging, still no joy. I have also set
> sip_ignore_remote_cause=true prior to entering the bridge, as well
> explicitly in vars.xml.
>
> By the way, I'm running in proxy-media mode, but I did try it with
> bypass-media as well. Same symptoms, same behaviour.
>
> Any help with this would be highly appreciated.
>
> Well, I do know that when you do a hangup in the dialplan you can pass an
optional cause as well:
<action application="hangup" data="USER_BUSY"/>
If you are doing the hanging up then you have a fair amount of control...
-MC
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From msc at freeswitch.org  Mon Jan 11 16:55:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 16:55:53 -0800
Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS
	based callcenter)
In-Reply-To: <20100111181203.GD10774@hijacked.us>
References: <20091219014359.GA21798@hijacked.us>
	<855e4dcf1001041842u46c6d36q9c2e4ece2ced7380@mail.gmail.com>
	<20100111181203.GD10774@hijacked.us>
Message-ID: <87f2f3b91001111655x3c785b6au3a279a32f3eec6da@mail.gmail.com>

On Mon, Jan 11, 2010 at 10:12 AM, Andrew Thompson <andrew at hijacked.us>wrote:

> On Tue, Jan 05, 2010 at 03:42:57PM +1300, Tim Uckun wrote:
> > > http://opencsm.org/downloads/spice-telephony-0.9.6.tar.gz
> > >
> >
> > There seems to be something wrong with both opencsm.org and
> wiki.opencsm.org.
> >
> > Just thought I'd let you know.
> >
>
> Yeah, my company wanted me to move it, so I re-hosted it as a github
> project (with a new name):
>
> http://github.com/Vagabond/OpenACD
>
> I hadn't announced the change yet because I've been away for the last
> week and didn't have time. The wiki contents haven't been moved over
> yet, but they needed some cleanup anyway.
>
> On the other hand attended transfer support finally materialized.
>
Thanks Andrew!
-MC
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From msc at freeswitch.org  Mon Jan 11 16:57:02 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 16:57:02 -0800
Subject: [Freeswitch-users] Sending call notification tone to a busy FXS
	port
In-Reply-To: <08abf4fa1e11a07bda4381fda8f93879@thom.fr.eu.org>
References: <08abf4fa1e11a07bda4381fda8f93879@thom.fr.eu.org>
Message-ID: <87f2f3b91001111657g51f63d55yce889283fee655f6@mail.gmail.com>

On Mon, Jan 11, 2010 at 4:53 AM, Fran?ois Legal <devel at thom.fr.eu.org>wrote:

> Hello,
>
> I was just wondering if it is possible (and how) to send a call
> notification tone to a phone connected to an FXS port and which is already
> in communication.
>
> What is the interface? ATA or a TDM card? What model? Need to make sure
this feature is supported by your specific device.
-MC
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From carlos.talbot at gmail.com  Mon Jan 11 17:22:12 2010
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Mon, 11 Jan 2010 19:22:12 -0600
Subject: [Freeswitch-users] FSComm Windows build
Message-ID: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>

FYI,

there's a Windows pre-compiled binary of FSComm now available for those who
want to check it.

http://files.freeswitch.org/windows_installer/FSComm.exe

regards,

Carlos
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From msc at freeswitch.org  Mon Jan 11 17:42:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 17:42:00 -0800
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
Message-ID: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>

We want to let everyone know that FreeSWITCH now supports the Broadsoft SCA
method of doing shared lines. The story is here:

http://www.freeswitch.org/node/227

Tony and Brian spent many hours laboring over this, so please be sure to
show your appreciation to them for this new feature and all of the great
things they do for the FreeSWITCH community and VoIP in general!

-Michael
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From jcasale at activenetwerx.com  Mon Jan 11 18:24:50 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 12 Jan 2010 02:24:50 +0000
Subject: [Freeswitch-users] Gateway Configuration
In-Reply-To: <87f2f3b91001111644o766fccdap5ccf4fcde3a7536b@mail.gmail.com>
References: <CA5A491E9DEFBE4CB777DE97E21575E9074E14@prato.activenetwerx.local>
	<87f2f3b91001111644o766fccdap5ccf4fcde3a7536b@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9077A6F@prato.activenetwerx.local>

>I've only got a PAP2T (2 FXS) but from what you describe I'd say that it makes
>sense for the FXO port to be "phone_#_of_pstn" and FXS to be "ext_to_dial."
>I believe FS tries to be endpoint-type agnostic in this scenario. It's a SIP
>call in and gets routed in the dialplan. 

This is what I thought as well, where I am unclear is how you register this device
as a gateway then as it isn't like a regular sip provider by means of a gateway
definition? The gateway definition requires a username and password for example
so how do you create the basic definition to allow the spa3102 to push the call into
the public context?

I can make an outbound call as I just construct a dialplan like:
"sofia/internal/$1 at spa3102.domain.local:5060"

>>Incoming sip did's and a zap line I had all were configured so that
>>they entered the public context filtered by <phone_#>.
>If a call is coming in from an actual PSTN line then hitting the public context makes a lot of sense.? 

>Out of curiosity, what is your application?

We use this one pstn line as backup when voip is down for whatever reason. The
company # given out is the single pstn line, its routed on busy to the sip provider
for a series of other lines. As we expect the pstn to always be working, if the
voip is up and more people call we get the calls. If voip is down, at least we have
1 line always working w/o any manual intervention. Outgoing always tries sip first
and then routes to the pstn line if required.

Thanks,
jlc



From Mailings at kh-dev.de  Mon Jan 11 19:01:27 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Tue, 12 Jan 2010 04:01:27 +0100
Subject: [Freeswitch-users] proxy_media seems to be broken
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403B145DC1941@srv02.khdev.corp>

Hi,

I just tested with the latest tarball (11. Jan).

And now my T.38 Fax config with proxy_media doesn't work anymore.

Here's the config:
Fax <-> Cisco SPA2102 <-> FS <-> Lancom SIP/ISDN-Gateway <-> ISDN
proxy_media and late negotiation is turned on.

Here's what happens:
- Fax makes call
- Lancom routes the call to ISDN
- Remote fax takes the call
- Lancom sends reinvite with T.38
- The Cisco just keeps on ringing without noticing that the call was answered

This config worked in older FS versions without problems.

Might this already be fixed in rev 16250 (bypass_media problem)?
Or did the config of proxy_media change?

Thanks, Klaus
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From msc at freeswitch.org  Mon Jan 11 19:26:34 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jan 2010 19:26:34 -0800
Subject: [Freeswitch-users] Gateway Configuration
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9077A6F@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E9074E14@prato.activenetwerx.local>
	<87f2f3b91001111644o766fccdap5ccf4fcde3a7536b@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E9077A6F@prato.activenetwerx.local>
Message-ID: <87f2f3b91001111926xcb72feepf147c45697b38c8f@mail.gmail.com>

On Mon, Jan 11, 2010 at 6:24 PM, Joseph L. Casale <jcasale at activenetwerx.com
> wrote:

> >I've only got a PAP2T (2 FXS) but from what you describe I'd say that it
> makes
> >sense for the FXO port to be "phone_#_of_pstn" and FXS to be
> "ext_to_dial."
> >I believe FS tries to be endpoint-type agnostic in this scenario. It's a
> SIP
> >call in and gets routed in the dialplan.
>
> This is what I thought as well, where I am unclear is how you register this
> device
> as a gateway then as it isn't like a regular sip provider by means of a
> gateway
> definition? The gateway definition requires a username and password for
> example
> so how do you create the basic definition to allow the spa3102 to push the
> call into
> the public context?
>

Please email me off list and we'll work out the particulars and then create
a more complete wiki entry.
-MC
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From djbinter at yahoo.com  Mon Jan 11 19:34:51 2010
From: djbinter at yahoo.com (DJB)
Date: Mon, 11 Jan 2010 19:34:51 -0800 (PST)
Subject: [Freeswitch-users] proxy_media seems to be broken
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F0403B145DC1941@srv02.khdev.corp>
References: <A2F1C0E212BBBA4E8C0E7BE568361F0403B145DC1941@srv02.khdev.corp>
Message-ID: <608345.82340.qm@web37502.mail.mud.yahoo.com>

Yes, it has been fixed in 16250.



________________________________
From: Klaus Hochlehnert <Mailings at kh-dev.de>
To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
Sent: Mon, January 11, 2010 7:01:27 PM
Subject: [Freeswitch-users] proxy_media seems to be broken

 
Hi,
 
I just tested with the latest tarball (11. Jan).
 
And now my T.38 Fax config with proxy_media
doesn?t work anymore.
 
Here?s the config:
Fax <-> Cisco SPA2102 <-> FS
<-> Lancom SIP/ISDN-Gateway <-> ISDN
proxy_media and late negotiation is turned
on.
 
Here?s what happens:
- Fax makes call
- Lancom routes the call to ISDN
- Remote fax takes the call
- Lancom sends reinvite with T.38
- The Cisco just keeps on ringing without
noticing that the call was answered
 
This config worked in older FS versions without
problems.
 
Might this already be fixed in rev 16250 (bypass_media
problem)?
Or did the config of proxy_media change?
 
Thanks, Klaus


      
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From a.alalousi at gmail.com  Mon Jan 11 23:56:36 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Tue, 12 Jan 2010 07:56:36 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
	<87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
Message-ID: <cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>

Hi Michael,

This is exactly what I'm doing, but it's just not happening.

Thanks,

Ahmed.


2010/1/12 Michael Collins <msc at freeswitch.org>

>
>
> On Mon, Jan 11, 2010 at 4:18 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>
>> Dear All,
>>
>> I posted a thread re the subject but didn't get any joy, so perhaps second
>> time lucky.
>>
>> I need to rewrite a couple of hangup causes to mean NORMAL_CONGESTION and
>> no matter which variables I set, this isn't happening. The idea is to
>> control what codes are returned to an end point after a successful bridge,
>> as well as deal with what codes are returned if the bridge is unsuccessful
>> (e.g. user_busy, originator_cancel ...etc).
>>
>> I've had limited success by setting hangup_after_bridge=false then
>> bridging to error/<required_code>. This, however only works when the B-leg
>> terminates the call after a successful answer. Any other codes are not
>> rewritten.
>>
>> I've also tried playing with the bridge_hangup_code and hangup_code
>> variables prior and after bridging, still no joy. I have also set
>> sip_ignore_remote_cause=true prior to entering the bridge, as well
>> explicitly in vars.xml.
>>
>> By the way, I'm running in proxy-media mode, but I did try it with
>> bypass-media as well. Same symptoms, same behaviour.
>>
>> Any help with this would be highly appreciated.
>>
>> Well, I do know that when you do a hangup in the dialplan you can pass an
> optional cause as well:
> <action application="hangup" data="USER_BUSY"/>
> If you are doing the hanging up then you have a fair amount of control...
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ahmed Naji
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From steveayre at gmail.com  Tue Jan 12 00:46:27 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 12 Jan 2010 08:46:27 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
	<87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
	<cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>
Message-ID: <eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>

Can you show us the dialplan extension you're trying?

Thanks,
-Steve

2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
> Hi Michael,
>
> This is exactly what I'm doing, but it's just not happening.
>
> Thanks,
>
> Ahmed.
>
>
> 2010/1/12 Michael Collins <msc at freeswitch.org>
>>
>>
>> On Mon, Jan 11, 2010 at 4:18 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:
>>>
>>> Dear All,
>>>
>>> I posted a thread re the subject but didn't get any joy, so perhaps
>>> second time lucky.
>>>
>>> I need to rewrite a couple of hangup causes to mean NORMAL_CONGESTION and
>>> no matter which variables I set, this isn't happening. The idea is to
>>> control what codes are returned to an end point after a successful bridge,
>>> as well as deal with what codes are returned if the bridge is unsuccessful
>>> (e.g. user_busy, originator_cancel ...etc).
>>>
>>> I've had limited success by setting hangup_after_bridge=false then
>>> bridging to error/<required_code>. This, however only works when the B-leg
>>> terminates the call after a successful answer. Any other codes are not
>>> rewritten.
>>>
>>> I've also tried playing with the bridge_hangup_code and hangup_code
>>> variables prior and after bridging, still no joy. I have also set
>>> sip_ignore_remote_cause=true prior to entering the bridge, as well
>>> explicitly in vars.xml.
>>>
>>> By the way, I'm running in proxy-media mode, but I did try it with
>>> bypass-media as well. Same symptoms, same behaviour.
>>>
>>> Any help with this would be highly appreciated.
>>>
>> Well, I do know that when you do a hangup in the dialplan you can pass an
>> optional cause as well:
>> <action application="hangup" data="USER_BUSY"/>
>> If you are doing the hanging up then you have a fair amount of control...
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From devel at thom.fr.eu.org  Tue Jan 12 01:25:11 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Tue, 12 Jan 2010 10:25:11 +0100
Subject: [Freeswitch-users] Sending call notification tone to a busy FXS
	port
In-Reply-To: <87f2f3b91001111657g51f63d55yce889283fee655f6@mail.gmail.com>
References: <08abf4fa1e11a07bda4381fda8f93879@thom.fr.eu.org>
	<87f2f3b91001111657g51f63d55yce889283fee655f6@mail.gmail.com>
Message-ID: <1b556d9a7c2b1d64fd745170e5b172c9@thom.fr.eu.org>



This is a TDM Sangoma A400. 

I will check with Sangoma. If it is
supported by HW, is it supported by openzap ? 

Fran?ois 

On Mon, 11 Jan
2010 16:57:02 -0800, Michael Collins  wrote: 

 On Mon, Jan 11, 2010 at
4:53 AM, Fran?ois Legal  wrote:
 Hello,

 I was just wondering if it is
possible (and how) to send a call
 notification tone to a phone connected
to an FXS port and which is already
 in communication.

 What is the
interface? ATA or a TDM card? What model? Need to make sure this feature is
supported by your specific device.
-MC  

 

Links:
------
[1]
mailto:devel at thom.fr.eu.org
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From mcampbellsmith at gmail.com  Tue Jan 12 01:32:28 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 12 Jan 2010 20:32:28 +1100
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
	<33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
Message-ID: <33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>

I've updated and tested bypass_media.

It works if I remove this line from the B leg dialplan (ie 2010 calls
1000 - this is in the 1000 section of the dialplan):
                        <action application="tone_detect" data="fax
1100 r +5000 transfer 'receivefax XML features' 1" />

Does bypass_media work with tone_detect?

Thanks!

On Tue, Jan 12, 2010 at 11:14 AM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Beauty! ?Thanks.. will update now.
>
> On Tue, Jan 12, 2010 at 11:08 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> fixed in 16250
>>
>> On Mon, Jan 11, 2010 at 5:22 PM, Eliot Gable <egable+freeswitch at gmail.com>
>> wrote:
>>>
>>> I can't say for certain. I know it's not in 16016. Wait for Anthony to
>>> fix the issue in the current version, then update and try again.
>>>
>>> On Mon, Jan 11, 2010 at 6:09 PM, Mark Campbell-Smith
>>> <mcampbellsmith at gmail.com> wrote:
>>> > Thanks Eliot. ?I'm using FreeSWITCH Version 1.0.trunk (16131). ? Do
>>> > you kno wif the fault was present in 16131?
>>> >
>>> > Cheers
>>> >
>>> > On Tue, Jan 12, 2010 at 10:02 AM, Eliot Gable
>>> > <egable+freeswitch at gmail.com> wrote:
>>> >> Assuming you're using SVN trunk, there is a bug that is likely causing
>>> >> your issue. Anthony is fixing it right now.
>>> >>
>>> >> On Mon, Jan 11, 2010 at 6:16 AM, Mark Campbell-Smith
>>> >> <mcampbellsmith at gmail.com> wrote:
>>> >>> Hi!
>>> >>> Hi!
>>> >>>
>>> >>> I am calling from extension 2010 to extension 1000. ?Both have ip
>>> >>> addesses 192.168.1.x.
>>> >>>
>>> >>> In the 2000 series dialplan (a separate context) I have the following
>>> >>> to try to enable bypass_media.
>>> >>>
>>> >>> ? ? ? ? ? ?<extension name="Local_1000_9xxx">
>>> >>> ? ? ? ? ? ? ? ? ?<condition field="destination_number"
>>> >>> expression="^(10[01][0-9]|9\d{3})$">
>>> >>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set"
>>> >>> data="dialed_extension=$1"/>
>>> >>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set"
>>> >>> data="proxy_media=false"/>
>>> >>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set"
>>> >>> data="bypass_media=true"/>
>>> >>> ? ? ? ? ? ? ? ? ? ? ? ?<action application="transfer"
>>> >>> data="${dialed_extension} XML default"/>
>>> >>> ? ? ? ? ? ? ? ? ?</condition>
>>> >>> ? ? ? ? ? ?</extension>
>>> >>>
>>> >>> Is this how bypass media should be enabled?
>>> >>>
>>> >>> This fails fo me (the calls hang up and no audio). ?The debug trace is
>>> >>> in http://pastebin.freeswitch.org/11737
>>> >>>
>>> >>> What have I done wrong?
>>> >>>
>>> >>> Thanks
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>
>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>>
>>> >>
>>> >>
>>> >>
>>> >> --
>>> >> Eliot Gable
>>> >>
>>> >> "We do not inherit the Earth from our ancestors: we borrow it from our
>>> >> children." ~David Brower
>>> >>
>>> >> "I decided the words were too conservative for me. We're not borrowing
>>> >> from our children, we're stealing from them--and it's not even
>>> >> considered to be a crime." ~David Brower
>>> >>
>>> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>>> >> live; not live to eat.) ~Marcus Tullius Cicero
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>>
>>>
>>> --
>>> Eliot Gable
>>>
>>> "We do not inherit the Earth from our ancestors: we borrow it from our
>>> children." ~David Brower
>>>
>>> "I decided the words were too conservative for me. We're not borrowing
>>> from our children, we're stealing from them--and it's not even
>>> considered to be a crime." ~David Brower
>>>
>>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>>> live; not live to eat.) ~Marcus Tullius Cicero
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>



From Claudio.Cavalera at italtel.it  Tue Jan 12 01:43:34 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Tue, 12 Jan 2010 10:43:34 +0100
Subject: [Freeswitch-users] playing with sessions in lua
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F362D9@BESONE.corp.dom>

Hello,
this should be simple in theory therefore I'm probably missing the right
way to do it.
I want to play with sessions in lua, bridge them, park them, etc...

example1: Consider this simple lua script in which i create two
sessions:

api = freeswitch.API();
api:execute("originate", "sofia/internal/1001%192.168.1.1 &park");
api:execute("originate", "sofia/internal/1002%192.168.1.1 &park");

now if i want to bridge them i suppose i should use something like

api:execute("uuid_bridge", "uuid_1 uuid_2");

but how do i get uuid_1 and uuid_2, i.e. the uuids of the two sessions?


example2: I could create sessions with

local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
local session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");

but then there is NOT a bridge API to bridge the sessions like:
bridge(session1, session2);

I admit I have not yet understood why such bridge possibility exist in
javascript but does not exist in lua.
http://wiki.freeswitch.org/wiki/Javascript_Misc_bridge
I guess there is a reason for this but I can't figure it out.


example3: yet another possibility
local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
session1:originate(session2, "sofia/internal/1002%192.168.1.1", 1000);

but it does not work either.

Besides with this third example something strange happen:

freeswitch at internal> show channels
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr
,dest,application,application_data,dialplan,context,read_codec,read_rate
,write_codec,write_rate,secure
1c5db2df-14ce-4516-94f2-bb7c087e0802,outbound,2010-01-12
10:50:59,1263289859,sofia/internal/1004,CS_REPORTING,FreeSWITCH,00000000
00,,1004,,,,default,PCMA,8000,PCMA,8000,

1 total.

freeswitch at internal> uuid_kill 1c5db2df-14ce-4516-94f2-bb7c087e0802
-ERR No Such Channel!

freeswitch at internal> show channels
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr
,dest,application,application_data,dialplan,context,read_codec,read_rate
,write_codec,write_rate,secure
1c5db2df-14ce-4516-94f2-bb7c087e0802,outbound,2010-01-12
10:50:59,1263289859,sofia/internal/1004,CS_REPORTING,FreeSWITCH,00000000
00,,1004,,,,default,PCMA,8000,PCMA,8000,

1 total.

freeswitch at internal>

If you are interested the full log is here:
http://pastebin.freeswitch.org/11757
but I admit i'm not on latest trunk yet!

Thanks.
Ciao,
Claudio


PS: Is there a reason why there is a uuid_park command but not
uuid_valet_park ?


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From a.alalousi at gmail.com  Tue Jan 12 01:52:08 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Tue, 12 Jan 2010 09:52:08 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
	<87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
	<cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>
	<eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>
Message-ID: <cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>

Here you go:

<condition field="destination_number" expression="^(?:5824)(44(1|2)\d+)$"
break="on-true">
  <action inline="true" application="set" data="bypass_media=false"/>
  <action inline="true" application="set" data="prpoxy_media=false"/>
  <action inline="true" application="set" data="disable_q850_reason=true"/>
  <action inline="true" application="set" data="hangup_after_bridge=false"/>
  <action inline="true" application="set"
data="sip_ignore_remote_cause=true"/>
  <action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1"/>
  <action inline="true" application="set"
data="bridge_hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
  <action inline="true" application="set"
data="hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
  <action application="bridge" data="error/normal_circuit_congestion"/>
  <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
</condition>

As you can see, I am trying to rewrite the hangup codes in a multitude of
ways and places, but still exhibit the same behaviour.

Any help appreciated.

Regards,

Ahmed.

2010/1/12 Steven Ayre <steveayre at gmail.com>

> Can you show us the dialplan extension you're trying?
>
> Thanks,
> -Steve
>
> 2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
> > Hi Michael,
> >
> > This is exactly what I'm doing, but it's just not happening.
> >
> > Thanks,
> >
> > Ahmed.
> >
> >
> > 2010/1/12 Michael Collins <msc at freeswitch.org>
> >>
> >>
> >> On Mon, Jan 11, 2010 at 4:18 AM, Ahmed Naji <a.alalousi at gmail.com>
> wrote:
> >>>
> >>> Dear All,
> >>>
> >>> I posted a thread re the subject but didn't get any joy, so perhaps
> >>> second time lucky.
> >>>
> >>> I need to rewrite a couple of hangup causes to mean NORMAL_CONGESTION
> and
> >>> no matter which variables I set, this isn't happening. The idea is to
> >>> control what codes are returned to an end point after a successful
> bridge,
> >>> as well as deal with what codes are returned if the bridge is
> unsuccessful
> >>> (e.g. user_busy, originator_cancel ...etc).
> >>>
> >>> I've had limited success by setting hangup_after_bridge=false then
> >>> bridging to error/<required_code>. This, however only works when the
> B-leg
> >>> terminates the call after a successful answer. Any other codes are not
> >>> rewritten.
> >>>
> >>> I've also tried playing with the bridge_hangup_code and hangup_code
> >>> variables prior and after bridging, still no joy. I have also set
> >>> sip_ignore_remote_cause=true prior to entering the bridge, as well
> >>> explicitly in vars.xml.
> >>>
> >>> By the way, I'm running in proxy-media mode, but I did try it with
> >>> bypass-media as well. Same symptoms, same behaviour.
> >>>
> >>> Any help with this would be highly appreciated.
> >>>
> >> Well, I do know that when you do a hangup in the dialplan you can pass
> an
> >> optional cause as well:
> >> <action application="hangup" data="USER_BUSY"/>
> >> If you are doing the hanging up then you have a fair amount of
> control...
> >> -MC
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Ahmed Naji
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From jason at jasonjgw.net  Tue Jan 12 01:53:32 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 12 Jan 2010 20:53:32 +1100
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
	<33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
	<33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>
Message-ID: <20100112095332.GA32294@jdc.jasonjgw.net>

Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> I've updated and tested bypass_media.
> 
> It works if I remove this line from the B leg dialplan (ie 2010 calls
> 1000 - this is in the 1000 section of the dialplan):
>                         <action application="tone_detect" data="fax
> 1100 r +5000 transfer 'receivefax XML features' 1" />
> 
> Does bypass_media work with tone_detect?

As I understand it, tone_detect detects tones in the RTP stream (i.e., in the
audio). For this to be possible, FreeSWITCH has to be in the audio path, hence
bypass media cannot be used

If this reasoning isn't obvious to you, then you've misunderstood what
tone_detect does or what bypass media is (the audio traffic flows directly
between the two endpoints without passing through the FreeSWITCH system that
establishes the connection, therefore FreeSWITCH can't process it to detect
tones and consequently bypass media and tone detection are inherently
incompatible.)




From mcampbellsmith at gmail.com  Tue Jan 12 02:23:33 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 12 Jan 2010 21:23:33 +1100
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <20100112095332.GA32294@jdc.jasonjgw.net>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
	<33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
	<33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>
	<20100112095332.GA32294@jdc.jasonjgw.net>
Message-ID: <33c87fa31001120223t13efc07agcc595e48405fd8bb@mail.gmail.com>

Hi Jason,

I have understood that.  Its not that a difficult concept to understand!

In the log I see:
2010-01-12 21:03:17.585598 [DEBUG] mod_dptools.c:818
sofia/internal/1000 at 192.168.1.120 SET
[ringback]=[v=-7;%(400,200,413,438);%(400,2000,413,438)]
2010-01-12 21:03:17.605591 [DEBUG] switch_core_session.c:1509
Application tone_detect Requires media! pre_answering channel
sofia/internal/1000 at 192.168.1.120

I thought the SIP re-Invite message can be used to update media
parameters, including IP address endpoints.  Does FS try too do this
in the case that tone_detect is used?

On Tue, Jan 12, 2010 at 8:53 PM, Jason White <jason at jasonjgw.net> wrote:
> Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>> I've updated and tested bypass_media.
>>
>> It works if I remove this line from the B leg dialplan (ie 2010 calls
>> 1000 - this is in the 1000 section of the dialplan):
>> ? ? ? ? ? ? ? ? ? ? ? ? <action application="tone_detect" data="fax
>> 1100 r +5000 transfer 'receivefax XML features' 1" />
>>
>> Does bypass_media work with tone_detect?
>
> As I understand it, tone_detect detects tones in the RTP stream (i.e., in the
> audio). For this to be possible, FreeSWITCH has to be in the audio path, hence
> bypass media cannot be used
>
> If this reasoning isn't obvious to you, then you've misunderstood what
> tone_detect does or what bypass media is (the audio traffic flows directly
> between the two endpoints without passing through the FreeSWITCH system that
> establishes the connection, therefore FreeSWITCH can't process it to detect
> tones and consequently bypass media and tone detection are inherently
> incompatible.)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From dome at tel.co.th  Tue Jan 12 03:13:12 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 12 Jan 2010 18:13:12 +0700
Subject: [Freeswitch-users] CDR and reporting.
Message-ID: <8ccbff061001120313u78701883lf9e212947af0528@mail.gmail.com>

Dear All,
           I have 200k cdr record daily. what's good solution to
record and report ?
Now I'm thinking about CDRTOOL from ag project and areski stat (now
part of a2billing).
My idea use mod_xml_cdr -> http (may be php, or perlembded on nginx) -> mysql

i want to report ASR, ACD ,
any comment ?

Dome C.



From sos at sokhapkin.dyndns.org  Tue Jan 12 03:39:37 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 12 Jan 2010 06:39:37 -0500
Subject: [Freeswitch-users] CDR and reporting.
In-Reply-To: <8ccbff061001120313u78701883lf9e212947af0528@mail.gmail.com>
References: <8ccbff061001120313u78701883lf9e212947af0528@mail.gmail.com>
Message-ID: <201001120639.38098.sos@sokhapkin.dyndns.org>

Look at default config for mod_cdr_csv, sql template. It can produce a 
sequence of SQL INSERT statements which can be later fed (after daily log 
rotation) to mysql client program.

I see no reason to employ xml for this task, the language hard to read and 
parse for both human and computer.

On Tuesday 12 January 2010, Dome Charoenyost wrote:
> Dear All,
>            I have 200k cdr record daily. what's good solution to
> record and report ?
> Now I'm thinking about CDRTOOL from ag project and areski stat (now
> part of a2billing).
> My idea use mod_xml_cdr -> http (may be php, or perlembded on nginx) ->
> mysql
>
> i want to report ASR, ACD ,
> any comment ?
>
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From Russell.Mosemann at cune.org  Tue Jan 12 04:25:47 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Tue, 12 Jan 2010 06:25:47 -0600
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com><87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com><cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com><eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>
	<cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>
Message-ID: <FDE78FE432FE49D091FA62A0BE7A3CCD@cune.pri>

>   <action inline="true" application="set" data="prpoxy_media=false"/>

prpoxy? ;-)

--
Russell Mosemann




From a.alalousi at gmail.com  Tue Jan 12 05:07:26 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Tue, 12 Jan 2010 13:07:26 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <FDE78FE432FE49D091FA62A0BE7A3CCD@cune.pri>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
	<87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
	<cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>
	<eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>
	<cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>
	<FDE78FE432FE49D091FA62A0BE7A3CCD@cune.pri>
Message-ID: <cd93b3701001120507j941367etc70d56cd92ccd398@mail.gmail.com>

;)


2010/1/12 Russell Mosemann <Russell.Mosemann at cune.org>

> >   <action inline="true" application="set" data="prpoxy_media=false"/>
>
> prpoxy? ;-)
>
> --
> Russell Mosemann
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From steveayre at gmail.com  Tue Jan 12 05:20:45 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 12 Jan 2010 13:20:45 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
	<87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
	<cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>
	<eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>
	<cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>
Message-ID: <eafa772f1001120520i719751b9hcbc22a969a89a585@mail.gmail.com>

- prpoxy_media should be proxy_media
- bypass_media and proxy_media shouldn't need setting to false -
that's their default (unless you're set one of them to true on the sip
profile?)
- why do you need to disable q850 reason?

I do something very similar - try this...

<condition field="destination_number" expression="^(?:5824)(44(1|2)\d+)$">
  <action application="set" data="continue_on_fail=true"/>
  <action application="set" data="hangup_after_bridge=true"/>
  <action application="set" data="sip_ignore_remote_cause=true"/>
  <action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1"/>
  <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
</condition>

By the way, you'll be unable to rewrite the hangup cause for a bridge
that gets a 180 or 183 packet from the gateway before getting a 4xx,
5xx or 6xx packet (because those bridges don't 'fail') - I've seen
this on a SIP provider before who gives 183 Session Progress before a
404 Not Found if the PSTN number dialled doesn't exist.

Regards,
-Steve


2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
> Here you go:
>
> <condition field="destination_number" expression="^(?:5824)(44(1|2)\d+)$"
> break="on-true">
> ? <action inline="true" application="set" data="bypass_media=false"/>
> ? <action inline="true" application="set" data="prpoxy_media=false"/>
> ? <action inline="true" application="set" data="disable_q850_reason=true"/>
> ? <action inline="true" application="set" data="hangup_after_bridge=false"/>
> ? <action inline="true" application="set"
> data="sip_ignore_remote_cause=true"/>
> ? <action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1"/>
> ? <action inline="true" application="set"
> data="bridge_hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
> ? <action inline="true" application="set"
> data="hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
> ? <action application="bridge" data="error/normal_circuit_congestion"/>
> ? <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
> </condition>
>
> As you can see, I am trying to rewrite the hangup codes in a multitude of
> ways and places, but still exhibit the same behaviour.
>
> Any help appreciated.
>
> Regards,
>
> Ahmed.
>
> 2010/1/12 Steven Ayre <steveayre at gmail.com>
>>
>> Can you show us the dialplan extension you're trying?
>>
>> Thanks,
>> -Steve
>>
>> 2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
>> > Hi Michael,
>> >
>> > This is exactly what I'm doing, but it's just not happening.
>> >
>> > Thanks,
>> >
>> > Ahmed.
>> >
>> >
>> > 2010/1/12 Michael Collins <msc at freeswitch.org>
>> >>
>> >>
>> >> On Mon, Jan 11, 2010 at 4:18 AM, Ahmed Naji <a.alalousi at gmail.com>
>> >> wrote:
>> >>>
>> >>> Dear All,
>> >>>
>> >>> I posted a thread re the subject but didn't get any joy, so perhaps
>> >>> second time lucky.
>> >>>
>> >>> I need to rewrite a couple of hangup causes to mean NORMAL_CONGESTION
>> >>> and
>> >>> no matter which variables I set, this isn't happening. The idea is to
>> >>> control what codes are returned to an end point after a successful
>> >>> bridge,
>> >>> as well as deal with what codes are returned if the bridge is
>> >>> unsuccessful
>> >>> (e.g. user_busy, originator_cancel ...etc).
>> >>>
>> >>> I've had limited success by setting hangup_after_bridge=false then
>> >>> bridging to error/<required_code>. This, however only works when the
>> >>> B-leg
>> >>> terminates the call after a successful answer. Any other codes are not
>> >>> rewritten.
>> >>>
>> >>> I've also tried playing with the bridge_hangup_code and hangup_code
>> >>> variables prior and after bridging, still no joy. I have also set
>> >>> sip_ignore_remote_cause=true prior to entering the bridge, as well
>> >>> explicitly in vars.xml.
>> >>>
>> >>> By the way, I'm running in proxy-media mode, but I did try it with
>> >>> bypass-media as well. Same symptoms, same behaviour.
>> >>>
>> >>> Any help with this would be highly appreciated.
>> >>>
>> >> Well, I do know that when you do a hangup in the dialplan you can pass
>> >> an
>> >> optional cause as well:
>> >> <action application="hangup" data="USER_BUSY"/>
>> >> If you are doing the hanging up then you have a fair amount of
>> >> control...
>> >> -MC
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > Ahmed Naji
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From pmhshz at gmail.com  Tue Jan 12 07:22:18 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Tue, 12 Jan 2010 20:52:18 +0530
Subject: [Freeswitch-users] Defunct process in ESL testserver example
Message-ID: <a792304b1001120722o2d4fbf91vb0456ef4f0436a5e@mail.gmail.com>

Hello everybody,

I am creating a C program of ESL outbound for call processing.
I am using testserver.c example, and till now it seems fine.

But i noticed that every call testserver process, a new process is being
created which I can see in Linux system with below command:
For example, when I make two calls and even after hangup, I saw three
process like below:
ps -A | grep testserver
 9345 pts/2    00:00:00 testserver
 9350 pts/2    00:00:00 testserver
 9357 pts/2    00:00:00 testserver

This get increased for every call i make.

I did some workout and placed below two lines (close & exit) at the end of
mycallback function, (as I found them on ivrd.c file):

    esl_disconnect(&handle);
    close(client_sock);
    exit(0);
}
But after that the process becomes defunct/zombie

 9440 pts/2    00:00:00 conflisten
 9442 pts/2    00:00:00 conflisten <defunct>
 9452 pts/2    00:00:00 conflisten <defunct>


Will anybody please suggest me how can I eliminate this process, which
remains in memory even after call hangup?

Thanks for any response.
MohammedShehzad
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From brian at freeswitch.org  Tue Jan 12 07:53:03 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 09:53:03 -0600
Subject: [Freeswitch-users] proxy_media seems to be broken
In-Reply-To: <608345.82340.qm@web37502.mail.mud.yahoo.com>
References: <A2F1C0E212BBBA4E8C0E7BE568361F0403B145DC1941@srv02.khdev.corp>
	<608345.82340.qm@web37502.mail.mud.yahoo.com>
Message-ID: <194CB7A1-D382-4F98-AB9A-21AFFABFFD6B@freeswitch.org>

And the tarball is updated already automatically too.  Please update to the latest FreeSWITCH... report any issues to jira if you have them in the future.

In the future please read thru the mailing list as this was discussed in two different threads yesterday with the details and the rev where it was fixed. 

Thanks,
/b

On Jan 11, 2010, at 9:34 PM, DJB wrote:

> Yes, it has been fixed in 16250.
> 

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From mailinglist at fribert.dk  Tue Jan 12 08:04:59 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Tue, 12 Jan 2010 17:04:59 +0100
Subject: [Freeswitch-users] Multi-Homed setup,
	starting over - still not working
Message-ID: <4B4CABBB020000E10000038E@mail.fribert.dk>

Hi Guys
 
I really would like to have this up and running, but I'm constantly running into things that doesn't work, and I have no idea where the problem is...
 
I've started from scratch with pfSense and Freeswitch.
I've followed the Multi Home tutorial here: http://wiki.freeswitch.org/wiki/Multi_home_tutorial
 
I've deleted the 'lan' profile, and altered the 'internal' profile as recommended, and I've gotten to the paragraph:
APPLYING YOUR CHANGES AND CHECKING YOUR WORK
When I start up my x-lite program I get this error:
2010-01-12 16:38:54.172731 [WARNING] sofia_reg.c:1755 Can't find user [1000 at 83.89.x.x]
You must define a domain called '83.89.x.xin your directory and add a user with the id="1000" attribute
and you must configure your device to use the proper domain in it's authentication credentials.
and you must configure your device to use the proper domain in it's authentication credentials.
 
83.89.x.x is my external IP, and not my internal IP???
 
Any help on this?
 
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From anthony.minessale at gmail.com  Tue Jan 12 08:34:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Jan 2010 10:34:23 -0600
Subject: [Freeswitch-users] Defunct process in ESL testserver example
In-Reply-To: <a792304b1001120722o2d4fbf91vb0456ef4f0436a5e@mail.gmail.com>
References: <a792304b1001120722o2d4fbf91vb0456ef4f0436a5e@mail.gmail.com>
Message-ID: <191c3a031001120834q36f72791s811b6352ab7c12eb@mail.gmail.com>

It's forking process code.
you need to ignore sigchld or create a sig handler for sigchld and use the
wait syscall to reap the process.


On Tue, Jan 12, 2010 at 9:22 AM, MohammedShehzad <pmhshz at gmail.com> wrote:

> Hello everybody,
>
> I am creating a C program of ESL outbound for call processing.
> I am using testserver.c example, and till now it seems fine.
>
> But i noticed that every call testserver process, a new process is being
> created which I can see in Linux system with below command:
> For example, when I make two calls and even after hangup, I saw three
> process like below:
> ps -A | grep testserver
>  9345 pts/2    00:00:00 testserver
>  9350 pts/2    00:00:00 testserver
>  9357 pts/2    00:00:00 testserver
>
> This get increased for every call i make.
>
> I did some workout and placed below two lines (close & exit) at the end of
> mycallback function, (as I found them on ivrd.c file):
>
>     esl_disconnect(&handle);
>     close(client_sock);
>     exit(0);
> }
> But after that the process becomes defunct/zombie
>
>  9440 pts/2    00:00:00 conflisten
>  9442 pts/2    00:00:00 conflisten <defunct>
>  9452 pts/2    00:00:00 conflisten <defunct>
>
>
> Will anybody please suggest me how can I eliminate this process, which
> remains in memory even after call hangup?
>
> Thanks for any response.
> MohammedShehzad
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at van.lammeren.net  Tue Jan 12 08:50:07 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Tue, 12 Jan 2010 11:50:07 -0500
Subject: [Freeswitch-users] How to provide dynamic directory information?
In-Reply-To: <87f2f3b91001111636j1613b6ak5b5de86804f2125e@mail.gmail.com>
References: <5d2828f1001111453t8044774w3cc882988e18ced@mail.gmail.com>
	<87f2f3b91001111636j1613b6ak5b5de86804f2125e@mail.gmail.com>
Message-ID: <5d2828f1001120850qb676df6wfecbb30c05e65f23@mail.gmail.com>

Hi Michael!

I've had a look at your blog. Good stuff!

Thanks!

On Mon, Jan 11, 2010 at 7:36 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Jan 11, 2010 at 2:53 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:
>
>> Hello!
>>
>> I'd like to be able to have FreeSWITCH check a database for authorization,
>> every time a user registers. There are some great examples on the wiki,
>> which use either MOD_XML_CURL or Lua to dynamically provide a dialplan, but
>> I cannot find an example for providing a directory.
>>
>> I am developing an application that will have thousands of users, and will
>> run on multiple FreeSWITCH servers behind a load balancer. Ideally,
>> FreeSWITCH would only look-up directory information, specifically, username
>> and password, whenever a user attempts to connect. The directory information
>> will be changing regularly, as users are added or removed from the system.
>>
>> Is this possible with FreeSWITCH? Or can only dialplan information be
>> provided dynamically?
>>
>> I've written a script in Lua that provides the XML data, such as that
>> found in the example /freeswitch/conf/directory/default/ folders, and I try
>> to call it with this bit of XML in /freeswitch/conf/directory/default.xml:
>>
>> <groups>
>> <group name="default">
>> <users>
>>  <param name="xml-handler-script" value="/directory.lua"/>
>> <param name="xml-handler-bindings" value="users"/>
>>  </users>
>> </groups>
>>
>> Is this the right approach? Am I going about this the right way?
>>
>
> You can bind "directory" as well as "dialplan" and a few others. I
> personally don't use xml_curl in production but for kicks I tried to learn
> it and I documented some of my journey on my personal blog. (
> http://telecommusings.blogspot.com/)
>
> xml_curl was designed to scale and be applied in your type of scenario.
> Raymond (intralanman on IRC) has played with it quite a bit as have a number
> of others.
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jcasale at activenetwerx.com  Tue Jan 12 09:29:18 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 12 Jan 2010 17:29:18 +0000
Subject: [Freeswitch-users] Multi-Homed setup,
 starting over - still not working
In-Reply-To: <4B4CABBB020000E10000038E@mail.fribert.dk>
References: <4B4CABBB020000E10000038E@mail.fribert.dk>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E907873D@prato.activenetwerx.local>

>Hi Guys
>?
>I really would like to have this up and running, but I'm constantly running into things that doesn't work, and I have no idea where the problem is...
>?
>I've started from scratch with pfSense and Freeswitch.
>I've followed the Multi Home tutorial here: http://wiki.freeswitch.org/wiki/Multi_home_tutorial
>?
>I've deleted the 'lan' profile, and altered the 'internal' profile as recommended, and I've gotten to the paragraph:
>
>APPLYING YOUR CHANGES AND CHECKING YOUR WORK
>
>When I start up my x-lite program I get this error:
>2010-01-12 16:38:54.172731 [WARNING] sofia_reg.c:1755 Can't find user [1000 at 83.89.x.x]
>You must define a domain called '83.89.x.xin your directory and add a user with the id="1000" attribute
>and you must configure your device to use the proper domain in it's authentication credentials.
>and you must configure your device to use the proper domain in it's authentication credentials.
>?
>83.89.x.x is my external IP, and not my internal IP???
>
>Any help on this?

This is because you haven't set your domain in vars.xml. The behavior is that
$${local_ip_v4} evals to your wan ip. This is the first step in that tutorial:)
http://wiki.freeswitch.org/wiki/Multi_home_tutorial#INTERNAL_LAN

Open vars.xml, make the line:
  <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
  <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>

Match your lan ip:
  <X-PRE-PROCESS cmd="set" data="domain=192.168.13.1"/>
  <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
?
restart fs, then goto the fs_cli and type `eval ${domain}` it should
come back with "your" lan ip.



From xanlich at gmail.com  Tue Jan 12 09:08:08 2010
From: xanlich at gmail.com (Chia-Yen Wu)
Date: Wed, 13 Jan 2010 01:08:08 +0800
Subject: [Freeswitch-users] Hello , about calculate numbers
Message-ID: <314dc3f81001120908l5772b13fqd676a3be367bb662@mail.gmail.com>

Hello everyone,
Is there anyway to calculate number in dialplan.xml?

like example:
<action application="set" data="var=1"/>
and with an action , I can do : var=var+1

I have tried mod_expr, but failed to catch the variable, like below:
<action application="set" data="var=${expr($var+1)}"/>
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From sos at sokhapkin.dyndns.org  Tue Jan 12 09:39:06 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 12 Jan 2010 12:39:06 -0500
Subject: [Freeswitch-users] Hello , about calculate numbers
In-Reply-To: <314dc3f81001120908l5772b13fqd676a3be367bb662@mail.gmail.com>
References: <314dc3f81001120908l5772b13fqd676a3be367bb662@mail.gmail.com>
Message-ID: <201001121239.06756.sos@sokhapkin.dyndns.org>

<action application="set" data="var=${expr(${var}+1)}"/>

On Tuesday 12 January 2010, Chia-Yen Wu wrote:
> Hello everyone,
> Is there anyway to calculate number in dialplan.xml?
>
> like example:
> <action application="set" data="var=1"/>
> and with an action , I can do : var=var+1
>
> I have tried mod_expr, but failed to catch the variable, like below:
> <action application="set" data="var=${expr($var+1)}"/>





From jcasale at activenetwerx.com  Tue Jan 12 10:20:21 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 12 Jan 2010 18:20:21 +0000
Subject: [Freeswitch-users] Fax Codecs
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E907880A@prato.activenetwerx.local>

While faxing is working, I am seeing these in the log when I
attempt to use rxfax:

switch_ivr_play_say.c:1154 Codec Activated L16 at 8000hz 1 channels 30ms
switch_core_io.c:652 sofia/internal/1002 at 192.168.13.1 receive message [TRANSCODING_NECESSARY]

So later I see:
switch_core_codec.c:128 sofia/internal/1002 at 192.168.13.1 Restore previous codec PCMU:0

So I am actually not clear on what's happening? I thought I had the ua set to PCMU but I
guess fs is seeing the incoming sip session as L16 which I read is bad for fax? What's
the significance of the last line where it states its restoring the old codec?

Can anyone shed some light on what's actually happening?

Thanks!
jlc




From steveu at coppice.org  Tue Jan 12 10:32:34 2010
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 13 Jan 2010 02:32:34 +0800
Subject: [Freeswitch-users] Fax Codecs
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E907880A@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E907880A@prato.activenetwerx.local>
Message-ID: <4B4CC042.3090404@coppice.org>

On 01/13/2010 02:20 AM, Joseph L. Casale wrote:
> While faxing is working, I am seeing these in the log when I
> attempt to use rxfax:
>
> switch_ivr_play_say.c:1154 Codec Activated L16 at 8000hz 1 channels 30ms
> switch_core_io.c:652 sofia/internal/1002 at 192.168.13.1 receive message [TRANSCODING_NECESSARY]
>
> So later I see:
> switch_core_codec.c:128 sofia/internal/1002 at 192.168.13.1 Restore previous codec PCMU:0
>
> So I am actually not clear on what's happening? I thought I had the ua set to PCMU but I
> guess fs is seeing the incoming sip session as L16 which I read is bad for fax? What's
> the significance of the last line where it states its restoring the old codec?
>
> Can anyone shed some light on what's actually happening?
>    
A-law and u-law are the right things for the transmission of FAX 
signals, but you can't actually do signal processing on an A-law or 
u-law signal. It needs to be converted to a 16 bit linear signal (i.e. 
L16) for that. What you see is the correct action.

Steve




From jmesquita at freeswitch.org  Tue Jan 12 11:51:12 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 12 Jan 2010 17:51:12 -0200
Subject: [Freeswitch-users] FSComm Windows build
In-Reply-To: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
References: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
Message-ID: <ebbc2d551001121151v5593ee83m96aabea917ea12f5@mail.gmail.com>

Thank you very much Carlos for your support.

All FSComm testers. Please, beware that FSComm has released its project and
not the real thing! We are still on the early ages of development and don't
expect everything to work! Nonetheless, what is not working has to be
reported. :-)

Thank you,

Jo?o Mesquita


On Mon, Jan 11, 2010 at 11:22 PM, Carlos Talbot <carlos.talbot at gmail.com>wrote:

> FYI,
>
> there's a Windows pre-compiled binary of FSComm now available for those who
> want to check it.
>
> http://files.freeswitch.org/windows_installer/FSComm.exe
>
> regards,
>
> Carlos
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From john at acsol.net  Tue Jan 12 12:08:53 2010
From: john at acsol.net (John)
Date: Tue, 12 Jan 2010 13:08:53 -0700
Subject: [Freeswitch-users] ShoreTel to FS connection
Message-ID: <4B4CD6D5.1010104@acsol.net>

I am trying to utilize a freeswitch system as a SIP trunk for ShoreTel. 
I have had to change the port from 5080 to 5060. When I dial through, I 
see on the FS console show the call; however I can't seem to change the 
public.xml dialplan to effect any changes in the behaviour. How can I 
tell which dialplan the external call is using?

  2010-01-12 13:04:17.705548 [NOTICE] switch_channel.c:613 New Channel 
sofia/external/+19705551212 at 192.168.155.13:5060 
[aa2f2fde-ffb5-11de-9590-015cd8b8c273]
2010-01-12 13:04:17.705548 [INFO] mod_dialplan_xml.c:408 Processing 
Extension 1002->5551212 in context public
2010-01-12 13:04:17.705548 [NOTICE] switch_core_state_machine.c:187 
Hangup sofia/external/+19705551212 at 192.168.155.13:5060 [CS_EXECUTE] 
[NORMAL_CLEARING

Thanks John



From peder at networkoblivion.com  Tue Jan 12 12:11:36 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 12 Jan 2010 14:11:36 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
Message-ID: <064e01ca93c3$71b259a0$55170ce0$@com>

Anybody know the settings on the SPA5xx to make SCA work?  I've tried all
sorts of combinations of shared/private on the ext and phone and set the
Server type to Broadsoft, etc and  it just never seems to work.  Do I need
the BLF info?  Do I set shared/private on the phone, or ext, or both?  Do I
need to enable SCA Line ID Mapping?  Or SCA Barge-In Enable?  Or both?

Peder





From brian at freeswitch.org  Tue Jan 12 12:17:07 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 14:17:07 -0600
Subject: [Freeswitch-users] ShoreTel to FS connection
In-Reply-To: <4B4CD6D5.1010104@acsol.net>
References: <4B4CD6D5.1010104@acsol.net>
Message-ID: <38D0A853-889C-4F53-B768-53C1283C8B5C@freeswitch.org>

Its looking for 5551212 in context public.  you can ONLY communicate with shoretel on port 5060

/b

On Jan 12, 2010, at 2:08 PM, John wrote:

> I am trying to utilize a freeswitch system as a SIP trunk for ShoreTel. 
> I have had to change the port from 5080 to 5060. When I dial through, I 
> see on the FS console show the call; however I can't seem to change the 
> public.xml dialplan to effect any changes in the behaviour. How can I 
> tell which dialplan the external call is using?
> 
>  2010-01-12 13:04:17.705548 [NOTICE] switch_channel.c:613 New Channel 
> sofia/external/+19705551212 at 192.168.155.13:5060 
> [aa2f2fde-ffb5-11de-9590-015cd8b8c273]
> 2010-01-12 13:04:17.705548 [INFO] mod_dialplan_xml.c:408 Processing 
> Extension 1002->5551212 in context public
> 2010-01-12 13:04:17.705548 [NOTICE] switch_core_state_machine.c:187 
> Hangup sofia/external/+19705551212 at 192.168.155.13:5060 [CS_EXECUTE] 
> [NORMAL_CLEARING
> 
> Thanks John
> 
> _____________




From brian at freeswitch.org  Tue Jan 12 12:19:13 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 14:19:13 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <064e01ca93c3$71b259a0$55170ce0$@com>
References: <064e01ca93c3$71b259a0$55170ce0$@com>
Message-ID: <DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>

Enable line.
Set to share.

On phone tab.. mark it shared and select the line.

Attendant console set broadsoft.

save.

reboot
try

/b






On Jan 12, 2010, at 2:11 PM, Peder wrote:

> Anybody know the settings on the SPA5xx to make SCA work?  I've tried all
> sorts of combinations of shared/private on the ext and phone and set the
> Server type to Broadsoft, etc and  it just never seems to work.  Do I need
> the BLF info?  Do I set shared/private on the phone, or ext, or both?  Do I
> need to enable SCA Line ID Mapping?  Or SCA Barge-In Enable?  Or both?
> 
> Peder




From peder at networkoblivion.com  Tue Jan 12 12:32:49 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 12 Jan 2010 14:32:49 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>
References: <064e01ca93c3$71b259a0$55170ce0$@com>
	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>
Message-ID: <066301ca93c6$6882a300$3987e900$@com>

We can answer and put it on hold and pickup from the other phone, but we
can't barge in.  Should that work?  It doesn't seem to.  Also, private hold
doesn't seem to work.  Whether I use private hold or regular hold, the other
phone can pickup the line.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, January 12, 2010 2:19 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cisco SPA5xx SCA

Enable line.
Set to share.

On phone tab.. mark it shared and select the line.

Attendant console set broadsoft.

save.

reboot
try

/b






On Jan 12, 2010, at 2:11 PM, Peder wrote:

> Anybody know the settings on the SPA5xx to make SCA work?  I've tried all
> sorts of combinations of shared/private on the ext and phone and set the
> Server type to Broadsoft, etc and  it just never seems to work.  Do I need
> the BLF info?  Do I set shared/private on the phone, or ext, or both?  Do
I
> need to enable SCA Line ID Mapping?  Or SCA Barge-In Enable?  Or both?
> 
> Peder


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From mike at van.lammeren.net  Tue Jan 12 12:33:08 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Tue, 12 Jan 2010 15:33:08 -0500
Subject: [Freeswitch-users] How to provide dynamic directory information?
In-Reply-To: <5d2828f1001120850qb676df6wfecbb30c05e65f23@mail.gmail.com>
References: <5d2828f1001111453t8044774w3cc882988e18ced@mail.gmail.com>
	<87f2f3b91001111636j1613b6ak5b5de86804f2125e@mail.gmail.com>
	<5d2828f1001120850qb676df6wfecbb30c05e65f23@mail.gmail.com>
Message-ID: <5d2828f1001121233h7a374f46hbf6964f2e1887b95@mail.gmail.com>

Hello!

I am now successfully pulling directory information from the database with a
Lua script.

I based my work on this section of the wiki:
http://wiki.freeswitch.org/wiki/Mod_lua#For_serving_configuration

There were a few odd things in the wiki that initially threw me for a loop,
and I have since improved the wiki entry linked above. I hope that it will
be easier for the next person!

There is definitely a learning curve to FreeSWITCH, but in the end,
FreeSWITCH always does the trick! Keep with it, and you will be rewarded!

Mike van Lammeren
<http://wiki.freeswitch.org/wiki/Mod_lua#For_serving_configuration>

On Tue, Jan 12, 2010 at 11:50 AM, Mike van Lammeren
<mike at van.lammeren.net>wrote:

> Hi Michael!
>
> I've had a look at your blog. Good stuff!
>
> Thanks!
>
> On Mon, Jan 11, 2010 at 7:36 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Mon, Jan 11, 2010 at 2:53 PM, Mike van Lammeren <mike at van.lammeren.net
>> > wrote:
>>
>>> Hello!
>>>
>>> I'd like to be able to have FreeSWITCH check a database for
>>> authorization, every time a user registers. There are some great examples on
>>> the wiki, which use either MOD_XML_CURL or Lua to dynamically provide a
>>> dialplan, but I cannot find an example for providing a directory.
>>>
>>> I am developing an application that will have thousands of users, and
>>> will run on multiple FreeSWITCH servers behind a load balancer. Ideally,
>>> FreeSWITCH would only look-up directory information, specifically, username
>>> and password, whenever a user attempts to connect. The directory information
>>> will be changing regularly, as users are added or removed from the system.
>>>
>>> Is this possible with FreeSWITCH? Or can only dialplan information be
>>> provided dynamically?
>>>
>>> I've written a script in Lua that provides the XML data, such as that
>>> found in the example /freeswitch/conf/directory/default/ folders, and I try
>>> to call it with this bit of XML in /freeswitch/conf/directory/default.xml:
>>>
>>> <groups>
>>> <group name="default">
>>> <users>
>>>  <param name="xml-handler-script" value="/directory.lua"/>
>>> <param name="xml-handler-bindings" value="users"/>
>>>  </users>
>>> </groups>
>>>
>>> Is this the right approach? Am I going about this the right way?
>>>
>>
>> You can bind "directory" as well as "dialplan" and a few others. I
>> personally don't use xml_curl in production but for kicks I tried to learn
>> it and I documented some of my journey on my personal blog. (
>> http://telecommusings.blogspot.com/)
>>
>> xml_curl was designed to scale and be applied in your type of scenario.
>> Raymond (intralanman on IRC) has played with it quite a bit as have a number
>> of others.
>> -MC
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From brian at freeswitch.org  Tue Jan 12 12:40:58 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 14:40:58 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <066301ca93c6$6882a300$3987e900$@com>
References: <064e01ca93c3$71b259a0$55170ce0$@com>
	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>
	<066301ca93c6$6882a300$3987e900$@com>
Message-ID: <5DE42506-5400-4B40-A522-D1FE5C5AC1EB@freeswitch.org>

Those phones do not support barge in, If they do I couldn't find it.  As for private hold how did you mark it private hold?

/b

On Jan 12, 2010, at 2:32 PM, Peder wrote:

> We can answer and put it on hold and pickup from the other phone, but we
> can't barge in.  Should that work?  It doesn't seem to.  Also, private hold
> doesn't seem to work.  Whether I use private hold or regular hold, the other
> phone can pickup the line.




From peder at networkoblivion.com  Tue Jan 12 12:50:40 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 12 Jan 2010 14:50:40 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <5DE42506-5400-4B40-A522-D1FE5C5AC1EB@freeswitch.org>
References: <064e01ca93c3$71b259a0$55170ce0$@com>	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>	<066301ca93c6$6882a300$3987e900$@com>
	<5DE42506-5400-4B40-A522-D1FE5C5AC1EB@freeswitch.org>
Message-ID: <066c01ca93c8$e67c2b80$b3748280$@com>

Barge:
Under Phone there is an "SCA Barge-In Enable:" line that defaults to no.  I
set it to yes, but when I hit it, I got "unauthorized".

Private Hold:
The soft keys say "PrvHld" when you are on a call.  It functioned the same
as the regular hold button.  From reading the Polycom SCA docs, it says that
private hold should block anybody else from grabbing it.



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, January 12, 2010 2:41 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cisco SPA5xx SCA

Those phones do not support barge in, If they do I couldn't find it.  As for
private hold how did you mark it private hold?

/b

On Jan 12, 2010, at 2:32 PM, Peder wrote:

> We can answer and put it on hold and pickup from the other phone, but we
> can't barge in.  Should that work?  It doesn't seem to.  Also, private
hold
> doesn't seem to work.  Whether I use private hold or regular hold, the
other
> phone can pickup the line.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From brian at freeswitch.org  Tue Jan 12 12:55:41 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 14:55:41 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <066c01ca93c8$e67c2b80$b3748280$@com>
References: <064e01ca93c3$71b259a0$55170ce0$@com>	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>	<066301ca93c6$6882a300$3987e900$@com>
	<5DE42506-5400-4B40-A522-D1FE5C5AC1EB@freeswitch.org>
	<066c01ca93c8$e67c2b80$b3748280$@com>
Message-ID: <D0EA8A36-6AA9-41C6-A9AF-3F374D24902A@freeswitch.org>

Can you post me a sip trace of this.. I only have a 501G without a display.  So I can't see this.

/b

On Jan 12, 2010, at 2:50 PM, Peder wrote:

> Barge:
> Under Phone there is an "SCA Barge-In Enable:" line that defaults to no.  I
> set it to yes, but when I hit it, I got "unauthorized".
> 
> Private Hold:
> The soft keys say "PrvHld" when you are on a call.  It functioned the same
> as the regular hold button.  From reading the Polycom SCA docs, it says that
> private hold should block anybody else from grabbing it.




From peder at networkoblivion.com  Tue Jan 12 13:16:13 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 12 Jan 2010 15:16:13 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <D0EA8A36-6AA9-41C6-A9AF-3F374D24902A@freeswitch.org>
References: <064e01ca93c3$71b259a0$55170ce0$@com>	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>	<066301ca93c6$6882a300$3987e900$@com>	<5DE42506-5400-4B40-A522-D1FE5C5AC1EB@freeswitch.org>	<066c01ca93c8$e67c2b80$b3748280$@com>
	<D0EA8A36-6AA9-41C6-A9AF-3F374D24902A@freeswitch.org>
Message-ID: <067201ca93cc$789d8100$69d88300$@com>

Will do.  Interesting note.  The Linksys SPA922 sort of supports SCA.  You
can do the shared line reg and it looks ok.  The Linksys can answer and then
put on hold and the other phone can pick it up.  The only issue is that the
Linksys can't pickup a call that the other phone puts on hold.  

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, January 12, 2010 2:56 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cisco SPA5xx SCA

Can you post me a sip trace of this.. I only have a 501G without a display.
So I can't see this.

/b

On Jan 12, 2010, at 2:50 PM, Peder wrote:

> Barge:
> Under Phone there is an "SCA Barge-In Enable:" line that defaults to no.
I
> set it to yes, but when I hit it, I got "unauthorized".
> 
> Private Hold:
> The soft keys say "PrvHld" when you are on a call.  It functioned the same
> as the regular hold button.  From reading the Polycom SCA docs, it says
that
> private hold should block anybody else from grabbing it.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From brian at freeswitch.org  Tue Jan 12 13:19:56 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 15:19:56 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <067201ca93cc$789d8100$69d88300$@com>
References: <064e01ca93c3$71b259a0$55170ce0$@com>	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>	<066301ca93c6$6882a300$3987e900$@com>	<5DE42506-5400-4B40-A522-D1FE5C5AC1EB@freeswitch.org>	<066c01ca93c8$e67c2b80$b3748280$@com>
	<D0EA8A36-6AA9-41C6-A9AF-3F374D24902A@freeswitch.org>
	<067201ca93cc$789d8100$69d88300$@com>
Message-ID: <5579C5DD-ED93-445A-AF56-61A9D3A3705B@freeswitch.org>

Yah I never said it would work with those :P

/b

On Jan 12, 2010, at 3:16 PM, Peder wrote:

> Will do.  Interesting note.  The Linksys SPA922 sort of supports SCA.  You
> can do the shared line reg and it looks ok.  The Linksys can answer and then
> put on hold and the other phone can pick it up.  The only issue is that the
> Linksys can't pickup a call that the other phone puts on hold.  




From anthony.minessale at gmail.com  Tue Jan 12 13:20:08 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Jan 2010 15:20:08 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <066301ca93c6$6882a300$3987e900$@com>
References: <064e01ca93c3$71b259a0$55170ce0$@com>
	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>
	<066301ca93c6$6882a300$3987e900$@com>
Message-ID: <191c3a031001121320s1840280aq2f99385b4b6102f7@mail.gmail.com>

Maybe you want to let us on your box to work on it or buy us your phones to
have in our lab.
Our phones did not support those options.





On Tue, Jan 12, 2010 at 2:32 PM, Peder <peder at networkoblivion.com> wrote:

> We can answer and put it on hold and pickup from the other phone, but we
> can't barge in.  Should that work?  It doesn't seem to.  Also, private hold
> doesn't seem to work.  Whether I use private hold or regular hold, the
> other
> phone can pickup the line.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
> Sent: Tuesday, January 12, 2010 2:19 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Cisco SPA5xx SCA
>
> Enable line.
> Set to share.
>
> On phone tab.. mark it shared and select the line.
>
> Attendant console set broadsoft.
>
> save.
>
> reboot
> try
>
> /b
>
>
>
>
>
>
> On Jan 12, 2010, at 2:11 PM, Peder wrote:
>
> > Anybody know the settings on the SPA5xx to make SCA work?  I've tried all
> > sorts of combinations of shared/private on the ext and phone and set the
> > Server type to Broadsoft, etc and  it just never seems to work.  Do I
> need
> > the BLF info?  Do I set shared/private on the phone, or ext, or both?  Do
> I
> > need to enable SCA Line ID Mapping?  Or SCA Barge-In Enable?  Or both?
> >
> > Peder
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From msc at freeswitch.org  Tue Jan 12 13:30:05 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Jan 2010 13:30:05 -0800
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <067201ca93cc$789d8100$69d88300$@com>
References: <064e01ca93c3$71b259a0$55170ce0$@com>
	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>
	<066301ca93c6$6882a300$3987e900$@com>
	<5DE42506-5400-4B40-A522-D1FE5C5AC1EB@freeswitch.org>
	<066c01ca93c8$e67c2b80$b3748280$@com>
	<D0EA8A36-6AA9-41C6-A9AF-3F374D24902A@freeswitch.org>
	<067201ca93cc$789d8100$69d88300$@com>
Message-ID: <87f2f3b91001121330t55d81787k3c7030e429a606b6@mail.gmail.com>

Peder,

Thanks for trying all this out in the real world. When you get it working
please let me know so that we can get proper documentation for these devices
in the wiki. I'd like to see the documentation for specific devices written
by people who are physically using them.

Thanks!
-MC

On Tue, Jan 12, 2010 at 1:16 PM, Peder <peder at networkoblivion.com> wrote:

> Will do.  Interesting note.  The Linksys SPA922 sort of supports SCA.  You
> can do the shared line reg and it looks ok.  The Linksys can answer and
> then
> put on hold and the other phone can pick it up.  The only issue is that the
> Linksys can't pickup a call that the other phone puts on hold.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
> Sent: Tuesday, January 12, 2010 2:56 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Cisco SPA5xx SCA
>
> Can you post me a sip trace of this.. I only have a 501G without a display.
> So I can't see this.
>
> /b
>
> On Jan 12, 2010, at 2:50 PM, Peder wrote:
>
> > Barge:
> > Under Phone there is an "SCA Barge-In Enable:" line that defaults to no.
> I
> > set it to yes, but when I hit it, I got "unauthorized".
> >
> > Private Hold:
> > The soft keys say "PrvHld" when you are on a call.  It functioned the
> same
> > as the regular hold button.  From reading the Polycom SCA docs, it says
> that
> > private hold should block anybody else from grabbing it.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Jan 12 13:34:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Jan 2010 15:34:49 -0600
Subject: [Freeswitch-users] Cisco SPA5xx SCA
In-Reply-To: <067201ca93cc$789d8100$69d88300$@com>
References: <064e01ca93c3$71b259a0$55170ce0$@com>
	<DC204982-BC34-4ACC-8512-CBBB90A64F7E@freeswitch.org>
	<066301ca93c6$6882a300$3987e900$@com>
	<5DE42506-5400-4B40-A522-D1FE5C5AC1EB@freeswitch.org>
	<066c01ca93c8$e67c2b80$b3748280$@com>
	<D0EA8A36-6AA9-41C6-A9AF-3F374D24902A@freeswitch.org>
	<067201ca93cc$789d8100$69d88300$@com>
Message-ID: <191c3a031001121334j6a0725aq74d64ca5cc1ea363@mail.gmail.com>

an isolated trace of that pickup not working with debug log and debug_sla=10
set in sofia.conf.xml under settings would be nice.


On Tue, Jan 12, 2010 at 3:16 PM, Peder <peder at networkoblivion.com> wrote:

> Will do.  Interesting note.  The Linksys SPA922 sort of supports SCA.  You
> can do the shared line reg and it looks ok.  The Linksys can answer and
> then
> put on hold and the other phone can pick it up.  The only issue is that the
> Linksys can't pickup a call that the other phone puts on hold.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
> Sent: Tuesday, January 12, 2010 2:56 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Cisco SPA5xx SCA
>
> Can you post me a sip trace of this.. I only have a 501G without a display.
> So I can't see this.
>
> /b
>
> On Jan 12, 2010, at 2:50 PM, Peder wrote:
>
> > Barge:
> > Under Phone there is an "SCA Barge-In Enable:" line that defaults to no.
> I
> > set it to yes, but when I hit it, I got "unauthorized".
> >
> > Private Hold:
> > The soft keys say "PrvHld" when you are on a call.  It functioned the
> same
> > as the regular hold button.  From reading the Polycom SCA docs, it says
> that
> > private hold should block anybody else from grabbing it.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From jerry.richards at teotech.com  Tue Jan 12 13:47:23 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 12 Jan 2010 13:47:23 -0800
Subject: [Freeswitch-users] PRI Goes Down Upon Calling Cell Phone
Message-ID: <24C642A4E3CB4C9F9865199154120E72@greyhawk.tonecommander.com>

 
I am having a problem where my PRI goes down after attempting to call a cell
phone from an internal phone.  I am running Freeswitch
(freeswitch-1.0.5-20100112-0400.tar.gz) with an A101D Sangoma Card and
wanpipe driver: wanpipe-3.5.10.tgz.  I am not using libpri.
 
The scenario is as follows:
 
1) Call desktop phone 3491006
2) Call completes normally
3) Call cell phone 4181432 (see http://pastebin.freeswitch.org/11767)
    ***** call fails; cell phone rings, but does not complete *****
4) Call desktop phone 3491006
    ***** call fails; far-end does not ring *****
5) Any subsequent calls through PRI fail
   
I notice the call to my cell phone receives a Q.931 PROGRESS message instead
of Q.931 ALERTING.  Has this issue already been identified as a bug?  Do you
know if this is a FS issue?  Or Wanpipe driver issue?
 
Best Regards,
Jerry




From msc at freeswitch.org  Tue Jan 12 14:09:38 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Jan 2010 14:09:38 -0800
Subject: [Freeswitch-users] PRI Goes Down Upon Calling Cell Phone
In-Reply-To: <24C642A4E3CB4C9F9865199154120E72@greyhawk.tonecommander.com>
References: <24C642A4E3CB4C9F9865199154120E72@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91001121409i141ac5fco8a2c50fed523dce1@mail.gmail.com>

Can you get a Q931 trace on this and put it up where we can download it? The
procedure is documented on Sangoma's website; see openzap wiki page under
debugging sangoma boost.

Thanks,
MC

On Tue, Jan 12, 2010 at 1:47 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> I am having a problem where my PRI goes down after attempting to call a
> cell
> phone from an internal phone.  I am running Freeswitch
> (freeswitch-1.0.5-20100112-0400.tar.gz) with an A101D Sangoma Card and
> wanpipe driver: wanpipe-3.5.10.tgz.  I am not using libpri.
>
> The scenario is as follows:
>
> 1) Call desktop phone 3491006
> 2) Call completes normally
> 3) Call cell phone 4181432 (see http://pastebin.freeswitch.org/11767)
>    ***** call fails; cell phone rings, but does not complete *****
> 4) Call desktop phone 3491006
>    ***** call fails; far-end does not ring *****
> 5) Any subsequent calls through PRI fail
>
> I notice the call to my cell phone receives a Q.931 PROGRESS message
> instead
> of Q.931 ALERTING.  Has this issue already been identified as a bug?  Do
> you
> know if this is a FS issue?  Or Wanpipe driver issue?
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jerry.richards at teotech.com  Tue Jan 12 15:10:25 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 12 Jan 2010 15:10:25 -0800
Subject: [Freeswitch-users] Inbound DTMF Not Recognized In Latest Version
Message-ID: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com>


I am having a problem where in inbound call from the PSTN going to the demo
IVR does not recognize DTMF digits.  I posted a log at
http://pastebin.freeswitch.org/11769.  I am running Freeswitch
(freeswitch-1.0.5-20100112-0400.tar.gz) with an A101D Sangoma Card and
wanpipe driver: wanpipe-3.5.10.tgz.  I am not using libpri.

It appears that each DTMF digit might be getting detected twice.  I tried to
dial "5401" and it thought I dialed "5544".

I am following the instructions at
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf, as shown
below.  Also, I tried moving the "start_dtmf" statement in between all the
statements within the condition tag, but it didn't make any difference.

<extension name="ivr_demo">
   <condition field="destination_number" expression="5000">
      <action application="answer"/>
      <action application="start_dtmf" />
      <action application="sleep" data="2000"/>
      <action application="ivr" data="demo_ivr"/>
   </condition>
</extension>

Also, I noticed at line #293, the log is saying:

[ERR] mod_sndfile.c:194 Error Opening File
[/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System
error : No such file or directory.]

Has anyone else encountered this issue?

Best Regards,
Jerry




From lart2150 at gmail.com  Tue Jan 12 15:08:27 2010
From: lart2150 at gmail.com (Brian Engert)
Date: Tue, 12 Jan 2010 17:08:27 -0600
Subject: [Freeswitch-users] Channel Variables with spaces
Message-ID: <a88e84c21001121508s191b89bdr6d67079cba62be34@mail.gmail.com>

I'm trying to set a channel variable with a space in it when I run the
originate command.  The command only seems to work when I escape the
space with a \ however that backspace get's passed on.  Some examples
are

originate {fax_ident=1231231234,fax_header=testing\
spaces}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)
originate {fax_ident=1231231234,fax_header=testing
spaces}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)

The first one will go through just fine but the fax at the other end
gets "testing\ spaces" the second one gives me
"2010-01-12 16:35:18.41325 [ERR] switch_ivr_originate.c:990 Parse Error!"

I think this could also be done as a profile but I would rather do it
from the call.  I've looked around the wiki and have not seen any
examples that use a space when doing the originate call.

- Brian



From brian at freeswitch.org  Tue Jan 12 15:25:58 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 17:25:58 -0600
Subject: [Freeswitch-users] Inbound DTMF Not Recognized In Latest Version
In-Reply-To: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com>
References: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com>
Message-ID: <44468DAE-9201-40AF-ACA3-E634E6682F04@freeswitch.org>

looks like you don't have the latest sound files installed.

/b

On Jan 12, 2010, at 5:10 PM, Jerry Richards wrote:

> [ERR] mod_sndfile.c:194 Error Opening File
> [/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] [System
> error : No such file or directory.]




From mike at jerris.com  Tue Jan 12 15:35:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 12 Jan 2010 18:35:20 -0500
Subject: [Freeswitch-users] Hello , about calculate numbers
In-Reply-To: <201001121239.06756.sos@sokhapkin.dyndns.org>
References: <314dc3f81001120908l5772b13fqd676a3be367bb662@mail.gmail.com>
	<201001121239.06756.sos@sokhapkin.dyndns.org>
Message-ID: <EDF668FE-A76D-4A5F-96D1-756A7ACD13A6@jerris.com>

Remember you can't do conditions on these vars being set unless you transfer back into the dialplan as these actions are not run immediately, but rather after dialplan parse, unless you use http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions

Mie

On Jan 12, 2010, at 12:39 PM, Sergey Okhapkin wrote:

> <action application="set" data="var=${expr(${var}+1)}"/>
> 
> On Tuesday 12 January 2010, Chia-Yen Wu wrote:
>> Hello everyone,
>> Is there anyway to calculate number in dialplan.xml?
>> 
>> like example:
>> <action application="set" data="var=1"/>
>> and with an action , I can do : var=var+1
>> 
>> I have tried mod_expr, but failed to catch the variable, like below:
>> <action application="set" data="var=${expr($var+1)}"/>
> 




From anthony.minessale at gmail.com  Tue Jan 12 15:35:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Jan 2010 17:35:38 -0600
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0F362D9@BESONE.corp.dom>
References: <A2167AE7633A5C4DAE56EEBD1428ECC0F362D9@BESONE.corp.dom>
Message-ID: <191c3a031001121535t4819ee44p74cc9f7cd4fe972f@mail.gmail.com>

On Tue, Jan 12, 2010 at 3:43 AM, Cavalera Claudio Luigi <
Claudio.Cavalera at italtel.it> wrote:

> Hello,
> this should be simple in theory therefore I'm probably missing the right
> way to do it.
> I want to play with sessions in lua, bridge them, park them, etc...
>
> example1: Consider this simple lua script in which i create two
> sessions:
>
> api = freeswitch.API();
> api:execute("originate", "sofia/internal/1001%192.168.1.1 &park");
> api:execute("originate", "sofia/internal/1002%192.168.1.1 &park");
>
>
capture the output from api:execute the uuid is in there



> now if i want to bridge them i suppose i should use something like
>
> api:execute("uuid_bridge", "uuid_1 uuid_2");
>
> but how do i get uuid_1 and uuid_2, i.e. the uuids of the two sessions?
>
>
> example2: I could create sessions with
>
> local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
> local session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");
>
> but then there is NOT a bridge API to bridge the sessions like:
> bridge(session1, session2);
>
> I admit I have not yet understood why such bridge possibility exist in
> javascript but does not exist in lua.
> http://wiki.freeswitch.org/wiki/Javascript_Misc_bridge
> I guess there is a reason for this but I can't figure it out.
>
>
>
because lua calls it freeswitch.bridge

session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1", session1);
freeswitch.bridge(session1, session2);





> example3: yet another possibility
> local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
> session1:originate(session2, "sofia/internal/1002%192.168.1.1", 1000);
>
> but it does not work either.
>
>
The above is gibberish try:

local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
session1:execute("bridge", "sofia/internal/1002%192.168.1.1");




> Besides with this third example something strange happen:
>
> freeswitch at internal> show channels
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr
> ,dest,application,application_data,dialplan,context,read_codec,read_rate
> ,write_codec,write_rate,secure
> 1c5db2df-14ce-4516-94f2-bb7c087e0802,outbound,2010-01-12
> 10:50:59,1263289859,sofia/internal/1004,CS_REPORTING,FreeSWITCH,00000000
> 00,,1004,,,,default,PCMA,8000,PCMA,8000,
>
> 1 total.
>
> freeswitch at internal> uuid_kill 1c5db2df-14ce-4516-94f2-bb7c087e0802
> -ERR No Such Channel!
>
> freeswitch at internal> show channels
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr
> ,dest,application,application_data,dialplan,context,read_codec,read_rate
> ,write_codec,write_rate,secure
> 1c5db2df-14ce-4516-94f2-bb7c087e0802,outbound,2010-01-12
> 10:50:59,1263289859,sofia/internal/1004,CS_REPORTING,FreeSWITCH,00000000
> 00,,1004,,,,default,PCMA,8000,PCMA,8000,
>
> 1 total.
>
> freeswitch at internal>
>
> If you are interested the full log is here:
> http://pastebin.freeswitch.org/11757
> but I admit i'm not on latest trunk yet!
>
> Thanks.
> Ciao,
> Claudio
>
>
> PS: Is there a reason why there is a uuid_park command but not
> uuid_valet_park ?
>
>
> Internet Email Confidentiality Footer
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>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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From brian at freeswitch.org  Tue Jan 12 15:37:42 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 17:37:42 -0600
Subject: [Freeswitch-users] Channel Variables with spaces
In-Reply-To: <a88e84c21001121508s191b89bdr6d67079cba62be34@mail.gmail.com>
References: <a88e84c21001121508s191b89bdr6d67079cba62be34@mail.gmail.com>
Message-ID: <0F330D84-2C15-47D3-986B-704370A6F745@freeswitch.org>

Have you tried to properly quote them?

/b

On Jan 12, 2010, at 5:08 PM, Brian Engert wrote:

> The first one will go through just fine but the fax at the other end
> gets "testing\ spaces" the second one gives me
> "2010-01-12 16:35:18.41325 [ERR] switch_ivr_originate.c:990 Parse Error!"




From msc at freeswitch.org  Tue Jan 12 15:40:43 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Jan 2010 15:40:43 -0800
Subject: [Freeswitch-users] Channel Variables with spaces
In-Reply-To: <a88e84c21001121508s191b89bdr6d67079cba62be34@mail.gmail.com>
References: <a88e84c21001121508s191b89bdr6d67079cba62be34@mail.gmail.com>
Message-ID: <87f2f3b91001121540j70367d55n1aa02aed36faccbf@mail.gmail.com>

On Tue, Jan 12, 2010 at 3:08 PM, Brian Engert <lart2150 at gmail.com> wrote:

> I'm trying to set a channel variable with a space in it when I run the
> originate command.  The command only seems to work when I escape the
> space with a \ however that backspace get's passed on.  Some examples
> are
>
> originate {fax_ident=1231231234,fax_header=testing\
> spaces}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)
> originate {fax_ident=1231231234,fax_header=testing
> spaces}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)
>
> The first one will go through just fine but the fax at the other end
> gets "testing\ spaces" the second one gives me
> "2010-01-12 16:35:18.41325 [ERR] switch_ivr_originate.c:990 Parse Error!"
>
> I think this could also be done as a profile but I would rather do it
> from the call.  I've looked around the wiki and have not seen any
> examples that use a space when doing the originate call.
>
> I'll give you the answer if you promise to put it into the wiki. ;)

Okay, use single quotes like this:

originate {fax_ident=1231231234,fax_header='testing
spaces'}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)

-MC
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From lart2150 at gmail.com  Tue Jan 12 16:18:14 2010
From: lart2150 at gmail.com (Brian Engert)
Date: Tue, 12 Jan 2010 18:18:14 -0600
Subject: [Freeswitch-users] Channel Variables with spaces
In-Reply-To: <87f2f3b91001121540j70367d55n1aa02aed36faccbf@mail.gmail.com>
References: <a88e84c21001121508s191b89bdr6d67079cba62be34@mail.gmail.com>
	<87f2f3b91001121540j70367d55n1aa02aed36faccbf@mail.gmail.com>
Message-ID: <a88e84c21001121618l59c196f0qe9655c7057d57f64@mail.gmail.com>

I'll try that tomorrow and update the wiki!  is it only single quotes
or should double quotes work as well?



On Tue, Jan 12, 2010 at 5:40 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Tue, Jan 12, 2010 at 3:08 PM, Brian Engert <lart2150 at gmail.com> wrote:
>>
>> I'm trying to set a channel variable with a space in it when I run the
>> originate command. ?The command only seems to work when I escape the
>> space with a \ however that backspace get's passed on. ?Some examples
>> are
>>
>> originate {fax_ident=1231231234,fax_header=testing\
>> spaces}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)
>> originate {fax_ident=1231231234,fax_header=testing
>> spaces}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)
>>
>> The first one will go through just fine but the fax at the other end
>> gets "testing\ spaces" the second one gives me
>> "2010-01-12 16:35:18.41325 [ERR] switch_ivr_originate.c:990 Parse Error!"
>>
>> I think this could also be done as a profile but I would rather do it
>> from the call. ?I've looked around the wiki and have not seen any
>> examples that use a space when doing the originate call.
>>
> I'll give you the answer if you promise to put it into the wiki. ;)
>
> Okay, use single quotes like this:
>
> originate {fax_ident=1231231234,fax_header='testing
> spaces'}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From john at acsol.net  Tue Jan 12 16:20:25 2010
From: john at acsol.net (John)
Date: Tue, 12 Jan 2010 17:20:25 -0700
Subject: [Freeswitch-users] ShoreTel to FS connection
In-Reply-To: <38D0A853-889C-4F53-B768-53C1283C8B5C@freeswitch.org>
References: <4B4CD6D5.1010104@acsol.net>
	<38D0A853-889C-4F53-B768-53C1283C8B5C@freeswitch.org>
Message-ID: <4B4D11C9.1060007@acsol.net>

5551212 is the number that I am dialing with the ShoreTel system. It is 
passing it to FS. Some how I need it to pass that number to the gateway 
defined on the FS.



On 1/12/2010 1:17 PM, Brian West wrote:
> Its looking for 5551212 in context public.  you can ONLY communicate with shoretel on port 5060
>
> /b
>
> On Jan 12, 2010, at 2:08 PM, John wrote:
>
>    
>> I am trying to utilize a freeswitch system as a SIP trunk for ShoreTel.
>> I have had to change the port from 5080 to 5060. When I dial through, I
>> see on the FS console show the call; however I can't seem to change the
>> public.xml dialplan to effect any changes in the behaviour. How can I
>> tell which dialplan the external call is using?
>>
>>   2010-01-12 13:04:17.705548 [NOTICE] switch_channel.c:613 New Channel
>> sofia/external/+19705551212 at 192.168.155.13:5060
>> [aa2f2fde-ffb5-11de-9590-015cd8b8c273]
>> 2010-01-12 13:04:17.705548 [INFO] mod_dialplan_xml.c:408 Processing
>> Extension 1002->5551212 in context public
>> 2010-01-12 13:04:17.705548 [NOTICE] switch_core_state_machine.c:187
>> Hangup sofia/external/+19705551212 at 192.168.155.13:5060 [CS_EXECUTE]
>> [NORMAL_CLEARING
>>
>> Thanks John
>>
>> _____________
>>      
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    




From brian at freeswitch.org  Tue Jan 12 16:25:34 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 18:25:34 -0600
Subject: [Freeswitch-users] Channel Variables with spaces
In-Reply-To: <a88e84c21001121618l59c196f0qe9655c7057d57f64@mail.gmail.com>
References: <a88e84c21001121508s191b89bdr6d67079cba62be34@mail.gmail.com>
	<87f2f3b91001121540j70367d55n1aa02aed36faccbf@mail.gmail.com>
	<a88e84c21001121618l59c196f0qe9655c7057d57f64@mail.gmail.com>
Message-ID: <77CA31CE-3238-45E7-826E-04BA9E7D8357@freeswitch.org>

Single
On Jan 12, 2010, at 6:18 PM, Brian Engert wrote:

> I'll try that tomorrow and update the wiki!  is it only single quotes
> or should double quotes work as well?




From brian at freeswitch.org  Tue Jan 12 16:26:11 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 18:26:11 -0600
Subject: [Freeswitch-users] ShoreTel to FS connection
In-Reply-To: <4B4D11C9.1060007@acsol.net>
References: <4B4CD6D5.1010104@acsol.net>
	<38D0A853-889C-4F53-B768-53C1283C8B5C@freeswitch.org>
	<4B4D11C9.1060007@acsol.net>
Message-ID: <4F42AEBC-7F10-4D33-B79A-F560F598637E@freeswitch.org>

The put a dialplan entry in public context to catch and redirect the number elsewhere or use bridge to send it out a gateway.

/b

On Jan 12, 2010, at 6:20 PM, John wrote:

> 5551212 is the number that I am dialing with the ShoreTel system. It is 
> passing it to FS. Some how I need it to pass that number to the gateway 
> defined on the FS.




From jeff at jefflenk.com  Tue Jan 12 18:35:25 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 12 Jan 2010 18:35:25 -0800 (PST)
Subject: [Freeswitch-users] FSComm Windows build
In-Reply-To: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
References: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
Message-ID: <1263350125191-4295967.post@n2.nabble.com>


Very Cool!

Just curious Carlos do you have any experience with x64 with regard to QT on
Win?

Thanks
Jeff


Carlos Talbot wrote:
> 
> FYI,
> 
> there's a Windows pre-compiled binary of FSComm now available for those
> who
> want to check it.
> 
> http://files.freeswitch.org/windows_installer/FSComm.exe
> 
> regards,
> 
> Carlos
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/FSComm-Windows-build-tp4289256p4295967.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From carlos.talbot at gmail.com  Tue Jan 12 19:18:26 2010
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Tue, 12 Jan 2010 21:18:26 -0600
Subject: [Freeswitch-users] FSComm Windows build
In-Reply-To: <1263350125191-4295967.post@n2.nabble.com>
References: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
	<1263350125191-4295967.post@n2.nabble.com>
Message-ID: <5800526b1001121918r15a336f5jbc5b724e66ac2825@mail.gmail.com>

Jeff,

not yet. I'm still linking with a 32bit QT library of 4.5.2 I compiled for
VS2008 sometime last year. I followed a guide similar to this one:
http://dcsoft.com/community_server/blogs/dcsoft/archive/2009/03/06/how-to-setup-qt-4-5-visual-studio-integration.aspx

At the time I didn't see a need for a 64bit version of FsGui. Since FSComm
has FreeSWITCH in the back end this kind of changes things.

Looks like someone put up a wiki on compiling QT for Win x64:
http://en.wikibooks.org/wiki/Opticks_Developer_Guide/Getting_Started/Building_Qt_From_Source

Now that QT 4.6 is out I might have to revisit a new library build with 64
bit in mind.

Carlos


On Tue, Jan 12, 2010 at 8:35 PM, Jeff Lenk <jeff at jefflenk.com> wrote:

>
> Very Cool!
>
> Just curious Carlos do you have any experience with x64 with regard to QT
> on
> Win?
>
> Thanks
> Jeff
>
>
> Carlos Talbot wrote:
> >
> > FYI,
> >
> > there's a Windows pre-compiled binary of FSComm now available for those
> > who
> > want to check it.
> >
> > http://files.freeswitch.org/windows_installer/FSComm.exe
> >
> > regards,
> >
> > Carlos
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://n2.nabble.com/FSComm-Windows-build-tp4289256p4295967.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From xanlich at gmail.com  Tue Jan 12 19:37:00 2010
From: xanlich at gmail.com (Chia-Yen Wu)
Date: Wed, 13 Jan 2010 11:37:00 +0800
Subject: [Freeswitch-users] Hello , about calculate numbers
In-Reply-To: <EDF668FE-A76D-4A5F-96D1-756A7ACD13A6@jerris.com>
References: <314dc3f81001120908l5772b13fqd676a3be367bb662@mail.gmail.com>
	<201001121239.06756.sos@sokhapkin.dyndns.org>
	<EDF668FE-A76D-4A5F-96D1-756A7ACD13A6@jerris.com>
Message-ID: <314dc3f81001121937l1b415c3amb715d50017850ebb@mail.gmail.com>

Thanks for reply, I have tried the new one and did transfer back
but it still dont work, when I was posting this question again with my test
commands.
suddenly I found out what the problem it is.

the variable cannot with number in it, like:

work:
<action application="set" data="var=1"/>
<action application="set" data="var=${expr(${var}+1)}"/>
fail:
<action application="set" data="var1=1"/>
<action application="set" data="var1=${expr(${var1}+1)}"/>


2010/1/13 Michael Jerris <mike at jerris.com>

> Remember you can't do conditions on these vars being set unless you
> transfer back into the dialplan as these actions are not run immediately,
> but rather after dialplan parse, unless you use
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>
> Mie
>
> On Jan 12, 2010, at 12:39 PM, Sergey Okhapkin wrote:
>
> > <action application="set" data="var=${expr(${var}+1)}"/>
> >
> > On Tuesday 12 January 2010, Chia-Yen Wu wrote:
> >> Hello everyone,
> >> Is there anyway to calculate number in dialplan.xml?
> >>
> >> like example:
> >> <action application="set" data="var=1"/>
> >> and with an action , I can do : var=var+1
> >>
> >> I have tried mod_expr, but failed to catch the variable, like below:
> >> <action application="set" data="var=${expr($var+1)}"/>
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mcampbellsmith at gmail.com  Tue Jan 12 20:04:24 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 13 Jan 2010 15:04:24 +1100
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <33c87fa31001120223t13efc07agcc595e48405fd8bb@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
	<33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
	<33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>
	<20100112095332.GA32294@jdc.jasonjgw.net>
	<33c87fa31001120223t13efc07agcc595e48405fd8bb@mail.gmail.com>
Message-ID: <33c87fa31001122004g3ad8fe11w701b14a2fcd03442@mail.gmail.com>

Hi All,

Does anyone know if tone_detect can be used with bypass_media?

I thought the SIP re-Invite message can be used to update media
parameters, including IP address endpoints.  Does FS try to do this
in the case that tone_detect is used?

In my case, the calls are dropped.

On Tue, Jan 12, 2010 at 9:23 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Hi Jason,
>
> I have understood that. ?Its not that a difficult concept to understand!
>
> In the log I see:
> 2010-01-12 21:03:17.585598 [DEBUG] mod_dptools.c:818
> sofia/internal/1000 at 192.168.1.120 SET
> [ringback]=[v=-7;%(400,200,413,438);%(400,2000,413,438)]
> 2010-01-12 21:03:17.605591 [DEBUG] switch_core_session.c:1509
> Application tone_detect Requires media! pre_answering channel
> sofia/internal/1000 at 192.168.1.120
>
> I thought the SIP re-Invite message can be used to update media
> parameters, including IP address endpoints. ?Does FS try too do this
> in the case that tone_detect is used?
>
> On Tue, Jan 12, 2010 at 8:53 PM, Jason White <jason at jasonjgw.net> wrote:
>> Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>>> I've updated and tested bypass_media.
>>>
>>> It works if I remove this line from the B leg dialplan (ie 2010 calls
>>> 1000 - this is in the 1000 section of the dialplan):
>>> ? ? ? ? ? ? ? ? ? ? ? ? <action application="tone_detect" data="fax
>>> 1100 r +5000 transfer 'receivefax XML features' 1" />
>>>
>>> Does bypass_media work with tone_detect?
>>
>> As I understand it, tone_detect detects tones in the RTP stream (i.e., in the
>> audio). For this to be possible, FreeSWITCH has to be in the audio path, hence
>> bypass media cannot be used
>>
>> If this reasoning isn't obvious to you, then you've misunderstood what
>> tone_detect does or what bypass media is (the audio traffic flows directly
>> between the two endpoints without passing through the FreeSWITCH system that
>> establishes the connection, therefore FreeSWITCH can't process it to detect
>> tones and consequently bypass media and tone detection are inherently
>> incompatible.)
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>



From mike at jerris.com  Tue Jan 12 20:09:29 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 12 Jan 2010 23:09:29 -0500
Subject: [Freeswitch-users] Hello , about calculate numbers
In-Reply-To: <314dc3f81001121937l1b415c3amb715d50017850ebb@mail.gmail.com>
References: <314dc3f81001120908l5772b13fqd676a3be367bb662@mail.gmail.com>
	<201001121239.06756.sos@sokhapkin.dyndns.org>
	<EDF668FE-A76D-4A5F-96D1-756A7ACD13A6@jerris.com>
	<314dc3f81001121937l1b415c3amb715d50017850ebb@mail.gmail.com>
Message-ID: <2C92D649-BFC9-4FF9-AA1F-E6D0FDB5CA1A@jerris.com>

Strange, could you post the debug log of this so we can see how it's  
expanded?

Mike

On Jan 12, 2010, at 10:37 PM, Chia-Yen Wu <xanlich at gmail.com> wrote:

> Thanks for reply, I have tried the new one and did transfer back
> but it still dont work, when I was posting this question again with  
> my test commands.
> suddenly I found out what the problem it is.
>
> the variable cannot with number in it, like:
>
> work:
> <action application="set" data="var=1"/>
> <action application="set" data="var=${expr(${var}+1)}"/>
> fail:
> <action application="set" data="var1=1"/>
> <action application="set" data="var1=${expr(${var1}+1)}"/>
>
>
> 2010/1/13 Michael Jerris <mike at jerris.com>
> Remember you can't do conditions on these vars being set unless you  
> transfer back into the dialplan as these actions are not run  
> immediately, but rather after dialplan parse, unless you use http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>
> Mie
>
> On Jan 12, 2010, at 12:39 PM, Sergey Okhapkin wrote:
>
> > <action application="set" data="var=${expr(${var}+1)}"/>
> >
> > On Tuesday 12 January 2010, Chia-Yen Wu wrote:
> >> Hello everyone,
> >> Is there anyway to calculate number in dialplan.xml?
> >>
> >> like example:
> >> <action application="set" data="var=1"/>
> >> and with an action , I can do : var=var+1
> >>
> >> I have tried mod_expr, but failed to catch the variable, like  
> below:
> >> <action application="set" data="var=${expr($var+1)}"/>
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From magesh.freeswitch at gmail.com  Tue Jan 12 20:21:20 2010
From: magesh.freeswitch at gmail.com (Magesh R)
Date: Tue, 12 Jan 2010 23:21:20 -0500
Subject: [Freeswitch-users] Sangoma PRI installation for FreeSWITCH
Message-ID: <369c72d81001122021o6885913cl618965791aec4621@mail.gmail.com>

Dear All,
         I have installed Sangoma PRI card in machine with following steps,
                 * wget
ftp://ftp.sangoma.com/linux/custom/3.5/wanpipe-3.5.8.7.tgz
                 * tar -xvfz wanpipe-3.5.8.7.tgz
                 * cd wanpipe-3.5.8.7
                 * make openzap
                 * make install
                 * make install_pri

           I have executed "wanrouter hwprobe" command it prints the
following details,

           1 . AFT-A102-SH : SLOT=4 : BUS=3 : IRQ=11 : CPU=A : PORT=1 :
HWEC=0 : V=36
           2 . AFT-A102-SH : SLOT=4 : BUS=3 : IRQ=11 : CPU=A : PORT=2 :
HWEC=0 : V=36

        Card Cnt: A101-2=1

       Next I have executed wancfg_fs script to configure the sangoma for
freeswitch.
It creates the following configuration files
                     * wanpipe1.conf
                     * wanpipe2.conf
                     * smg_prid.conf
                     * openzap.conf
                     * openzap.conf.xml
       I have attached those files.
       I have started the wanrouter and printed the wanrouter status,
              Wanrouter Status:

Device name | Protocol | Station | Status        |
wanpipe1    | AFT TE1  | N/A     | Connected     |
wanpipe2    | AFT TE1  | N/A     | Disconnected  |

      Next I have started the smg_ctrl, but it failed to start. It prints
the following things,
smg_ctrl start

Starting processes...
Loading SCTP...OK
Starting sangoma_prid...OK
sangoma_prid failed to start
check /var/log/sangoma_mgd.log for errors

Stopping running processes...
safe_sangoma not running...
sangoma_prid is stopped
Removing PID files...done

I have checked /var/log/sangoma_mgd.log file. But nothing was there.

Could any please tell me where I made mistake?

Thanks,
Mag
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From brian at freeswitch.org  Tue Jan 12 20:41:35 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Jan 2010 22:41:35 -0600
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <33c87fa31001122004g3ad8fe11w701b14a2fcd03442@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
	<33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
	<33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>
	<20100112095332.GA32294@jdc.jasonjgw.net>
	<33c87fa31001120223t13efc07agcc595e48405fd8bb@mail.gmail.com>
	<33c87fa31001122004g3ad8fe11w701b14a2fcd03442@mail.gmail.com>
Message-ID: <12C239EE-DAB3-4B52-9317-32399B5290E3@freeswitch.org>

NO.

/b

On Jan 12, 2010, at 10:04 PM, Mark Campbell-Smith wrote:

> 
> Does anyone know if tone_detect can be used with bypass_media?




From vgoget at yahoo.com  Tue Jan 12 16:54:54 2010
From: vgoget at yahoo.com (VG Oget)
Date: Tue, 12 Jan 2010 16:54:54 -0800 (PST)
Subject: [Freeswitch-users] Fw: SIP client authorization issues (Globarange
	phone)
Message-ID: <271101.94727.qm@web53107.mail.re2.yahoo.com>

Hi, I am trying to register a GlobaRange SIP phone to FreeSwitch (was able to do it with Asterisk some time ago). It is a locked VOIP phone to Joip.com. On Ubuntu with latest version as of a few days ago?

I redirected the traffic to Freeswitch (basically proxy.joip.com points and to the Freeswitch host) and also changed the SIP port to 23768 (what the phone wants). Very close to this: http://darkskiez.co.uk/index.php?page=Use_The_Panasonic_Globarange_With_Asterisk


I have Xlite successfully registering (SIP port 23768 and using proxy.joip.com not the IP address) so I?m confident the freeswitch config is close to being okay?

Now I have intercepted the traffic during registration. The issue (I think) is that the SIP authentication phases use a different username and I don?t know how to configure it.
I looked into this page extensively: http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide
And also tried to understand sip_suth_username but I cannot find the solution.

If someone can point me in the right direction, that would be perfect.

Thank you,
G.


--------------

Phone: 192.168.1.11
Freeswitch: 192.168.1.5 (=proxy.joip.com on local dns)
SIP port: 5060
Extension(changed): 1234567890 (joip username)
Authentication username (changed): 98765432

------------


REGISTER sip:proxy.joip.com:23768 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:6060;branch=z9hG4bK991d765d
Max-Forwards: 70
To: sip:1234567890 at proxy.joip.com
From: sip:1234567890 at proxy.joip.com;tag=3957008896
Call-ID: yyyy-xxxxxxxxxxxx at 192.168.1.11
CSeq: 1 REGISTER
Contact: sip:1234567890 at 192.168.1.11:6060;nat=3;deviceid=2
Expires: 304
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY
User-Agent: Panasonic GT1500/a13.32/xxxxxxxxxxxx
Content-Length: 0

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.11:6060;branch=z9hG4bK991d765d
From: <sip:1234567890 at proxy.joip.com>;tag=3957008896
To: <sip:1234567890 at proxy.joip.com>;tag=96BeFKm5U4jFH
Call-ID: yyyy-xxxxxxxxxxxx at 192.168.1.11
CSeq: 1 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.0.5-20100106-0400-hacked
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
WWW-Authenticate: Digest realm="proxy.joip.com", nonce="67132386-fe25-11de-97c3-c7963d15da66", algorithm=MD5, qop="auth"
Content-Length: 0

REGISTER sip:proxy.joip.com:23768 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:6060;branch=z9hG4bKe585be9f
Max-Forwards: 70
To: sip:1234567890 at proxy.joip.com
From: sip:1234567890 at proxy.joip.com;tag=3957008896
Call-ID: yyyy-xxxxxxxxxxxx at 192.168.1.11
CSeq: 2 REGISTER
Contact: sip:1234567890 at 192.168.1.11:6060;nat=3;deviceid=2
Expires: 304
Authorization: Digest realm="proxy.joip.com", nonce="67132386-fe25-11de-97c3-c7963d15da66", algorithm=MD5, qop=auth, cnonce="7FDF680B", nc=00000001, uri="sip:proxy.joip.com:23768", username="98765432", 

response="7bd8c1898e9c01ee1817146ddc86c8b4"
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY
User-Agent: Panasonic GT1500/a13.32/xxxxxxxxxxxx
Content-Length: 0

SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.11:6060;branch=z9hG4bKe585be9f
From: <sip:1234567890 at proxy.joip.com>;tag=3957008896
To: <sip:1234567890 at proxy.joip.com>;tag=ag56ge58rD91c
Call-ID: yyyy-xxxxxxxxxxxx at 192.168.1.11
CSeq: 2 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.0.5-20100106-0400-hacked
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Length: 0


----




      Get your preferred Email name!
Now you can @ymail.com and @rocketmail.com. 
http://mail.promotions.yahoo.com/newdomains/aa/



From mike at jerris.com  Tue Jan 12 20:46:00 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 12 Jan 2010 23:46:00 -0500
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <33c87fa31001122004g3ad8fe11w701b14a2fcd03442@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<bf6923ed1001111502p5c4883far2173a8e2bcd443bc@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
	<33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
	<33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>
	<20100112095332.GA32294@jdc.jasonjgw.net>
	<33c87fa31001120223t13efc07agcc595e48405fd8bb@mail.gmail.com>
	<33c87fa31001122004g3ad8fe11w701b14a2fcd03442@mail.gmail.com>
Message-ID: <03E1D9E8-ADB2-4D3D-BCDC-3F54ADD8879A@jerris.com>

As soon as it pulls in the media (in this case, to do the tone detect), its going to give up on bypass media unless you adjust settings to behave otherwise. (such as http://wiki.freeswitch.org/wiki/Variable_bypass_media_after_bridge)

Mike

On Jan 12, 2010, at 11:04 PM, Mark Campbell-Smith wrote:

> Hi All,
> 
> Does anyone know if tone_detect can be used with bypass_media?
> 
> I thought the SIP re-Invite message can be used to update media
> parameters, including IP address endpoints.  Does FS try to do this
> in the case that tone_detect is used?
> 
> In my case, the calls are dropped.
> 
> On Tue, Jan 12, 2010 at 9:23 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>> Hi Jason,
>> 
>> I have understood that.  Its not that a difficult concept to understand!
>> 
>> In the log I see:
>> 2010-01-12 21:03:17.585598 [DEBUG] mod_dptools.c:818
>> sofia/internal/1000 at 192.168.1.120 SET
>> [ringback]=[v=-7;%(400,200,413,438);%(400,2000,413,438)]
>> 2010-01-12 21:03:17.605591 [DEBUG] switch_core_session.c:1509
>> Application tone_detect Requires media! pre_answering channel
>> sofia/internal/1000 at 192.168.1.120
>> 
>> I thought the SIP re-Invite message can be used to update media
>> parameters, including IP address endpoints.  Does FS try too do this
>> in the case that tone_detect is used?
>> 
>> On Tue, Jan 12, 2010 at 8:53 PM, Jason White <jason at jasonjgw.net> wrote:
>>> Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>>>> I've updated and tested bypass_media.
>>>> 
>>>> It works if I remove this line from the B leg dialplan (ie 2010 calls
>>>> 1000 - this is in the 1000 section of the dialplan):
>>>>                         <action application="tone_detect" data="fax
>>>> 1100 r +5000 transfer 'receivefax XML features' 1" />
>>>> 
>>>> Does bypass_media work with tone_detect?
>>> 
>>> As I understand it, tone_detect detects tones in the RTP stream (i.e., in the
>>> audio). For this to be possible, FreeSWITCH has to be in the audio path, hence
>>> bypass media cannot be used
>>> 
>>> If this reasoning isn't obvious to you, then you've misunderstood what
>>> tone_detect does or what bypass media is (the audio traffic flows directly
>>> between the two endpoints without passing through the FreeSWITCH system that
>>> establishes the connection, therefore FreeSWITCH can't process it to detect
>>> tones and consequently bypass media and tone detection are inherently
>>> incompatible.)
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jeff at jefflenk.com  Tue Jan 12 20:51:14 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 12 Jan 2010 20:51:14 -0800 (PST)
Subject: [Freeswitch-users] FSComm Windows build
In-Reply-To: <5800526b1001121918r15a336f5jbc5b724e66ac2825@mail.gmail.com>
References: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
	<1263350125191-4295967.post@n2.nabble.com>
	<5800526b1001121918r15a336f5jbc5b724e66ac2825@mail.gmail.com>
Message-ID: <1263358274180-4311097.post@n2.nabble.com>


Carlos,

Thanks for those links!

I am currenltly using prebuilt 4.6 32 bit libs and was hoping Nokia prebuilt
the x64 too. I guess I might have to look into building myself as well.

-Jeff


Jeff,

not yet. I'm still linking with a 32bit QT library of 4.5.2 I compiled for
VS2008 sometime last year. I followed a guide similar to this one:
http://dcsoft.com/community_server/blogs/dcsoft/archive/2009/03/06/how-to-setup-qt-4-5-visual-studio-integration.aspx

At the time I didn't see a need for a 64bit version of FsGui. Since FSComm
has FreeSWITCH in the back end this kind of changes things.

Looks like someone put up a wiki on compiling QT for Win x64:
http://en.wikibooks.org/wiki/Opticks_Developer_Guide/Getting_Started/Building_Qt_From_Source

Now that QT 4.6 is out I might have to revisit a new library build with 64
bit in mind.

Carlos

-- 
View this message in context: http://n2.nabble.com/FSComm-Windows-build-tp4289256p4311097.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From pmhshz at gmail.com  Tue Jan 12 20:56:44 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Wed, 13 Jan 2010 10:26:44 +0530
Subject: [Freeswitch-users] Defunct process in ESL testserver example
In-Reply-To: <191c3a031001120834q36f72791s811b6352ab7c12eb@mail.gmail.com>
References: <a792304b1001120722o2d4fbf91vb0456ef4f0436a5e@mail.gmail.com>
	<191c3a031001120834q36f72791s811b6352ab7c12eb@mail.gmail.com>
Message-ID: <a792304b1001122056m1885bf01ka5e2d64c9c183710@mail.gmail.com>

Thanks Anthony,

I added the below line before esl_listen to ignore the sigchild, and it
resolved the problem:
signal(SIGCHLD, SIG_IGN);
esl_listen(ip, port, mycallback);

-MohammedShehzad

On Tue, Jan 12, 2010 at 10:04 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> It's forking process code.
> you need to ignore sigchld or create a sig handler for sigchld and use the
> wait syscall to reap the process.
>
>
> On Tue, Jan 12, 2010 at 9:22 AM, MohammedShehzad <pmhshz at gmail.com> wrote:
>
>> Hello everybody,
>>
>> I am creating a C program of ESL outbound for call processing.
>> I am using testserver.c example, and till now it seems fine.
>>
>> But i noticed that every call testserver process, a new process is being
>> created which I can see in Linux system with below command:
>> For example, when I make two calls and even after hangup, I saw three
>> process like below:
>> ps -A | grep testserver
>>  9345 pts/2    00:00:00 testserver
>>  9350 pts/2    00:00:00 testserver
>>  9357 pts/2    00:00:00 testserver
>>
>> This get increased for every call i make.
>>
>> I did some workout and placed below two lines (close & exit) at the end of
>> mycallback function, (as I found them on ivrd.c file):
>>
>>     esl_disconnect(&handle);
>>     close(client_sock);
>>     exit(0);
>> }
>> But after that the process becomes defunct/zombie
>>
>>  9440 pts/2    00:00:00 conflisten
>>  9442 pts/2    00:00:00 conflisten <defunct>
>>  9452 pts/2    00:00:00 conflisten <defunct>
>>
>>
>> Will anybody please suggest me how can I eliminate this process, which
>> remains in memory even after call hangup?
>>
>> Thanks for any response.
>> MohammedShehzad
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From mcampbellsmith at gmail.com  Tue Jan 12 21:02:01 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 13 Jan 2010 16:02:01 +1100
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <03E1D9E8-ADB2-4D3D-BCDC-3F54ADD8879A@jerris.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
	<33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
	<33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>
	<20100112095332.GA32294@jdc.jasonjgw.net>
	<33c87fa31001120223t13efc07agcc595e48405fd8bb@mail.gmail.com>
	<33c87fa31001122004g3ad8fe11w701b14a2fcd03442@mail.gmail.com>
	<03E1D9E8-ADB2-4D3D-BCDC-3F54ADD8879A@jerris.com>
Message-ID: <33c87fa31001122102v573ebe3aj365915e850d5b926@mail.gmail.com>

ahha... thanks Mike.

So I could do something like this, if 1000 is the b-leg?:

                  <condition field="destination_number"
expression="^(10[01][0-9])$">
                        <action application="set" data="dialed_extension=$1"/>
                        <action application="set"
data="bypass_media_after_bridge=true"/>
                        <action application="tone_detect" data="fax
1100 r +5000 transfer 'receivefax XML features' 1" />
                        <action application="bridge"
data="user/${dialed_extension}@${domain}"/>
                  </condition>

Looks interesting - I'll have to give it a whirl later ....

On Wed, Jan 13, 2010 at 3:46 PM, Michael Jerris <mike at jerris.com> wrote:
> As soon as it pulls in the media (in this case, to do the tone detect), its going to give up on bypass media unless you adjust settings to behave otherwise. (such as http://wiki.freeswitch.org/wiki/Variable_bypass_media_after_bridge)
>
> Mike
>
> On Jan 12, 2010, at 11:04 PM, Mark Campbell-Smith wrote:
>
>> Hi All,
>>
>> Does anyone know if tone_detect can be used with bypass_media?
>>
>> I thought the SIP re-Invite message can be used to update media
>> parameters, including IP address endpoints. ?Does FS try to do this
>> in the case that tone_detect is used?
>>
>> In my case, the calls are dropped.
>>
>> On Tue, Jan 12, 2010 at 9:23 PM, Mark Campbell-Smith
>> <mcampbellsmith at gmail.com> wrote:
>>> Hi Jason,
>>>
>>> I have understood that. ?Its not that a difficult concept to understand!
>>>
>>> In the log I see:
>>> 2010-01-12 21:03:17.585598 [DEBUG] mod_dptools.c:818
>>> sofia/internal/1000 at 192.168.1.120 SET
>>> [ringback]=[v=-7;%(400,200,413,438);%(400,2000,413,438)]
>>> 2010-01-12 21:03:17.605591 [DEBUG] switch_core_session.c:1509
>>> Application tone_detect Requires media! pre_answering channel
>>> sofia/internal/1000 at 192.168.1.120
>>>
>>> I thought the SIP re-Invite message can be used to update media
>>> parameters, including IP address endpoints. ?Does FS try too do this
>>> in the case that tone_detect is used?
>>>
>>> On Tue, Jan 12, 2010 at 8:53 PM, Jason White <jason at jasonjgw.net> wrote:
>>>> Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>>>>> I've updated and tested bypass_media.
>>>>>
>>>>> It works if I remove this line from the B leg dialplan (ie 2010 calls
>>>>> 1000 - this is in the 1000 section of the dialplan):
>>>>> ? ? ? ? ? ? ? ? ? ? ? ? <action application="tone_detect" data="fax
>>>>> 1100 r +5000 transfer 'receivefax XML features' 1" />
>>>>>
>>>>> Does bypass_media work with tone_detect?
>>>>
>>>> As I understand it, tone_detect detects tones in the RTP stream (i.e., in the
>>>> audio). For this to be possible, FreeSWITCH has to be in the audio path, hence
>>>> bypass media cannot be used
>>>>
>>>> If this reasoning isn't obvious to you, then you've misunderstood what
>>>> tone_detect does or what bypass media is (the audio traffic flows directly
>>>> between the two endpoints without passing through the FreeSWITCH system that
>>>> establishes the connection, therefore FreeSWITCH can't process it to detect
>>>> tones and consequently bypass media and tone detection are inherently
>>>> incompatible.)
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mike at jerris.com  Tue Jan 12 21:14:55 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 00:14:55 -0500
Subject: [Freeswitch-users] Bypass_media mode
In-Reply-To: <33c87fa31001122102v573ebe3aj365915e850d5b926@mail.gmail.com>
References: <33c87fa31001110316v352a57afpd2d45e32dc248f32@mail.gmail.com>
	<33c87fa31001111509y323767eegf1b7fcf4da5502d3@mail.gmail.com>
	<bf6923ed1001111522y7e824d13k88523b42f929ca39@mail.gmail.com>
	<191c3a031001111608l5c568548xe2b9ea82b3c08c88@mail.gmail.com>
	<33c87fa31001111614m2030385fwd6f712a0fd72a094@mail.gmail.com>
	<33c87fa31001120132n699aafa2g67519776f96b3a6f@mail.gmail.com>
	<20100112095332.GA32294@jdc.jasonjgw.net>
	<33c87fa31001120223t13efc07agcc595e48405fd8bb@mail.gmail.com>
	<33c87fa31001122004g3ad8fe11w701b14a2fcd03442@mail.gmail.com>
	<03E1D9E8-ADB2-4D3D-BCDC-3F54ADD8879A@jerris.com>
	<33c87fa31001122102v573ebe3aj365915e850d5b926@mail.gmail.com>
Message-ID: <ABAE5818-5254-4DCA-8AAB-36E0D065EA5B@jerris.com>

I doubt that works as you will still need media after it has been bridged.  I was just giving an example of something that could be used.  You might be able to set that var, set tone detect, answer, play fake ringback for a short time (as long as you have you tone detect checking for tone), and then bridge.

Mike

On Jan 13, 2010, at 12:02 AM, Mark Campbell-Smith wrote:

> ahha... thanks Mike.
> 
> So I could do something like this, if 1000 is the b-leg?:
> 
>                  <condition field="destination_number"
> expression="^(10[01][0-9])$">
>                        <action application="set" data="dialed_extension=$1"/>
>                        <action application="set"
> data="bypass_media_after_bridge=true"/>
>                        <action application="tone_detect" data="fax
> 1100 r +5000 transfer 'receivefax XML features' 1" />
>                        <action application="bridge"
> data="user/${dialed_extension}@${domain}"/>
>                  </condition>
> 
> Looks interesting - I'll have to give it a whirl later ....




From xanlich at gmail.com  Wed Jan 13 00:48:40 2010
From: xanlich at gmail.com (Chia-Yen Wu)
Date: Wed, 13 Jan 2010 16:48:40 +0800
Subject: [Freeswitch-users] Hello , about calculate numbers
In-Reply-To: <2C92D649-BFC9-4FF9-AA1F-E6D0FDB5CA1A@jerris.com>
References: <314dc3f81001120908l5772b13fqd676a3be367bb662@mail.gmail.com>
	<201001121239.06756.sos@sokhapkin.dyndns.org>
	<EDF668FE-A76D-4A5F-96D1-756A7ACD13A6@jerris.com>
	<314dc3f81001121937l1b415c3amb715d50017850ebb@mail.gmail.com>
	<2C92D649-BFC9-4FF9-AA1F-E6D0FDB5CA1A@jerris.com>
Message-ID: <314dc3f81001130048k3fec1deaocf444a30aeb5a9ce@mail.gmail.com>

Sorry, my bad, I finally found out that I got an incorrect condition and the
stage goes wrong.
Everything is working fine now, thanks for help!


2010/1/13 Michael Jerris <mike at jerris.com>

> Strange, could you post the debug log of this so we can see how it's
> expanded?
>
> Mike
>
>
> On Jan 12, 2010, at 10:37 PM, Chia-Yen Wu <xanlich at gmail.com> wrote:
>
> Thanks for reply, I have tried the new one and did transfer back
> but it still dont work, when I was posting this question again with my test
> commands.
> suddenly I found out what the problem it is.
>
> the variable cannot with number in it, like:
>
> work:
> <action application="set" data="var=1"/>
> <action application="set" data="var=${expr(${var}+1)}"/>
> fail:
> <action application="set" data="var1=1"/>
> <action application="set" data="var1=${expr(${var1}+1)}"/>
>
>
> 2010/1/13 Michael Jerris < <mike at jerris.com>mike at jerris.com>
>
>> Remember you can't do conditions on these vars being set unless you
>> transfer back into the dialplan as these actions are not run immediately,
>> but rather after dialplan parse, unless you use
>> <http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions>
>> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>>
>> Mie
>>
>> On Jan 12, 2010, at 12:39 PM, Sergey Okhapkin wrote:
>>
>> > <action application="set" data="var=${expr(${var}+1)}"/>
>> >
>> > On Tuesday 12 January 2010, Chia-Yen Wu wrote:
>> >> Hello everyone,
>> >> Is there anyway to calculate number in dialplan.xml?
>> >>
>> >> like example:
>> >> <action application="set" data="var=1"/>
>> >> and with an action , I can do : var=var+1
>> >>
>> >> I have tried mod_expr, but failed to catch the variable, like below:
>> >> <action application="set" data="var=${expr($var+1)}"/>
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Wed Jan 13 01:13:03 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 13 Jan 2010 14:43:03 +0530
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
Message-ID: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>

Hi all,

I've done a sample program (In perl ESL) , which play a file to the caller
and then it will call recvEvent() and print the event name. I've handled
signals also.

When I send SIGINT to my program (Perl), the signal handler is called and I
can see the print output. But in the same time,  I received
SERVER_DISCONNECTED from freeswitch as event.

I don't know why I received SERVER_DISCONNECTED from freeswitch. Is it
because, the recvEvent() from perl internally  calls the recvevent function
in the Esl.c and when it waits to receive the information from socket, the
signal occurred???

Please clarify me!!

Here is my program
require ESL;
use IO::Socket::INET;
use Data::Dumper;

my $ip = "192.168.1.222";
my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
Proto => 'tcp',  Listen => 1,  Reuse => 1 );
die "Could not create socket: $!\n" unless $sock;
&register_Signals();

for(;;) {
    my $new_sock = $sock->accept();
    next if (not defined ($new_sock));
    my $pid = fork();
    if ($pid) {
        close($new_sock);
        next;
    }
    print "CHILD PID: $$\n";
    my $host = $new_sock->sockhost();
    my $fd = fileno($new_sock);

    my $con = new ESL::ESLconnection($fd);
    my $info = $con->getInfo();

    my $uuid = $info->getHeader("unique-id");

    printf "Connected call %s, from %s\n", $uuid,
$info->getHeader("caller-caller-id-number");
    my $r=$con->execute("answer");
    print Dumper $r;
    $con->events("plain","all");
    my
$re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");

    while($con->connected()) {
        my $e = $con->recvEvent();

        if ($e) {
            my $name = $e->getHeader("event-name");
            print "EVENT [$name]\n";
            if ($name eq "DTMF") {
                my $digit = $e->getHeader("dtmf-digit");
                print "$digit\n";
            }
        }
    }
    close($new_sock);
}
sub register_Signals() {
    foreach ( keys %SIG ) {
        $SIG{$_} = 'sig_Handler';
    }
}

sub sig_Handler() {
    my $handle=$_[0];
    if($handle eq "INT") {
        print "$$: SIGNAL SIG$handle is generated\n";
    }
}
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From tayeb.meftah at gmail.com  Wed Jan 13 01:56:30 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Wed, 13 Jan 2010 10:56:30 +0100
Subject: [Freeswitch-users] unable to dial out troug siemens Hie9200
	softswitch
Message-ID: <4B4D98CE.1060808@gmail.com>

hi dear all,
we have a siemens HIE 9200 softswitch that support SIP, SIP-T and H.248
we are trying to interconnect freeswitch with it
if HIE9200 dial out troug freeswitch, is passing the call, but without RTP
but if the freeswitch dial out troug the HIE9200, the call is unable to 
pass with error 500
the trace is here:

http://siplabs.net/tracebin/fs-siemens-500.pcap
http://siplabs.net/tracebin/fs-siemens-rtp.pcap
unfortunatly the freeswitch don't support H.248 otherwise i will control 
the media gateway (HIG 1100)
thanks for any help



From Claudio.Cavalera at italtel.it  Wed Jan 13 02:22:29 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Wed, 13 Jan 2010 11:22:29 +0100
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <191c3a031001121535t4819ee44p74cc9f7cd4fe972f@mail.gmail.com>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F362E0@BESONE.corp.dom>

Thanks a lot Anthony,
some comments inline (and please forgive me for my broken email client).
 
>>  example1: Consider this simple lua script in which i create two
sessions:

 >>  api = freeswitch.API();
 >>  api:execute("originate", "sofia/internal/1001%192.168.1.1 &park");
 >>  api:execute("originate", "sofia/internal/1002%192.168.1.1 &park");

 >  capture the output from api:execute the uuid is in there

Thx a lot,
this was one piece i was missing although it's already on the wiki here:
http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls 


 > because lua calls it freeswitch.bridge

 > session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
 > session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1",
session1);
 > freeswitch.bridge(session1, session2); 
 
good to now, there isn't any example of freeswitch.bridge in the wiki
and i would like to add one.
Where I could find the full api of
freeswitch.Session( ) ?
because I've seen this working also without "session1" in the second
line:
session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");
freeswitch.bridge(session1, session2); 

 also is there any difference between  freeswitch.bridge and
freeswitch.execute(uuid_bridge ...) ?
 
 >> example3: yet another possibility
 >> local session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
 >> session1:originate(session2, "sofia/internal/1002%192.168.1.1",
1000);
 >> but it does not work either.

 > The above is gibberish try:
 > local session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
 > session1:execute("bridge", "sofia/internal/1002%192.168.1.1"); 
 
Okay i will report also this bridge example on the wiki which was
missing.
But does session:originate make sense in some cases or not? Otherwise
i'm going to remove this line on the wiki
 http://wiki.freeswitch.org/wiki/Mod_lua#session:originate
<http://wiki.freeswitch.org/wiki/Mod_lua#session:originate> 
 
Thanks,
Claudio


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From david.villasmil.work at gmail.com  Wed Jan 13 03:11:34 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 13 Jan 2010 12:11:34 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
Message-ID: <5F707113-F78F-44C0-96A4-2C211F1C4791@gmail.com>

is it possible? can i bridge to multiple b-sides and m?ltiple





From david.villasmil.work at gmail.com  Wed Jan 13 03:15:14 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 13 Jan 2010 12:15:14 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
Message-ID: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>

hello

is it possible to bridge multiple b-legs and provide all audio  
(progress) until there is an answer on one channel?

thanks guys

David



From Prometheus001 at gmx.net  Wed Jan 13 03:26:13 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 13 Jan 2010 12:26:13 +0100
Subject: [Freeswitch-users] Multiple registrations: Only calling the first
 element in the list in this mode
Message-ID: <4B4DADD5.3010507@gmx.net>

Hello,

I habe the following behaviour

when I call a user which is registered twice with 2 phones via
    bridge user/100 at domain
both phones are ringing. This is correct as I allow multiple
registrations in a profile

However when I call multiple endpoints via
    bridge user/100 at domain,user/101 at domain,user/102 at domain
only one phone with number100 is ringing.

Console log shows "Only calling the first element in the list in this
mode.":
2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734 variable
string 0 = [presence_id=100 at domain]
2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734 variable
string 1 = [transfer_fallback_extension=100]
2010-01-12 19:52:18.236361 [WARNING] switch_ivr_originate.c:2048 Only
calling the first element in the list in this mode.
2010-01-12 19:52:18.236361 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip:100 at 10.11.12.203:2048
[9b95fcdc-ffab-11de-9ba2-13b2daa7ce61]

Is there any way to work around this? I need all phones to be ringing in
this scenario.

Best regards
Peter



From mailinglist at fribert.dk  Wed Jan 13 03:44:37 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Wed, 13 Jan 2010 12:44:37 +0100
Subject: [Freeswitch-users] Svar: Re:  Multi-Homed setup,
	starting over	 - still not working
Message-ID: <4B4DC035020000E1000003A2@mail.fribert.dk>

Hi Joseph

Oh yes I have :-)
But are you sayng that "eval ${domain}" will do more than is done when I restart the fs?
I don't see that anywhere in the guide :-D

I'll try it out asap!

Thanks
Fribse


>>> "Joseph L. Casale"  12-01-10 18:41 >>>
>Hi Guys
> 
>I really would like to have this up and running, but I'm constantly running into things that doesn't work, and I have no idea where the problem is...
> 
>I've started from scratch with pfSense and Freeswitch.
>I've followed the Multi Home tutorial here: http://wiki.freeswitch.org/wiki/Multi_home_tutorial
> 
>I've deleted the 'lan' profile, and altered the 'internal' profile as recommended, and I've gotten to the paragraph:
>
>APPLYING YOUR CHANGES AND CHECKING YOUR WORK
>
>When I start up my x-lite program I get this error:
>2010-01-12 16:38:54.172731 [WARNING] sofia_reg.c:1755 Can't find user [1000 at 83.89.x.x]
>You must define a domain called '83.89.x.xin your directory and add a user with the id="1000" attribute
>and you must configure your device to use the proper domain in it's authentication credentials.
>and you must configure your device to use the proper domain in it's authentication credentials.
> 
>83.89.x.x is my external IP, and not my internal IP???
>
>Any help on this?

This is because you haven't set your domain in vars.xml. The behavior is that
$${local_ip_v4} evals to your wan ip. This is the first step in that tutorial:)
http://wiki.freeswitch.org/wiki/Multi_home_tutorial#INTERNAL_LAN

Open vars.xml, make the line:
  
  

Match your lan ip:
  
  
 
restart fs, then goto the fs_cli and type `eval ${domain}` it should
come back with "your" lan ip.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From sos at sokhapkin.dyndns.org  Wed Jan 13 04:46:47 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 07:46:47 -0500
Subject: [Freeswitch-users] 10 seconds delay on fallback
Message-ID: <201001130746.47968.sos@sokhapkin.dyndns.org>

To implement fallback to backup PSTN routes and do LCR my dialplan executes 
the following commands:

set execute_on_answer=set hangup_after_bridge=true
set some custom channel variables
bridge sofia/gateway1/number
set some custom channel variables
bridge sofia/gateway2/number
....

This generally works fine if gateway1 returns SIP error, dialplan sets channel 
variables to another values and calls gateway2 immediately. But if GW1 
responds with 183 early media and SIP error after that, bad thing happens - 
dialplan continues immediately, sets channel variables, executes bridge 
application, but INVITE to GW2 is sent after 10 seconds delay. Caller gets 10 
extra seconds of post dial delay. I suspect the delay happens because of 
attempt to read audio frame, switch_ivr_originate() has the following lines:

if (switch_channel_media_ready(caller_channel)) {
	tstatus = switch_core_session_read_frame()
      ...

Any advice how to avoid extra delay in this situation?



From rupa at rupa.com  Wed Jan 13 04:50:21 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 13 Jan 2010 06:50:21 -0600
Subject: [Freeswitch-users] Multiple registrations: Only calling the
	first element in the list in this mode
In-Reply-To: <4B4DADD5.3010507@gmx.net>
References: <4B4DADD5.3010507@gmx.net>
Message-ID: <b39273531001130450k16af27d2v209a300f441d39d8@mail.gmail.com>

Try:

bridge user/100 at domain:_:user/101 at domain:_:user/102 at domain

Then document it up if it works.

On Wed, Jan 13, 2010 at 5:26 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello,
>
> I habe the following behaviour
>
> when I call a user which is registered twice with 2 phones via
>    bridge user/100 at domain
> both phones are ringing. This is correct as I allow multiple
> registrations in a profile
>
> However when I call multiple endpoints via
>    bridge user/100 at domain,user/101 at domain,user/102 at domain
> only one phone with number100 is ringing.
>
> Console log shows "Only calling the first element in the list in this
> mode.":
> 2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734 variable
> string 0 = [presence_id=100 at domain]
> 2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734 variable
> string 1 = [transfer_fallback_extension=100]
> 2010-01-12 19:52:18.236361 [WARNING] switch_ivr_originate.c:2048 Only
> calling the first element in the list in this mode.
> 2010-01-12 19:52:18.236361 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/sip:100 at 10.11.12.203:2048
> [9b95fcdc-ffab-11de-9ba2-13b2daa7ce61]
>
> Is there any way to work around this? I need all phones to be ringing in
> this scenario.
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From a.alalousi at gmail.com  Wed Jan 13 05:51:54 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Wed, 13 Jan 2010 13:51:54 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <eafa772f1001120520i719751b9hcbc22a969a89a585@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
	<87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
	<cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>
	<eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>
	<cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>
	<eafa772f1001120520i719751b9hcbc22a969a89a585@mail.gmail.com>
Message-ID: <cd93b3701001130551v26fe6355w77f532a86ba4e08a@mail.gmail.com>

Steve/All,

Thanks for all your feedback, this thread can be closed.

Here is some feedback:

Re: Steve's queries:

- "pproxy" was a typo on my part, but it should not affect anything. I set
them both to false in vars.xml, and don't over-ride them anywhere.
- true re: default values for proxy_media & bypass_media being false. I am
explicitly setting them here out of 1) paranoia and 2) I like to make sure I
know my variable values and not leave them to defaults - relics of being a
developer
- I was disabling the q850 code as part of my attempts to crack this nut.

Re: the solution, I've managed to rewrite some of the codes with a call to
the hangup app, which is what Steve is also using, and his findings re:
bridges getting 183 and 180 before 4xx.

I wonder if it's possible to rewrite causes from such bridges by executing a
JS or similar app attached to the bridge. I'll report on this as and when.

Regards,

Ahmed.




2010/1/12 Steven Ayre <steveayre at gmail.com>

> - prpoxy_media should be proxy_media
> - bypass_media and proxy_media shouldn't need setting to false -
> that's their default (unless you're set one of them to true on the sip
> profile?)
> - why do you need to disable q850 reason?
>
> I do something very similar - try this...
>
> <condition field="destination_number" expression="^(?:5824)(44(1|2)\d+)$">
>   <action application="set" data="continue_on_fail=true"/>
>  <action application="set" data="hangup_after_bridge=true"/>
>  <action application="set" data="sip_ignore_remote_cause=true"/>
>   <action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1"/>
>   <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
> </condition>
>
> By the way, you'll be unable to rewrite the hangup cause for a bridge
> that gets a 180 or 183 packet from the gateway before getting a 4xx,
> 5xx or 6xx packet (because those bridges don't 'fail') - I've seen
> this on a SIP provider before who gives 183 Session Progress before a
> 404 Not Found if the PSTN number dialled doesn't exist.
>
> Regards,
> -Steve
>
>
> 2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
> > Here you go:
> >
> > <condition field="destination_number" expression="^(?:5824)(44(1|2)\d+)$"
> > break="on-true">
> >   <action inline="true" application="set" data="bypass_media=false"/>
> >   <action inline="true" application="set" data="prpoxy_media=false"/>
> >   <action inline="true" application="set"
> data="disable_q850_reason=true"/>
> >   <action inline="true" application="set"
> data="hangup_after_bridge=false"/>
> >   <action inline="true" application="set"
> > data="sip_ignore_remote_cause=true"/>
> >   <action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1"/>
> >   <action inline="true" application="set"
> > data="bridge_hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
> >   <action inline="true" application="set"
> > data="hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
> >   <action application="bridge" data="error/normal_circuit_congestion"/>
> >   <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
> > </condition>
> >
> > As you can see, I am trying to rewrite the hangup codes in a multitude of
> > ways and places, but still exhibit the same behaviour.
> >
> > Any help appreciated.
> >
> > Regards,
> >
> > Ahmed.
> >
> > 2010/1/12 Steven Ayre <steveayre at gmail.com>
> >>
> >> Can you show us the dialplan extension you're trying?
> >>
> >> Thanks,
> >> -Steve
> >>
> >> 2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
> >> > Hi Michael,
> >> >
> >> > This is exactly what I'm doing, but it's just not happening.
> >> >
> >> > Thanks,
> >> >
> >> > Ahmed.
> >> >
> >> >
> >> > 2010/1/12 Michael Collins <msc at freeswitch.org>
> >> >>
> >> >>
> >> >> On Mon, Jan 11, 2010 at 4:18 AM, Ahmed Naji <a.alalousi at gmail.com>
> >> >> wrote:
> >> >>>
> >> >>> Dear All,
> >> >>>
> >> >>> I posted a thread re the subject but didn't get any joy, so perhaps
> >> >>> second time lucky.
> >> >>>
> >> >>> I need to rewrite a couple of hangup causes to mean
> NORMAL_CONGESTION
> >> >>> and
> >> >>> no matter which variables I set, this isn't happening. The idea is
> to
> >> >>> control what codes are returned to an end point after a successful
> >> >>> bridge,
> >> >>> as well as deal with what codes are returned if the bridge is
> >> >>> unsuccessful
> >> >>> (e.g. user_busy, originator_cancel ...etc).
> >> >>>
> >> >>> I've had limited success by setting hangup_after_bridge=false then
> >> >>> bridging to error/<required_code>. This, however only works when the
> >> >>> B-leg
> >> >>> terminates the call after a successful answer. Any other codes are
> not
> >> >>> rewritten.
> >> >>>
> >> >>> I've also tried playing with the bridge_hangup_code and hangup_code
> >> >>> variables prior and after bridging, still no joy. I have also set
> >> >>> sip_ignore_remote_cause=true prior to entering the bridge, as well
> >> >>> explicitly in vars.xml.
> >> >>>
> >> >>> By the way, I'm running in proxy-media mode, but I did try it with
> >> >>> bypass-media as well. Same symptoms, same behaviour.
> >> >>>
> >> >>> Any help with this would be highly appreciated.
> >> >>>
> >> >> Well, I do know that when you do a hangup in the dialplan you can
> pass
> >> >> an
> >> >> optional cause as well:
> >> >> <action application="hangup" data="USER_BUSY"/>
> >> >> If you are doing the hanging up then you have a fair amount of
> >> >> control...
> >> >> -MC
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> >
> >> > --
> >> > Ahmed Naji
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Ahmed Naji
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From kond at nstel.ru  Wed Jan 13 06:10:02 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Wed, 13 Jan 2010 17:10:02 +0300
Subject: [Freeswitch-users] sip trunk question: why call through external
	profile is challenged?
Message-ID: <20100113141003.970B311F32@mail.nstel.ru>

Hi all!

 

I'm brand new to FreeSwitch, but have some experience with SipX.

Our company is evaluating FS.

For test purposes I set up FS on a virtual machine (vmware esxi). I use
CentOS.

The FS version I use is 1.0.5-20100110-0400.

 

I have a question regarding sip trunk between FS and SipX.

I created the following GW in external profile: 

[freeswitch at freeswitch external]$ cat sipx-lab.xml | grep -v '<!--'

<include>

  <gateway name="sipx4.lab.nstel.ru">

  <param name="username" value="zxcv"/>

  <param name="password" value="2007"/>

  <param name="register" value="false"/>

  </gateway>

</include>

 

External.xml file is not modified after installation.

 

I see this gateway via fs_cli:

freeswitch at internal> sofia status gateway sipx4.lab.nstel.ru

============================================================================
=====================

Name            sipx4.lab.nstel.ru

Profile         external

Scheme          Digest

Realm           sipx4.lab.nstel.ru

Username        zxcv

Password        yes

>From            <sip:zxcv at sipx4.lab.nstel.ru;transport=udp>

Contact
<sip:gw+sipx4.lab.nstel.ru at 172.23.22.49:5080;transport=udp;gw=sipx4.lab.nste
l.ru>

Exten           zxcv

To              sip:zxcv at sipx4.lab.nstel.ru

Proxy           sip:sipx4.lab.nstel.ru

Context         public

Expires         3600

Freq            3600

Ping            0

PingFreq        0

PingState       0/0/0

State           NOREG

Status          UP

CallsIN         0

CallsOUT        0

============================================================================
=====================

 

I created new FS extension 2853. I registers (xlite) and I can call it.

 

Now I want to call FS user from sipx.

 

>From the sipx side one can configure link to FS differently. There are two
options:

1. Call FS directly trough sipxproxy (the core part of sipx works as sip
proxy, not as b2bua)

2. Call trough embedded b2bua, named sipxbridge.

 

When a call is going trough sipxbridge, it is successfully landed at FS
extension 2853.

But when a call is going from the sipxproxy, it is challenged with Status
407 "proxy authentication required', and then call fails.

(I'm not sure if sipx should handle this challenge, but this is separate
question for the sipx forum).

At the default log level I see nothing in the freeswitch.log.

 

So the question is why one call through external profile is being challenged
while the other is not?

I suspect that the reason is in the difference in the two Invite messages:

 

Here is challenged Invite:

INVITE sip:2853 at fs.lab.nstel.ru:5080;transport=udp SIP/2.0

Record-Route:
<sip:172.23.12.104:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EOTRiZjI3MjgtYWMxN
zBjZTktMTNjNC0yZjZkZWMtNjNkNmUwN2UtMmY2ZGVj.900_ntap%2Aid%7EMjIyNjAtMQ%60%60
%213ba6d85f6946c4c6001bee1d3b54474f>

From:
"testphone3857"<sip:3857 at lab.nstel.ru>;tag=94bf2728-ac170ce9-13c4-2f6dec-63d
6e07e-2f6dec

To: <sip:2853 at lab.nstel.ru>

Call-Id: 94bed2a0-ac170ce9-13c4-2f6dec-68ffdb81-2f6dec at lab.nstel.ru

Cseq: 2 INVITE

Via: SIP/2.0/UDP
172.23.12.104;branch=z9hG4bK-sipXecs-000eaf48a36fe4029c7cde004a6f44424847

Via: SIP/2.0/TCP
172.23.12.104;branch=z9hG4bK-sipXecs-000bf3008cfa3f3e70470789c75232ba9499~ac
7fd729330fd563f83cacd941311e75;id=22260-1

Via: SIP/2.0/UDP 172.23.12.233:5060;branch=z9hG4bK-2f6dec-b9456412-7ad07784

Max-Forwards: 18

Supported: replaces

User-Agent: LG-Nortel LIP 6812 v1.2.38sp SN/00405A18B634

Contact: <sip:3857 at 172.23.12.233:5060;x-sipX-nonat>

Proxy-Authorization: Digest
username="3857",realm="lab.nstel.ru",nonce="e2152722611af1bbe59f3a8eb31b8eb8
4b4db2ca",uri="sip:2853 at lab.nstel.ru",response="a09c34fbd897783c9b9af50bd044
ccaa",algorithm=MD5

Content-Type: application/sdp

Content-Length: 301

Date: Wed, 13 Jan 2010 11:47:22 GMT

Expires: 60

X-Sipx-Handled: X172.23.12.104-81.211.30.104

 

v=0

o=LGEIPP 16246 16247 IN IP4 172.23.12.233

s=SIP Call

c=IN IP4 172.23.12.233

t=0 0

m=audio 23008 RTP/AVP 0 8 18 4 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:18 annexb=no

a=fmtp:101 0-11

a=sendrecv

 

 

And here is successful Invite:

INVITE sip:2853 at fs.lab.nstel.ru;user=phone SIP/2.0

Call-ID: 94bec3b8-ac170ce9-13c4-2f6d47-2fee09d0-2f6d47 at lab.nstel.ru.0

CSeq: 1 INVITE

From: "testphone3857" <sip:3857 at 172.23.12.104>;tag=816640414244159961

To: <sip:2853 at fs.lab.nstel.ru;user=phone>

Via: SIP/2.0/UDP
172.23.12.104:5080;branch=z9hG4bK885dde2c8680f5315845cd3350b8b605373534

Max-Forwards: 70

User-Agent: sipXecs/4.0.2 sipXecs/sipxbridge (Linux)

P-Asserted-Identity: <sip:1004 at 172.23.12.104>

Contact: <sip:1004 at 172.23.12.104:5080;transport=udp>

Route: <sip:172.23.22.49:5080;transport=udp;lr>

Session-Expires: 1800;refresher=uac

Allow: INVITE,BYE,ACK,CANCEL,OPTIONS

Content-Type: application/sdp

Content-Length: 315

 

v=0

o=sipxbridge 6640787141824452741 1 IN IP4 172.23.12.104

s=SIP Call

c=IN IP4 172.23.12.104

t=0 0

m=audio 30248 RTP/AVP 0 8 18 4 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:18 annexb=no

a=fmtp:101 0-11

a=sendrecv

 

Can somebody please if it is a FS configuration problem or a software
problem or is it a problem on the sipx side?

 

Thanks in advance,

Nikolay.

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From juanbackson at gmail.com  Wed Jan 13 06:19:33 2010
From: juanbackson at gmail.com (Juan Backson)
Date: Wed, 13 Jan 2010 22:19:33 +0800
Subject: [Freeswitch-users] need help with switch_ivr_originate
Message-ID: <27c25bc41001130619g3213e37bo8f79e6ee1984701a@mail.gmail.com>

Hi,

I tried to use the function switch_ivr_originate to originate a call instead
of using the "bridge" app.

However, I am getting "ORIGINATOR CANCEL" error after switch_ivr_originate
is executed.

Does anyone know why would freeswitch cancel the call ?

I have checked all sample usage of switch_ivr_originate in the code but no
luck.

thx,
jb
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From steveayre at gmail.com  Wed Jan 13 06:25:32 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 13 Jan 2010 14:25:32 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <cd93b3701001130551v26fe6355w77f532a86ba4e08a@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
	<87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
	<cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>
	<eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>
	<cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>
	<eafa772f1001120520i719751b9hcbc22a969a89a585@mail.gmail.com>
	<cd93b3701001130551v26fe6355w77f532a86ba4e08a@mail.gmail.com>
Message-ID: <eafa772f1001130625r14c40cb7q584f5f84a175e21b@mail.gmail.com>

"I wonder if it's possible to rewrite causes from such bridges by
executing a JS or similar app attached to the bridge. I'll report on
this as and when."

If you run a LUA (or JS, etc) script where the hangup app currently
is, then you'll be able to check the current hangup cause at that
point within the script and conditionally run the hangup app from your
script with the cause you want to rewrite to.

-Steve

2010/1/13 Ahmed Naji <a.alalousi at gmail.com>:
> Steve/All,
>
> Thanks for all your feedback, this thread can be closed.
>
> Here is some feedback:
>
> Re: Steve's queries:
>
> - "pproxy" was a typo on my part, but it should not affect anything. I set
> them both to false in vars.xml, and don't over-ride them anywhere.
> - true re: default values for proxy_media & bypass_media being false. I am
> explicitly setting them here out of 1) paranoia and 2) I like to make sure I
> know my variable values and not leave them to defaults - relics of being a
> developer
> - I was disabling the q850 code as part of my attempts to crack this nut.
>
> Re: the solution, I've managed to rewrite some of the codes with a call to
> the hangup app, which is what Steve is also using, and his findings re:
> bridges getting 183 and 180 before 4xx.
>
> I wonder if it's possible to rewrite causes from such bridges by executing a
> JS or similar app attached to the bridge. I'll report on this as and when.
>
> Regards,
>
> Ahmed.
>
>
>
>
> 2010/1/12 Steven Ayre <steveayre at gmail.com>
>>
>> - prpoxy_media should be proxy_media
>> - bypass_media and proxy_media shouldn't need setting to false -
>> that's their default (unless you're set one of them to true on the sip
>> profile?)
>> - why do you need to disable q850 reason?
>>
>> I do something very similar - try this...
>>
>> <condition field="destination_number" expression="^(?:5824)(44(1|2)\d+)$">
>> ?<action application="set" data="continue_on_fail=true"/>
>> ?<action application="set" data="hangup_after_bridge=true"/>
>> ?<action application="set" data="sip_ignore_remote_cause=true"/>
>> ?<action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1"/>
>> ?<action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
>> </condition>
>>
>> By the way, you'll be unable to rewrite the hangup cause for a bridge
>> that gets a 180 or 183 packet from the gateway before getting a 4xx,
>> 5xx or 6xx packet (because those bridges don't 'fail') - I've seen
>> this on a SIP provider before who gives 183 Session Progress before a
>> 404 Not Found if the PSTN number dialled doesn't exist.
>>
>> Regards,
>> -Steve
>>
>>
>> 2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
>> > Here you go:
>> >
>> > <condition field="destination_number"
>> > expression="^(?:5824)(44(1|2)\d+)$"
>> > break="on-true">
>> > ? <action inline="true" application="set" data="bypass_media=false"/>
>> > ? <action inline="true" application="set" data="prpoxy_media=false"/>
>> > ? <action inline="true" application="set"
>> > data="disable_q850_reason=true"/>
>> > ? <action inline="true" application="set"
>> > data="hangup_after_bridge=false"/>
>> > ? <action inline="true" application="set"
>> > data="sip_ignore_remote_cause=true"/>
>> > ? <action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1"/>
>> > ? <action inline="true" application="set"
>> > data="bridge_hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
>> > ? <action inline="true" application="set"
>> > data="hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
>> > ? <action application="bridge" data="error/normal_circuit_congestion"/>
>> > ? <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
>> > </condition>
>> >
>> > As you can see, I am trying to rewrite the hangup codes in a multitude
>> > of
>> > ways and places, but still exhibit the same behaviour.
>> >
>> > Any help appreciated.
>> >
>> > Regards,
>> >
>> > Ahmed.
>> >
>> > 2010/1/12 Steven Ayre <steveayre at gmail.com>
>> >>
>> >> Can you show us the dialplan extension you're trying?
>> >>
>> >> Thanks,
>> >> -Steve
>> >>
>> >> 2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
>> >> > Hi Michael,
>> >> >
>> >> > This is exactly what I'm doing, but it's just not happening.
>> >> >
>> >> > Thanks,
>> >> >
>> >> > Ahmed.
>> >> >
>> >> >
>> >> > 2010/1/12 Michael Collins <msc at freeswitch.org>
>> >> >>
>> >> >>
>> >> >> On Mon, Jan 11, 2010 at 4:18 AM, Ahmed Naji <a.alalousi at gmail.com>
>> >> >> wrote:
>> >> >>>
>> >> >>> Dear All,
>> >> >>>
>> >> >>> I posted a thread re the subject but didn't get any joy, so perhaps
>> >> >>> second time lucky.
>> >> >>>
>> >> >>> I need to rewrite a couple of hangup causes to mean
>> >> >>> NORMAL_CONGESTION
>> >> >>> and
>> >> >>> no matter which variables I set, this isn't happening. The idea is
>> >> >>> to
>> >> >>> control what codes are returned to an end point after a successful
>> >> >>> bridge,
>> >> >>> as well as deal with what codes are returned if the bridge is
>> >> >>> unsuccessful
>> >> >>> (e.g. user_busy, originator_cancel ...etc).
>> >> >>>
>> >> >>> I've had limited success by setting hangup_after_bridge=false then
>> >> >>> bridging to error/<required_code>. This, however only works when
>> >> >>> the
>> >> >>> B-leg
>> >> >>> terminates the call after a successful answer. Any other codes are
>> >> >>> not
>> >> >>> rewritten.
>> >> >>>
>> >> >>> I've also tried playing with the bridge_hangup_code and hangup_code
>> >> >>> variables prior and after bridging, still no joy. I have also set
>> >> >>> sip_ignore_remote_cause=true prior to entering the bridge, as well
>> >> >>> explicitly in vars.xml.
>> >> >>>
>> >> >>> By the way, I'm running in proxy-media mode, but I did try it with
>> >> >>> bypass-media as well. Same symptoms, same behaviour.
>> >> >>>
>> >> >>> Any help with this would be highly appreciated.
>> >> >>>
>> >> >> Well, I do know that when you do a hangup in the dialplan you can
>> >> >> pass
>> >> >> an
>> >> >> optional cause as well:
>> >> >> <action application="hangup" data="USER_BUSY"/>
>> >> >> If you are doing the hanging up then you have a fair amount of
>> >> >> control...
>> >> >> -MC
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> > Ahmed Naji
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Ahmed Naji
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Ahmed Naji
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From dftoro at yahoo.com  Wed Jan 13 06:44:13 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Wed, 13 Jan 2010 06:44:13 -0800 (PST)
Subject: [Freeswitch-users] sangoma echo cancellation software
In-Reply-To: <4B05F78E.3080007@coppice.org>
Message-ID: <796826.47173.qm@web33506.mail.mud.yahoo.com>

greetings,
?
I have sangoma analog card without echo cancellation hardware, I have problems with echo on calls, does FreeSwitch support echo cancellation software on windows ?
?
Thanks

Diego Toro
http://lacarretade.blogspot.com


      
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From a.alalousi at gmail.com  Wed Jan 13 07:13:09 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Wed, 13 Jan 2010 15:13:09 +0000
Subject: [Freeswitch-users] Help rewriting hangup causes
In-Reply-To: <eafa772f1001130625r14c40cb7q584f5f84a175e21b@mail.gmail.com>
References: <cd93b3701001110418m5114270cmc36e3b5f79161e7b@mail.gmail.com>
	<87f2f3b91001111655t6a2571a2u9b3222700664409a@mail.gmail.com>
	<cd93b3701001112356n530839c2qeab122ea522e9756@mail.gmail.com>
	<eafa772f1001120046i6b2e03a2td3bbbc0636dde579@mail.gmail.com>
	<cd93b3701001120152y2a7518c4x1004e45c76eb70c9@mail.gmail.com>
	<eafa772f1001120520i719751b9hcbc22a969a89a585@mail.gmail.com>
	<cd93b3701001130551v26fe6355w77f532a86ba4e08a@mail.gmail.com>
	<eafa772f1001130625r14c40cb7q584f5f84a175e21b@mail.gmail.com>
Message-ID: <cd93b3701001130713x5078457fpbf56b5ee45238e6e@mail.gmail.com>

Precisely my line of thought Steve. I'm almost done with a LUA to do the
call setup/tare-down and manipulate hangup causes as needed.

While I'm at it, I think I may as well handle codec prefs ...etc. within the
same script.

Will let you and the others how things progress.

BR,

Ahmed.

2010/1/13 Steven Ayre <steveayre at gmail.com>

> "I wonder if it's possible to rewrite causes from such bridges by
> executing a JS or similar app attached to the bridge. I'll report on
> this as and when."
>
> If you run a LUA (or JS, etc) script where the hangup app currently
> is, then you'll be able to check the current hangup cause at that
> point within the script and conditionally run the hangup app from your
> script with the cause you want to rewrite to.
>
> -Steve
>
> 2010/1/13 Ahmed Naji <a.alalousi at gmail.com>:
> > Steve/All,
> >
> > Thanks for all your feedback, this thread can be closed.
> >
> > Here is some feedback:
> >
> > Re: Steve's queries:
> >
> > - "pproxy" was a typo on my part, but it should not affect anything. I
> set
> > them both to false in vars.xml, and don't over-ride them anywhere.
> > - true re: default values for proxy_media & bypass_media being false. I
> am
> > explicitly setting them here out of 1) paranoia and 2) I like to make
> sure I
> > know my variable values and not leave them to defaults - relics of being
> a
> > developer
> > - I was disabling the q850 code as part of my attempts to crack this nut.
> >
> > Re: the solution, I've managed to rewrite some of the codes with a call
> to
> > the hangup app, which is what Steve is also using, and his findings re:
> > bridges getting 183 and 180 before 4xx.
> >
> > I wonder if it's possible to rewrite causes from such bridges by
> executing a
> > JS or similar app attached to the bridge. I'll report on this as and
> when.
> >
> > Regards,
> >
> > Ahmed.
> >
> >
> >
> >
> > 2010/1/12 Steven Ayre <steveayre at gmail.com>
> >>
> >> - prpoxy_media should be proxy_media
> >> - bypass_media and proxy_media shouldn't need setting to false -
> >> that's their default (unless you're set one of them to true on the sip
> >> profile?)
> >> - why do you need to disable q850 reason?
> >>
> >> I do something very similar - try this...
> >>
> >> <condition field="destination_number"
> expression="^(?:5824)(44(1|2)\d+)$">
> >>  <action application="set" data="continue_on_fail=true"/>
> >>  <action application="set" data="hangup_after_bridge=true"/>
> >>  <action application="set" data="sip_ignore_remote_cause=true"/>
> >>  <action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1"/>
> >>  <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
> >> </condition>
> >>
> >> By the way, you'll be unable to rewrite the hangup cause for a bridge
> >> that gets a 180 or 183 packet from the gateway before getting a 4xx,
> >> 5xx or 6xx packet (because those bridges don't 'fail') - I've seen
> >> this on a SIP provider before who gives 183 Session Progress before a
> >> 404 Not Found if the PSTN number dialled doesn't exist.
> >>
> >> Regards,
> >> -Steve
> >>
> >>
> >> 2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
> >> > Here you go:
> >> >
> >> > <condition field="destination_number"
> >> > expression="^(?:5824)(44(1|2)\d+)$"
> >> > break="on-true">
> >> >   <action inline="true" application="set" data="bypass_media=false"/>
> >> >   <action inline="true" application="set" data="prpoxy_media=false"/>
> >> >   <action inline="true" application="set"
> >> > data="disable_q850_reason=true"/>
> >> >   <action inline="true" application="set"
> >> > data="hangup_after_bridge=false"/>
> >> >   <action inline="true" application="set"
> >> > data="sip_ignore_remote_cause=true"/>
> >> >   <action application="bridge" data="sofia/gateway/sip.xx.xx.com/$1
> "/>
> >> >   <action inline="true" application="set"
> >> > data="bridge_hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
> >> >   <action inline="true" application="set"
> >> > data="hangup_cause=NORMAL_CIRCUIT_CONGESTION"/>
> >> >   <action application="bridge"
> data="error/normal_circuit_congestion"/>
> >> >   <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
> >> > </condition>
> >> >
> >> > As you can see, I am trying to rewrite the hangup codes in a multitude
> >> > of
> >> > ways and places, but still exhibit the same behaviour.
> >> >
> >> > Any help appreciated.
> >> >
> >> > Regards,
> >> >
> >> > Ahmed.
> >> >
> >> > 2010/1/12 Steven Ayre <steveayre at gmail.com>
> >> >>
> >> >> Can you show us the dialplan extension you're trying?
> >> >>
> >> >> Thanks,
> >> >> -Steve
> >> >>
> >> >> 2010/1/12 Ahmed Naji <a.alalousi at gmail.com>:
> >> >> > Hi Michael,
> >> >> >
> >> >> > This is exactly what I'm doing, but it's just not happening.
> >> >> >
> >> >> > Thanks,
> >> >> >
> >> >> > Ahmed.
> >> >> >
> >> >> >
> >> >> > 2010/1/12 Michael Collins <msc at freeswitch.org>
> >> >> >>
> >> >> >>
> >> >> >> On Mon, Jan 11, 2010 at 4:18 AM, Ahmed Naji <a.alalousi at gmail.com
> >
> >> >> >> wrote:
> >> >> >>>
> >> >> >>> Dear All,
> >> >> >>>
> >> >> >>> I posted a thread re the subject but didn't get any joy, so
> perhaps
> >> >> >>> second time lucky.
> >> >> >>>
> >> >> >>> I need to rewrite a couple of hangup causes to mean
> >> >> >>> NORMAL_CONGESTION
> >> >> >>> and
> >> >> >>> no matter which variables I set, this isn't happening. The idea
> is
> >> >> >>> to
> >> >> >>> control what codes are returned to an end point after a
> successful
> >> >> >>> bridge,
> >> >> >>> as well as deal with what codes are returned if the bridge is
> >> >> >>> unsuccessful
> >> >> >>> (e.g. user_busy, originator_cancel ...etc).
> >> >> >>>
> >> >> >>> I've had limited success by setting hangup_after_bridge=false
> then
> >> >> >>> bridging to error/<required_code>. This, however only works when
> >> >> >>> the
> >> >> >>> B-leg
> >> >> >>> terminates the call after a successful answer. Any other codes
> are
> >> >> >>> not
> >> >> >>> rewritten.
> >> >> >>>
> >> >> >>> I've also tried playing with the bridge_hangup_code and
> hangup_code
> >> >> >>> variables prior and after bridging, still no joy. I have also set
> >> >> >>> sip_ignore_remote_cause=true prior to entering the bridge, as
> well
> >> >> >>> explicitly in vars.xml.
> >> >> >>>
> >> >> >>> By the way, I'm running in proxy-media mode, but I did try it
> with
> >> >> >>> bypass-media as well. Same symptoms, same behaviour.
> >> >> >>>
> >> >> >>> Any help with this would be highly appreciated.
> >> >> >>>
> >> >> >> Well, I do know that when you do a hangup in the dialplan you can
> >> >> >> pass
> >> >> >> an
> >> >> >> optional cause as well:
> >> >> >> <action application="hangup" data="USER_BUSY"/>
> >> >> >> If you are doing the hanging up then you have a fair amount of
> >> >> >> control...
> >> >> >> -MC
> >> >> >>
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >>
> >> >> >
> >> >> >
> >> >> >
> >> >> > --
> >> >> > Ahmed Naji
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > --
> >> > Ahmed Naji
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Ahmed Naji
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From jeff at jefflenk.com  Wed Jan 13 07:20:12 2010
From: jeff at jefflenk.com (Jeff Lenk )
Date: Wed, 13 Jan 2010 15:20:12 +0000
Subject: [Freeswitch-users] sangoma echo cancellation software
Message-ID: <COL117-DS9F4F51BA87712328B80E3B96B0@phx.gbl>

No such luck - I dont know any solution other than get the other card


-----Original Message-----
From: Diego Toro
Sent: 1/13/2010 2:44:13 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] sangoma echo cancellation software
greetings,

I have sangoma analog card without echo cancellation hardware, I have problems with echo on calls, does FreeSwitch support echo cancellation software on windows ?

Thanks

Diego Toro
http://lacarretade.blogspot.com


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From jcasale at activenetwerx.com  Wed Jan 13 07:30:10 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 13 Jan 2010 15:30:10 +0000
Subject: [Freeswitch-users] Svar: Re:  Multi-Homed setup,
	starting over	 - still not working
In-Reply-To: <4B4DC035020000E1000003A2@mail.fribert.dk>
References: <4B4DC035020000E1000003A2@mail.fribert.dk>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9079C65@prato.activenetwerx.local>

>But are you sayng that "eval ${domain}" will do more than is done when I restart the fs?
>I don't see that anywhere in the guide :-D
>
>I'll try it out asap!

What I am saying is it "evaluates" that placeholder and shows
its value. You can use it to show the current value to verify
it matches what you intend, which is for it to be the same as
your LAN ip.



From devel at thom.fr.eu.org  Wed Jan 13 07:32:16 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Wed, 13 Jan 2010 16:32:16 +0100
Subject: [Freeswitch-users] =?utf-8?q?playback_and_play=5Fand=5Fget=5Fdigi?=
 =?utf-8?q?ts_strange_misunderstanding?=
Message-ID: <cdeeb122e98434d6303df4b05f8cbada@thom.fr.eu.org>



Hello, 

trying to make so dialplan extensions that use the playbvack
and play_and_get_digits applications, but I'm having trouble with the file
name specification. 

The files I want to play are in the french language
(fr/fr/julie as configured in lang/fr/fr.xml) 

My extension is as follows
: 

The channel is using a bit rate of 8000 Hz, so by the set
default_language=fr I would expect freeswitch to playback the file at
${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/8000/ringing_disabled.wav
whereas it tries to playback the file at
${FREESWITCH_PATH}/sounds/en/us/callie/misc/ringing_disabled.wav 

I have
the same with play_and_get_digits application. 

What am I doing wrong ?


Fran?ois
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From Claudio.Cavalera at italtel.it  Wed Jan 13 07:51:10 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Wed, 13 Jan 2010 16:51:10 +0100
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0F362E0@BESONE.corp.dom>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F362E4@BESONE.corp.dom>

Also after
api = freeswitch.API();
what is the difference between
api:execute()
 
and
api:executeString() ?
 
I've found example of both uses in
http://wiki.freeswitch.org/wiki/Mod_lua
and in 
http://wiki.freeswitch.org/wiki/Make_API_calls_directly_from_Lua_code
 
Thanks,
Claudio
 
 


________________________________

	From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Cavalera Claudio Luigi
	Sent: Wednesday, January 13, 2010 11:22 AM
	To: freeswitch-users at lists.freeswitch.org
	Subject: Re: [Freeswitch-users] playing with sessions in lua
	
	
	Thanks a lot Anthony,
	some comments inline (and please forgive me for my broken email
client).
	 
	>>  example1: Consider this simple lua script in which i create
two sessions:
	
	 >>  api = freeswitch.API();
	 >>  api:execute("originate", "sofia/internal/1001%192.168.1.1
&park");
	 >>  api:execute("originate", "sofia/internal/1002%192.168.1.1
&park");
	
	 >  capture the output from api:execute the uuid is in there
	
	Thx a lot,
	this was one piece i was missing although it's already on the
wiki here:
	http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls 
	


	 > because lua calls it freeswitch.bridge
	
	 > session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
	 > session2 =
freeswitch.Session("sofia/internal/1002%192.168.1.1", session1);
	 > freeswitch.bridge(session1, session2); 
	 
	good to now, there isn't any example of freeswitch.bridge in the
wiki and i would like to add one.
	Where I could find the full api of
	freeswitch.Session( ) ?
	because I've seen this working also without "session1" in the
second line:
	session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
	session2 =
freeswitch.Session("sofia/internal/1002%192.168.1.1");
	freeswitch.bridge(session1, session2); 
	
	 also is there any difference between  freeswitch.bridge and
freeswitch.execute(uuid_bridge ...) ?
	 
	 >> example3: yet another possibility
	 >> local session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
	 >> session1:originate(session2,
"sofia/internal/1002%192.168.1.1", 1000);
	 >> but it does not work either.
	
	 > The above is gibberish try:
	 > local session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
	 > session1:execute("bridge",
"sofia/internal/1002%192.168.1.1"); 
	 
	Okay i will report also this bridge example on the wiki which
was missing.
	But does session:originate make sense in some cases or not?
Otherwise i'm going to remove this line on the wiki
	 http://wiki.freeswitch.org/wiki/Mod_lua#session:originate
<http://wiki.freeswitch.org/wiki/Mod_lua#session:originate> 
	 
	Thanks,
	Claudio

	 

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From scott.torr.fs at letterboxes.org  Wed Jan 13 07:56:03 2010
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Thu, 14 Jan 2010 02:56:03 +1100
Subject: [Freeswitch-users] Inbound DTMF Not Recognized In Latest Version
In-Reply-To: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com>
References: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com>
Message-ID: <1263398163.29661.1354446823@webmail.messagingengine.com>

Hi Jerrey,

>From your debug information:
2010-01-12 14:50:27.778790 [DEBUG] switch_ivr_async.c:1532 DTMF
DETECTED: [5]

This is the FS software 'inband' detection of the digit 5 DTMF tone as a
result of issuing 'start_dtmf'.


2010-01-12 14:50:28.038716 [DEBUG] mod_openzap.c:672 queue DTMF [5]

mod_openzap.c -- OPENZAP Endpoint Module
This is the A101D Sangoma Card detection of the digit 5 DTMF tone. I
assume your card has the 'DTMF decoding and tone recognition' optional
DSP daughterboard installed.


So there is no need to use <action application="start_dtmf" /> which
will start the FS software 'inband' DTMF tone detection algorithm.


If someone would like to confirm this, please do.


regards,
Scott Torr


On Tue, 12 Jan 2010 15:10 -0800, "Jerry Richards"
<jerry.richards at teotech.com> wrote:
> 
> I am having a problem where in inbound call from the PSTN going to the
> demo
> IVR does not recognize DTMF digits.  I posted a log at
> http://pastebin.freeswitch.org/11769.  I am running Freeswitch
> (freeswitch-1.0.5-20100112-0400.tar.gz) with an A101D Sangoma Card and
> wanpipe driver: wanpipe-3.5.10.tgz.  I am not using libpri.
> 
> It appears that each DTMF digit might be getting detected twice.  I tried
> to
> dial "5401" and it thought I dialed "5544".
> 
> I am following the instructions at
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf, as shown
> below.  Also, I tried moving the "start_dtmf" statement in between all
> the
> statements within the condition tag, but it didn't make any difference.
> 
> <extension name="ivr_demo">
>    <condition field="destination_number" expression="5000">
>       <action application="answer"/>
>       <action application="start_dtmf" />
>       <action application="sleep" data="2000"/>
>       <action application="ivr" data="demo_ivr"/>
>    </condition>
> </extension>
> 
> Also, I noticed at line #293, the log is saying:
> 
> [ERR] mod_sndfile.c:194 Error Opening File
> [/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav]
> [System
> error : No such file or directory.]
> 
> Has anyone else encountered this issue?
> 
> Best Regards,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jerry.richards at teotech.com  Wed Jan 13 08:00:29 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 13 Jan 2010 08:00:29 -0800
Subject: [Freeswitch-users] Inbound DTMF Not Recognized In Latest Version
In-Reply-To: <44468DAE-9201-40AF-ACA3-E634E6682F04@freeswitch.org>
References: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com>
	<44468DAE-9201-40AF-ACA3-E634E6682F04@freeswitch.org>
Message-ID: <46044FFE91874AA79BA08DC8EE31159E@greyhawk.tonecommander.com>

These are my install steps.  

1) ./bootstrap.sh
2) ./configure
3) uncomment ../../libs/openzap/mod_openzap in ./modules.conf
4) make
5) make install sounds-install moh-install hd-sounds-install hd-moh-install
uhd-sounds-install uhd-moh-install
6) contrib/trixter/makemodconf.pl modules.conf >
/usr/local/freeswitch/conf/autoload_configs/modules.conf.xml 

Are these correct?

Thanks,
Jerry


-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Tuesday, January 12, 2010 3:26 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Inbound DTMF Not Recognized In Latest
Version

looks like you don't have the latest sound files installed.

/b

On Jan 12, 2010, at 5:10 PM, Jerry Richards wrote:

> [ERR] mod_sndfile.c:194 Error Opening File 
> [/usr/local/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav] 
> [System error : No such file or directory.]







From brian at freeswitch.org  Wed Jan 13 08:00:55 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 10:00:55 -0600
Subject: [Freeswitch-users] Inbound DTMF Not Recognized In Latest Version
In-Reply-To: <1263398163.29661.1354446823@webmail.messagingengine.com>
References: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com>
	<1263398163.29661.1354446823@webmail.messagingengine.com>
Message-ID: <2A012832-A0B1-4DBC-ABB3-85FCBB71810A@freeswitch.org>

If you're using OpenZAP you don't need start_dtmf thats mostly for RTP streams that are inband and OpenZAP does the DTMF bits for you if you start that you'll end up with double digits most likely.

/b

On Jan 13, 2010, at 9:56 AM, Scott Torr wrote:

> If someone would like to confirm this, please do.




From brian at freeswitch.org  Wed Jan 13 08:04:06 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 10:04:06 -0600
Subject: [Freeswitch-users] Inbound DTMF Not Recognized In Latest Version
In-Reply-To: <46044FFE91874AA79BA08DC8EE31159E@greyhawk.tonecommander.com>
References: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com>
	<44468DAE-9201-40AF-ACA3-E634E6682F04@freeswitch.org>
	<46044FFE91874AA79BA08DC8EE31159E@greyhawk.tonecommander.com>
Message-ID: <5553C06E-B1B2-428B-B140-CF3044169470@freeswitch.org>

Yes but you can just do uhd-sounds-install uhd-moh-install they include everything below it.

/b

On Jan 13, 2010, at 10:00 AM, Jerry Richards wrote:

> These are my install steps.  
> 
> 1) ./bootstrap.sh
> 2) ./configure
> 3) uncomment ../../libs/openzap/mod_openzap in ./modules.conf
> 4) make
> 5) make install sounds-install moh-install hd-sounds-install hd-moh-install
> uhd-sounds-install uhd-moh-install
> 6) contrib/trixter/makemodconf.pl modules.conf >
> /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml 
> 
> Are these correct?
> 
> Thanks,
> Jerry




From moises.silva at gmail.com  Wed Jan 13 08:13:14 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 13 Jan 2010 11:13:14 -0500
Subject: [Freeswitch-users] Sangoma PRI installation for FreeSWITCH
In-Reply-To: <369c72d81001122021o6885913cl618965791aec4621@mail.gmail.com>
References: <369c72d81001122021o6885913cl618965791aec4621@mail.gmail.com>
Message-ID: <c4d05cbe1001130813q26ba6fc9r846aaad7cf01c5f5@mail.gmail.com>

Hi Magesh,

On Tue, Jan 12, 2010 at 11:21 PM, Magesh R <magesh.freeswitch at gmail.com>wrote:

>       Next I have started the smg_ctrl, but it failed to start. It prints
> the following things,
> smg_ctrl start
>
> Starting processes...
> Loading SCTP...OK
> Starting sangoma_prid...OK
> sangoma_prid failed to start
> check /var/log/sangoma_mgd.log for errors
>
> Stopping running processes...
> safe_sangoma not running...
> sangoma_prid is stopped
> Removing PID files...done
>
> I have checked /var/log/sangoma_mgd.log file. But nothing was there.
>
> Could any please tell me where I made mistake?
>

Check that you have local2.* /var/log/sangoma_mgd.log in your syslog
configuration (which Linux distribution are you using?), and then restart
the syslog service, otherwise nothing will be logged.

If you still can't get the logs after doing that, you can try to start
manually sangoma_prid by doing this:

# export LD_LIBRARY_PATH=/usr/lib/sangoma_prid
# /usr/sbin/sangoma_prid &> /tmp/sangoma_prid.log

This will try to start sangoma_prid and save all the output to
/tmp/sangoma_prid.log, try doing that and then paste (
http://pastebin.freeswitch.org/) the resulting log.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From jerry.richards at teotech.com  Wed Jan 13 08:15:58 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 13 Jan 2010 08:15:58 -0800
Subject: [Freeswitch-users] Sangoma PRI installation for FreeSWITCH
In-Reply-To: <369c72d81001122021o6885913cl618965791aec4621@mail.gmail.com>
References: <369c72d81001122021o6885913cl618965791aec4621@mail.gmail.com>
Message-ID: <B4A27408A33F4E74A4DBEA492B906C20@greyhawk.tonecommander.com>

Here are my instructions to install the wanpipe driver and I have not had
your problem:
 
1)       Download the following Wanpipe driver tarball wanpipe-<version>.tgz

2)       Store the wanpipe-<version>.tgz file in the /opt folder.

3)       tar xvfz wanpipe-<version>.tgz

4)       cd wanpipe-<version>

5)       make openzap

6)       make install

7)       make install_pri

8)       wanrouter hwprobe     // confirms successful wanpipe installation

9)       /usr/sbin/wancfg_fs   // starts wanpipe configuration utility

10)   1=NO            // Change Freeswitch Configuration Directory

11)   1=T1            // Select Media Type

12)   1=YES (keep)    // Configure Port 1 T1, B8ZS, ESF

13)   1=NORMAL        // Select Clock

14)   1=NATIONAL      // Select Switchtype

15)   1=CPE           // Select Signaling Type

16)   1               // Input Group #

17)   1=YES           // Enable H/W DTMF Detect

18)   1=YES           // Enable FAX Detect

19)   1=YES=CONTINUE  // Configuration Complete

20)   1=YES           // Save Configuration

21)   1=YES           // Wanrouter start on reboot

22)   1=YES           // smg_ctrl start/stop on wanrouter start

23)   Done.

 
Jerry


  _____  

From: Magesh R [mailto:magesh.freeswitch at gmail.com] 
Sent: Tuesday, January 12, 2010 8:21 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Sangoma PRI installation for FreeSWITCH


Dear All,
         I have installed Sangoma PRI card in machine with following steps,
                 * wget
ftp://ftp.sangoma.com/linux/custom/3.5/wanpipe-3.5.8.7.tgz
                 * tar -xvfz wanpipe-3.5.8.7.tgz
                 * cd wanpipe-3.5.8.7
                 * make openzap
                 * make install
                 * make install_pri
      
           I have executed "wanrouter hwprobe" command it prints the
following details,
    
           1 . AFT-A102-SH : SLOT=4 : BUS=3 : IRQ=11 : CPU=A : PORT=1 :
HWEC=0 : V=36
           2 . AFT-A102-SH : SLOT=4 : BUS=3 : IRQ=11 : CPU=A : PORT=2 :
HWEC=0 : V=36
        
        Card Cnt: A101-2=1

       Next I have executed wancfg_fs script to configure the sangoma for
freeswitch. 
It creates the following configuration files
                     * wanpipe1.conf 
                     * wanpipe2.conf
                     * smg_prid.conf
                     * openzap.conf
                     * openzap.conf.xml 
       I have attached those files.
       I have started the wanrouter and printed the wanrouter status,
              Wanrouter Status:

Device name | Protocol | Station | Status        |
wanpipe1    | AFT TE1  | N/A     | Connected     |
wanpipe2    | AFT TE1  | N/A     | Disconnected  |

      Next I have started the smg_ctrl, but it failed to start. It prints
the following things,
smg_ctrl start

Starting processes...
Loading SCTP...OK
Starting sangoma_prid...OK
sangoma_prid failed to start
check /var/log/sangoma_mgd.log for errors

Stopping running processes...
safe_sangoma not running...
sangoma_prid is stopped
Removing PID files...done

I have checked /var/log/sangoma_mgd.log file. But nothing was there.

Could any please tell me where I made mistake?

Thanks,
Mag


              



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From anthony.minessale at gmail.com  Wed Jan 13 08:18:42 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Jan 2010 10:18:42 -0600
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
References: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
Message-ID: <191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>

you would have to register signals in your child process too

On Wed, Jan 13, 2010 at 3:13 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi all,
>
> I've done a sample program (In perl ESL) , which play a file to the caller
> and then it will call recvEvent() and print the event name. I've handled
> signals also.
>
> When I send SIGINT to my program (Perl), the signal handler is called and I
> can see the print output. But in the same time,  I received
> SERVER_DISCONNECTED from freeswitch as event.
>
> I don't know why I received SERVER_DISCONNECTED from freeswitch. Is it
> because, the recvEvent() from perl internally  calls the recvevent function
> in the Esl.c and when it waits to receive the information from socket, the
> signal occurred???
>
> Please clarify me!!
>
> Here is my program
> require ESL;
> use IO::Socket::INET;
> use Data::Dumper;
>
> my $ip = "192.168.1.222";
> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
> Proto => 'tcp',  Listen => 1,  Reuse => 1 );
> die "Could not create socket: $!\n" unless $sock;
> &register_Signals();
>
> for(;;) {
>     my $new_sock = $sock->accept();
>     next if (not defined ($new_sock));
>     my $pid = fork();
>     if ($pid) {
>         close($new_sock);
>         next;
>     }
>     print "CHILD PID: $$\n";
>     my $host = $new_sock->sockhost();
>     my $fd = fileno($new_sock);
>
>     my $con = new ESL::ESLconnection($fd);
>     my $info = $con->getInfo();
>
>     my $uuid = $info->getHeader("unique-id");
>
>     printf "Connected call %s, from %s\n", $uuid,
> $info->getHeader("caller-caller-id-number");
>     my $r=$con->execute("answer");
>     print Dumper $r;
>     $con->events("plain","all");
>     my
> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>
>     while($con->connected()) {
>         my $e = $con->recvEvent();
>
>         if ($e) {
>             my $name = $e->getHeader("event-name");
>             print "EVENT [$name]\n";
>             if ($name eq "DTMF") {
>                 my $digit = $e->getHeader("dtmf-digit");
>                 print "$digit\n";
>             }
>         }
>     }
>     close($new_sock);
> }
> sub register_Signals() {
>     foreach ( keys %SIG ) {
>         $SIG{$_} = 'sig_Handler';
>     }
> }
>
> sub sig_Handler() {
>     my $handle=$_[0];
>     if($handle eq "INT") {
>         print "$$: SIGNAL SIG$handle is generated\n";
>     }
> }
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From john at acsol.net  Wed Jan 13 08:38:38 2010
From: john at acsol.net (John)
Date: Wed, 13 Jan 2010 09:38:38 -0700
Subject: [Freeswitch-users] Sip provider Using IP authenication
Message-ID: <4B4DF70E.50007@acsol.net>

I'm am attempting to use Voipinnovations.com (M2Tel) as a SIP provider. 
They do not utilize username/password; how would I go about creating a 
gateway entry for them without username/password? Thanks



From gmaruzz at celliax.org  Wed Jan 13 08:40:27 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 13 Jan 2010 17:40:27 +0100
Subject: [Freeswitch-users] GSM modem for Skype
In-Reply-To: <SNT134-w3842E3232C8D3C80F105EAF47E0@phx.gbl>
References: <mailman.4361.1261716035.2495.freeswitch-users@lists.freeswitch.org>
	<SNT134-w3842E3232C8D3C80F105EAF47E0@phx.gbl>
Message-ID: <7b197bef1001130840k6a2aac9bk9e173c8a79fa3376@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#


2009/12/25 ??? <qinglan_zeng at hotmail.com>:
> Hello All,
>
> I had a GSM VoIP Gateway which can be connected to service providers who use
> SIP protocal. I remember somebody in Freeswitch community mentioned there is
> a software which can connect GSM network with Skype network. I'm not sure if
> this software can work with my GSM VOIP gateway or not.  If somebody can
> send me such software that would be much appriciated and even license fee
> required would be accepted. I hope I can have a testing on this or I can
> send this gateway to those guys who have this kind of software for testing.
>
> Thanks and Merry Christmas.
>
> Daniel Zeng
>
>
>
> ________________________________
> ??????????MSN??? ?????
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From gmartinr82 at gmail.com  Wed Jan 13 08:41:36 2010
From: gmartinr82 at gmail.com (Martin Rodriguez)
Date: Wed, 13 Jan 2010 13:41:36 -0300
Subject: [Freeswitch-users] FSComm Windows build
In-Reply-To: <1263358274180-4311097.post@n2.nabble.com>
References: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
	<1263350125191-4295967.post@n2.nabble.com>
	<5800526b1001121918r15a336f5jbc5b724e66ac2825@mail.gmail.com>
	<1263358274180-4311097.post@n2.nabble.com>
Message-ID: <fc0ed37b1001130841k11b3ae8ekaeefb227fb4a227f@mail.gmail.com>

Hi List:

Test Installation of this compilation but wanting to run it generates
the following error. I have installed Windows XP SP3.

Attached screenshot.

Bye
Martin Rodriguez


2010/1/13 Jeff Lenk <jeff at jefflenk.com>:
>
> Carlos,
>
> Thanks for those links!
>
> I am currenltly using prebuilt 4.6 32 bit libs and was hoping Nokia prebuilt
> the x64 too. I guess I might have to look into building myself as well.
>
> -Jeff
>
>
> Jeff,
>
> not yet. I'm still linking with a 32bit QT library of 4.5.2 I compiled for
> VS2008 sometime last year. I followed a guide similar to this one:
> http://dcsoft.com/community_server/blogs/dcsoft/archive/2009/03/06/how-to-setup-qt-4-5-visual-studio-integration.aspx
>
> At the time I didn't see a need for a 64bit version of FsGui. Since FSComm
> has FreeSWITCH in the back end this kind of changes things.
>
> Looks like someone put up a wiki on compiling QT for Win x64:
> http://en.wikibooks.org/wiki/Opticks_Developer_Guide/Getting_Started/Building_Qt_From_Source
>
> Now that QT 4.6 is out I might have to revisit a new library build with 64
> bit in mind.
>
> Carlos
>
> --
> View this message in context: http://n2.nabble.com/FSComm-Windows-build-tp4289256p4311097.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Jan 13 08:45:15 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 10:45:15 -0600
Subject: [Freeswitch-users] Sip provider Using IP authenication
In-Reply-To: <4B4DF70E.50007@acsol.net>
References: <4B4DF70E.50007@acsol.net>
Message-ID: <103A262C-672A-445E-832D-386EF944FA0E@freeswitch.org>

You don't.  You just send an invite in the form of sofia/profile/number at ip

/b

On Jan 13, 2010, at 10:38 AM, John wrote:

> I'm am attempting to use Voipinnovations.com (M2Tel) as a SIP provider. 
> They do not utilize username/password; how would I go about creating a 
> gateway entry for them without username/password? Thanks

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From rupa at rupa.com  Wed Jan 13 08:56:44 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 13 Jan 2010 10:56:44 -0600
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0F362E4@BESONE.corp.dom>
References: <A2167AE7633A5C4DAE56EEBD1428ECC0F362E0@BESONE.corp.dom>
	<A2167AE7633A5C4DAE56EEBD1428ECC0F362E4@BESONE.corp.dom>
Message-ID: <b39273531001130856l6f72fe4bmcc61c41b04b2a7c1@mail.gmail.com>

execute("api", "args")
executeString("api args")

one needs the api name as a separate arg

On Wed, Jan 13, 2010 at 9:51 AM, Cavalera Claudio Luigi <
Claudio.Cavalera at italtel.it> wrote:

>  Also after
> api = freeswitch.API();
> what is the difference between
> api:execute()
>
> and
> api:executeString() ?
>
> I've found example of both uses in http://wiki.freeswitch.org/wiki/Mod_lua
> and in
> http://wiki.freeswitch.org/wiki/Make_API_calls_directly_from_Lua_code
>
> Thanks,
> Claudio
>
>
>
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cavalera
> Claudio Luigi
> *Sent:* Wednesday, January 13, 2010 11:22 AM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] playing with sessions in lua
>
>  Thanks a lot Anthony,
> some comments inline (and please forgive me for my broken email client).
>
> >>  example1: Consider this simple lua script in which i create two
> sessions:
>
>  >>  api = freeswitch.API();
>  >>  api:execute("originate", "sofia/internal/1001%192.168.1.1 &park");
>  >>  api:execute("originate", "sofia/internal/1002%192.168.1.1 &park");
>
>  >  capture the output from api:execute the uuid is in there
>
> Thx a lot,
> this was one piece i was missing although it's already on the wiki here:
> http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls
>
>
>  > because lua calls it freeswitch.bridge
>
>  > session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
>  > session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1",
> session1);
>  > freeswitch.bridge(session1, session2);
>
> good to now, there isn't any example of freeswitch.bridge in the wiki and i
> would like to add one.
> Where I could find the full api of
> freeswitch.Session( ) ?
> because I've seen this working also without "session1" in the second line:
> session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
> session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");
> freeswitch.bridge(session1, session2);
>
>  also is there any difference between  freeswitch.bridge
> and freeswitch.execute(uuid_bridge ...) ?
>
>  >> example3: yet another possibility
>  >> local session1 =
> freeswitch.Session("sofia/internal/1001%192.168.1.1");
>  >> session1:originate(session2, "sofia/internal/1002%192.168.1.1", 1000);
>  >> but it does not work either.
>
>  > The above is gibberish try:
>  > local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
>  > session1:execute("bridge", "sofia/internal/1002%192.168.1.1");
>
> Okay i will report also this bridge example on the wiki which was missing.
> But does session:originate make sense in some cases or not? Otherwise i'm
> going to remove this line on the wiki
>  http://wiki.freeswitch.org/wiki/Mod_lua#session:originate
>
> Thanks,
> Claudio
>
>
>
> Internet Email Confidentiality Footer
>
>
> ********************************************************************************************************************************************
>
> La presente comunicazione, con le informazioni in essa contenute e ogni
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> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From john at acsol.net  Wed Jan 13 09:02:36 2010
From: john at acsol.net (John)
Date: Wed, 13 Jan 2010 10:02:36 -0700
Subject: [Freeswitch-users] Sip provider Using IP authenication
In-Reply-To: <103A262C-672A-445E-832D-386EF944FA0E@freeswitch.org>
References: <4B4DF70E.50007@acsol.net>
	<103A262C-672A-445E-832D-386EF944FA0E@freeswitch.org>
Message-ID: <4B4DFCAC.6070804@acsol.net>

So you don't need a gateway.xml file in the external defining it? You 
would just need a bridge statement in the dialplan?

Thanks John

On 1/13/2010 9:45 AM, Brian West wrote:
> You don't.  You just send an invite in the form of 
> sofia/profile/number at ip
>
> /b
>
> On Jan 13, 2010, at 10:38 AM, John wrote:
>
>> I'm am attempting to use Voipinnovations.com 
>> <http://Voipinnovations.com/> (M2Tel) as a SIP provider.
>> They do not utilize username/password; how would I go about creating a
>> gateway entry for them without username/password? Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From brian at freeswitch.org  Wed Jan 13 09:06:13 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 11:06:13 -0600
Subject: [Freeswitch-users] Sip provider Using IP authenication
In-Reply-To: <4B4DFCAC.6070804@acsol.net>
References: <4B4DF70E.50007@acsol.net>
	<103A262C-672A-445E-832D-386EF944FA0E@freeswitch.org>
	<4B4DFCAC.6070804@acsol.net>
Message-ID: <632BEFBA-74BA-4479-8C31-029AF7A0A4AB@freeswitch.org>


On Jan 13, 2010, at 11:02 AM, John wrote:

> So you don't need a gateway.xml file in the external defining it?

Nope.

> You would just need a bridge statement in the dialplan?

Yep.

> 
> Thanks John 




From mailinglist at fribert.dk  Wed Jan 13 09:08:36 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Wed, 13 Jan 2010 18:08:36 +0100
Subject: [Freeswitch-users] Svar: Re:  Multi-Homed setup,
	starting	 over	 - still not working
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9079C65@prato.activenetwerx.local>
References: <4B4DC035020000E1000003A2@mail.fribert.dk>
	<CA5A491E9DEFBE4CB777DE97E21575E9079C65@prato.activenetwerx.local>
Message-ID: <4B4E0C24020000E1000003A7@mail.fribert.dk>

Hi Joseph
 
Hmm, you were right, I hadn't set it there, as the guide explains me that I should set it in internal.xml.
After setting it in vars.xml it authenticates as well.
 
So I'm contiuing the quest :-)
 
Thankyou very much for your assistance!
 
/Fribse


>>> 13-01-2010 kl. 16:30 skrev "Joseph L. Casale" <jcasale at activenetwerx.com> i meddelelsen <CA5A491E9DEFBE4CB777DE97E21575E9079C65 at prato.activenetwerx.local>:

>But are you sayng that "eval ${domain}" will do more than is done when I restart the fs?
>I don't see that anywhere in the guide :-D
>
>I'll try it out asap!

What I am saying is it "evaluates" that placeholder and shows
its value. You can use it to show the current value to verify
it matches what you intend, which is for it to be the same as
your LAN ip.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From jerry.richards at teotech.com  Wed Jan 13 09:15:53 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 13 Jan 2010 09:15:53 -0800
Subject: [Freeswitch-users] Inbound DTMF Not Recognized In Latest Version
In-Reply-To: <2A012832-A0B1-4DBC-ABB3-85FCBB71810A@freeswitch.org>
References: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com><1263398163.29661.1354446823@webmail.messagingengine.com>
	<2A012832-A0B1-4DBC-ABB3-85FCBB71810A@freeswitch.org>
Message-ID: <E3ACF9CD600448E789F5D7178E4AE3DB@greyhawk.tonecommander.com>


Okay, cool.  Now it's working.  It is possible I screwed up the XML files
when I upgraded (since there were several new paths relating to accessing
sound files), so I rebuilt/reinstalled it from scratch and then more
carefully edited the XML files to preserve the new sound paths.  I also took
out the "start_dtmf".  So this my issue resolved.

Thanks,
Jerry


-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Wednesday, January 13, 2010 8:01 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Inbound DTMF Not Recognized In Latest
Version

If you're using OpenZAP you don't need start_dtmf thats mostly for RTP
streams that are inband and OpenZAP does the DTMF bits for you if you start
that you'll end up with double digits most likely.

/b

On Jan 13, 2010, at 9:56 AM, Scott Torr wrote:

> If someone would like to confirm this, please do.







From mike at jerris.com  Wed Jan 13 09:21:01 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 12:21:01 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
Message-ID: <EA8724C8-84A8-44DB-9F76-EFA517E76402@jerris.com>

Are you asking if you can mux all of the progress audio from multiple b-legs?  if so, no, and why would you want to?

Mike

On Jan 13, 2010, at 6:15 AM, David Villasmil wrote:

> hello
> 
> is it possible to bridge multiple b-legs and provide all audio  
> (progress) until there is an answer on one channel?




From mike at jerris.com  Wed Jan 13 09:29:35 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 12:29:35 -0500
Subject: [Freeswitch-users] sip trunk question: why call through
	external profile is challenged?
In-Reply-To: <20100113141003.970B311F32@mail.nstel.ru>
References: <20100113141003.970B311F32@mail.nstel.ru>
Message-ID: <98955F81-E9C1-4926-A648-AD49FB9D38A4@jerris.com>

Look at how sipx sets up the users when they build the extensions and such for conferences, there was something special here, but I can't recall what.

Mike

On Jan 13, 2010, at 9:10 AM, Nikolay Kondratyev wrote:

> Hi all!
>  
> I?m brand new to FreeSwitch, but have some experience with SipX.
> Our company is evaluating FS.
> For test purposes I set up FS on a virtual machine (vmware esxi). I use CentOS.
> The FS version I use is 1.0.5-20100110-0400.
>  
> I have a question regarding sip trunk between FS and SipX.
> I created the following GW in external profile:
> [freeswitch at freeswitch external]$ cat sipx-lab.xml | grep -v '<!--'
> <include>
>   <gateway name="sipx4.lab.nstel.ru">
>   <param name="username" value="zxcv"/>
>   <param name="password" value="2007"/>
>   <param name="register" value="false"/>
>   </gateway>
> </include>
>  
> External.xml file is not modified after installation.
>  
> I see this gateway via fs_cli:
> freeswitch at internal> sofia status gateway sipx4.lab.nstel.ru
> =================================================================================================
> Name            sipx4.lab.nstel.ru
> Profile         external
> Scheme          Digest
> Realm           sipx4.lab.nstel.ru
> Username        zxcv
> Password        yes
> From            <sip:zxcv at sipx4.lab.nstel.ru;transport=udp>
> Contact         <sip:gw+sipx4.lab.nstel.ru at 172.23.22.49:5080;transport=udp;gw=sipx4.lab.nstel.ru>
> Exten           zxcv
> To              sip:zxcv at sipx4.lab.nstel.ru
> Proxy           sip:sipx4.lab.nstel.ru
> Context         public
> Expires         3600
> Freq            3600
> Ping            0
> PingFreq        0
> PingState       0/0/0
> State           NOREG
> Status          UP
> CallsIN         0
> CallsOUT        0
> =================================================================================================
>  
> I created new FS extension 2853. I registers (xlite) and I can call it.
>  
> Now I want to call FS user from sipx.
>  
> From the sipx side one can configure link to FS differently. There are two options:
> 1. Call FS directly trough sipxproxy (the core part of sipx works as sip proxy, not as b2bua)
> 2. Call trough embedded b2bua, named sipxbridge.
>  
> When a call is going trough sipxbridge, it is successfully landed at FS extension 2853.
> But when a call is going from the sipxproxy, it is challenged with Status 407 ?proxy authentication required?, and then call fails.
> (I?m not sure if sipx should handle this challenge, but this is separate question for the sipx forum).
> At the default log level I see nothing in the freeswitch.log.
>  
> So the question is why one call through external profile is being challenged while the other is not?
> I suspect that the reason is in the difference in the two Invite messages:
>  
> Here is challenged Invite:
> INVITE sip:2853 at fs.lab.nstel.ru:5080;transport=udp SIP/2.0
> Record-Route: <sip:172.23.12.104:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EOTRiZjI3MjgtYWMxNzBjZTktMTNjNC0yZjZkZWMtNjNkNmUwN2UtMmY2ZGVj.900_ntap%2Aid%7EMjIyNjAtMQ%60%60%213ba6d85f6946c4c6001bee1d3b54474f>
> From: "testphone3857"<sip:3857 at lab.nstel.ru>;tag=94bf2728-ac170ce9-13c4-2f6dec-63d6e07e-2f6dec
> To: <sip:2853 at lab.nstel.ru>
> Call-Id: 94bed2a0-ac170ce9-13c4-2f6dec-68ffdb81-2f6dec at lab.nstel.ru
> Cseq: 2 INVITE
> Via: SIP/2.0/UDP 172.23.12.104;branch=z9hG4bK-sipXecs-000eaf48a36fe4029c7cde004a6f44424847
> Via: SIP/2.0/TCP 172.23.12.104;branch=z9hG4bK-sipXecs-000bf3008cfa3f3e70470789c75232ba9499~ac7fd729330fd563f83cacd941311e75;id=22260-1
> Via: SIP/2.0/UDP 172.23.12.233:5060;branch=z9hG4bK-2f6dec-b9456412-7ad07784
> Max-Forwards: 18
> Supported: replaces
> User-Agent: LG-Nortel LIP 6812 v1.2.38sp SN/00405A18B634
> Contact: <sip:3857 at 172.23.12.233:5060;x-sipX-nonat>
> Proxy-Authorization: Digest username="3857",realm="lab.nstel.ru",nonce="e2152722611af1bbe59f3a8eb31b8eb84b4db2ca",uri="sip:2853 at lab.nstel.ru",response="a09c34fbd897783c9b9af50bd044ccaa",algorithm=MD5
> Content-Type: application/sdp
> Content-Length: 301
> Date: Wed, 13 Jan 2010 11:47:22 GMT
> Expires: 60
> X-Sipx-Handled: X172.23.12.104-81.211.30.104
>  
> v=0
> o=LGEIPP 16246 16247 IN IP4 172.23.12.233
> s=SIP Call
> c=IN IP4 172.23.12.233
> t=0 0
> m=audio 23008 RTP/AVP 0 8 18 4 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:18 annexb=no
> a=fmtp:101 0-11
> a=sendrecv
>  
>  
> And here is successful Invite:
> INVITE sip:2853 at fs.lab.nstel.ru;user=phone SIP/2.0
> Call-ID: 94bec3b8-ac170ce9-13c4-2f6d47-2fee09d0-2f6d47 at lab.nstel.ru.0
> CSeq: 1 INVITE
> From: "testphone3857" <sip:3857 at 172.23.12.104>;tag=816640414244159961
> To: <sip:2853 at fs.lab.nstel.ru;user=phone>
> Via: SIP/2.0/UDP 172.23.12.104:5080;branch=z9hG4bK885dde2c8680f5315845cd3350b8b605373534
> Max-Forwards: 70
> User-Agent: sipXecs/4.0.2 sipXecs/sipxbridge (Linux)
> P-Asserted-Identity: <sip:1004 at 172.23.12.104>
> Contact: <sip:1004 at 172.23.12.104:5080;transport=udp>
> Route: <sip:172.23.22.49:5080;transport=udp;lr>
> Session-Expires: 1800;refresher=uac
> Allow: INVITE,BYE,ACK,CANCEL,OPTIONS
> Content-Type: application/sdp
> Content-Length: 315
>  
> v=0
> o=sipxbridge 6640787141824452741 1 IN IP4 172.23.12.104
> s=SIP Call
> c=IN IP4 172.23.12.104
> t=0 0
> m=audio 30248 RTP/AVP 0 8 18 4 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:18 annexb=no
> a=fmtp:101 0-11
> a=sendrecv
>  
> Can somebody please if it is a FS configuration problem or a software problem or is it a problem on the sipx side?
>  
> Thanks in advance,
> Nikolay.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Wed Jan 13 09:33:42 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 12:33:42 -0500
Subject: [Freeswitch-users] Inbound DTMF Not Recognized In Latest Version
In-Reply-To: <E3ACF9CD600448E789F5D7178E4AE3DB@greyhawk.tonecommander.com>
References: <4456D6578DAF40E39E49416B7D4735AC@greyhawk.tonecommander.com><1263398163.29661.1354446823@webmail.messagingengine.com>
	<2A012832-A0B1-4DBC-ABB3-85FCBB71810A@freeswitch.org>
	<E3ACF9CD600448E789F5D7178E4AE3DB@greyhawk.tonecommander.com>
Message-ID: <7F7949F8-EE9A-4D9B-8594-7C85607E2112@jerris.com>

The way I changed the configs, they should all still work fine with old configs, just not if sounds directory is moved out of prefix, which there is not a way to do yet.

Mike

On Jan 13, 2010, at 12:15 PM, Jerry Richards wrote:

> 
> Okay, cool.  Now it's working.  It is possible I screwed up the XML files
> when I upgraded (since there were several new paths relating to accessing
> sound files), so I rebuilt/reinstalled it from scratch and then more
> carefully edited the XML files to preserve the new sound paths.  I also took
> out the "start_dtmf".  So this my issue resolved.
> 
> Thanks,
> Jerry
> 
> 
> -----Original Message-----
> From: Brian West [mailto:brian at freeswitch.org] 
> Sent: Wednesday, January 13, 2010 8:01 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Inbound DTMF Not Recognized In Latest
> Version
> 
> If you're using OpenZAP you don't need start_dtmf thats mostly for RTP
> streams that are inband and OpenZAP does the DTMF bits for you if you start
> that you'll end up with double digits most likely.
> 
> /b
> 
> On Jan 13, 2010, at 9:56 AM, Scott Torr wrote:
> 
>> If someone would like to confirm this, please do.
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jeff at jefflenk.com  Wed Jan 13 09:38:13 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 13 Jan 2010 11:38:13 -0600
Subject: [Freeswitch-users] FSComm Windows build
In-Reply-To: <fc0ed37b1001130841k11b3ae8ekaeefb227fb4a227f@mail.gmail.com>
References: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>,
	<1263350125191-4295967.post@n2.nabble.com>,
	<5800526b1001121918r15a336f5jbc5b724e66ac2825@mail.gmail.com>,
	<1263358274180-4311097.post@n2.nabble.com>,
	<fc0ed37b1001130841k11b3ae8ekaeefb227fb4a227f@mail.gmail.com>
Message-ID: <COL117-W3855379D4087BF8CA339A8B96B0@phx.gbl>


Use the prebuilt install from Carlos

 

http://files.freeswitch.org/windows_installer/

 

This will list all the dependents that you need

 


 
> Date: Wed, 13 Jan 2010 13:41:36 -0300
> From: gmartinr82 at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FSComm Windows build
> 
> Hi List:
> 
> Test Installation of this compilation but wanting to run it generates
> the following error. I have installed Windows XP SP3.
> 
> Attached screenshot.
> 
> Bye
> Martin Rodriguez
> 
> 
> 2010/1/13 Jeff Lenk <jeff at jefflenk.com>:
> >
> > Carlos,
> >
> > Thanks for those links!
> >
> > I am currenltly using prebuilt 4.6 32 bit libs and was hoping Nokia prebuilt
> > the x64 too. I guess I might have to look into building myself as well.
> >
> > -Jeff
> >
> >
> > Jeff,
> >
> > not yet. I'm still linking with a 32bit QT library of 4.5.2 I compiled for
> > VS2008 sometime last year. I followed a guide similar to this one:
> > http://dcsoft.com/community_server/blogs/dcsoft/archive/2009/03/06/how-to-setup-qt-4-5-visual-studio-integration.aspx
> >
> > At the time I didn't see a need for a 64bit version of FsGui. Since FSComm
> > has FreeSWITCH in the back end this kind of changes things.
> >
> > Looks like someone put up a wiki on compiling QT for Win x64:
> > http://en.wikibooks.org/wiki/Opticks_Developer_Guide/Getting_Started/Building_Qt_From_Source
> >
> > Now that QT 4.6 is out I might have to revisit a new library build with 64
> > bit in mind.
> >
> > Carlos
> >
> > --
> > View this message in context: http://n2.nabble.com/FSComm-Windows-build-tp4289256p4311097.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
http://clk.atdmt.com/GBL/go/196390706/direct/01/
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From carlos.talbot at gmail.com  Wed Jan 13 09:56:03 2010
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Wed, 13 Jan 2010 11:56:03 -0600
Subject: [Freeswitch-users] FSComm Windows build
In-Reply-To: <fc0ed37b1001130841k11b3ae8ekaeefb227fb4a227f@mail.gmail.com>
References: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
	<1263350125191-4295967.post@n2.nabble.com>
	<5800526b1001121918r15a336f5jbc5b724e66ac2825@mail.gmail.com>
	<1263358274180-4311097.post@n2.nabble.com>
	<fc0ed37b1001130841k11b3ae8ekaeefb227fb4a227f@mail.gmail.com>
Message-ID: <5800526b1001130956g9f00a5ai2020c9d00d89992e@mail.gmail.com>

D'oh! I forgot to include the VS 2008 redistributable package like I do with
FreeSWITCH. You'll have to manually install one of the following below until
I've updated the install.

32bit version
http://www.microsoft.com/downloads/details.aspx?FamilyID=9b2da534-3e03-4391-8a4d-074b9f2bc1bf&displaylang=en


regards,

Carlos

On Wed, Jan 13, 2010 at 10:41 AM, Martin Rodriguez <gmartinr82 at gmail.com>wrote:

> Hi List:
>
> Test Installation of this compilation but wanting to run it generates
> the following error. I have installed Windows XP SP3.
>
> Attached screenshot.
>
> Bye
> Martin Rodriguez
>
>
> 2010/1/13 Jeff Lenk <jeff at jefflenk.com>:
> >
> > Carlos,
> >
> > Thanks for those links!
> >
> > I am currenltly using prebuilt 4.6 32 bit libs and was hoping Nokia
> prebuilt
> > the x64 too. I guess I might have to look into building myself as well.
> >
> > -Jeff
> >
> >
> > Jeff,
> >
> > not yet. I'm still linking with a 32bit QT library of 4.5.2 I compiled
> for
> > VS2008 sometime last year. I followed a guide similar to this one:
> >
> http://dcsoft.com/community_server/blogs/dcsoft/archive/2009/03/06/how-to-setup-qt-4-5-visual-studio-integration.aspx
> >
> > At the time I didn't see a need for a 64bit version of FsGui. Since
> FSComm
> > has FreeSWITCH in the back end this kind of changes things.
> >
> > Looks like someone put up a wiki on compiling QT for Win x64:
> >
> http://en.wikibooks.org/wiki/Opticks_Developer_Guide/Getting_Started/Building_Qt_From_Source
> >
> > Now that QT 4.6 is out I might have to revisit a new library build with
> 64
> > bit in mind.
> >
> > Carlos
> >
> > --
> > View this message in context:
> http://n2.nabble.com/FSComm-Windows-build-tp4289256p4311097.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch-users at digitaldan.com  Wed Jan 13 10:01:50 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Wed, 13 Jan 2010 11:01:50 -0700 (MST)
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <21466640.91263405615737.JavaMail.root@zimbra>
Message-ID: <10842021.111263405710193.JavaMail.root@zimbra>

Hi guys, I'm having an issue with DTMF detection from our cisco 5300 gateways. DTMF detection has been working great when someone was manually inputing the digits (ie there are small pauses between the DTMF digits), but when the other end is sending those digits from say a speed dial or programmed dial string and can send the DTMF digits one right after the other we start to drop one or more digits about 75% of the time. 


I upgraded to trunk this morning and have tried everything that was listed on http://wiki.freeswitch.org/wiki/RTP_Issues . I have a pcap output of the SIP and RTP data, it looks fine to me, I can see the DTMF payloads coming as they should be. Any ideas? I would be fine sending or posting the pcap file to someone privately. 


Thanks 
Dan- 
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From mailinglist at fribert.dk  Wed Jan 13 10:06:59 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Wed, 13 Jan 2010 19:06:59 +0100
Subject: [Freeswitch-users] Svar: Re:  Multi-Homed setup,
	starting		 over	 - still not working
In-Reply-To: <4B4E0C24020000E1000003A7@mail.fribert.dk>
References: <4B4DC035020000E1000003A2@mail.fribert.dk>
	<CA5A491E9DEFBE4CB777DE97E21575E9079C65@prato.activenetwerx.local>
	<4B4E0C24020000E1000003A7@mail.fribert.dk>
Message-ID: <4B4E19D3020000E1000003AB@mail.fribert.dk>

Hi again
 
Ok, got a bit further, just had a success, I called out, weeheee :-D
 
Now I want to call in, and it seems the SIP provider I'm using has multiple IP's, I'm not sure if that's the issue here?
I've created the dialplan part explained in the guide, and placed it in dialplan/public as explained in IN-COMING CALLS, RINGING A GROUP

<include>
  <extension name="Privat">
    <condition field="destination_number" expression="^(4692xxxx)$">  (my phonenumber)
      <action application="set" data="domain_name=10.11.12.25"/>
      <action application="transfer" data="*8201 XML default"/>
    </condition>
  </extension>
</include>
I've joined the two soft phones I have to the group.
 
When I call in I get this displayed:
2010-01-13 18:57:51.396370 [NOTICE] switch_channel.c:602 New Channel sofia/external/2680xxxx at 87.54.25.116 [fedccd2a-6d00-df11-aee0-000c29b7b4cb]
2010-01-13 18:57:51.396370 [NOTICE] sofia.c:3419 Hangup sofia/external/26802233 at 87.54.25.116 [CS_NEW] [INCOMPATIBLE_DESTINATION]
2010-01-13 18:57:51.399569 [NOTICE] switch_core_session.c:1085 Session 19 (sofia/external/2680xxxx at 87.54.25.116) Ended
2010-01-13 18:57:51.399569 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/2680xxxx at 87.54.25.116 [CS_DESTROY]

The 87.54.25.116 is from musimi.dk, but outgoing is handled by 87.54.25.114 (which musimi.dk translates to).

2680xxxx is my cellphones number.

Hmm, help :-)

>>> 13-01-2010 kl. 18:08 skrev "mailinglist" <mailinglist at fribert.dk> i meddelelsen <4B4E0C24020000E1000003A7 at mail.fribert.dk>:

Hi Joseph
 
Hmm, you were right, I hadn't set it there, as the guide explains me that I should set it in internal.xml.
After setting it in vars.xml it authenticates as well.
 
So I'm contiuing the quest :-)
 
Thankyou very much for your assistance!
 
/Fribse


>>> 13-01-2010 kl. 16:30 skrev "Joseph L. Casale" <jcasale at activenetwerx.com> i meddelelsen <CA5A491E9DEFBE4CB777DE97E21575E9079C65 at prato.activenetwerx.local>:

>But are you sayng that "eval ${domain}" will do more than is done when I restart the fs?
>I don't see that anywhere in the guide :-D
>
>I'll try it out asap!

What I am saying is it "evaluates" that placeholder and shows
its value. You can use it to show the current value to verify
it matches what you intend, which is for it to be the same as
your LAN ip.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From msc at freeswitch.org  Wed Jan 13 10:14:25 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Jan 2010 10:14:25 -0800
Subject: [Freeswitch-users] CDR and reporting.
In-Reply-To: <201001120639.38098.sos@sokhapkin.dyndns.org>
References: <8ccbff061001120313u78701883lf9e212947af0528@mail.gmail.com>
	<201001120639.38098.sos@sokhapkin.dyndns.org>
Message-ID: <87f2f3b91001131014w4d19027bl4e90895954a40bbd@mail.gmail.com>

Also, you can have a cron job collect these more frequently than once per
day. There are folks out there we grab them several times per minute and
process them so that it is almost real-time. Of course, using mod_xml_cdr
requires more effort up front but it *IS* real-time and more scalable.

-MC

On Tue, Jan 12, 2010 at 3:39 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Look at default config for mod_cdr_csv, sql template. It can produce a
> sequence of SQL INSERT statements which can be later fed (after daily log
> rotation) to mysql client program.
>
> I see no reason to employ xml for this task, the language hard to read and
> parse for both human and computer.
>
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From devel at thom.fr.eu.org  Wed Jan 13 10:25:20 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Wed, 13 Jan 2010 19:25:20 +0100
Subject: [Freeswitch-users] No hangup tone after zap channel closed,
	tones in general
Message-ID: <8b18ffe229516c90285de8a54a00e845@thom.fr.eu.org>



Hello, 

How shall I do to get a hangup tone on an FXS port after the
channel is closed ? 

Is there something specific to configure in openzap ?


For instance, if I call the following extension from an FXS port : 

I
get no tone after the hangup application is called. 

I also wonder if
there is some documentation on the tones.conf file format, and about the
variables uk-ring, us-ring, bong-ring, sit in vars.xml (when are they used,
what is the syntax). I could not find any info on wiki. 

Thank you for
helping 

Fran?ois
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From msc at freeswitch.org  Wed Jan 13 10:41:40 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Jan 2010 10:41:40 -0800
Subject: [Freeswitch-users] Fw: SIP client authorization issues
	(Globarange phone)
In-Reply-To: <271101.94727.qm@web53107.mail.re2.yahoo.com>
References: <271101.94727.qm@web53107.mail.re2.yahoo.com>
Message-ID: <87f2f3b91001131041t1aea3a89n68c6f6948db1253d@mail.gmail.com>

On Tue, Jan 12, 2010 at 4:54 PM, VG Oget <vgoget at yahoo.com> wrote:

> Hi, I am trying to register a GlobaRange SIP phone to FreeSwitch (was able
> to do it with Asterisk some time ago). It is a locked VOIP phone to
> Joip.com. On Ubuntu with latest version as of a few days ago?
>
> I redirected the traffic to Freeswitch (basically proxy.joip.com points
> and to the Freeswitch host) and also changed the SIP port to 23768 (what the
> phone wants). Very close to this:
> http://darkskiez.co.uk/index.php?page=Use_The_Panasonic_Globarange_With_Asterisk
>
>
> I have Xlite successfully registering (SIP port 23768 and using
> proxy.joip.com not the IP address) so I?m confident the freeswitch config
> is close to being okay?
>
> Now I have intercepted the traffic during registration. The issue (I think)
> is that the SIP authentication phases use a different username and I don?t
> know how to configure it.
> I looked into this page extensively:
> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide
> And also tried to understand sip_suth_username but I cannot find the
> solution.
>
> If someone can point me in the right direction, that would be perfect.
>
> Were these the exact same credentials that you used when you tested with
x-lite? Grab a SIP trace of that registration happening and compare it to
this one.
-MC
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From carlos.talbot at gmail.com  Wed Jan 13 10:42:47 2010
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Wed, 13 Jan 2010 12:42:47 -0600
Subject: [Freeswitch-users] FSComm Windows build
In-Reply-To: <fc0ed37b1001130841k11b3ae8ekaeefb227fb4a227f@mail.gmail.com>
References: <5800526b1001111722i23a569beu713d4e394618803@mail.gmail.com>
	<1263350125191-4295967.post@n2.nabble.com>
	<5800526b1001121918r15a336f5jbc5b724e66ac2825@mail.gmail.com>
	<1263358274180-4311097.post@n2.nabble.com>
	<fc0ed37b1001130841k11b3ae8ekaeefb227fb4a227f@mail.gmail.com>
Message-ID: <5800526b1001131042sef0ffbfvdfd8e28efe57a137@mail.gmail.com>

I just finished uploading a new version of FSComm that includes a 32 and 64
bit build with respective QT and FreeSWITCH libraries. It also now includes
the Microsoft redistributable package. The new executable is 15MB in size.
I'm not sure when the CDN sync will kick in...

regards,

Carlos

On Wed, Jan 13, 2010 at 10:41 AM, Martin Rodriguez <gmartinr82 at gmail.com>wrote:

> Hi List:
>
> Test Installation of this compilation but wanting to run it generates
> the following error. I have installed Windows XP SP3.
>
> Attached screenshot.
>
> Bye
> Martin Rodriguez
>
>
> 2010/1/13 Jeff Lenk <jeff at jefflenk.com>:
> >
> > Carlos,
> >
> > Thanks for those links!
> >
> > I am currenltly using prebuilt 4.6 32 bit libs and was hoping Nokia
> prebuilt
> > the x64 too. I guess I might have to look into building myself as well.
> >
> > -Jeff
> >
> >
> > Jeff,
> >
> > not yet. I'm still linking with a 32bit QT library of 4.5.2 I compiled
> for
> > VS2008 sometime last year. I followed a guide similar to this one:
> >
> http://dcsoft.com/community_server/blogs/dcsoft/archive/2009/03/06/how-to-setup-qt-4-5-visual-studio-integration.aspx
> >
> > At the time I didn't see a need for a 64bit version of FsGui. Since
> FSComm
> > has FreeSWITCH in the back end this kind of changes things.
> >
> > Looks like someone put up a wiki on compiling QT for Win x64:
> >
> http://en.wikibooks.org/wiki/Opticks_Developer_Guide/Getting_Started/Building_Qt_From_Source
> >
> > Now that QT 4.6 is out I might have to revisit a new library build with
> 64
> > bit in mind.
> >
> > Carlos
> >
> > --
> > View this message in context:
> http://n2.nabble.com/FSComm-Windows-build-tp4289256p4311097.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mailinglist at fribert.dk  Wed Jan 13 11:10:02 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Wed, 13 Jan 2010 20:10:02 +0100
Subject: [Freeswitch-users] Svar: Re:  Multi-Homed setup,
	starting		 over	 - still not working
In-Reply-To: <4B4E19D3020000E1000003AB@mail.fribert.dk>
References: <4B4DC035020000E1000003A2@mail.fribert.dk>
	<CA5A491E9DEFBE4CB777DE97E21575E9079C65@prato.activenetwerx.local>
	<4B4E0C24020000E1000003A7@mail.fribert.dk>
	<4B4E19D3020000E1000003AB@mail.fribert.dk>
Message-ID: <4B4E289A020000E1000003B0@mail.fribert.dk>

hey, I got it solved. Apparently my voip provider used a G729 Codec, which freeswitch didn't like.
Of course the VOIP provider has 4, and just the one not supported in this setup is used as default :-o
 
Now I can call in, WOW, cool :-)
 
Small steps here...


>>> 13-01-2010 kl. 19:06 skrev "mailinglist" <mailinglist at fribert.dk> i meddelelsen <4B4E19D3020000E1000003AB at mail.fribert.dk>:

Hi again

Ok, got a bit further, just had a success, I called out, weeheee :-D

Now I want to call in, and it seems the SIP provider I'm using has multiple IP's, I'm not sure if that's the issue here?
I've created the dialplan part explained in the guide, and placed it in dialplan/public as explained in IN-COMING CALLS, RINGING A GROUP

<include>
  <extension name="Privat">
    <condition field="destination_number" expression="^(4692xxxx)$">  (my phonenumber)
      <action application="set" data="domain_name=10.11.12.25"/>
      <action application="transfer" data="*8201 XML default"/>
    </condition>
  </extension>
</include>
I've joined the two soft phones I have to the group.

When I call in I get this displayed:
2010-01-13 18:57:51.396370 [NOTICE] switch_channel.c:602 New Channel sofia/external/2680xxxx at 87.54.25.116 [fedccd2a-6d00-df11-aee0-000c29b7b4cb]
2010-01-13 18:57:51.396370 [NOTICE] sofia.c:3419 Hangup sofia/external/26802233 at 87.54.25.116 [CS_NEW] [INCOMPATIBLE_DESTINATION]
2010-01-13 18:57:51.399569 [NOTICE] switch_core_session.c:1085 Session 19 (sofia/external/2680xxxx at 87.54.25.116) Ended
2010-01-13 18:57:51.399569 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/2680xxxx at 87.54.25.116 [CS_DESTROY]

The 87.54.25.116 is from musimi.dk, but outgoing is handled by 87.54.25.114 (which musimi.dk translates to).

2680xxxx is my cellphones number.

Hmm, help :-)

>>> 13-01-2010 kl. 18:08 skrev "mailinglist" <mailinglist at fribert.dk> i meddelelsen <4B4E0C24020000E1000003A7 at mail.fribert.dk>:

Hi Joseph

Hmm, you were right, I hadn't set it there, as the guide explains me that I should set it in internal.xml.
After setting it in vars.xml it authenticates as well.

So I'm contiuing the quest :-)

Thankyou very much for your assistance!

/Fribse


>>> 13-01-2010 kl. 16:30 skrev "Joseph L. Casale" <jcasale at activenetwerx.com> i meddelelsen <CA5A491E9DEFBE4CB777DE97E21575E9079C65 at prato.activenetwerx.local>:

>But are you sayng that "eval ${domain}" will do more than is done when I restart the fs?
>I don't see that anywhere in the guide :-D
>
>I'll try it out asap!

What I am saying is it "evaluates" that placeholder and shows
its value. You can use it to show the current value to verify
it matches what you intend, which is for it to be the same as
your LAN ip.

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From jerry.richards at teotech.com  Wed Jan 13 11:12:54 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 13 Jan 2010 11:12:54 -0800
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <10842021.111263405710193.JavaMail.root@zimbra>
References: <21466640.91263405615737.JavaMail.root@zimbra>
	<10842021.111263405710193.JavaMail.root@zimbra>
Message-ID: <47564CF8463D463E8816123D6AF5B30F@greyhawk.tonecommander.com>

I would recommend you use out-of-band DTMF based on RFC 2833.  In band DTMF
is usually not reliable in short bursts.
 
Jerry


  _____  

From: Dan [mailto:freeswitch-users at digitaldan.com] 
Sent: Wednesday, January 13, 2010 10:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Cisco media gateway and DTMF


Hi guys, I'm having an issue with DTMF detection from our cisco 5300
gateways.  DTMF detection has been working great when someone was manually
inputing the digits (ie there are small pauses between the DTMF digits),
but when the other end is sending those digits from say a speed dial or
programmed dial string and can send the DTMF digits one right after the
other we start to drop one or more digits about 75% of the time.    

I upgraded to trunk this morning and have tried everything that was listed
on http://wiki.freeswitch.org/wiki/RTP_Issues .  I have a pcap output of the
SIP and RTP data, it looks fine to me, I can see the DTMF payloads coming as
they should be.  Any ideas?  I would be fine sending or posting the pcap
file to someone privately. 

Thanks
Dan-

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From mike at van.lammeren.net  Wed Jan 13 11:30:06 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Wed, 13 Jan 2010 14:30:06 -0500
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0F362E0@BESONE.corp.dom>
References: <191c3a031001121535t4819ee44p74cc9f7cd4fe972f@mail.gmail.com>
	<A2167AE7633A5C4DAE56EEBD1428ECC0F362E0@BESONE.corp.dom>
Message-ID: <5d2828f1001131130s6ef166c0wc98f70023420c0fd@mail.gmail.com>

Hello!

Before you remove "session:originate" from the wiki, you should take a look
at this:
http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocumented_Session_Functions.3F

<http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocumented_Session_Functions.3F>There
is, in fact, a function called "originate".

Mike van Lammeren


On Wed, Jan 13, 2010 at 5:22 AM, Cavalera Claudio Luigi <
Claudio.Cavalera at italtel.it> wrote:

>  Thanks a lot Anthony,
> some comments inline (and please forgive me for my broken email client).
>
> >>  example1: Consider this simple lua script in which i create two
> sessions:
>
>  >>  api = freeswitch.API();
>  >>  api:execute("originate", "sofia/internal/1001%192.168.1.1 &park");
>  >>  api:execute("originate", "sofia/internal/1002%192.168.1.1 &park");
>
>  >  capture the output from api:execute the uuid is in there
>
> Thx a lot,
> this was one piece i was missing although it's already on the wiki here:
> http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls
>
>
>  > because lua calls it freeswitch.bridge
>
>  > session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
>  > session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1",
> session1);
>  > freeswitch.bridge(session1, session2);
>
> good to now, there isn't any example of freeswitch.bridge in the wiki and i
> would like to add one.
> Where I could find the full api of
> freeswitch.Session( ) ?
> because I've seen this working also without "session1" in the second line:
> session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
> session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");
> freeswitch.bridge(session1, session2);
>
>  also is there any difference between  freeswitch.bridge
> and freeswitch.execute(uuid_bridge ...) ?
>
>  >> example3: yet another possibility
>  >> local session1 =
> freeswitch.Session("sofia/internal/1001%192.168.1.1");
>  >> session1:originate(session2, "sofia/internal/1002%192.168.1.1", 1000);
>  >> but it does not work either.
>
>  > The above is gibberish try:
>  > local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
>  > session1:execute("bridge", "sofia/internal/1002%192.168.1.1");
>
> Okay i will report also this bridge example on the wiki which was missing.
> But does session:originate make sense in some cases or not? Otherwise i'm
> going to remove this line on the wiki
>  http://wiki.freeswitch.org/wiki/Mod_lua#session:originate
>
> Thanks,
> Claudio
>
>
>
> Internet Email Confidentiality Footer
>
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From yehavi.bourvine at gmail.com  Wed Jan 13 11:32:17 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 13 Jan 2010 21:32:17 +0200
Subject: [Freeswitch-users] How to build FreeSwitch with non-default SSL
	library?
Message-ID: <d361ca7b1001131132o1043cce4p31af0f95f804bd7b@mail.gmail.com>

Hello,

  I suspect that the Openssl library used on fedora is a modified one
and has some interoperability problems with phones. I would like to build
freeswitch with a library from openssl.org which is installed
under /usr/local. Is there an easy way to tell  "configure" script to use
that version instead?

                                 Thanks, __Yehavi:
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From freeswitch-users at digitaldan.com  Wed Jan 13 11:36:26 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Wed, 13 Jan 2010 12:36:26 -0700 (MST)
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <47564CF8463D463E8816123D6AF5B30F@greyhawk.tonecommander.com>
Message-ID: <26415028.141263411386237.JavaMail.root@zimbra>

Sorry, I should have made that more clear, this is using 2833. As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through). 
----- Original Message ----- 
From: "Jerry Richards" <jerry.richards at teotech.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 12:12:54 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 


I would recommend you use out-of-band DTMF based on RFC 2833. In band DTMF is usually not reliable in short bursts. 

Jerry 




From: Dan [mailto:freeswitch-users at digitaldan.com] 
Sent: Wednesday, January 13, 2010 10:02 AM 
To: freeswitch-users at lists.freeswitch.org 
Subject: [Freeswitch-users] Cisco media gateway and DTMF 



Hi guys, I'm having an issue with DTMF detection from our cisco 5300 gateways. DTMF detection has been working great when someone was manually inputing the digits (ie there are small pauses between the DTMF digits), but when the other end is sending those digits from say a speed dial or programmed dial string and can send the DTMF digits one right after the other we start to drop one or more digits about 75% of the time. 


I upgraded to trunk this morning and have tried everything that was listed on http://wiki.freeswitch.org/wiki/RTP_Issues . I have a pcap output of the SIP and RTP data, it looks fine to me, I can see the DTMF payloads coming as they should be. Any ideas? I would be fine sending or posting the pcap file to someone privately. 


Thanks 
Dan- 
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From brian at freeswitch.org  Wed Jan 13 11:39:52 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 13:39:52 -0600
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <26415028.141263411386237.JavaMail.root@zimbra>
References: <26415028.141263411386237.JavaMail.root@zimbra>
Message-ID: <84ACB346-D901-4B2A-AFD0-CDA482D1B731@freeswitch.org>

Cisco gateways can detect DTMF at random if you have kids screaming or loud noises in the background... very unreliable.

/b

On Jan 13, 2010, at 1:36 PM, Dan wrote:

> Sorry, I should have made that more clear, this is using 2833.  As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through).  

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From freeswitch-users at digitaldan.com  Wed Jan 13 12:12:14 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Wed, 13 Jan 2010 13:12:14 -0700 (MST)
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <24348778.171263412569609.JavaMail.root@zimbra>
Message-ID: <1256032.191263413534672.JavaMail.root@zimbra>

Thanks for your reply, 


Wireshark shows all of the digits coming through cleanly, in the case below (also on pastebin) its 0951*72789, freeswitch does not seem to be processing one or more of the rtp events. Is there some specific debugging I should turn on that might give me some more info on why freeswitch is not processing some of these events? 


http://pastebin.freeswitch.org/11780 



449 12.280148 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
450 12.285108 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
451 12.290082 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
452 12.335129 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
453 12.340124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
454 12.345124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
455 12.365162 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21862, Time=359229277, Mark 
456 12.380167 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21863, Time=359229605 
461 12.385248 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
462 12.385253 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
465 12.390163 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
466 12.435211 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
467 12.440201 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
468 12.445205 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
469 12.480251 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21870, Time=359230405 
470 12.485275 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
471 12.485284 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
472 12.490251 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
475 12.535277 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
476 12.540287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
477 12.545287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
478 12.580337 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21877, Time=359231197 
484 12.585337 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
485 12.585343 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
486 12.590362 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
492 12.635393 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
493 12.640368 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
494 12.645369 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
495 12.665406 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21884, Time=359231677, Mark 
496 12.680431 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21885, Time=359232005 
497 12.680438 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
502 12.685408 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
506 12.690412 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
507 12.735450 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
508 12.740453 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
509 12.745469 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
511 12.780526 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21892, Time=359232805 
512 12.780534 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
513 12.785502 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
514 12.790492 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
515 12.835544 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
516 12.840537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
517 12.845537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
523 12.880582 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21899, Time=359233605 
524 12.885593 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
525 12.885599 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
526 12.890580 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
528 12.935618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
529 12.940618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
530 12.945618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
531 12.965656 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21906, Time=359234077, Mark 
532 12.980659 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21907, Time=359234413 
538 12.985690 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
539 12.985695 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
540 12.990657 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
541 13.035703 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
542 13.040701 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
543 13.045700 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
544 13.080762 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21914, Time=359235205 
545 13.080768 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
546 13.085736 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
547 13.090798 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
549 13.135809 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
550 13.140780 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
551 13.145793 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
554 13.180827 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21921, Time=359236005 
558 13.185847 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
559 13.185854 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
560 13.190821 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
566 13.235872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
567 13.240864 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
568 13.245872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) ----- Original Message ----- 
From: "Brian West" <brian at freeswitch.org> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 12:39:52 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

Cisco gateways can detect DTMF at random if you have kids screaming or loud noises in the background... very unreliable. 


/b 



On Jan 13, 2010, at 1:36 PM, Dan wrote: 


Sorry, I should have made that more clear, this is using 2833. As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through). 

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From david.villasmil.work at gmail.com  Wed Jan 13 12:20:31 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 13 Jan 2010 21:20:31 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <EA8724C8-84A8-44DB-9F76-EFA517E76402@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<EA8724C8-84A8-44DB-9F76-EFA517E76402@jerris.com>
Message-ID: <9853f4ff1001131220u72ab5177x287a6dcb84f9b185@mail.gmail.com>

Thanks for answering,

Well, imagine I have a content provider which will deliver custom ringbacks
via SIP INVITES, they point would be to receive A-leg then bridge to 2
B-legs and deliver both incoming rops to A-side. Another option is to fork
the call with bridge, the bad thing is that as soon as FS receives
progress/audio from 1 leg, FS discards the other one, not good for me :)


Thanks, and I hope you can enlighten me!

David


On Wed, Jan 13, 2010 at 6:21 PM, Michael Jerris <mike at jerris.com> wrote:

> Are you asking if you can mux all of the progress audio from multiple
> b-legs?  if so, no, and why would you want to?
>
> Mike
>
> On Jan 13, 2010, at 6:15 AM, David Villasmil wrote:
>
> > hello
> >
> > is it possible to bridge multiple b-legs and provide all audio
> > (progress) until there is an answer on one channel?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From david.villasmil.work at gmail.com  Wed Jan 13 12:23:20 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 13 Jan 2010 21:23:20 +0100
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0F362D9@BESONE.corp.dom>
References: <A2167AE7633A5C4DAE56EEBD1428ECC0F362D9@BESONE.corp.dom>
Message-ID: <9853f4ff1001131223g68ff7baev3622d19deac45677@mail.gmail.com>

Hello,

That's easy ;) (I'm pretty new)

var = session:getVariable("uuid")

but you need to have 2 sessions.

Hope this helps

David

On Tue, Jan 12, 2010 at 10:43 AM, Cavalera Claudio Luigi <
Claudio.Cavalera at italtel.it> wrote:

> Hello,
> this should be simple in theory therefore I'm probably missing the right
> way to do it.
> I want to play with sessions in lua, bridge them, park them, etc...
>
> example1: Consider this simple lua script in which i create two
> sessions:
>
> api = freeswitch.API();
> api:execute("originate", "sofia/internal/1001%192.168.1.1 &park");
> api:execute("originate", "sofia/internal/1002%192.168.1.1 &park");
>
> now if i want to bridge them i suppose i should use something like
>
> api:execute("uuid_bridge", "uuid_1 uuid_2");
>
> but how do i get uuid_1 and uuid_2, i.e. the uuids of the two sessions?
>
>
> example2: I could create sessions with
>
> local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
> local session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");
>
> but then there is NOT a bridge API to bridge the sessions like:
> bridge(session1, session2);
>
> I admit I have not yet understood why such bridge possibility exist in
> javascript but does not exist in lua.
> http://wiki.freeswitch.org/wiki/Javascript_Misc_bridge
> I guess there is a reason for this but I can't figure it out.
>
>
> example3: yet another possibility
> local session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
> session1:originate(session2, "sofia/internal/1002%192.168.1.1", 1000);
>
> but it does not work either.
>
> Besides with this third example something strange happen:
>
> freeswitch at internal> show channels
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr
> ,dest,application,application_data,dialplan,context,read_codec,read_rate
> ,write_codec,write_rate,secure
> 1c5db2df-14ce-4516-94f2-bb7c087e0802,outbound,2010-01-12
> 10:50:59,1263289859,sofia/internal/1004,CS_REPORTING,FreeSWITCH,00000000
> 00,,1004,,,,default,PCMA,8000,PCMA,8000,
>
> 1 total.
>
> freeswitch at internal> uuid_kill 1c5db2df-14ce-4516-94f2-bb7c087e0802
> -ERR No Such Channel!
>
> freeswitch at internal> show channels
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr
> ,dest,application,application_data,dialplan,context,read_codec,read_rate
> ,write_codec,write_rate,secure
> 1c5db2df-14ce-4516-94f2-bb7c087e0802,outbound,2010-01-12
> 10:50:59,1263289859,sofia/internal/1004,CS_REPORTING,FreeSWITCH,00000000
> 00,,1004,,,,default,PCMA,8000,PCMA,8000,
>
> 1 total.
>
> freeswitch at internal>
>
> If you are interested the full log is here:
> http://pastebin.freeswitch.org/11757
> but I admit i'm not on latest trunk yet!
>
> Thanks.
> Ciao,
> Claudio
>
>
> PS: Is there a reason why there is a uuid_park command but not
> uuid_valet_park ?
>
>
> Internet Email Confidentiality Footer
>
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From mike at jerris.com  Wed Jan 13 12:41:51 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 15:41:51 -0500
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <1256032.191263413534672.JavaMail.root@zimbra>
References: <1256032.191263413534672.JavaMail.root@zimbra>
Message-ID: <692070C5-1915-4CDA-B3A3-C7BB3D0E2CA9@jerris.com>

What are you doing with these digits on the freeswitch side, what is it thinking is passed, and do you see anything in debug logs.  You also could uncomment the #define DEBUG_2833  from switch_rtp.c for more info.

Mike

On Jan 13, 2010, at 3:12 PM, Dan wrote:

> Thanks for your reply,
> 
> Wireshark shows all of the digits coming through cleanly, in the case below (also on pastebin)  its 0951*72789, freeswitch does not seem to be processing one or more of the rtp events.  Is there some specific debugging I should turn on that might give me some more info on why freeswitch is not processing some of these events?
> 
> http://pastebin.freeswitch.org/11780
> 
>     449 12.280148   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Zero 0
>     450 12.285108   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Zero 0
>     451 12.290082   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Zero 0
>     452 12.335129   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end)
>     453 12.340124   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end)
>     454 12.345124   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end)
>     455 12.365162   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21862, Time=359229277, Mark 
>     456 12.380167   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21863, Time=359229605 
>     461 12.385248   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9
>     462 12.385253   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9
>     465 12.390163   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9
>     466 12.435211   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end)
>     467 12.440201   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end)
>     468 12.445205   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end)
>     469 12.480251   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21870, Time=359230405 
>     470 12.485275   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Five 5
>     471 12.485284   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Five 5
>     472 12.490251   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Five 5
>     475 12.535277   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Five 5 (end)
>     476 12.540287   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Five 5 (end)
>     477 12.545287   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Five 5 (end)
>     478 12.580337   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21877, Time=359231197 
>     484 12.585337   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF One 1
>     485 12.585343   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF One 1
>     486 12.590362   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF One 1
>     492 12.635393   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF One 1 (end)
>     493 12.640368   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF One 1 (end)
>     494 12.645369   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF One 1 (end)
>     495 12.665406   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21884, Time=359231677, Mark 
>     496 12.680431   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21885, Time=359232005 
>     497 12.680438   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Star *
>     502 12.685408   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Star *
>     506 12.690412   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Star *
>     507 12.735450   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Star * (end)
>     508 12.740453   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Star * (end)
>     509 12.745469   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Star * (end)
>     511 12.780526   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21892, Time=359232805 
>     512 12.780534   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7
>     513 12.785502   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7
>     514 12.790492   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7
>     515 12.835544   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end)
>     516 12.840537   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end)
>     517 12.845537   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end)
>     523 12.880582   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21899, Time=359233605 
>     524 12.885593   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Two 2
>     525 12.885599   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Two 2
>     526 12.890580   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Two 2
>     528 12.935618   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Two 2 (end)
>     529 12.940618   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Two 2 (end)
>     530 12.945618   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Two 2 (end)
>     531 12.965656   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21906, Time=359234077, Mark 
>     532 12.980659   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21907, Time=359234413 
>     538 12.985690   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7
>     539 12.985695   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7
>     540 12.990657   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7
>     541 13.035703   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end)
>     542 13.040701   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end)
>     543 13.045700   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end)
>     544 13.080762   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21914, Time=359235205 
>     545 13.080768   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Eight 8
>     546 13.085736   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Eight 8
>     547 13.090798   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Eight 8
>     549 13.135809   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end)
>     550 13.140780   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end)
>     551 13.145793   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end)
>     554 13.180827   192.168.3.21          192.168.6.130         RTP      PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21921, Time=359236005 
>     558 13.185847   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9
>     559 13.185854   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9
>     560 13.190821   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9
>     566 13.235872   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end)
>     567 13.240864   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end)
>     568 13.245872   192.168.3.21          192.168.6.130         RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end)
> ----- Original Message -----
> From: "Brian West" <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wednesday, January 13, 2010 12:39:52 PM
> Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF
> 
> Cisco gateways can detect DTMF at random if you have kids screaming or loud noises in the background... very unreliable.
> 
> /b
> 
> On Jan 13, 2010, at 1:36 PM, Dan wrote:
> 
> Sorry, I should have made that more clear, this is using 2833.  As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through).  
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From mike at jerris.com  Wed Jan 13 12:43:59 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 15:43:59 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001131220u72ab5177x287a6dcb84f9b185@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<EA8724C8-84A8-44DB-9F76-EFA517E76402@jerris.com>
	<9853f4ff1001131220u72ab5177x287a6dcb84f9b185@mail.gmail.com>
Message-ID: <3BED4A88-87DB-4D9A-979D-6DBFBEA21778@jerris.com>

Muxing 2 ringtones together would result in complete nonsense, especially in the case of custom ringback.  How could this ever be usable?

Mike

On Jan 13, 2010, at 3:20 PM, David Villasmil wrote:

> Thanks for answering,
> 
> Well, imagine I have a content provider which will deliver custom ringbacks via SIP INVITES, they point would be to receive A-leg then bridge to 2 B-legs and deliver both incoming rops to A-side. Another option is to fork the call with bridge, the bad thing is that as soon as FS receives progress/audio from 1 leg, FS discards the other one, not good for me :)
> 
> 
> Thanks, and I hope you can enlighten me!
> 
> David
> 
> 
> On Wed, Jan 13, 2010 at 6:21 PM, Michael Jerris <mike at jerris.com> wrote:
> Are you asking if you can mux all of the progress audio from multiple b-legs?  if so, no, and why would you want to?
> 
> Mike
> 
> On Jan 13, 2010, at 6:15 AM, David Villasmil wrote:
> 
> > hello
> >
> > is it possible to bridge multiple b-legs and provide all audio
> > (progress) until there is an answer on one channel?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
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From freeswitch-users at digitaldan.com  Wed Jan 13 12:54:25 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Wed, 13 Jan 2010 13:54:25 -0700 (MST)
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <692070C5-1915-4CDA-B3A3-C7BB3D0E2CA9@jerris.com>
Message-ID: <18293564.271263416065447.JavaMail.root@zimbra>

Thanks, I will try and turn up the debugging on it. I have a lua script that is calling session:getDigits(20,"#",15000) . When I watch it under fs_cli I'm getting this output 

----- Original Message ----- 
From: "Michael Jerris" <mike at jerris.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 1:41:51 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

What are you doing with these digits on the freeswitch side, what is it thinking is passed, and do you see anything in debug logs. You also could uncomment the #define DEBUG_2833 from switch_rtp.c for more info. 


Mike 




On Jan 13, 2010, at 3:12 PM, Dan wrote: 




Thanks for your reply, 


Wireshark shows all of the digits coming through cleanly, in the case below (also on pastebin) its 0951*72789, freeswitch does not seem to be processing one or more of the rtp events. Is there some specific debugging I should turn on that might give me some more info on why freeswitch is not processing some of these events? 


http://pastebin.freeswitch.org/11780 



449 12.280148 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
450 12.285108 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
451 12.290082 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
452 12.335129 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
453 12.340124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
454 12.345124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
455 12.365162 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21862, Time=359229277, Mark 
456 12.380167 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21863, Time=359229605 
461 12.385248 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
462 12.385253 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
465 12.390163 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
466 12.435211 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
467 12.440201 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
468 12.445205 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
469 12.480251 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21870, Time=359230405 
470 12.485275 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
471 12.485284 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
472 12.490251 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
475 12.535277 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
476 12.540287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
477 12.545287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
478 12.580337 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21877, Time=359231197 
484 12.585337 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
485 12.585343 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
486 12.590362 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
492 12.635393 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
493 12.640368 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
494 12.645369 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
495 12.665406 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21884, Time=359231677, Mark 
496 12.680431 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21885, Time=359232005 
497 12.680438 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
502 12.685408 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
506 12.690412 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
507 12.735450 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
508 12.740453 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
509 12.745469 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
511 12.780526 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21892, Time=359232805 
512 12.780534 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
513 12.785502 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
514 12.790492 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
515 12.835544 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
516 12.840537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
517 12.845537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
523 12.880582 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21899, Time=359233605 
524 12.885593 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
525 12.885599 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
526 12.890580 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
528 12.935618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
529 12.940618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
530 12.945618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
531 12.965656 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21906, Time=359234077, Mark 
532 12.980659 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21907, Time=359234413 
538 12.985690 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
539 12.985695 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
540 12.990657 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
541 13.035703 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
542 13.040701 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
543 13.045700 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
544 13.080762 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21914, Time=359235205 
545 13.080768 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
546 13.085736 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
547 13.090798 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
549 13.135809 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
550 13.140780 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
551 13.145793 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
554 13.180827 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21921, Time=359236005 
558 13.185847 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
559 13.185854 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
560 13.190821 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
566 13.235872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
567 13.240864 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
568 13.245872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) ----- Original Message ----- 
From: "Brian West" < brian at freeswitch.org > 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 12:39:52 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

Cisco gateways can detect DTMF at random if you have kids screaming or loud noises in the background... very unreliable. 


/b 



On Jan 13, 2010, at 1:36 PM, Dan wrote: 


Sorry, I should have made that more clear, this is using 2833. As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through). 

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From freeswitch-users at digitaldan.com  Wed Jan 13 12:56:16 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Wed, 13 Jan 2010 13:56:16 -0700 (MST)
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <692070C5-1915-4CDA-B3A3-C7BB3D0E2CA9@jerris.com>
Message-ID: <8181470.301263416176200.JavaMail.root@zimbra>

Thanks, I will try and turn up the debugging on it. I have a lua script that is calling session:getDigits(20,"#",15000) . When I watch it under fs_cli I'm getting this output 



2010-01-13 10:44:15.750329 [NOTICE] switch_cpp.cpp:1129 main: Getting digits 
2010-01-13 10:44:17.586676 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 0:400 
2010-01-13 10:44:17.790705 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 5:400 
2010-01-13 10:44:17.910737 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 1:400 
2010-01-13 10:44:18.114779 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 7:400 
2010-01-13 10:44:18.234812 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 2:400 
2010-01-13 10:44:18.462832 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 8:400 
2010-01-13 10:44:18.582852 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 9:400 
2010-01-13 10:44:23.311740 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF #:400 
2010-01-13 10:44:23.311740 [DEBUG] switch_cpp.cpp:735 getDigits dtmf_buf: 0517289 




It will drop any one or more numbers from any part of that sequence. 


Thanks 
Dan 
(sorry for the duped message) ----- Original Message ----- 
From: "Michael Jerris" <mike at jerris.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 1:41:51 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

What are you doing with these digits on the freeswitch side, what is it thinking is passed, and do you see anything in debug logs. You also could uncomment the #define DEBUG_2833 from switch_rtp.c for more info. 


Mike 




On Jan 13, 2010, at 3:12 PM, Dan wrote: 




Thanks for your reply, 


Wireshark shows all of the digits coming through cleanly, in the case below (also on pastebin) its 0951*72789, freeswitch does not seem to be processing one or more of the rtp events. Is there some specific debugging I should turn on that might give me some more info on why freeswitch is not processing some of these events? 


http://pastebin.freeswitch.org/11780 



449 12.280148 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
450 12.285108 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
451 12.290082 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
452 12.335129 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
453 12.340124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
454 12.345124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
455 12.365162 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21862, Time=359229277, Mark 
456 12.380167 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21863, Time=359229605 
461 12.385248 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
462 12.385253 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
465 12.390163 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
466 12.435211 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
467 12.440201 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
468 12.445205 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
469 12.480251 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21870, Time=359230405 
470 12.485275 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
471 12.485284 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
472 12.490251 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
475 12.535277 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
476 12.540287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
477 12.545287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
478 12.580337 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21877, Time=359231197 
484 12.585337 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
485 12.585343 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
486 12.590362 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
492 12.635393 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
493 12.640368 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
494 12.645369 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
495 12.665406 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21884, Time=359231677, Mark 
496 12.680431 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21885, Time=359232005 
497 12.680438 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
502 12.685408 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
506 12.690412 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
507 12.735450 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
508 12.740453 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
509 12.745469 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
511 12.780526 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21892, Time=359232805 
512 12.780534 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
513 12.785502 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
514 12.790492 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
515 12.835544 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
516 12.840537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
517 12.845537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
523 12.880582 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21899, Time=359233605 
524 12.885593 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
525 12.885599 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
526 12.890580 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
528 12.935618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
529 12.940618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
530 12.945618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
531 12.965656 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21906, Time=359234077, Mark 
532 12.980659 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21907, Time=359234413 
538 12.985690 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
539 12.985695 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
540 12.990657 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
541 13.035703 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
542 13.040701 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
543 13.045700 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
544 13.080762 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21914, Time=359235205 
545 13.080768 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
546 13.085736 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
547 13.090798 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
549 13.135809 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
550 13.140780 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
551 13.145793 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
554 13.180827 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21921, Time=359236005 
558 13.185847 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
559 13.185854 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
560 13.190821 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
566 13.235872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
567 13.240864 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
568 13.245872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) ----- Original Message ----- 
From: "Brian West" < brian at freeswitch.org > 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 12:39:52 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

Cisco gateways can detect DTMF at random if you have kids screaming or loud noises in the background... very unreliable. 


/b 



On Jan 13, 2010, at 1:36 PM, Dan wrote: 


Sorry, I should have made that more clear, this is using 2833. As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through). 

_______________________________________________ 
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FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 
_______________________________________________ 
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
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FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
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From msc at freeswitch.org  Wed Jan 13 14:36:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Jan 2010 14:36:23 -0800
Subject: [Freeswitch-users] 10 seconds delay on fallback
In-Reply-To: <201001130746.47968.sos@sokhapkin.dyndns.org>
References: <201001130746.47968.sos@sokhapkin.dyndns.org>
Message-ID: <87f2f3b91001131436k68fcc2f3kfa4ce66f4fd5c72@mail.gmail.com>

Which rev of FreeSWITCH are you running? Also, collect a full debug trace of
a working vs. non-working call so that you can compare what's happening. Put
those on pastebin so others can have a look.
-MC

On Wed, Jan 13, 2010 at 4:46 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> To implement fallback to backup PSTN routes and do LCR my dialplan executes
> the following commands:
>
> set execute_on_answer=set hangup_after_bridge=true
> set some custom channel variables
> bridge sofia/gateway1/number
> set some custom channel variables
> bridge sofia/gateway2/number
> ....
>
> This generally works fine if gateway1 returns SIP error, dialplan sets
> channel
> variables to another values and calls gateway2 immediately. But if GW1
> responds with 183 early media and SIP error after that, bad thing happens -
> dialplan continues immediately, sets channel variables, executes bridge
> application, but INVITE to GW2 is sent after 10 seconds delay. Caller gets
> 10
> extra seconds of post dial delay. I suspect the delay happens because of
> attempt to read audio frame, switch_ivr_originate() has the following
> lines:
>
> if (switch_channel_media_ready(caller_channel)) {
>        tstatus = switch_core_session_read_frame()
>      ...
>
> Any advice how to avoid extra delay in this situation?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mouncifbb at gmail.com  Wed Jan 13 15:00:32 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Wed, 13 Jan 2010 18:00:32 -0500
Subject: [Freeswitch-users] rtp-autoflush
Message-ID: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>

We are running freeswitch with bypass_media=true, if we change the following
settings :
rtp-autoflush-during-bridge and  rtp-autoflush
does it affect anything?

Thanks
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From anthony.minessale at gmail.com  Wed Jan 13 15:08:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Jan 2010 17:08:51 -0600
Subject: [Freeswitch-users] rtp-autoflush
In-Reply-To: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>
References: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>
Message-ID: <191c3a031001131508i713ef7b8r48ba21c875e4e723@mail.gmail.com>

no, as the name implies bypass_media does not touch media whatsoever and
hense rtp related settings do not come into play.


On Wed, Jan 13, 2010 at 5:00 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> We are running freeswitch with bypass_media=true, if we change the
> following settings :
> rtp-autoflush-during-bridge and  rtp-autoflush
> does it affect anything?
>
> Thanks
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mouncifbb at gmail.com  Wed Jan 13 15:20:13 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Wed, 13 Jan 2010 18:20:13 -0500
Subject: [Freeswitch-users] rtp-autoflush
In-Reply-To: <191c3a031001131508i713ef7b8r48ba21c875e4e723@mail.gmail.com>
References: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>
	<191c3a031001131508i713ef7b8r48ba21c875e4e723@mail.gmail.com>
Message-ID: <ade741d61001131520x1390ba23ud33dc4f3daba58f6@mail.gmail.com>

thank you, also is it okay to set bypass_media_after_bridge=true and
bypass_media=true at the same time?

On Wed, Jan 13, 2010 at 6:08 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> no, as the name implies bypass_media does not touch media whatsoever and
> hense rtp related settings do not come into play.
>
>
> On Wed, Jan 13, 2010 at 5:00 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>
>> We are running freeswitch with bypass_media=true, if we change the
>> following settings :
>> rtp-autoflush-during-bridge and  rtp-autoflush
>> does it affect anything?
>>
>> Thanks
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Wed Jan 13 15:30:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Jan 2010 17:30:33 -0600
Subject: [Freeswitch-users] rtp-autoflush
In-Reply-To: <ade741d61001131520x1390ba23ud33dc4f3daba58f6@mail.gmail.com>
References: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>
	<191c3a031001131508i713ef7b8r48ba21c875e4e723@mail.gmail.com>
	<ade741d61001131520x1390ba23ud33dc4f3daba58f6@mail.gmail.com>
Message-ID: <191c3a031001131530u6b105b35jcb09aa989a6ecce7@mail.gmail.com>

its better to use one of the other but it probably is ok

On Wed, Jan 13, 2010 at 5:20 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> thank you, also is it okay to set bypass_media_after_bridge=true and
> bypass_media=true at the same time?
>
>
> On Wed, Jan 13, 2010 at 6:08 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> no, as the name implies bypass_media does not touch media whatsoever and
>> hense rtp related settings do not come into play.
>>
>>
>> On Wed, Jan 13, 2010 at 5:00 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>>
>>> We are running freeswitch with bypass_media=true, if we change the
>>> following settings :
>>>  rtp-autoflush-during-bridge and  rtp-autoflush
>>> does it affect anything?
>>>
>>> Thanks
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From sos at sokhapkin.dyndns.org  Wed Jan 13 15:36:42 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 18:36:42 -0500
Subject: [Freeswitch-users] 10 seconds delay on fallback
In-Reply-To: <87f2f3b91001131436k68fcc2f3kfa4ce66f4fd5c72@mail.gmail.com>
References: <201001130746.47968.sos@sokhapkin.dyndns.org>
	<87f2f3b91001131436k68fcc2f3kfa4ce66f4fd5c72@mail.gmail.com>
Message-ID: <201001131836.42349.sos@sokhapkin.dyndns.org>

freeswitch at internal> version
FreeSWITCH Version 1.0.5pre10 (16012M)

The problem happens with bypass_media=true.

The lines in log related to the call are (sorry, need to hide IP addresses and 
numbers):

recv 1080 bytes from udp/[X.X.X.X]:5060 at 03:27:02.529350:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
...
2010-01-12 22:27:02.529311 [INFO] sofia.c:509 Update Callee ID to "XXXXX" 
<sip:XXXXXX at XXXXXX>
2010-01-12 22:27:02.529311 [DEBUG] sofia.c:3806 Channel sofia/cwu/XXXXXX at XXXXX 
entering state [proceeding][183]

2010-01-12 22:27:02.529311 [NOTICE] sofia.c:3885 Pre-Answer 2010-01-12 
22:27:02.529311 [DEBUG] switch_channel.c:2020 Send signal XXXXXXXXX [BREAK]
2010-01-12 22:27:02.529311 [DEBUG] sofia.c:3898 XXXXXXXXXXXXX receive message 
[PROGRESS]
2010-01-12 22:27:02.529311 [INFO] sofia.c:3898 Sending early media
2010-01-12 22:27:02.529311 [NOTICE] mod_sofia.c:1765 Pre-Answer XXXXXXXXXXX!
2010-01-12 22:27:02.529311 [DEBUG] switch_core_session.c:700 Send signal 
XXXXXXXXXX [BREAK]
2010-01-12 22:27:02.529311 [DEBUG] switch_core_session.c:639 Send signal 
XXXXXXXXXXXX [BREAK]
   ------------------------------------------------------------------------
recv 716 bytes from udp/[XXXXXXXXX]:5060 at 03:27:02.608352:
   ------------------------------------------------------------------------
   SIP/2.0 480 Temporarily Unavailable
....
2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXX SET 
[A2B_id]=[86339421]
EXECUTE XXXXXXXXXX set(A2B_tp_id_trunk=126)
2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXXXXXXXXX SET 
[A2B_tp_id_trunk]=[126]
EXECUTE XXXXXXXXX sched_hangup(+10800)
2010-01-12 22:27:02.611340 [DEBUG] switch_scheduler.c:214 Added task 109951 
switch_ivr_schedule_hangup (8e75053d-e864-4f34-b768-b41b6839dadd) to run at 
1263364022
EXECUTE XXXXXXXXXXX set(bypass_media=true)
2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXXXXXX SET 
[bypass_media]=[true]
EXECUTE XXXXXXXXX 
bridge([sip_contact_user=XXXXXX,sip_auth_username=,sip_auth_password=]sofia/cwu/XXXXXXX at XXXXXXX)
2010-01-12 22:27:02.612344 [DEBUG] switch_ivr.c:1199 XXXXXXXX receive message 
[MEDIA]
2010-01-12 22:27:02.612344 [DEBUG] switch_core_session.c:639 Send signal 
XXXXXXXXXXXX [BREAK]

10 seconds later:

2010-01-12 22:27:12.614323 [NOTICE] switch_channel.c:613 New Channel 
sofia/cwu/XXXXXXX at XXXXXXX [68cb4360-8da5-49fc-8fbd-f4a4e
4351a94]





On Wednesday 13 January 2010, Michael Collins wrote:
> Which rev of FreeSWITCH are you running? Also, collect a full debug trace
> of a working vs. non-working call so that you can compare what's happening.
> Put those on pastebin so others can have a look.
> -MC
>
> On Wed, Jan 13, 2010 at 4:46 AM, Sergey Okhapkin
>
> <sos at sokhapkin.dyndns.org>wrote:
> > To implement fallback to backup PSTN routes and do LCR my dialplan
> > executes the following commands:
> >
> > set execute_on_answer=set hangup_after_bridge=true
> > set some custom channel variables
> > bridge sofia/gateway1/number
> > set some custom channel variables
> > bridge sofia/gateway2/number
> > ....
> >
> > This generally works fine if gateway1 returns SIP error, dialplan sets
> > channel
> > variables to another values and calls gateway2 immediately. But if GW1
> > responds with 183 early media and SIP error after that, bad thing happens
> > - dialplan continues immediately, sets channel variables, executes bridge
> > application, but INVITE to GW2 is sent after 10 seconds delay. Caller
> > gets 10
> > extra seconds of post dial delay. I suspect the delay happens because of
> > attempt to read audio frame, switch_ivr_originate() has the following
> > lines:
> >
> > if (switch_channel_media_ready(caller_channel)) {
> >        tstatus = switch_core_session_read_frame()
> >      ...
> >
> > Any advice how to avoid extra delay in this situation?
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From david.villasmil.work at gmail.com  Wed Jan 13 18:10:23 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 14 Jan 2010 03:10:23 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <3BED4A88-87DB-4D9A-979D-6DBFBEA21778@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<EA8724C8-84A8-44DB-9F76-EFA517E76402@jerris.com>
	<9853f4ff1001131220u72ab5177x287a6dcb84f9b185@mail.gmail.com>
	<3BED4A88-87DB-4D9A-979D-6DBFBEA21778@jerris.com>
Message-ID: <9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>

MIke,

This is done on a daily basis by i.e. mobile companies, you dial a customer
number and you hear some music whilst hearing the ringing at the same time.

If it can not be done by muxing both rtps, can it be done the other way,
then?: (Another option is to fork the call with bridge, the bad thing is
that as soon as FS receives progress/audio from 1 leg, FS discards the other
one, not good for me )

Thanks


david

On Wed, Jan 13, 2010 at 9:43 PM, Michael Jerris <mike at jerris.com> wrote:

> Muxing 2 ringtones together would result in complete nonsense, especially
> in the case of custom ringback.  How could this ever be usable?
>
> Mike
>
> On Jan 13, 2010, at 3:20 PM, David Villasmil wrote:
>
> Thanks for answering,
>
> Well, imagine I have a content provider which will deliver custom ringbacks
> via SIP INVITES, they point would be to receive A-leg then bridge to 2
> B-legs and deliver both incoming rops to A-side. Another option is to fork
> the call with bridge, the bad thing is that as soon as FS receives
> progress/audio from 1 leg, FS discards the other one, not good for me :)
>
>
> Thanks, and I hope you can enlighten me!
>
> David
>
>
> On Wed, Jan 13, 2010 at 6:21 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> Are you asking if you can mux all of the progress audio from multiple
>> b-legs?  if so, no, and why would you want to?
>>
>> Mike
>>
>> On Jan 13, 2010, at 6:15 AM, David Villasmil wrote:
>>
>> > hello
>> >
>> > is it possible to bridge multiple b-legs and provide all audio
>> > (progress) until there is an answer on one channel?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Jan 13 18:42:13 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 20:42:13 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<EA8724C8-84A8-44DB-9F76-EFA517E76402@jerris.com>
	<9853f4ff1001131220u72ab5177x287a6dcb84f9b185@mail.gmail.com>
	<3BED4A88-87DB-4D9A-979D-6DBFBEA21778@jerris.com>
	<9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
Message-ID: <C3D4ADB0-7A80-45A9-91E1-FFABB987851B@freeswitch.org>

It can not be done currently and I don't expect this to ever be done.  The specs says the first target out of all invites to provide media wins and the others are canceled.  That is why it behaves the way it does.

You can ignore_early_media=true and set ringback=blah.wav if you wish to provide caller ringback.

/b

On Jan 13, 2010, at 8:10 PM, David Villasmil wrote:

> MIke,
> 
> This is done on a daily basis by i.e. mobile companies, you dial a customer number and you hear some music whilst hearing the ringing at the same time.
> 
> If it can not be done by muxing both rtps, can it be done the other way, then?: (Another option is to fork the call with bridge, the bad thing is that as soon as FS receives progress/audio from 1 leg, FS discards the other one, not good for me )
> 
> Thanks
> 
> 
> david




From mouncifbb at gmail.com  Wed Jan 13 18:46:15 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Wed, 13 Jan 2010 21:46:15 -0500
Subject: [Freeswitch-users] rtp-autoflush
In-Reply-To: <191c3a031001131530u6b105b35jcb09aa989a6ecce7@mail.gmail.com>
References: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>
	<191c3a031001131508i713ef7b8r48ba21c875e4e723@mail.gmail.com>
	<ade741d61001131520x1390ba23ud33dc4f3daba58f6@mail.gmail.com>
	<191c3a031001131530u6b105b35jcb09aa989a6ecce7@mail.gmail.com>
Message-ID: <ade741d61001131846v5379fe23ua550e7293515e84d@mail.gmail.com>

One more question, will it participate in codec negotiation sent by the
inbound voip provider?
Right now I have it setup this way:

Incoming Inbound DID (voip provider --> freeswitch(media bypass mode) ->
endpoint sip device.

Thanks again!

On Wed, Jan 13, 2010 at 6:30 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> its better to use one of the other but it probably is ok
>
>
> On Wed, Jan 13, 2010 at 5:20 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>
>> thank you, also is it okay to set bypass_media_after_bridge=true and
>> bypass_media=true at the same time?
>>
>>
>> On Wed, Jan 13, 2010 at 6:08 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> no, as the name implies bypass_media does not touch media whatsoever and
>>> hense rtp related settings do not come into play.
>>>
>>>
>>> On Wed, Jan 13, 2010 at 5:00 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>>>
>>>> We are running freeswitch with bypass_media=true, if we change the
>>>> following settings :
>>>>  rtp-autoflush-during-bridge and  rtp-autoflush
>>>> does it affect anything?
>>>>
>>>> Thanks
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sos at sokhapkin.dyndns.org  Wed Jan 13 19:05:43 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 22:05:43 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <C3D4ADB0-7A80-45A9-91E1-FFABB987851B@freeswitch.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
	<C3D4ADB0-7A80-45A9-91E1-FFABB987851B@freeswitch.org>
Message-ID: <201001132205.43306.sos@sokhapkin.dyndns.org>

Well, the question is - what is "media"? To me media is what is returned 
by "200 OK" response to INVITE. 18X provisional responses are NOT media, they 
are early media indications (well, even with RTP stream inside), which shall 
be sent back to the caller, but should NOT be accounted in the call 
processing. Early media is too early to be accounted. Only responses with SIP 
code 200 or more matter.

On Wednesday 13 January 2010, Brian West wrote:
> It can not be done currently and I don't expect this to ever be done.  The
> specs says the first target out of all invites to provide media wins and
> the others are canceled.  That is why it behaves the way it does.
>
> You can ignore_early_media=true and set ringback=blah.wav if you wish to
> provide caller ringback.
>
> /b
>
> On Jan 13, 2010, at 8:10 PM, David Villasmil wrote:
> > MIke,
> >
> > This is done on a daily basis by i.e. mobile companies, you dial a
> > customer number and you hear some music whilst hearing the ringing at the
> > same time.
> >
> > If it can not be done by muxing both rtps, can it be done the other way,
> > then?: (Another option is to fork the call with bridge, the bad thing is
> > that as soon as FS receives progress/audio from 1 leg, FS discards the
> > other one, not good for me )
> >
> > Thanks
> >
> >
> > david
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From brian at freeswitch.org  Wed Jan 13 19:11:38 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 21:11:38 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001132205.43306.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
	<C3D4ADB0-7A80-45A9-91E1-FFABB987851B@freeswitch.org>
	<201001132205.43306.sos@sokhapkin.dyndns.org>
Message-ID: <7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>

By all accounts its still Media and the first one to provide it in a forked dial is to be connected to the channel of the calling party even if its early... the call answer time is not started till the 200 is received.  

I'm not talking about billing or answered time either.. i'm talking pure early media and how it is to be handled in FreeSWITCH.

That my friend is in the specs to behave like that.

/b



On Jan 13, 2010, at 9:05 PM, Sergey Okhapkin wrote:

> Well, the question is - what is "media"? To me media is what is returned 
> by "200 OK" response to INVITE. 18X provisional responses are NOT media, they 
> are early media indications (well, even with RTP stream inside), which shall 
> be sent back to the caller, but should NOT be accounted in the call 
> processing. Early media is too early to be accounted. Only responses with SIP 
> code 200 or more matter.




From mike at jerris.com  Wed Jan 13 19:12:48 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 22:12:48 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<EA8724C8-84A8-44DB-9F76-EFA517E76402@jerris.com>
	<9853f4ff1001131220u72ab5177x287a6dcb84f9b185@mail.gmail.com>
	<3BED4A88-87DB-4D9A-979D-6DBFBEA21778@jerris.com>
	<9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
Message-ID: <FA68C6E7-732F-4104-BFF1-23333AD2A68E@jerris.com>

I have never seen a carrier do this and I still doubt that it would be  
at all usable.  What happens now is we essentially fork the dial,  
first with media wins.  Alternatively you can wait for first answer,  
and provide your own ringback tones.  Can you describe a bit more your  
use case as I just don't get it.  With what you describe, I imagine a  
call where one leg gets a sit tone due to problem with the number and  
it's muxed with a ringtone of the other b leg, or ring and busy at the  
same time.

Mike

On Jan 13, 2010, at 9:10 PM, David Villasmil <david.villasmil.work at gmail.com 
 > wrote:

> MIke,
>
> This is done on a daily basis by i.e. mobile companies, you dial a  
> customer number and you hear some music whilst hearing the ringing  
> at the same time.
>
> If it can not be done by muxing both rtps, can it be done the other  
> way, then?: (Another option is to fork the call with bridge, the bad  
> thing is that as soon as FS receives progress/audio from 1 leg, FS  
> discards the other one, not good for me )
>
> Thanks
>
>
> david
>
> On Wed, Jan 13, 2010 at 9:43 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> Muxing 2 ringtones together would result in complete nonsense,  
> especially in the case of custom ringback.  How could this ever be  
> usable?
>
> Mike
>
> On Jan 13, 2010, at 3:20 PM, David Villasmil wrote:
>
>> Thanks for answering,
>>
>> Well, imagine I have a content provider which will deliver custom  
>> ringbacks via SIP INVITES, they point would be to receive A-leg  
>> then bridge to 2 B-legs and deliver both incoming rops to A-side.  
>> Another option is to fork the call with bridge, the bad thing is  
>> that as soon as FS receives progress/audio from 1 leg, FS discards  
>> the other one, not good for me :)
>>
>>
>> Thanks, and I hope you can enlighten me!
>>
>> David
>>
>>
>> On Wed, Jan 13, 2010 at 6:21 PM, Michael Jerris <mike at jerris.com>  
>> wrote:
>> Are you asking if you can mux all of the progress audio from  
>> multiple b-legs?  if so, no, and why would you want to?
>>
>> Mike
>>
>> On Jan 13, 2010, at 6:15 AM, David Villasmil wrote:
>>
>> > hello
>> >
>> > is it possible to bridge multiple b-legs and provide all audio
>> > (progress) until there is an answer on one channel?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From null at invalid.name  Wed Jan 13 17:48:03 2010
From: null at invalid.name (Dan Lane)
Date: Thu, 14 Jan 2010 01:48:03 +0000
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <3BED4A88-87DB-4D9A-979D-6DBFBEA21778@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<EA8724C8-84A8-44DB-9F76-EFA517E76402@jerris.com>
	<9853f4ff1001131220u72ab5177x287a6dcb84f9b185@mail.gmail.com>
	<3BED4A88-87DB-4D9A-979D-6DBFBEA21778@jerris.com>
Message-ID: <b042b23b1001131748t229a2206y479e56666b037378@mail.gmail.com>

On Wed, Jan 13, 2010 at 8:43 PM, Michael Jerris <mike at jerris.com> wrote:
> Muxing 2 ringtones together would result in complete nonsense, especially in
> the case of custom ringback. ?How could this ever be usable?

It sounds like he wants to mux two audio streams so he can have the
normal ringback tone overlaid on a novelty ringback tone so as not to
confuse the caller (I understand this is quite a normal way of doing
such things)

The easiest way to do this (and the way I do it) is to record a normal
ringback tone then use sox to combine the custom ringback with the
normal one and use the resulting file as your ringback. You get fine
control over the timing and volume this way too.



From sos at sokhapkin.dyndns.org  Wed Jan 13 19:29:58 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 22:29:58 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132205.43306.sos@sokhapkin.dyndns.org>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
Message-ID: <201001132229.58830.sos@sokhapkin.dyndns.org>

If you wish FreeSWITCH to be well adopted by the community, then FS should 
follow the real world "rules" but not specs...

Don't you agree that 99% of SIP servers are set up to interconnect 
with "buggy" PSTN and should follow PSTN rules, but not SIP specs?

Some background - I did run asterisk for years, but switched to FS recently 
because of critical asterisk problems with SIP handling when asterisk is not 
in the media path. I spend a lot of time porting the billing system to FS. 
And I did it. What I got? Critical problems with FS early media handling.  
Hopefully I can switch back to asterisk if FS problems with early media will 
begin to draw customers away.

On Wednesday 13 January 2010, Brian West wrote:
> By all accounts its still Media and the first one to provide it in a forked
> dial is to be connected to the channel of the calling party even if its
> early... the call answer time is not started till the 200 is received.
>
> I'm not talking about billing or answered time either.. i'm talking pure
> early media and how it is to be handled in FreeSWITCH.
>
> That my friend is in the specs to behave like that.
>
> /b
>
> On Jan 13, 2010, at 9:05 PM, Sergey Okhapkin wrote:
> > Well, the question is - what is "media"? To me media is what is returned
> > by "200 OK" response to INVITE. 18X provisional responses are NOT media,
> > they are early media indications (well, even with RTP stream inside),
> > which shall be sent back to the caller, but should NOT be accounted in
> > the call processing. Early media is too early to be accounted. Only
> > responses with SIP code 200 or more matter.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From mike at jerris.com  Wed Jan 13 19:30:40 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 22:30:40 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
	<C3D4ADB0-7A80-45A9-91E1-FFABB987851B@freeswitch.org>
	<201001132205.43306.sos@sokhapkin.dyndns.org>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
Message-ID: <174D1183-2731-454B-AFEB-9F92F6101BB1@jerris.com>

And to reiterate, you can control this to wait for first answer, at  
your option.  These 2 modes are for totally different use cases.

Mike

On Jan 13, 2010, at 10:11 PM, Brian West <brian at freeswitch.org> wrote:

> By all accounts its still Media and the first one to provide it in a  
> forked dial is to be connected to the channel of the calling party  
> even if its early... the call answer time is not started till the  
> 200 is received.
>
> I'm not talking about billing or answered time either.. i'm talking  
> pure early media and how it is to be handled in FreeSWITCH.
>
> That my friend is in the specs to behave like that.
>
> /b
>
>
>
> On Jan 13, 2010, at 9:05 PM, Sergey Okhapkin wrote:
>
>> Well, the question is - what is "media"? To me media is what is  
>> returned
>> by "200 OK" response to INVITE. 18X provisional responses are NOT  
>> media, they
>> are early media indications (well, even with RTP stream inside),  
>> which shall
>> be sent back to the caller, but should NOT be accounted in the call
>> processing. Early media is too early to be accounted. Only  
>> responses with SIP
>> code 200 or more matter.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From dave at 3c.co.uk  Wed Jan 13 19:32:05 2010
From: dave at 3c.co.uk (David Knell)
Date: Thu, 14 Jan 2010 03:32:05 +0000
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
	<C3D4ADB0-7A80-45A9-91E1-FFABB987851B@freeswitch.org>
	<201001132205.43306.sos@sokhapkin.dyndns.org>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
Message-ID: <1263439925.11216.36.camel@local.freepabx.com>

What would happen if several outdials were made from a conference,
rather than using a forked dial?

--Dave

> By all accounts its still Media and the first one to provide it in a forked dial is to be connected to the channel of the calling party even if its early... the call answer time is not started till the 200 is received.  
> 
> I'm not talking about billing or answered time either.. i'm talking pure early media and how it is to be handled in FreeSWITCH.
> 
> That my friend is in the specs to behave like that.
> 
> /b
> 
> 
> 
> On Jan 13, 2010, at 9:05 PM, Sergey Okhapkin wrote:
> 
> > Well, the question is - what is "media"? To me media is what is returned 
> > by "200 OK" response to INVITE. 18X provisional responses are NOT media, they 
> > are early media indications (well, even with RTP stream inside), which shall 
> > be sent back to the caller, but should NOT be accounted in the call 
> > processing. Early media is too early to be accounted. Only responses with SIP 
> > code 200 or more matter.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From anthony.minessale at gmail.com  Wed Jan 13 19:33:15 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Jan 2010 21:33:15 -0600
Subject: [Freeswitch-users] rtp-autoflush
In-Reply-To: <ade741d61001131846v5379fe23ua550e7293515e84d@mail.gmail.com>
References: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>
	<191c3a031001131508i713ef7b8r48ba21c875e4e723@mail.gmail.com>
	<ade741d61001131520x1390ba23ud33dc4f3daba58f6@mail.gmail.com>
	<191c3a031001131530u6b105b35jcb09aa989a6ecce7@mail.gmail.com>
	<ade741d61001131846v5379fe23ua550e7293515e84d@mail.gmail.com>
Message-ID: <191c3a031001131933g2c290efds5b8342d1ce55ca0c@mail.gmail.com>

Not if you set inbound-bypass-media = true in the sip profile or set
late-negotiaation = true and set bypass_media channel var to true from the
dp

On Jan 13, 2010 8:51 PM, "Mouncif Benniane" <mouncifbb at gmail.com> wrote:

One more question, will it participate in codec negotiation sent by the
inbound voip provider?
Right now I have it setup this way:

Incoming Inbound DID (voip provider --> freeswitch(media bypass mode) ->
endpoint sip device.

Thanks again!

On Wed, Jan 13, 2010 at 6:30 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote: > > its be...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From brian at freeswitch.org  Wed Jan 13 19:39:56 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 21:39:56 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <1263439925.11216.36.camel@local.freepabx.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
	<C3D4ADB0-7A80-45A9-91E1-FFABB987851B@freeswitch.org>
	<201001132205.43306.sos@sokhapkin.dyndns.org>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
	<1263439925.11216.36.camel@local.freepabx.com>
Message-ID: <26079A18-5968-4E1E-A276-9DE889E83FA4@freeswitch.org>

That would work but its sub optimal.  What he wants to accomplish is not do able and not going to ever happen at this rate.

/b

On Jan 13, 2010, at 9:32 PM, David Knell wrote:

> What would happen if several outdials were made from a conference,
> rather than using a forked dial?
> 
> --Dave




From brian at freeswitch.org  Wed Jan 13 19:44:27 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 21:44:27 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001132229.58830.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132205.43306.sos@sokhapkin.dyndns.org>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
	<201001132229.58830.sos@sokhapkin.dyndns.org>
Message-ID: <6D061C37-43A6-48D1-A8E3-549FC3E8679E@freeswitch.org>

Can you elaborate on these "Critical" issues you seem to be having?  Why aren't you opening a jira for them if they are that critical to your needs?

/b


On Jan 13, 2010, at 9:29 PM, Sergey Okhapkin wrote:

> If you wish FreeSWITCH to be well adopted by the community, then FS should 
> follow the real world "rules" but not specs...
> 
> Don't you agree that 99% of SIP servers are set up to interconnect 
> with "buggy" PSTN and should follow PSTN rules, but not SIP specs?
> 
> Some background - I did run asterisk for years, but switched to FS recently 
> because of critical asterisk problems with SIP handling when asterisk is not 
> in the media path. I spend a lot of time porting the billing system to FS. 
> And I did it. What I got? Critical problems with FS early media handling.  
> Hopefully I can switch back to asterisk if FS problems with early media will 
> begin to draw customers away.




From sos at sokhapkin.dyndns.org  Wed Jan 13 19:45:38 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 22:45:38 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <174D1183-2731-454B-AFEB-9F92F6101BB1@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
	<174D1183-2731-454B-AFEB-9F92F6101BB1@jerris.com>
Message-ID: <201001132245.38117.sos@sokhapkin.dyndns.org>

How to wait for first answer and pass early media back to the caller?

On Wednesday 13 January 2010, Michael Jerris wrote:
> And to reiterate, you can control this to wait for first answer, at
> your option.  These 2 modes are for totally different use cases.
>
> Mike
>
> On Jan 13, 2010, at 10:11 PM, Brian West <brian at freeswitch.org> wrote:
> > By all accounts its still Media and the first one to provide it in a
> > forked dial is to be connected to the channel of the calling party
> > even if its early... the call answer time is not started till the
> > 200 is received.
> >
> > I'm not talking about billing or answered time either.. i'm talking
> > pure early media and how it is to be handled in FreeSWITCH.
> >
> > That my friend is in the specs to behave like that.
> >
> > /b
> >
> > On Jan 13, 2010, at 9:05 PM, Sergey Okhapkin wrote:
> >> Well, the question is - what is "media"? To me media is what is
> >> returned
> >> by "200 OK" response to INVITE. 18X provisional responses are NOT
> >> media, they
> >> are early media indications (well, even with RTP stream inside),
> >> which shall
> >> be sent back to the caller, but should NOT be accounted in the call
> >> processing. Early media is too early to be accounted. Only
> >> responses with SIP
> >> code 200 or more matter.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From mike at jerris.com  Wed Jan 13 19:47:53 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 22:47:53 -0500
Subject: [Freeswitch-users] 10 seconds delay on fallback
In-Reply-To: <201001131836.42349.sos@sokhapkin.dyndns.org>
References: <201001130746.47968.sos@sokhapkin.dyndns.org>
	<87f2f3b91001131436k68fcc2f3kfa4ce66f4fd5c72@mail.gmail.com>
	<201001131836.42349.sos@sokhapkin.dyndns.org>
Message-ID: <27ADEBDB-E9C6-42AF-9A19-1CDF918A5F6E@jerris.com>

Try out a more recent trunk, specifically, svn r16193 had an important fix for signaling handling in bypass and proxy media modes.

Mike

On Jan 13, 2010, at 6:36 PM, Sergey Okhapkin wrote:

> freeswitch at internal> version
> FreeSWITCH Version 1.0.5pre10 (16012M)
> 
> The problem happens with bypass_media=true.
> 
> The lines in log related to the call are (sorry, need to hide IP addresses and 
> numbers):
> 
> recv 1080 bytes from udp/[X.X.X.X]:5060 at 03:27:02.529350:
>   ------------------------------------------------------------------------
>   SIP/2.0 183 Session Progress
> ...
> 2010-01-12 22:27:02.529311 [INFO] sofia.c:509 Update Callee ID to "XXXXX" 
> <sip:XXXXXX at XXXXXX>
> 2010-01-12 22:27:02.529311 [DEBUG] sofia.c:3806 Channel sofia/cwu/XXXXXX at XXXXX 
> entering state [proceeding][183]
> 
> 2010-01-12 22:27:02.529311 [NOTICE] sofia.c:3885 Pre-Answer 2010-01-12 
> 22:27:02.529311 [DEBUG] switch_channel.c:2020 Send signal XXXXXXXXX [BREAK]
> 2010-01-12 22:27:02.529311 [DEBUG] sofia.c:3898 XXXXXXXXXXXXX receive message 
> [PROGRESS]
> 2010-01-12 22:27:02.529311 [INFO] sofia.c:3898 Sending early media
> 2010-01-12 22:27:02.529311 [NOTICE] mod_sofia.c:1765 Pre-Answer XXXXXXXXXXX!
> 2010-01-12 22:27:02.529311 [DEBUG] switch_core_session.c:700 Send signal 
> XXXXXXXXXX [BREAK]
> 2010-01-12 22:27:02.529311 [DEBUG] switch_core_session.c:639 Send signal 
> XXXXXXXXXXXX [BREAK]
>   ------------------------------------------------------------------------
> recv 716 bytes from udp/[XXXXXXXXX]:5060 at 03:27:02.608352:
>   ------------------------------------------------------------------------
>   SIP/2.0 480 Temporarily Unavailable
> ....
> 2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXX SET 
> [A2B_id]=[86339421]
> EXECUTE XXXXXXXXXX set(A2B_tp_id_trunk=126)
> 2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXXXXXXXXX SET 
> [A2B_tp_id_trunk]=[126]
> EXECUTE XXXXXXXXX sched_hangup(+10800)
> 2010-01-12 22:27:02.611340 [DEBUG] switch_scheduler.c:214 Added task 109951 
> switch_ivr_schedule_hangup (8e75053d-e864-4f34-b768-b41b6839dadd) to run at 
> 1263364022
> EXECUTE XXXXXXXXXXX set(bypass_media=true)
> 2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXXXXXX SET 
> [bypass_media]=[true]
> EXECUTE XXXXXXXXX 
> bridge([sip_contact_user=XXXXXX,sip_auth_username=,sip_auth_password=]sofia/cwu/XXXXXXX at XXXXXXX)
> 2010-01-12 22:27:02.612344 [DEBUG] switch_ivr.c:1199 XXXXXXXX receive message 
> [MEDIA]
> 2010-01-12 22:27:02.612344 [DEBUG] switch_core_session.c:639 Send signal 
> XXXXXXXXXXXX [BREAK]
> 
> 10 seconds later:
> 
> 2010-01-12 22:27:12.614323 [NOTICE] switch_channel.c:613 New Channel 
> sofia/cwu/XXXXXXX at XXXXXXX [68cb4360-8da5-49fc-8fbd-f4a4e
> 4351a94]
> 
> 
> 
> 
> 
> On Wednesday 13 January 2010, Michael Collins wrote:
>> Which rev of FreeSWITCH are you running? Also, collect a full debug trace
>> of a working vs. non-working call so that you can compare what's happening.
>> Put those on pastebin so others can have a look.
>> -MC
>> 
>> On Wed, Jan 13, 2010 at 4:46 AM, Sergey Okhapkin
>> 
>> <sos at sokhapkin.dyndns.org>wrote:
>>> To implement fallback to backup PSTN routes and do LCR my dialplan
>>> executes the following commands:
>>> 
>>> set execute_on_answer=set hangup_after_bridge=true
>>> set some custom channel variables
>>> bridge sofia/gateway1/number
>>> set some custom channel variables
>>> bridge sofia/gateway2/number
>>> ....
>>> 
>>> This generally works fine if gateway1 returns SIP error, dialplan sets
>>> channel
>>> variables to another values and calls gateway2 immediately. But if GW1
>>> responds with 183 early media and SIP error after that, bad thing happens
>>> - dialplan continues immediately, sets channel variables, executes bridge
>>> application, but INVITE to GW2 is sent after 10 seconds delay. Caller
>>> gets 10
>>> extra seconds of post dial delay. I suspect the delay happens because of
>>> attempt to read audio frame, switch_ivr_originate() has the following
>>> lines:
>>> 
>>> if (switch_channel_media_ready(caller_channel)) {
>>>       tstatus = switch_core_session_read_frame()
>>>     ...
>>> 
>>> Any advice how to avoid extra delay in this situation?
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Wed Jan 13 19:49:12 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 22:49:12 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001132229.58830.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132205.43306.sos@sokhapkin.dyndns.org>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
	<201001132229.58830.sos@sokhapkin.dyndns.org>
Message-ID: <464CD1B0-95D0-4D6D-B330-87B059FB76A6@jerris.com>

What exactly are your problems?   Is this just the "10 second" thread you posted today?

Mike

On Jan 13, 2010, at 10:29 PM, Sergey Okhapkin wrote:

> If you wish FreeSWITCH to be well adopted by the community, then FS should 
> follow the real world "rules" but not specs...
> 
> Don't you agree that 99% of SIP servers are set up to interconnect 
> with "buggy" PSTN and should follow PSTN rules, but not SIP specs?
> 
> Some background - I did run asterisk for years, but switched to FS recently 
> because of critical asterisk problems with SIP handling when asterisk is not 
> in the media path. I spend a lot of time porting the billing system to FS. 
> And I did it. What I got? Critical problems with FS early media handling.  
> Hopefully I can switch back to asterisk if FS problems with early media will 
> begin to draw customers away.




From brian at freeswitch.org  Wed Jan 13 19:53:23 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 21:53:23 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001132245.38117.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
	<174D1183-2731-454B-AFEB-9F92F6101BB1@jerris.com>
	<201001132245.38117.sos@sokhapkin.dyndns.org>
Message-ID: <02DBC6DF-7B8A-435A-B89D-367BE0FE80C7@freeswitch.org>

Can you elaborate what you mean?  I'm guessing you want to fork dial X calls and pass a mux version of all that early media back to the A-Leg.  In which case that is 100% impossible and impractical.  This scenario never happens on the PSTN. 

Now if you want to ignore the early media and provide your own ringback and pass media once the call is answered thats doable. 

http://wiki.freeswitch.org/wiki/Channel_Variables#ringback
&&
http://wiki.freeswitch.org/wiki/Channel_Variables#ignore_early_media

If this isn't the case on either one please clarify.

/b


On Jan 13, 2010, at 9:45 PM, Sergey Okhapkin wrote:

> How to wait for first answer and pass early media back to the caller?
> 
> On Wednesday 13 January 2010, Michael Jerris wrote:
>> And to reiterate, you can control this to wait for first answer, at
>> your option.  These 2 modes are for totally different use cases.
>> 
>> Mike
> 




From sos at sokhapkin.dyndns.org  Wed Jan 13 19:55:23 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 22:55:23 -0500
Subject: [Freeswitch-users] 10 seconds delay on fallback
In-Reply-To: <27ADEBDB-E9C6-42AF-9A19-1CDF918A5F6E@jerris.com>
References: <201001130746.47968.sos@sokhapkin.dyndns.org>
	<201001131836.42349.sos@sokhapkin.dyndns.org>
	<27ADEBDB-E9C6-42AF-9A19-1CDF918A5F6E@jerris.com>
Message-ID: <201001132255.23846.sos@sokhapkin.dyndns.org>

Thank you for the heads up, will do.

On Wednesday 13 January 2010, Michael Jerris wrote:
> Try out a more recent trunk, specifically, svn r16193 had an important fix
> for signaling handling in bypass and proxy media modes.
>
> Mike
>
> On Jan 13, 2010, at 6:36 PM, Sergey Okhapkin wrote:
> > freeswitch at internal> version
> > FreeSWITCH Version 1.0.5pre10 (16012M)
> >
> > The problem happens with bypass_media=true.
> >
> > The lines in log related to the call are (sorry, need to hide IP
> > addresses and numbers):
> >
> > recv 1080 bytes from udp/[X.X.X.X]:5060 at 03:27:02.529350:
> >  
> > ------------------------------------------------------------------------
> > SIP/2.0 183 Session Progress
> > ...
> > 2010-01-12 22:27:02.529311 [INFO] sofia.c:509 Update Callee ID to "XXXXX"
> > <sip:XXXXXX at XXXXXX>
> > 2010-01-12 22:27:02.529311 [DEBUG] sofia.c:3806 Channel
> > sofia/cwu/XXXXXX at XXXXX entering state [proceeding][183]
> >
> > 2010-01-12 22:27:02.529311 [NOTICE] sofia.c:3885 Pre-Answer 2010-01-12
> > 22:27:02.529311 [DEBUG] switch_channel.c:2020 Send signal XXXXXXXXX
> > [BREAK] 2010-01-12 22:27:02.529311 [DEBUG] sofia.c:3898 XXXXXXXXXXXXX
> > receive message [PROGRESS]
> > 2010-01-12 22:27:02.529311 [INFO] sofia.c:3898 Sending early media
> > 2010-01-12 22:27:02.529311 [NOTICE] mod_sofia.c:1765 Pre-Answer
> > XXXXXXXXXXX! 2010-01-12 22:27:02.529311 [DEBUG] switch_core_session.c:700
> > Send signal XXXXXXXXXX [BREAK]
> > 2010-01-12 22:27:02.529311 [DEBUG] switch_core_session.c:639 Send signal
> > XXXXXXXXXXXX [BREAK]
> >  
> > ------------------------------------------------------------------------
> > recv 716 bytes from udp/[XXXXXXXXX]:5060 at 03:27:02.608352:
> >  
> > ------------------------------------------------------------------------
> > SIP/2.0 480 Temporarily Unavailable
> > ....
> > 2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXX SET
> > [A2B_id]=[86339421]
> > EXECUTE XXXXXXXXXX set(A2B_tp_id_trunk=126)
> > 2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXXXXXXXXX SET
> > [A2B_tp_id_trunk]=[126]
> > EXECUTE XXXXXXXXX sched_hangup(+10800)
> > 2010-01-12 22:27:02.611340 [DEBUG] switch_scheduler.c:214 Added task
> > 109951 switch_ivr_schedule_hangup (8e75053d-e864-4f34-b768-b41b6839dadd)
> > to run at 1263364022
> > EXECUTE XXXXXXXXXXX set(bypass_media=true)
> > 2010-01-12 22:27:02.611340 [DEBUG] mod_dptools.c:818 XXXXXXXXX SET
> > [bypass_media]=[true]
> > EXECUTE XXXXXXXXX
> > bridge([sip_contact_user=XXXXXX,sip_auth_username=,sip_auth_password=]sof
> >ia/cwu/XXXXXXX at XXXXXXX) 2010-01-12 22:27:02.612344 [DEBUG]
> > switch_ivr.c:1199 XXXXXXXX receive message [MEDIA]
> > 2010-01-12 22:27:02.612344 [DEBUG] switch_core_session.c:639 Send signal
> > XXXXXXXXXXXX [BREAK]
> >
> > 10 seconds later:
> >
> > 2010-01-12 22:27:12.614323 [NOTICE] switch_channel.c:613 New Channel
> > sofia/cwu/XXXXXXX at XXXXXXX [68cb4360-8da5-49fc-8fbd-f4a4e
> > 4351a94]
> >
> > On Wednesday 13 January 2010, Michael Collins wrote:
> >> Which rev of FreeSWITCH are you running? Also, collect a full debug
> >> trace of a working vs. non-working call so that you can compare what's
> >> happening. Put those on pastebin so others can have a look.
> >> -MC
> >>
> >> On Wed, Jan 13, 2010 at 4:46 AM, Sergey Okhapkin
> >>
> >> <sos at sokhapkin.dyndns.org>wrote:
> >>> To implement fallback to backup PSTN routes and do LCR my dialplan
> >>> executes the following commands:
> >>>
> >>> set execute_on_answer=set hangup_after_bridge=true
> >>> set some custom channel variables
> >>> bridge sofia/gateway1/number
> >>> set some custom channel variables
> >>> bridge sofia/gateway2/number
> >>> ....
> >>>
> >>> This generally works fine if gateway1 returns SIP error, dialplan sets
> >>> channel
> >>> variables to another values and calls gateway2 immediately. But if GW1
> >>> responds with 183 early media and SIP error after that, bad thing
> >>> happens - dialplan continues immediately, sets channel variables,
> >>> executes bridge application, but INVITE to GW2 is sent after 10 seconds
> >>> delay. Caller gets 10
> >>> extra seconds of post dial delay. I suspect the delay happens because
> >>> of attempt to read audio frame, switch_ivr_originate() has the
> >>> following lines:
> >>>
> >>> if (switch_channel_media_ready(caller_channel)) {
> >>>       tstatus = switch_core_session_read_frame()
> >>>     ...
> >>>
> >>> Any advice how to avoid extra delay in this situation?
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> >>>s http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From mouncifbb at gmail.com  Wed Jan 13 19:54:57 2010
From: mouncifbb at gmail.com (Mouncifbb)
Date: Wed, 13 Jan 2010 22:54:57 -0500
Subject: [Freeswitch-users] rtp-autoflush
In-Reply-To: <191c3a031001131933g2c290efds5b8342d1ce55ca0c@mail.gmail.com>
References: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>
	<191c3a031001131508i713ef7b8r48ba21c875e4e723@mail.gmail.com>
	<ade741d61001131520x1390ba23ud33dc4f3daba58f6@mail.gmail.com>
	<191c3a031001131530u6b105b35jcb09aa989a6ecce7@mail.gmail.com>
	<ade741d61001131846v5379fe23ua550e7293515e84d@mail.gmail.com>
	<191c3a031001131933g2c290efds5b8342d1ce55ca0c@mail.gmail.com>
Message-ID: <86CE8043-D1F0-4D75-8EE9-777AA6BBA27B@gmail.com>

Is the external.XML profile suited to be used for my scenario when  
bridging to the sip endpoint ?

Sent from my iPhone

On Jan 13, 2010, at 10:33 PM, Anthony Minessale <anthony.minessale at gmail.com 
 > wrote:

> Not if you set inbound-bypass-media = true in the sip profile or set  
> late-negotiaation = true and set bypass_media channel var to true  
> from the dp
>
>> On Jan 13, 2010 8:51 PM, "Mouncif Benniane" <mouncifbb at gmail.com>  
>> wrote:
>>
>> One more question, will it participate in codec negotiation sent by  
>> the inbound voip provider?
>> Right now I have it setup this way:
>>
>> Incoming Inbound DID (voip provider --> freeswitch(media bypass  
>> mode) ->  endpoint sip device.
>>
>> Thanks again!
>> On Wed, Jan 13, 2010 at 6:30 PM, Anthony Minessale <anthony.minessale at gmail.com 
>> > wrote: > > its be...
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From mouncifbb at gmail.com  Wed Jan 13 20:00:51 2010
From: mouncifbb at gmail.com (Mouncifbb)
Date: Wed, 13 Jan 2010 23:00:51 -0500
Subject: [Freeswitch-users] rtp-autoflush
In-Reply-To: <191c3a031001131933g2c290efds5b8342d1ce55ca0c@mail.gmail.com>
References: <ade741d61001131500v7555e06fj913d4a09bd21a7a1@mail.gmail.com>
	<191c3a031001131508i713ef7b8r48ba21c875e4e723@mail.gmail.com>
	<ade741d61001131520x1390ba23ud33dc4f3daba58f6@mail.gmail.com>
	<191c3a031001131530u6b105b35jcb09aa989a6ecce7@mail.gmail.com>
	<ade741d61001131846v5379fe23ua550e7293515e84d@mail.gmail.com>
	<191c3a031001131933g2c290efds5b8342d1ce55ca0c@mail.gmail.com>
Message-ID: <76CD79BB-64BD-4C2F-B97E-F80DC74EAB4E@gmail.com>

I forgot to add the sip end point device is not local it's another sip  
proxy on public IP who is suppose to trust FS invite without auth

Thanks

Sent from my iPhone

On Jan 13, 2010, at 10:33 PM, Anthony Minessale <anthony.minessale at gmail.com 
 > wrote:

> Not if you set inbound-bypass-media = true in the sip profile or set  
> late-negotiaation = true and set bypass_media channel var to true  
> from the dp
>
>> On Jan 13, 2010 8:51 PM, "Mouncif Benniane" <mouncifbb at gmail.com>  
>> wrote:
>>
>> One more question, will it participate in codec negotiation sent by  
>> the inbound voip provider?
>> Right now I have it setup this way:
>>
>> Incoming Inbound DID (voip provider --> freeswitch(media bypass  
>> mode) ->  endpoint sip device.
>>
>> Thanks again!
>> On Wed, Jan 13, 2010 at 6:30 PM, Anthony Minessale <anthony.minessale at gmail.com 
>> > wrote: > > its be...
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From sos at sokhapkin.dyndns.org  Wed Jan 13 20:08:03 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 23:08:03 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <6D061C37-43A6-48D1-A8E3-549FC3E8679E@freeswitch.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132229.58830.sos@sokhapkin.dyndns.org>
	<6D061C37-43A6-48D1-A8E3-549FC3E8679E@freeswitch.org>
Message-ID: <201001132308.03170.sos@sokhapkin.dyndns.org>

Critical issues are when SIP error come after 18X provisional response.

- if bypass_media is false then dialplan stops and leg a is explicitly hang up 
(switch_ivr_bridge.c, line 513).
- if bypass_media is true, then dialplan continue, but there is 10 seconds 
delay before next bridge application sends INVITE to gateway ( 
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024354.html ) 
I didn't track down yet why this happens looking at FS sources.

Why I didn't open a bug on jira? Because FS behaves according to the design 
and specs :-) But not according to real world requirements... 


On Wednesday 13 January 2010, Brian West wrote:
> Can you elaborate on these "Critical" issues you seem to be having?  Why
> aren't you opening a jira for them if they are that critical to your needs?
>
> /b
>
> On Jan 13, 2010, at 9:29 PM, Sergey Okhapkin wrote:
> > If you wish FreeSWITCH to be well adopted by the community, then FS
> > should follow the real world "rules" but not specs...
> >
> > Don't you agree that 99% of SIP servers are set up to interconnect
> > with "buggy" PSTN and should follow PSTN rules, but not SIP specs?
> >
> > Some background - I did run asterisk for years, but switched to FS
> > recently because of critical asterisk problems with SIP handling when
> > asterisk is not in the media path. I spend a lot of time porting the
> > billing system to FS. And I did it. What I got? Critical problems with FS
> > early media handling. Hopefully I can switch back to asterisk if FS
> > problems with early media will begin to draw customers away.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From sos at sokhapkin.dyndns.org  Wed Jan 13 20:11:06 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 23:11:06 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <464CD1B0-95D0-4D6D-B330-87B059FB76A6@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132229.58830.sos@sokhapkin.dyndns.org>
	<464CD1B0-95D0-4D6D-B330-87B059FB76A6@jerris.com>
Message-ID: <201001132311.06257.sos@sokhapkin.dyndns.org>

Just "10 seconds" could easily draw away customer's $$$. And seems like the 
process already began... :-(

On Wednesday 13 January 2010, Michael Jerris wrote:
> What exactly are your problems?   Is this just the "10 second" thread you
> posted today?
>
> Mike
>
> On Jan 13, 2010, at 10:29 PM, Sergey Okhapkin wrote:
> > If you wish FreeSWITCH to be well adopted by the community, then FS
> > should follow the real world "rules" but not specs...
> >
> > Don't you agree that 99% of SIP servers are set up to interconnect
> > with "buggy" PSTN and should follow PSTN rules, but not SIP specs?
> >
> > Some background - I did run asterisk for years, but switched to FS
> > recently because of critical asterisk problems with SIP handling when
> > asterisk is not in the media path. I spend a lot of time porting the
> > billing system to FS. And I did it. What I got? Critical problems with FS
> > early media handling. Hopefully I can switch back to asterisk if FS
> > problems with early media will begin to draw customers away.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From brian at freeswitch.org  Wed Jan 13 20:13:30 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Jan 2010 22:13:30 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001132308.03170.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132229.58830.sos@sokhapkin.dyndns.org>
	<6D061C37-43A6-48D1-A8E3-549FC3E8679E@freeswitch.org>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
Message-ID: <5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>

This is exactly why you open a jira.  But we have zero control over the far end sending a 18X then an error.  That is NOT something we can fix. As for the others I suspect they have been fixed... what Rev are you running?

/b

On Jan 13, 2010, at 10:08 PM, Sergey Okhapkin wrote:

> Critical issues are when SIP error come after 18X provisional response.
> 
> - if bypass_media is false then dialplan stops and leg a is explicitly hang up 
> (switch_ivr_bridge.c, line 513).
> - if bypass_media is true, then dialplan continue, but there is 10 seconds 
> delay before next bridge application sends INVITE to gateway ( 
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024354.html ) 
> I didn't track down yet why this happens looking at FS sources.
> 
> Why I didn't open a bug on jira? Because FS behaves according to the design 
> and specs :-) But not according to real world requirements... 

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From sos at sokhapkin.dyndns.org  Wed Jan 13 20:26:41 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 13 Jan 2010 23:26:41 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
Message-ID: <201001132326.41354.sos@sokhapkin.dyndns.org>

I run FreeSWITCH Version 1.0.5pre10 (16012M), sorry but I don't feel 
comfortable running SVN trunk on production servers... Perhaps because of my 
experience with asterisk. Neither version after 1.4.18 did work OK to me.

On Wednesday 13 January 2010, Brian West wrote:
> This is exactly why you open a jira.  But we have zero control over the far
> end sending a 18X then an error.  That is NOT something we can fix. As for
> the others I suspect they have been fixed... what Rev are you running?
>
> /b
>
> On Jan 13, 2010, at 10:08 PM, Sergey Okhapkin wrote:
> > Critical issues are when SIP error come after 18X provisional response.
> >
> > - if bypass_media is false then dialplan stops and leg a is explicitly
> > hang up (switch_ivr_bridge.c, line 513).
> > - if bypass_media is true, then dialplan continue, but there is 10
> > seconds delay before next bridge application sends INVITE to gateway (
> > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
> >4.html ) I didn't track down yet why this happens looking at FS sources.
> >
> > Why I didn't open a bug on jira? Because FS behaves according to the
> > design and specs :-) But not according to real world requirements...





From dave at 3c.co.uk  Wed Jan 13 20:28:18 2010
From: dave at 3c.co.uk (David Knell)
Date: Thu, 14 Jan 2010 04:28:18 +0000
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <26079A18-5968-4E1E-A276-9DE889E83FA4@freeswitch.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<9853f4ff1001131810y669bb7e3uee6b9937f59e8215@mail.gmail.com>
	<C3D4ADB0-7A80-45A9-91E1-FFABB987851B@freeswitch.org>
	<201001132205.43306.sos@sokhapkin.dyndns.org>
	<7B165AA4-4F8F-488F-950A-D8B2556D4E57@freeswitch.org>
	<1263439925.11216.36.camel@local.freepabx.com>
	<26079A18-5968-4E1E-A276-9DE889E83FA4@freeswitch.org>
Message-ID: <1263443298.11216.45.camel@local.freepabx.com>

Well, surely if it would work, then it's feasible and there's no reason
why it shouldn't happen.  Whether or not a solution will achieve the
desired end result in an optimal fashion is - for most people -
subordinate to getting their problem solved.

Don't forget the two rules of optimisation:
1.  Don't.
2.  (for experts only) Don't yet.

--Dave

> That would work but its sub optimal.  What he wants to accomplish is not do able and not going to ever happen at this rate.
> 
> /b
> 
> On Jan 13, 2010, at 9:32 PM, David Knell wrote:
> 
> > What would happen if several outdials were made from a conference,
> > rather than using a forked dial?
> > 
> > --Dave
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From mike at jerris.com  Wed Jan 13 20:47:53 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 23:47:53 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001132308.03170.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132229.58830.sos@sokhapkin.dyndns.org>
	<6D061C37-43A6-48D1-A8E3-549FC3E8679E@freeswitch.org>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
Message-ID: <632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>


On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:

> Critical issues are when SIP error come after 18X provisional response.
> 
> - if bypass_media is false then dialplan stops and leg a is explicitly hang up 
> (switch_ivr_bridge.c, line 513).

 behavior can be modified with continue_on_fail and hangup_after_bridge channel vars, perhaps ignore_early_media as well

> - if bypass_media is true, then dialplan continue, but there is 10 seconds 
> delay before next bridge application sends INVITE to gateway ( 
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024354.html ) 
> I didn't track down yet why this happens looking at FS sources.

see response on that thread

> 
> Why I didn't open a bug on jira? Because FS behaves according to the design 
> and specs :-) But not according to real world requirements... 


Really, people are trying to help you and your going to be snarky in response?

> 
> 
> On Wednesday 13 January 2010, Brian West wrote:
>> Can you elaborate on these "Critical" issues you seem to be having?  Why
>> aren't you opening a jira for them if they are that critical to your needs?


Mike




From mike at jerris.com  Wed Jan 13 20:48:34 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Jan 2010 23:48:34 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001132326.41354.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
	<201001132326.41354.sos@sokhapkin.dyndns.org>
Message-ID: <52E8BE9E-B6D6-411C-9080-697AF24017B4@jerris.com>

You already are running on trunk.

Mike

On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote:

> I run FreeSWITCH Version 1.0.5pre10 (16012M), sorry but I don't feel 
> comfortable running SVN trunk on production servers... Perhaps because of my 
> experience with asterisk. Neither version after 1.4.18 did work OK to me.
> 
> On Wednesday 13 January 2010, Brian West wrote:
>> This is exactly why you open a jira.  But we have zero control over the far
>> end sending a 18X then an error.  That is NOT something we can fix. As for
>> the others I suspect they have been fixed... what Rev are you running?
>> 
>> /b




From anthony.minessale at gmail.com  Wed Jan 13 21:15:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Jan 2010 23:15:38 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001132114s18ed2baw2ff802dbd615fe01@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
	<201001132326.41354.sos@sokhapkin.dyndns.org>
	<52E8BE9E-B6D6-411C-9080-697AF24017B4@jerris.com>
	<191c3a031001132114s18ed2baw2ff802dbd615fe01@mail.gmail.com>
Message-ID: <191c3a031001132115u2d88bab7p7cbb27ab1eaad466@mail.gmail.com>

Perhaps best not to help him anymore without an apology for the snap
judgement and comparison to asterisk clearly designed to push our buttons.
We'll be here when you realize we were trying to help you but I can't
promise we will still have paitence......

On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike at jerris.com> wrote:

You already are running on trunk.

Mike

On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote: > I run FreeSWITCH
Version 1.0.5pre10 (16012M...

_______________________________________________ FreeSWITCH-users mailing
list FreeSWITCH-users at lists...
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From lakindia89 at gmail.com  Wed Jan 13 23:57:54 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 14 Jan 2010 13:27:54 +0530
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>
References: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
	<191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>
Message-ID: <7d79b3931001132357i36bbb482jdf2bbdd3aea2a583@mail.gmail.com>

I taught the signal handler will be inherited by the child process. It also
does like that.
After making a call, If I press ctrl + c, the above program printed
PARENT PID: Signal SIGINT is generated
CHILD PID: Signal SIGINT is generated.

So I think the sigal handlers will be inherited to the child.
Anyway I'll also try registering signal handlers in child also, and then
I'll come back with that result.

Thanks....
On Wed, Jan 13, 2010 at 9:48 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you would have to register signals in your child process too
>
>   On Wed, Jan 13, 2010 at 3:13 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>>  Hi all,
>>
>> I've done a sample program (In perl ESL) , which play a file to the caller
>> and then it will call recvEvent() and print the event name. I've handled
>> signals also.
>>
>> When I send SIGINT to my program (Perl), the signal handler is called and
>> I can see the print output. But in the same time,  I received
>> SERVER_DISCONNECTED from freeswitch as event.
>>
>> I don't know why I received SERVER_DISCONNECTED from freeswitch. Is it
>> because, the recvEvent() from perl internally  calls the recvevent function
>> in the Esl.c and when it waits to receive the information from socket, the
>> signal occurred???
>>
>> Please clarify me!!
>>
>> Here is my program
>> require ESL;
>> use IO::Socket::INET;
>> use Data::Dumper;
>>
>> my $ip = "192.168.1.222";
>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
>> Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>> die "Could not create socket: $!\n" unless $sock;
>> &register_Signals();
>>
>> for(;;) {
>>     my $new_sock = $sock->accept();
>>     next if (not defined ($new_sock));
>>     my $pid = fork();
>>     if ($pid) {
>>         close($new_sock);
>>         next;
>>     }
>>     print "CHILD PID: $$\n";
>>     my $host = $new_sock->sockhost();
>>     my $fd = fileno($new_sock);
>>
>>     my $con = new ESL::ESLconnection($fd);
>>     my $info = $con->getInfo();
>>
>>     my $uuid = $info->getHeader("unique-id");
>>
>>     printf "Connected call %s, from %s\n", $uuid,
>> $info->getHeader("caller-caller-id-number");
>>     my $r=$con->execute("answer");
>>     print Dumper $r;
>>     $con->events("plain","all");
>>     my
>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>
>>     while($con->connected()) {
>>         my $e = $con->recvEvent();
>>
>>         if ($e) {
>>             my $name = $e->getHeader("event-name");
>>             print "EVENT [$name]\n";
>>             if ($name eq "DTMF") {
>>                 my $digit = $e->getHeader("dtmf-digit");
>>                 print "$digit\n";
>>             }
>>         }
>>     }
>>     close($new_sock);
>> }
>> sub register_Signals() {
>>     foreach ( keys %SIG ) {
>>         $SIG{$_} = 'sig_Handler';
>>     }
>> }
>>
>> sub sig_Handler() {
>>     my $handle=$_[0];
>>     if($handle eq "INT") {
>>         print "$$: SIGNAL SIG$handle is generated\n";
>>     }
>> }
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Thu Jan 14 00:02:32 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 14 Jan 2010 13:32:32 +0530
Subject: [Freeswitch-users] Sangoma PRI installation for FreeSWITCH
In-Reply-To: <B4A27408A33F4E74A4DBEA492B906C20@greyhawk.tonecommander.com>
References: <369c72d81001122021o6885913cl618965791aec4621@mail.gmail.com>
	<B4A27408A33F4E74A4DBEA492B906C20@greyhawk.tonecommander.com>
Message-ID: <7d79b3931001140002w638d2c0bn2f1d9b6f9b51bd1d@mail.gmail.com>

Thanks Jerry, for making the wancfg_fs configuration available.

On Wed, Jan 13, 2010 at 9:45 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  Here are my instructions to install the wanpipe driver and I have not had
> your problem:
>
>
> 1)       Download the following Wanpipe driver tarball
> wanpipe-<version>.tgz
>
> 2)       Store the wanpipe-<version>.tgz file in the /opt folder.
>
> 3)       tar xvfz wanpipe-<version>.tgz
>
> 4)       cd wanpipe-<version>
>
> 5)       make openzap
>
> 6)       make install
>
> 7)       make install_pri
>
> 8)       wanrouter hwprobe     // confirms successful wanpipe installation
>
> 9)       /usr/sbin/wancfg_fs   // starts wanpipe configuration utility
>
> 10)   1=NO            // Change Freeswitch Configuration Directory
>
> 11)   1=T1            // Select Media Type
>
> 12)   1=YES (keep)    // Configure Port 1 T1, B8ZS, ESF
>
> 13)   1=NORMAL        // Select Clock
>
> 14)   1=NATIONAL      // Select Switchtype
>
> 15)   1=CPE           // Select Signaling Type
>
> 16)   1               // Input Group #
>
> 17)   1=YES           // Enable H/W DTMF Detect
>
> 18)   1=YES           // Enable FAX Detect
>
> 19)   1=YES=CONTINUE  // Configuration Complete
>
> 20)   1=YES           // Save Configuration
>
> 21)   1=YES           // Wanrouter start on reboot
>
> 22)   1=YES           // smg_ctrl start/stop on wanrouter start
>
> 23)   Done.
>
> Jerry
>
>  ------------------------------
> *From:* Magesh R [mailto:magesh.freeswitch at gmail.com]
> *Sent:* Tuesday, January 12, 2010 8:21 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Sangoma PRI installation for FreeSWITCH
>
> Dear All,
>          I have installed Sangoma PRI card in machine with following steps,
>                  * wget
> ftp://ftp.sangoma.com/linux/custom/3.5/wanpipe-3.5.8.7.tgz
>                  * tar -xvfz wanpipe-3.5.8.7.tgz
>                  * cd wanpipe-3.5.8.7
>                  * make openzap
>                  * make install
>                  * make install_pri
>
>            I have executed "wanrouter hwprobe" command it prints the
> following details,
>
>            1 . AFT-A102-SH : SLOT=4 : BUS=3 : IRQ=11 : CPU=A : PORT=1 :
> HWEC=0 : V=36
>            2 . AFT-A102-SH : SLOT=4 : BUS=3 : IRQ=11 : CPU=A : PORT=2 :
> HWEC=0 : V=36
>
>         Card Cnt: A101-2=1
>
>        Next I have executed wancfg_fs script to configure the sangoma for
> freeswitch.
> It creates the following configuration files
>                      * wanpipe1.conf
>                      * wanpipe2.conf
>                      * smg_prid.conf
>                      * openzap.conf
>                      * openzap.conf.xml
>        I have attached those files.
>        I have started the wanrouter and printed the wanrouter status,
>               Wanrouter Status:
>
> Device name | Protocol | Station | Status        |
> wanpipe1    | AFT TE1  | N/A     | Connected     |
> wanpipe2    | AFT TE1  | N/A     | Disconnected  |
>
>
>       Next I have started the smg_ctrl, but it failed to start. It prints
> the following things,
> smg_ctrl start
>
> Starting processes...
> Loading SCTP...OK
> Starting sangoma_prid...OK
> sangoma_prid failed to start
> check /var/log/sangoma_mgd.log for errors
>
> Stopping running processes...
> safe_sangoma not running...
> sangoma_prid is stopped
> Removing PID files...done
>
> I have checked /var/log/sangoma_mgd.log file. But nothing was there.
>
> Could any please tell me where I made mistake?
>
> Thanks,
> Mag
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kawarod at laposte.net  Thu Jan 14 00:17:50 2010
From: kawarod at laposte.net (rod)
Date: Thu, 14 Jan 2010 12:17:50 +0400
Subject: [Freeswitch-users] Eavesdrop in LUA
Message-ID: <4B4ED32E.30706@laposte.net>

Hi all,

I'm trying to do this in LUA:
     A call B

and I'd like to setup a new call to C with eavesdrop of A conversation 
with B.

I have no idea how to do this if someone can help.
I switched to LUA cause I see no way to achieve this with dialplan 
(snippets are welcome).

regards,
rod



From devel at thom.fr.eu.org  Thu Jan 14 00:23:09 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Thu, 14 Jan 2010 09:23:09 +0100
Subject: [Freeswitch-users]
 =?utf-8?q?playback_and_play=5Fand=5Fget=5Fdigi?=
 =?utf-8?q?ts_strange_misunderstanding?=
Message-ID: <6d698be09281bb173cf0de33c1c2e44b@thom.fr.eu.org>

 

So I could (kind of) solve this by myself. There is in vars.xml the
variable ${sound_prefix}. I did set it properly to my french sound path and
then it worked. 

However, for the sake of discussion, I did try this with
1.0.5pre8 and the result was different : 

with the extension 

FS was
trying to play the file
${FREESWITCH_PATH}/sounds/en/us/callie/misc/ringing_disabled.wav 

with the
extension 

FS was trying to load
${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/ringing_disabled.wav 

Now with
the latest snapshot, with either one of the 2 mentionned extensions, FS
tries to play the file in
${sound_prefix}/filepath/${codec_bit_rate}/filename whereas 1.0.5pre8 did
not add the codec_bit_rate in the path but took care of the
default_language variable. 

Fran?ois 

On Wed, 13 Jan 2010 16:32:16 +0100,
Fran?ois Legal wrote:  

Hello, 

trying to make so dialplan extensions
that use the playback and play_and_get_digits applications, but I'm having
trouble with the file name specification. 

The files I want to play are in
the french language (fr/fr/julie as configured in lang/fr/fr.xml) 

My
extension is as follows : 

The channel is using a bit rate of 8000 Hz, so
by the set default_language=fr I would expect freeswitch to playback the
file at
${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/8000/ringing_disabled.wav
whereas it tries to playback the file at
${FREESWITCH_PATH}/sounds/en/us/callie/misc/8000/ringing_disabled.wav 

I
have the same with play_and_get_digits application. 

What am I doing wrong
? 

Fran?ois  

 
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From kond at nstel.ru  Thu Jan 14 00:26:40 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Thu, 14 Jan 2010 11:26:40 +0300
Subject: [Freeswitch-users] sip trunk question: why call throughexternal
	profile is challenged?
In-Reply-To: <98955F81-E9C1-4926-A648-AD49FB9D38A4@jerris.com>
Message-ID: <20100114082640.2C5D311F32@mail.nstel.ru>

Mike, thanks for the reply.

 

Mmm. looks like I need more detailed instructions where to dig.

Is there a way to turn off "challenging" completely?

I thought that <param name="auth-calls" value="false"/> should do it, but
alas.

By the way should this parameter be visible in either "sofia status profile
external" or "sofia status gateway sipx4.lab.nstel.ru" ? I don't see it.

 

I attached traces of failed and successful calls.

 

Thanks and regards,

Nikolay. 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Wednesday, January 13, 2010 8:30 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] sip trunk question: why call throughexternal
profile is challenged?

 

Look at how sipx sets up the users when they build the extensions and such
for conferences, there was something special here, but I can't recall what.

 

Mike

 

On Jan 13, 2010, at 9:10 AM, Nikolay Kondratyev wrote:





Hi all!

 

I'm brand new to FreeSwitch, but have some experience with SipX.

Our company is evaluating FS.

For test purposes I set up FS on a virtual machine (vmware esxi). I use
CentOS.

The FS version I use is 1.0.5-20100110-0400.

 

I have a question regarding sip trunk between FS and SipX.

I created the following GW in external profile: 

[freeswitch at freeswitch external]$ cat sipx-lab.xml | grep -v '<!--'

<include>

  <gateway name="sipx4.lab.nstel.ru">

  <param name="username" value="zxcv"/>

  <param name="password" value="2007"/>

  <param name="register" value="false"/>

  </gateway>

</include>

 

External.xml file is not modified after installation.

 

I see this gateway via fs_cli:

freeswitch at internal> sofia status gateway sipx4.lab.nstel.ru

============================================================================
=====================

Name            sipx4.lab.nstel.ru

Profile         external

Scheme          Digest

Realm           sipx4.lab.nstel.ru

Username        zxcv

Password        yes

>From            <sip:zxcv at sipx4.lab.nstel.ru;transport=udp>

Contact
<sip:gw+sipx4.lab.nstel.ru at 172.23.22.49:5080;transport=udp;gw=sipx4.lab.nste
l.ru>

Exten           zxcv

To              sip:zxcv at sipx4.lab.nstel.ru

Proxy           sip:sipx4.lab.nstel.ru

Context         public

Expires         3600

Freq            3600

Ping            0

PingFreq        0

PingState       0/0/0

State           NOREG

Status          UP

CallsIN         0

CallsOUT        0

============================================================================
=====================

 

I created new FS extension 2853. I registers (xlite) and I can call it.

 

Now I want to call FS user from sipx.

 

>From the sipx side one can configure link to FS differently. There are two
options:

1. Call FS directly trough sipxproxy (the core part of sipx works as sip
proxy, not as b2bua)

2. Call trough embedded b2bua, named sipxbridge.

 

When a call is going trough sipxbridge, it is successfully landed at FS
extension 2853.

But when a call is going from the sipxproxy, it is challenged with Status
407 "proxy authentication required', and then call fails.

(I'm not sure if sipx should handle this challenge, but this is separate
question for the sipx forum).

At the default log level I see nothing in the freeswitch.log.

 

So the question is why one call through external profile is being challenged
while the other is not?

I suspect that the reason is in the difference in the two Invite messages:

 

Here is challenged Invite:

INVITE sip:2853 at fs.lab.nstel.ru:5080;transport=udp SIP/2.0

Record-Route:
<sip:172.23.12.104:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EOTRiZjI3MjgtYWMxN
zBjZTktMTNjNC0yZjZkZWMtNjNkNmUwN2UtMmY2ZGVj.900_ntap%2Aid%7EMjIyNjAtMQ%60%60
%213ba6d85f6946c4c6001bee1d3b54474f>

From:
"testphone3857"<sip:3857 at lab.nstel.ru>;tag=94bf2728-ac170ce9-13c4-2f6dec-63d
6e07e-2f6dec

To: <sip:2853 at lab.nstel.ru>

Call-Id: 94bed2a0-ac170ce9-13c4-2f6dec-68ffdb81-2f6dec at lab.nstel.ru

Cseq: 2 INVITE

Via: SIP/2.0/UDP
172.23.12.104;branch=z9hG4bK-sipXecs-000eaf48a36fe4029c7cde004a6f44424847

Via: SIP/2.0/TCP
172.23.12.104;branch=z9hG4bK-sipXecs-000bf3008cfa3f3e70470789c75232ba9499~ac
7fd729330fd563f83cacd941311e75;id=22260-1

Via: SIP/2.0/UDP 172.23.12.233:5060;branch=z9hG4bK-2f6dec-b9456412-7ad07784

Max-Forwards: 18

Supported: replaces

User-Agent: LG-Nortel LIP 6812 v1.2.38sp SN/00405A18B634

Contact: <sip:3857 at 172.23.12.233:5060;x-sipX-nonat>

Proxy-Authorization: Digest
username="3857",realm="lab.nstel.ru",nonce="e2152722611af1bbe59f3a8eb31b8eb8
4b4db2ca",uri="sip:2853 at lab.nstel.ru",response="a09c34fbd897783c9b9af50bd044
ccaa",algorithm=MD5

Content-Type: application/sdp

Content-Length: 301

Date: Wed, 13 Jan 2010 11:47:22 GMT

Expires: 60

X-Sipx-Handled: X172.23.12.104-81.211.30.104

 

v=0

o=LGEIPP 16246 16247 IN IP4 172.23.12.233

s=SIP Call

c=IN IP4 172.23.12.233

t=0 0

m=audio 23008 RTP/AVP 0 8 18 4 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:18 annexb=no

a=fmtp:101 0-11

a=sendrecv

 

 

And here is successful Invite:

INVITE sip:2853 at fs.lab.nstel.ru;user=phone SIP/2.0

Call-ID: 94bec3b8-ac170ce9-13c4-2f6d47-2fee09d0-2f6d47 at lab.nstel.ru.0

CSeq: 1 INVITE

From: "testphone3857" <sip:3857 at 172.23.12.104>;tag=816640414244159961

To: <sip:2853 at fs.lab.nstel.ru;user=phone>

Via: SIP/2.0/UDP
172.23.12.104:5080;branch=z9hG4bK885dde2c8680f5315845cd3350b8b605373534

Max-Forwards: 70

User-Agent: sipXecs/4.0.2 sipXecs/sipxbridge (Linux)

P-Asserted-Identity: <sip:1004 at 172.23.12.104>

Contact: <sip:1004 at 172.23.12.104:5080;transport=udp>

Route: <sip:172.23.22.49:5080;transport=udp;lr>

Session-Expires: 1800;refresher=uac

Allow: INVITE,BYE,ACK,CANCEL,OPTIONS

Content-Type: application/sdp

Content-Length: 315

 

v=0

o=sipxbridge 6640787141824452741 1 IN IP4 172.23.12.104

s=SIP Call

c=IN IP4 172.23.12.104

t=0 0

m=audio 30248 RTP/AVP 0 8 18 4 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:18 annexb=no

a=fmtp:101 0-11

a=sendrecv

 

Can somebody please if it is a FS configuration problem or a software
problem or is it a problem on the sipx side?

 

Thanks in advance,

Nikolay.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From john_re at fastmail.us  Thu Jan 14 01:06:52 2010
From: john_re at fastmail.us (giovanni_re)
Date: Thu, 14 Jan 2010 01:06:52 -0800
Subject: [Freeswitch-users] Attend CampKDE Jan 15-18 via Voice over Internet
	(VOIP), BerkeleyTIP
Message-ID: <1263460012.26773.1354547409@webmail.messagingengine.com>

This weekend is CampKDE in San Diego, California, USA.  For those who
would like to participate, but can't physically attend, the BerkeleyTIP
group will (unofficially, as a public service) provide assistance to
everyone who would like to communicate via voice about &/or with the
conference, & other KDE community members, globally.  [VOIP
communication is typically free (no money $ cost) to people with
internet access.]

Join #berkeleytip on irc.freenode.net, & we will do our best (depending
on availability of BTIP people) to help people get on VOIP in order to
talk about KDE, or whatever.

You are also invited to join the BerkeleyTIP mailing list, & ask any
help or other questions there.
http://groups.google.com/group/BerkTIPGlobal

Here's the announcement there:
CampKDE Jan 15-18 SanDiego, VOIP & Ride Share; BerkeleyTIP  
http://groups.google.com/group/BerkTIPGlobal/t/53cbf00b987b66e0

I'll be announcing this on some KDE related mailing lists.

When (if) I get time, I'll locate the CampKDE organizers, & email them
telling them BTIP is helping KDE community members with this
unofficially, & asking them if they'd like to try to do something
officially in arranging global VOIP for the KDE community interested in
CampKDE.

Here is the VOIP HowTo page for BerkeleyTIP:
http://sites.google.com/site/berkeleytip/remote-attendance

There are 2 ways to join the VOIP conference: 1) Computer VOIP globally
using a sw client & headset & mic, & 2) Dial in with land or cell phone
to a phone # in Washington state, USA (free, no $ money cost, if you
have free long distance phone service).

BerkeleyTIP:
http://sites.google.com/site/berkeleytip/

=====
So, join on the #berkeleytip freenode irc channel, or the mailing list,
& we'll do our best to help you get on the global conference to discuss
KDE via voice communication. :)



From devel at thom.fr.eu.org  Thu Jan 14 01:08:28 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Thu, 14 Jan 2010 10:08:28 +0100
Subject: [Freeswitch-users] Issue with openzap and DTMF
Message-ID: <2bf9ca9a45058c72a1f8d13070cb582e@thom.fr.eu.org>



Hello, 

yesterday, while trying some new extensions with
play_and_get_dtmf, I noticed a strange behaviour of freeswitch. 

The
configuration is as follows : 

An FXS port is bridged to an FXO (say FXO1)
port for calling (through my carrier) another FXO (say FXO2) port. The call
incoming on FXO2 reaches some extension which does answer then
play_and_get_dtmf (or read). So I dial the digits I want, and for each
dialed digit, I get it twice in the resultant variable. 

Now if I reach
the extension by calling in from a cell phone, I get exactly the digits I
dialed. 

This seems to me as a problem in openzap. 

I just wanted to
report. 

Fran?ois
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From yehavi.bourvine at gmail.com  Thu Jan 14 01:14:58 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 14 Jan 2010 11:14:58 +0200
Subject: [Freeswitch-users] How to build FreeSwitch with non-default SSL
	library?
In-Reply-To: <d361ca7b1001131132o1043cce4p31af0f95f804bd7b@mail.gmail.com>
References: <d361ca7b1001131132o1043cce4p31af0f95f804bd7b@mail.gmail.com>
Message-ID: <d361ca7b1001140114s53c25d28h2cbf496440833941@mail.gmail.com>

Answwring myself:  "make distclean", set LDFLAGS=-L/path-to-ssl-lib and then
configure and make.

                       __Yehavi:

2010/1/13 Yehavi Bourvine <yehavi.bourvine at gmail.com>

>  Hello,
>
>   I suspect that the Openssl library used on fedora is a modified one
> and has some interoperability problems with phones. I would like to build
> freeswitch with a library from openssl.org which is installed
> under /usr/local. Is there an easy way to tell  "configure" script to use
> that version instead?
>
>                                  Thanks, __Yehavi:
>
>
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From jingwei.yang at gmail.com  Thu Jan 14 01:24:04 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 14 Jan 2010 17:24:04 +0800
Subject: [Freeswitch-users]  Not able to capture the custom event
Message-ID: <13529f9d1001140124t42e9dfe5u3e70a6976aa032a8@mail.gmail.com>

Hi Guys,

I'm testing to see whether a custom event can be triggered and captured.
Here's the extension:

    <extension name="test_ivr">
      <condition field="destination_number" expression="^777$">
        <action application="event" data="Event-Name=*CUSTOM*
,Event-Subclass=myevent::ACDnotify"/>
        <action application="transfer" data="9999 XML default" />
      </condition>
    </extension>

If I telnet and use "events plain all", I can get the event. However, if I
use "events plain CUSTOM" or "event plain CUSTOM", it doesn't get captured.
Is there anything wrong with the command?

Regards,
-Jingwei
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From david.villasmil.work at gmail.com  Thu Jan 14 02:26:45 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 14 Jan 2010 11:26:45 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001132115u2d88bab7p7cbb27ab1eaad466@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
	<201001132326.41354.sos@sokhapkin.dyndns.org>
	<52E8BE9E-B6D6-411C-9080-697AF24017B4@jerris.com>
	<191c3a031001132114s18ed2baw2ff802dbd615fe01@mail.gmail.com>
	<191c3a031001132115u2d88bab7p7cbb27ab1eaad466@mail.gmail.com>
Message-ID: <9853f4ff1001140226y2b035f37rfcafb03190435ec1@mail.gmail.com>

Hello again,

Using ingore early media only ignores ALL media, that's not what I need. At
least in europe i've seen it many times, MNOs provide a service with which
you can have a song played to the caller while the call is connecting to
your cell phone. This is basically what I'm trying to achieve.

The content provider is in possesions of the media and it require us to fork
the call and send a SIP INVITE to the on one leg and the call to the
destination number on another leg. they will only provide a progress, no
answer.

I CAN do it locally by playing the file as a custom ringback but that's not
the standard in terms of commercial use of the content.

Is there any way to modify the behaviour of fs when i receives media? Let's
say i.e. it doesn't drop the other leg, but provides the first early media
it receives and just wait for some channel to answer? I will not have both
media but it would work.

thanks

David

On Thu, Jan 14, 2010 at 6:15 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Perhaps best not to help him anymore without an apology for the snap
> judgement and comparison to asterisk clearly designed to push our buttons.
> We'll be here when you realize we were trying to help you but I can't
> promise we will still have paitence......
>
> On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike at jerris.com> wrote:
>
> You already are running on trunk.
>
> Mike
>
> On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote: > I run FreeSWITCH
> Version 1.0.5pre10 (16012M...
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Claudio.Cavalera at italtel.it  Thu Jan 14 03:23:05 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Thu, 14 Jan 2010 12:23:05 +0100
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <5d2828f1001131130s6ef166c0wc98f70023420c0fd@mail.gmail.com>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F362E8@BESONE.corp.dom>

Hi Mike,
in fact i've completed that page with the list of available  session
functions.
 
I've not removed "session:originate" yet, but it would be better if
someone could provide an example in order to write an example in the
wiki.
I've added this valuable example also with the help of rupa
http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.Session
 
I would like to write something also about api_on_answer to use an api
instead of a dialplan application.
 
BRs,
Claudio


________________________________

	From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike
van Lammeren
	Sent: Wednesday, January 13, 2010 8:30 PM
	To: freeswitch-users at lists.freeswitch.org
	Subject: Re: [Freeswitch-users] playing with sessions in lua
	
	
	Hello! 

	Before you remove "session:originate" from the wiki, you should
take a look at this:
	
http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocument
ed_Session_Functions.3F

	
<http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocumen
ted_Session_Functions.3F> There is, in fact, a function called
"originate".

	Mike van Lammeren


	On Wed, Jan 13, 2010 at 5:22 AM, Cavalera Claudio Luigi
<Claudio.Cavalera at italtel.it> wrote:
	

		Thanks a lot Anthony,
		some comments inline (and please forgive me for my
broken email client).
		 
		
		>>  example1: Consider this simple lua script in which i
create two sessions:
		
		 >>  api = freeswitch.API();
		 >>  api:execute("originate",
"sofia/internal/1001%192.168.1.1 &park");
		 >>  api:execute("originate",
"sofia/internal/1002%192.168.1.1 &park");
		
		 >  capture the output from api:execute the uuid is in
there
		
		
		Thx a lot,
		this was one piece i was missing although it's already
on the wiki here:
	
http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls 
		


		 > because lua calls it freeswitch.bridge
		
		 > session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
		 > session2 =
freeswitch.Session("sofia/internal/1002%192.168.1.1", session1);
		 > freeswitch.bridge(session1, session2); 
		 
		good to now, there isn't any example of
freeswitch.bridge in the wiki and i would like to add one.
		Where I could find the full api of
		freeswitch.Session( ) ?
		because I've seen this working also without "session1"
in the second line:
		
		session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
		
		session2 =
freeswitch.Session("sofia/internal/1002%192.168.1.1");
		freeswitch.bridge(session1, session2); 
		
		 also is there any difference between  freeswitch.bridge
and freeswitch.execute(uuid_bridge ...) ?
		 
		 >> example3: yet another possibility
		 >> local session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
		 >> session1:originate(session2,
"sofia/internal/1002%192.168.1.1", 1000);
		 >> but it does not work either.
		
		 > The above is gibberish try:
		 > local session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
		 > session1:execute("bridge",
"sofia/internal/1002%192.168.1.1"); 
		 
		Okay i will report also this bridge example on the wiki
which was missing.
		But does session:originate make sense in some cases or
not? Otherwise i'm going to remove this line on the wiki
	
http://wiki.freeswitch.org/wiki/Mod_lua#session:originate
<http://wiki.freeswitch.org/wiki/Mod_lua#session:originate> 
		 
		Thanks,
		Claudio

		 

		Internet Email Confidentiality Footer

	
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		La presente comunicazione, con le informazioni in essa
contenute e ogni documento o file allegato, e' rivolta unicamente alla/e
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riceverla. Se non siete i destinatari/autorizzati siete avvisati che
qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul
contenuto di tali informazioni e' vietata e potrebbe essere contro la
legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione
dei dati personali). Se avete ricevuto questa comunicazione per errore,
vi preghiamo di darne immediata notizia al mittente e di distruggere il
messaggio originale e ogni file allegato senza farne copia alcuna o
riprodurne in alcun modo il contenuto. 
		
		This e-mail and its attachments are intended for the
addressee(s) only and are confidential and/or may contain legally
privileged information. If you have received this message by mistake or
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and point out the error which has occurred. 
		
	
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From sos at sokhapkin.dyndns.org  Thu Jan 14 04:19:12 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Jan 2010 07:19:12 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
Message-ID: <201001140719.12363.sos@sokhapkin.dyndns.org>

Case 1 (bypass_media is off) is already on jira, 
http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test 
installation with latest trunk to reproduce case 2, when bypass_media=true 
and 10 seconds delay happens when 18X and then error are received on leg b.

On Wednesday 13 January 2010, Michael Jerris wrote:
> On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> > Critical issues are when SIP error come after 18X provisional response.
> >
> > - if bypass_media is false then dialplan stops and leg a is explicitly
> > hang up (switch_ivr_bridge.c, line 513).
>
>  behavior can be modified with continue_on_fail and hangup_after_bridge
> channel vars, perhaps ignore_early_media as well
>
> > - if bypass_media is true, then dialplan continue, but there is 10
> > seconds delay before next bridge application sends INVITE to gateway (
> > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
> >4.html ) I didn't track down yet why this happens looking at FS sources.
>
> see response on that thread
>
> > Why I didn't open a bug on jira? Because FS behaves according to the
> > design and specs :-) But not according to real world requirements...
>
> Really, people are trying to help you and your going to be snarky in
> response?
>
> > On Wednesday 13 January 2010, Brian West wrote:
> >> Can you elaborate on these "Critical" issues you seem to be having?  Why
> >> aren't you opening a jira for them if they are that critical to your
> >> needs?
>
> Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From brian at freeswitch.org  Thu Jan 14 06:50:46 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Jan 2010 08:50:46 -0600
Subject: [Freeswitch-users] ESL Mono/Managed
Message-ID: <1E6792AC-17F2-4DE9-9798-431350A5E77D@freeswitch.org>

I need someone that knows C# to write up some examples using ESL in C#

cd libs/esl
make managedmod
cd managed

Look at the perl example single_command.pl
require ESL;

my $command = shift;
my $args = join(" ", @ARGV);

my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
my $e = $con->api($command, $args);
print $e->getBody();


Need to write the same thing in C# and commit it as an example.  Also other examples exist in the perl directory those should be translated also if possible.

If you're interested please find me on IRC (bkw_)

Thanks,
Brina




From costa.zikalala at gmail.com  Thu Jan 14 06:54:57 2010
From: costa.zikalala at gmail.com (Costa Zikalala)
Date: Thu, 14 Jan 2010 16:54:57 +0200
Subject: [Freeswitch-users] Mod_Fax
Message-ID: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>

Hi All

Has anyone worked on a email2fax script for mod_fax?
If not how much would it cost for some genius here to quickly whip-up one?

Ideally both email2fax and fax2email should come standard with mod_fax or is
this not possible?

Thanks
Costa
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From brian at freeswitch.org  Thu Jan 14 07:09:40 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Jan 2010 09:09:40 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001140719.12363.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
Message-ID: <11677234-5A33-42BE-A440-6BBC859CE91E@freeswitch.org>

The first question is why are you using Proxy Media?  Secondly I'm almost positive that issue was resolved.  Thirdly 1.0.5pre10 was just an SVN snapshot!  We have since fixed a multitude of tiny issues here and there since that time one being a bug related to proxy media mode.

Thanks,
/b

On Jan 14, 2010, at 6:19 AM, Sergey Okhapkin wrote:

> Case 1 (bypass_media is off) is already on jira, 
> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test 
> installation with latest trunk to reproduce case 2, when bypass_media=true 
> and 10 seconds delay happens when 18X and then error are received on leg b.

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From kawarod at laposte.net  Thu Jan 14 07:09:59 2010
From: kawarod at laposte.net (rod)
Date: Thu, 14 Jan 2010 19:09:59 +0400
Subject: [Freeswitch-users] Eavesdrop in LUA
In-Reply-To: <4B4ED32E.30706@laposte.net>
References: <4B4ED32E.30706@laposte.net>
Message-ID: <4B4F33C7.6020403@laposte.net>

Hi all,

I have an incomplete solution for those interested.

I did it like this in dialplan:
        <action application="set" data="api_on_answer=originate 
sofia/internal/2000 at 10.10.10.10 &eavesdrop(${uuid})"/>-->
        <action application="bridge" 
data="{ecoute=${caller_id_number}}sofia/gateway/${caller_id_number}/${destination_number}"/>

so when a call is setup, FS initiate a new call to 2000 and eavesdrop 
the call.
But I have a small problem, the callee receives no sound until the 
eavesdropper send a SIP reply, so there is a 2-3 seconds delay before 
caller and callee can talk each other.

rod


rod a ?crit :
> Hi all,
>
> I'm trying to do this in LUA:
>      A call B
>
> and I'd like to setup a new call to C with eavesdrop of A conversation 
> with B.
>
> I have no idea how to do this if someone can help.
> I switched to LUA cause I see no way to achieve this with dialplan 
> (snippets are welcome).
>
> regards,
> rod
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   



From mike at van.lammeren.net  Thu Jan 14 07:15:52 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 14 Jan 2010 10:15:52 -0500
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0F362E8@BESONE.corp.dom>
References: <5d2828f1001131130s6ef166c0wc98f70023420c0fd@mail.gmail.com>
	<A2167AE7633A5C4DAE56EEBD1428ECC0F362E8@BESONE.corp.dom>
Message-ID: <5d2828f1001140715m2aba7ed6n3cef077c64943e76@mail.gmail.com>

Hi Claudio!

Thanks for the additions to the wiki! Every little bit helps.

I don't think I explained myself well, earlier. The point I was trying to
make about the wiki is that rather than remove the section about
"originate", it would be better to make an entry like "originate -- Does
anyone know what this does?"

Mike van Lammeren


On Thu, Jan 14, 2010 at 6:23 AM, Cavalera Claudio Luigi <
Claudio.Cavalera at italtel.it> wrote:

>  Hi Mike,
> in fact i've completed that page with the list of available  session
> functions.
>
> I've not removed "session:originate" yet, but it would be better if someone
> could provide an example in order to write an example in the wiki.
> I've added this valuable example also with the help of rupa
> http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.Session
>
> I would like to write something also about api_on_answer to use an api
> instead of a dialplan application.
>
> BRs,
> Claudio
>
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Mike van
> Lammeren
> *Sent:* Wednesday, January 13, 2010 8:30 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] playing with sessions in lua
>
> Hello!
>
> Before you remove "session:originate" from the wiki, you should take a look
> at this:
>
> http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocumented_Session_Functions.3F
>
> <http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocumented_Session_Functions.3F>There
> is, in fact, a function called "originate".
>
> Mike van Lammeren
>
>
> On Wed, Jan 13, 2010 at 5:22 AM, Cavalera Claudio Luigi <
> Claudio.Cavalera at italtel.it> wrote:
>
>>  Thanks a lot Anthony,
>> some comments inline (and please forgive me for my broken email client).
>>
>>  >>  example1: Consider this simple lua script in which i create two
>> sessions:
>>
>>  >>  api = freeswitch.API();
>>  >>  api:execute("originate", "sofia/internal/1001%192.168.1.1 &park");
>>  >>  api:execute("originate", "sofia/internal/1002%192.168.1.1 &park");
>>
>>  >  capture the output from api:execute the uuid is in there
>>
>> Thx a lot,
>> this was one piece i was missing although it's already on the wiki here:
>> http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls
>>
>>
>>  > because lua calls it freeswitch.bridge
>>
>>  > session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
>>  > session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1",
>> session1);
>>  > freeswitch.bridge(session1, session2);
>>
>> good to now, there isn't any example of freeswitch.bridge in the wiki and
>> i would like to add one.
>> Where I could find the full api of
>> freeswitch.Session( ) ?
>> because I've seen this working also without "session1" in the second line:
>>  session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
>> session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");
>> freeswitch.bridge(session1, session2);
>>
>>  also is there any difference between  freeswitch.bridge
>> and freeswitch.execute(uuid_bridge ...) ?
>>
>>   >> example3: yet another possibility
>>  >> local session1 =
>> freeswitch.Session("sofia/internal/1001%192.168.1.1");
>>  >> session1:originate(session2, "sofia/internal/1002%192.168.1.1",
>> 1000);
>>  >> but it does not work either.
>>
>>  > The above is gibberish try:
>>  > local session1 =
>> freeswitch.Session("sofia/internal/1001%192.168.1.1");
>>  > session1:execute("bridge", "sofia/internal/1002%192.168.1.1");
>>
>> Okay i will report also this bridge example on the wiki which was missing.
>> But does session:originate make sense in some cases or not? Otherwise i'm
>> going to remove this line on the wiki
>>  http://wiki.freeswitch.org/wiki/Mod_lua#session:originate
>>
>> Thanks,
>> Claudio
>>
>>
>>
>> Internet Email Confidentiality Footer
>>
>>
>> ********************************************************************************************************************************************
>>
>> La presente comunicazione, con le informazioni in essa contenute e ogni
>> documento o file allegato, e' rivolta unicamente alla/e persona/e cui e'
>> indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete
>> i destinatari/autorizzati siete avvisati che qualsiasi azione, copia,
>> comunicazione, divulgazione o simili basate sul contenuto di tali
>> informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P.,
>> D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se
>> avete ricevuto questa comunicazione per errore, vi preghiamo di darne
>> immediata notizia al mittente e di distruggere il messaggio originale e ogni
>> file allegato senza farne copia alcuna o riprodurne in alcun modo il
>> contenuto.
>>
>> This e-mail and its attachments are intended for the addressee(s) only and
>> are confidential and/or may contain legally privileged information. If you
>> have received this message by mistake or are not one of the addressees
>> above, you may take no action based on it, and you may not copy or show it
>> to anyone; please reply to this e-mail and point out the error which has
>> occurred.
>>
>> ********************************************************************************************************************************************
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
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From sos at sokhapkin.dyndns.org  Thu Jan 14 07:20:06 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Jan 2010 10:20:06 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
Message-ID: <201001141020.06137.sos@sokhapkin.dyndns.org>

With trunk version and bypass_media=true the behavior is different - leg a is 
terminated after 20 seconds of wait, dialplan can't continue. Dialplan and FS 
log are at http://pastebin.freeswitch.org/11794 , note timestamps of lines 
378 and 379.

FS is running on 192.168.1.2:5060, SIP client on 192.168.1.2:5066, 
103 at 192.168.1.254 responds with early media and 480.

Let me know if you need additional information.

On Wednesday 13 January 2010, Michael Jerris wrote:
> On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> > Critical issues are when SIP error come after 18X provisional response.
> >
> > - if bypass_media is false then dialplan stops and leg a is explicitly
> > hang up (switch_ivr_bridge.c, line 513).
>
>  behavior can be modified with continue_on_fail and hangup_after_bridge
> channel vars, perhaps ignore_early_media as well
>
> > - if bypass_media is true, then dialplan continue, but there is 10
> > seconds delay before next bridge application sends INVITE to gateway (
> > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
> >4.html ) I didn't track down yet why this happens looking at FS sources.
>
> see response on that thread
>
> > Why I didn't open a bug on jira? Because FS behaves according to the
> > design and specs :-) But not according to real world requirements...
>
> Really, people are trying to help you and your going to be snarky in
> response?
>
> > On Wednesday 13 January 2010, Brian West wrote:
> >> Can you elaborate on these "Critical" issues you seem to be having?  Why
> >> aren't you opening a jira for them if they are that critical to your
> >> needs?
>
> Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From mike at van.lammeren.net  Thu Jan 14 07:44:04 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 14 Jan 2010 10:44:04 -0500
Subject: [Freeswitch-users] Question about a1-hash
Message-ID: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>

Hello!

I have written a Lua script to connect to a database and provide directory
information for phones registering with FreeSWITCH.

My problem is that I store an MD5 hash of the passwords in the database, so
I wish there was a way to get FreeSWITCH to authenticate using the MD5 hash
of the password provided by the phone, and not the password itself.

According to the
wiki<http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide>,
it is possible to pass in a parameter called *a1-hash* instead of the
username and password. The a1-hash parameter is an MD5 hash of a string
comprising the username, domain and password, separated by
colons. Unfortunately, I can't generate that string, since I don't have the
raw password, just the MD5 hash.

I would have my Lua script do the authentication, but cannot because
FreeSWITCH doesn't pass the user's password to the script.

The best solution I can think of is to enter the MD5 hash of the password in
the phone.

Does anyone have a better idea?


Mike van Lammeren
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From sos at sokhapkin.dyndns.org  Thu Jan 14 07:46:58 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Jan 2010 10:46:58 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001141020.06137.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
	<201001141020.06137.sos@sokhapkin.dyndns.org>
Message-ID: <201001141046.58561.sos@sokhapkin.dyndns.org>

I repeated the test without bypass_media. Leg a is terminated as soon as 480 
received from leg b and dialplan doesn't continue. Let me know if you want me 
to post FS log to pastebin.

On Thursday 14 January 2010, Sergey Okhapkin wrote:
> With trunk version and bypass_media=true the behavior is different - leg a
> is terminated after 20 seconds of wait, dialplan can't continue. Dialplan
> and FS log are at http://pastebin.freeswitch.org/11794 , note timestamps of
> lines 378 and 379.
>
> FS is running on 192.168.1.2:5060, SIP client on 192.168.1.2:5066,
> 103 at 192.168.1.254 responds with early media and 480.
>
> Let me know if you need additional information.
>
> On Wednesday 13 January 2010, Michael Jerris wrote:
> > On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> > > Critical issues are when SIP error come after 18X provisional response.
> > >
> > > - if bypass_media is false then dialplan stops and leg a is explicitly
> > > hang up (switch_ivr_bridge.c, line 513).
> >
> >  behavior can be modified with continue_on_fail and hangup_after_bridge
> > channel vars, perhaps ignore_early_media as well
> >
> > > - if bypass_media is true, then dialplan continue, but there is 10
> > > seconds delay before next bridge application sends INVITE to gateway (
> > > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024
> > >35 4.html ) I didn't track down yet why this happens looking at FS
> > > sources.
> >
> > see response on that thread
> >
> > > Why I didn't open a bug on jira? Because FS behaves according to the
> > > design and specs :-) But not according to real world requirements...
> >
> > Really, people are trying to help you and your going to be snarky in
> > response?
> >
> > > On Wednesday 13 January 2010, Brian West wrote:
> > >> Can you elaborate on these "Critical" issues you seem to be having? 
> > >> Why aren't you opening a jira for them if they are that critical to
> > >> your needs?
> >
> > Mike
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From brian at freeswitch.org  Thu Jan 14 08:00:00 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Jan 2010 10:00:00 -0600
Subject: [Freeswitch-users] Question about a1-hash
In-Reply-To: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>
References: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>
Message-ID: <7677225F-6384-415B-AF98-09DED8A1F033@freeswitch.org>

We don't have the password so we can't pass it to you please read: http://en.wikipedia.org/wiki/Digest_access_authentication

Its how the authentication is done and we are never given the text of the password you are however given the details so you can calculate the response and verify it without having to know the password.

/b

On Jan 14, 2010, at 9:44 AM, Mike van Lammeren wrote:

> Hello!
> 
> I have written a Lua script to connect to a database and provide directory information for phones registering with FreeSWITCH.
> 
> My problem is that I store an MD5 hash of the passwords in the database, so I wish there was a way to get FreeSWITCH to authenticate using the MD5 hash of the password provided by the phone, and not the password itself.
> 
> According to the wiki, it is possible to pass in a parameter called a1-hash instead of the username and password. The a1-hash parameter is an MD5 hash of a string comprising the username, domain and password, separated by colons. Unfortunately, I can't generate that string, since I don't have the raw password, just the MD5 hash.
> 
> I would have my Lua script do the authentication, but cannot because FreeSWITCH doesn't pass the user's password to the script.
> 
> The best solution I can think of is to enter the MD5 hash of the password in the phone.
> 
> Does anyone have a better idea?
> 
> 
> Mike van Lammeren

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From mike at jerris.com  Thu Jan 14 08:08:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Jan 2010 11:08:37 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001141046.58561.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
	<201001141020.06137.sos@sokhapkin.dyndns.org>
	<201001141046.58561.sos@sokhapkin.dyndns.org>
Message-ID: <B2801B0F-47E5-43C8-919A-671D5733C53C@jerris.com>

Lets move this all to jira so that we can properly track the issue.  We will need full debug logs with siptrace, on current svn trunk along with dialplan and such to reproduce the issue.

Mike

On Jan 14, 2010, at 10:46 AM, Sergey Okhapkin wrote:

> I repeated the test without bypass_media. Leg a is terminated as soon as 480 
> received from leg b and dialplan doesn't continue. Let me know if you want me 
> to post FS log to pastebin.
> 
> On Thursday 14 January 2010, Sergey Okhapkin wrote:
>> With trunk version and bypass_media=true the behavior is different - leg a
>> is terminated after 20 seconds of wait, dialplan can't continue. Dialplan
>> and FS log are at http://pastebin.freeswitch.org/11794 , note timestamps of
>> lines 378 and 379.
>> 
>> FS is running on 192.168.1.2:5060, SIP client on 192.168.1.2:5066,
>> 103 at 192.168.1.254 responds with early media and 480.
>> 
>> Let me know if you need additional information.




From mike at jerris.com  Thu Jan 14 08:12:59 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Jan 2010 11:12:59 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001140226y2b035f37rfcafb03190435ec1@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
	<201001132326.41354.sos@sokhapkin.dyndns.org>
	<52E8BE9E-B6D6-411C-9080-697AF24017B4@jerris.com>
	<191c3a031001132114s18ed2baw2ff802dbd615fe01@mail.gmail.com>
	<191c3a031001132115u2d88bab7p7cbb27ab1eaad466@mail.gmail.com>
	<9853f4ff1001140226y2b035f37rfcafb03190435ec1@mail.gmail.com>
Message-ID: <43692088-738A-475C-8D60-E40D296ECEB1@jerris.com>

Okay, so that is a very interesting use case.   Seems a bizarre way to do this for the carrier, but interesting none the less.  I'd say the chances of actually muxing the early media is small, what might be possible would be something to say which b legs media to pass along to the a leg.  I have not looked at all at how complicated this is, it will be down deep somewhere in switch_ivr_originate code around where we if for ignore_early_media.  This code is pretty complex, I can't say that we will ever actually add this functionality, but at least now I won't blow it off as complete nonsense.

Mike



On Jan 14, 2010, at 5:26 AM, David Villasmil wrote:

> Hello again,
> 
> Using ingore early media only ignores ALL media, that's not what I need. At least in europe i've seen it many times, MNOs provide a service with which you can have a song played to the caller while the call is connecting to your cell phone. This is basically what I'm trying to achieve.
> 
> The content provider is in possesions of the media and it require us to fork the call and send a SIP INVITE to the on one leg and the call to the destination number on another leg. they will only provide a progress, no answer.
> 
> I CAN do it locally by playing the file as a custom ringback but that's not the standard in terms of commercial use of the content.
> 
> Is there any way to modify the behaviour of fs when i receives media? Let's say i.e. it doesn't drop the other leg, but provides the first early media it receives and just wait for some channel to answer? I will not have both media but it would work.
> 
> thanks
> 
> David
> 
> On Thu, Jan 14, 2010 at 6:15 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> Perhaps best not to help him anymore without an apology for the snap judgement and comparison to asterisk clearly designed to push our buttons.
> We'll be here when you realize we were trying to help you but I can't promise we will still have paitence......
> 
> 
>> On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike at jerris.com> wrote:
>> 
>> You already are running on trunk.
>> 
>> Mike
>> On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote: > I run FreeSWITCH Version 1.0.5pre10 (16012M...
>> 
>> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists...
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From freeswitch at aastral.net  Thu Jan 14 08:15:06 2010
From: freeswitch at aastral.net (Bill W)
Date: Thu, 14 Jan 2010 11:15:06 -0500
Subject: [Freeswitch-users] Question about a1-hash
In-Reply-To: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>
References: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>
Message-ID: <4B4F430A.8080006@aastral.net>

Why don't you just store the a1-hash in the database instead of the 
password?

-Bill W.


Mike van Lammeren wrote:
> Hello!
> 
> I have written a Lua script to connect to a database and provide 
> directory information for phones registering with FreeSWITCH.
> 
> My problem is that I store an MD5 hash of the passwords in the database, 
> so I wish there was a way to get FreeSWITCH to authenticate using the 
> MD5 hash of the password provided by the phone, and not the password itself.
> 
> According to the wiki 
> <http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide>, it is 
> possible to pass in a parameter called /a1-hash/ instead of the username 
> and password. The a1-hash parameter is an MD5 hash of a string 
> comprising the username, domain and password, separated by 
> colons. Unfortunately, I can't generate that string, since I don't have 
> the raw password, just the MD5 hash.
> 
> I would have my Lua script do the authentication, but cannot because 
> FreeSWITCH doesn't pass the user's password to the script.
> 
> The best solution I can think of is to enter the MD5 hash of the 
> password in the phone.
> 
> Does anyone have a better idea?
> 
> 
> Mike van Lammeren
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Thu Jan 14 08:22:21 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 10:22:21 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001140719.12363.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>

Sergey,

The bug you reference was closed because proxy_media mode by design send's
the B leg's codec to the A
LEG so if there is a failure condition there is no easy graceful way to back
out and try another call.
I will try to make it possible if yo post a bounty, I estimate a minimum of
$1000USD in consulting time.

The other one I might look at once you have apologized.

I can probably add some code to make the bridge exit without terminating the
A leg even when hangup_after_bridge=true in the case where the the B leg is
not answered but right now I am sort of annoyed with your attitude in this
thread.


On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Case 1 (bypass_media is off) is already on jira,
> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
> installation with latest trunk to reproduce case 2, when bypass_media=true
> and 10 seconds delay happens when 18X and then error are received on leg b.
>
> On Wednesday 13 January 2010, Michael Jerris wrote:
> > On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> > > Critical issues are when SIP error come after 18X provisional response.
> > >
> > > - if bypass_media is false then dialplan stops and leg a is explicitly
> > > hang up (switch_ivr_bridge.c, line 513).
> >
> >  behavior can be modified with continue_on_fail and hangup_after_bridge
> > channel vars, perhaps ignore_early_media as well
> >
> > > - if bypass_media is true, then dialplan continue, but there is 10
> > > seconds delay before next bridge application sends INVITE to gateway (
> > >
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
> > >4.html ) I didn't track down yet why this happens looking at FS sources.
> >
> > see response on that thread
> >
> > > Why I didn't open a bug on jira? Because FS behaves according to the
> > > design and specs :-) But not according to real world requirements...
> >
> > Really, people are trying to help you and your going to be snarky in
> > response?
> >
> > > On Wednesday 13 January 2010, Brian West wrote:
> > >> Can you elaborate on these "Critical" issues you seem to be having?
>  Why
> > >> aren't you opening a jira for them if they are that critical to your
> > >> needs?
> >
> > Mike
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at van.lammeren.net  Thu Jan 14 08:24:42 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 14 Jan 2010 11:24:42 -0500
Subject: [Freeswitch-users] Question about a1-hash
In-Reply-To: <7677225F-6384-415B-AF98-09DED8A1F033@freeswitch.org>
References: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>
	<7677225F-6384-415B-AF98-09DED8A1F033@freeswitch.org>
Message-ID: <5d2828f1001140824u416ec5ebh2b2636af4a02ae33@mail.gmail.com>

That's awesome! I should have noticed those 32-character strings in the
parameters passed to the script. Thanks!

It's a little off-topic, but I'm glad to see someone using digest
authentication. It's too bad that it was un-supported by browsers for so
long, that no one touched it for web apps. The choice is either use basic
authentication, which is plaintext, or switch to https. With https, not
everyone realizes that the web server, and any apps, can see the password in
plain text.

Mike van Lammeren


On Thu, Jan 14, 2010 at 11:00 AM, Brian West <brian at freeswitch.org> wrote:

> We don't have the password so we can't pass it to you please read:
> http://en.wikipedia.org/wiki/Digest_access_authentication
>
> Its how the authentication is done and we are never given the text of the
> password you are however given the details so you can calculate the response
> and verify it without having to know the password.
>
> /b
>
> On Jan 14, 2010, at 9:44 AM, Mike van Lammeren wrote:
>
> Hello!
>
> I have written a Lua script to connect to a database and provide directory
> information for phones registering with FreeSWITCH.
>
> My problem is that I store an MD5 hash of the passwords in the database, so
> I wish there was a way to get FreeSWITCH to authenticate using the MD5 hash
> of the password provided by the phone, and not the password itself.
>
> According to the wiki<http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide>,
> it is possible to pass in a parameter called *a1-hash* instead of the
> username and password. The a1-hash parameter is an MD5 hash of a string
> comprising the username, domain and password, separated by
> colons. Unfortunately, I can't generate that string, since I don't have the
> raw password, just the MD5 hash.
>
> I would have my Lua script do the authentication, but cannot because
> FreeSWITCH doesn't pass the user's password to the script.
>
> The best solution I can think of is to enter the MD5 hash of the password
> in the phone.
>
> Does anyone have a better idea?
>
>
> Mike van Lammeren
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Jan 14 08:26:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 10:26:52 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
	<191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
Message-ID: <191c3a031001140826q2d014484v8807bf3b98b78f63@mail.gmail.com>

David,

Your request is somewhat challenging because we have to make sure FreeSWITCH
is agnostic in protocols and that aside, we also have a very complicated and
feature-rich originate API that does not currently support sending audio
from one leg to the A leg while it's trying to call 10 other B legs.  Please
try to understand that this request is a very special side case and we would
have to do many hours of work to make it possible.  I am not sure if you are
simply asking if its possible or if you want us to implement it for you but
I am afraid it would fall under commercial support to undertake that unique
of a feature that only helps a very small percentage of our user base
considering all the work we already have to do every day.



On Thu, Jan 14, 2010 at 10:22 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Sergey,
>
> The bug you reference was closed because proxy_media mode by design send's
> the B leg's codec to the A
> LEG so if there is a failure condition there is no easy graceful way to
> back out and try another call.
> I will try to make it possible if yo post a bounty, I estimate a minimum of
> $1000USD in consulting time.
>
> The other one I might look at once you have apologized.
>
> I can probably add some code to make the bridge exit without terminating
> the A leg even when hangup_after_bridge=true in the case where the the B leg
> is not answered but right now I am sort of annoyed with your attitude in
> this thread.
>
>
>
> On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin <sos at sokhapkin.dyndns.org
> > wrote:
>
>> Case 1 (bypass_media is off) is already on jira,
>> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
>> installation with latest trunk to reproduce case 2, when bypass_media=true
>> and 10 seconds delay happens when 18X and then error are received on leg
>> b.
>>
>> On Wednesday 13 January 2010, Michael Jerris wrote:
>> > On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
>> > > Critical issues are when SIP error come after 18X provisional
>> response.
>> > >
>> > > - if bypass_media is false then dialplan stops and leg a is explicitly
>> > > hang up (switch_ivr_bridge.c, line 513).
>> >
>> >  behavior can be modified with continue_on_fail and hangup_after_bridge
>> > channel vars, perhaps ignore_early_media as well
>> >
>> > > - if bypass_media is true, then dialplan continue, but there is 10
>> > > seconds delay before next bridge application sends INVITE to gateway (
>> > >
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
>> > >4.html ) I didn't track down yet why this happens looking at FS
>> sources.
>> >
>> > see response on that thread
>> >
>> > > Why I didn't open a bug on jira? Because FS behaves according to the
>> > > design and specs :-) But not according to real world requirements...
>> >
>> > Really, people are trying to help you and your going to be snarky in
>> > response?
>> >
>> > > On Wednesday 13 January 2010, Brian West wrote:
>> > >> Can you elaborate on these "Critical" issues you seem to be having?
>>  Why
>> > >> aren't you opening a jira for them if they are that critical to your
>> > >> needs?
>> >
>> > Mike
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at van.lammeren.net  Thu Jan 14 08:27:26 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 14 Jan 2010 11:27:26 -0500
Subject: [Freeswitch-users] Question about a1-hash
In-Reply-To: <4B4F430A.8080006@aastral.net>
References: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>
	<4B4F430A.8080006@aastral.net>
Message-ID: <5d2828f1001140827w386fe2acw478ae378b0ecfe53@mail.gmail.com>

Hi Bill!

That's a perfectly reasonable suggestion, but for this solution, I need a
username/password combo that spans domains. Also, the password is used in
other places too.

Thanks for the reply!

Mike van Lammeren


On Thu, Jan 14, 2010 at 11:15 AM, Bill W <freeswitch at aastral.net> wrote:

> Why don't you just store the a1-hash in the database instead of the
> password?
>
> -Bill W.
>
>
> Mike van Lammeren wrote:
> > Hello!
> >
> > I have written a Lua script to connect to a database and provide
> > directory information for phones registering with FreeSWITCH.
> >
> > My problem is that I store an MD5 hash of the passwords in the database,
> > so I wish there was a way to get FreeSWITCH to authenticate using the
> > MD5 hash of the password provided by the phone, and not the password
> itself.
> >
> > According to the wiki
> > <http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide>, it is
> > possible to pass in a parameter called /a1-hash/ instead of the username
> > and password. The a1-hash parameter is an MD5 hash of a string
> > comprising the username, domain and password, separated by
> > colons. Unfortunately, I can't generate that string, since I don't have
> > the raw password, just the MD5 hash.
> >
> > I would have my Lua script do the authentication, but cannot because
> > FreeSWITCH doesn't pass the user's password to the script.
> >
> > The best solution I can think of is to enter the MD5 hash of the
> > password in the phone.
> >
> > Does anyone have a better idea?
> >
> >
> > Mike van Lammeren
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Thu Jan 14 08:37:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 10:37:45 -0600
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <5d2828f1001140715m2aba7ed6n3cef077c64943e76@mail.gmail.com>
References: <5d2828f1001131130s6ef166c0wc98f70023420c0fd@mail.gmail.com>
	<A2167AE7633A5C4DAE56EEBD1428ECC0F362E8@BESONE.corp.dom>
	<5d2828f1001140715m2aba7ed6n3cef077c64943e76@mail.gmail.com>
Message-ID: <191c3a031001140837m7cffcdd5w71886d6c8ba1dafe@mail.gmail.com>

Don't forget that lua,perl,python,managed,java all share the same exact C++
source file with swig so the same exact api applies to all of the above.

On Thu, Jan 14, 2010 at 9:15 AM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Hi Claudio!
>
> Thanks for the additions to the wiki! Every little bit helps.
>
> I don't think I explained myself well, earlier. The point I was trying to
> make about the wiki is that rather than remove the section about
> "originate", it would be better to make an entry like "originate -- Does
> anyone know what this does?"
>
> Mike van Lammeren
>
>
> On Thu, Jan 14, 2010 at 6:23 AM, Cavalera Claudio Luigi <
> Claudio.Cavalera at italtel.it> wrote:
>
>>  Hi Mike,
>> in fact i've completed that page with the list of available  session
>> functions.
>>
>> I've not removed "session:originate" yet, but it would be better if
>> someone could provide an example in order to write an example in the wiki.
>> I've added this valuable example also with the help of rupa
>> http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.Session
>>
>> I would like to write something also about api_on_answer to use an api
>> instead of a dialplan application.
>>
>> BRs,
>> Claudio
>>
>>  ------------------------------
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Mike van
>> Lammeren
>> *Sent:* Wednesday, January 13, 2010 8:30 PM
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] playing with sessions in lua
>>
>> Hello!
>>
>> Before you remove "session:originate" from the wiki, you should take a
>> look at this:
>>
>> http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocumented_Session_Functions.3F
>>
>> <http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocumented_Session_Functions.3F>There
>> is, in fact, a function called "originate".
>>
>> Mike van Lammeren
>>
>>
>> On Wed, Jan 13, 2010 at 5:22 AM, Cavalera Claudio Luigi <
>> Claudio.Cavalera at italtel.it> wrote:
>>
>>>  Thanks a lot Anthony,
>>> some comments inline (and please forgive me for my broken email client).
>>>
>>>  >>  example1: Consider this simple lua script in which i create two
>>> sessions:
>>>
>>>  >>  api = freeswitch.API();
>>>  >>  api:execute("originate", "sofia/internal/1001%192.168.1.1 &park");
>>>  >>  api:execute("originate", "sofia/internal/1002%192.168.1.1 &park");
>>>
>>>  >  capture the output from api:execute the uuid is in there
>>>
>>> Thx a lot,
>>> this was one piece i was missing although it's already on the wiki here:
>>> http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls
>>>
>>>
>>>  > because lua calls it freeswitch.bridge
>>>
>>>  > session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
>>>  > session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1",
>>> session1);
>>>  > freeswitch.bridge(session1, session2);
>>>
>>> good to now, there isn't any example of freeswitch.bridge in the wiki and
>>> i would like to add one.
>>> Where I could find the full api of
>>> freeswitch.Session( ) ?
>>> because I've seen this working also without "session1" in the second
>>> line:
>>>  session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
>>> session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");
>>> freeswitch.bridge(session1, session2);
>>>
>>>  also is there any difference between  freeswitch.bridge
>>> and freeswitch.execute(uuid_bridge ...) ?
>>>
>>>   >> example3: yet another possibility
>>>  >> local session1 =
>>> freeswitch.Session("sofia/internal/1001%192.168.1.1");
>>>  >> session1:originate(session2, "sofia/internal/1002%192.168.1.1",
>>> 1000);
>>>  >> but it does not work either.
>>>
>>>  > The above is gibberish try:
>>>  > local session1 =
>>> freeswitch.Session("sofia/internal/1001%192.168.1.1");
>>>  > session1:execute("bridge", "sofia/internal/1002%192.168.1.1");
>>>
>>> Okay i will report also this bridge example on the wiki which was
>>> missing.
>>> But does session:originate make sense in some cases or not? Otherwise i'm
>>> going to remove this line on the wiki
>>>  http://wiki.freeswitch.org/wiki/Mod_lua#session:originate
>>>
>>> Thanks,
>>> Claudio
>>>
>>>
>>>
>>> Internet Email Confidentiality Footer
>>>
>>>
>>> ********************************************************************************************************************************************
>>>
>>> La presente comunicazione, con le informazioni in essa contenute e ogni
>>> documento o file allegato, e' rivolta unicamente alla/e persona/e cui e'
>>> indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete
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>>> D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se
>>> avete ricevuto questa comunicazione per errore, vi preghiamo di darne
>>> immediata notizia al mittente e di distruggere il messaggio originale e ogni
>>> file allegato senza farne copia alcuna o riprodurne in alcun modo il
>>> contenuto.
>>>
>>> This e-mail and its attachments are intended for the addressee(s) only
>>> and are confidential and/or may contain legally privileged information. If
>>> you have received this message by mistake or are not one of the addressees
>>> above, you may take no action based on it, and you may not copy or show it
>>> to anyone; please reply to this e-mail and point out the error which has
>>> occurred.
>>>
>>> ********************************************************************************************************************************************
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From david.villasmil.work at gmail.com  Thu Jan 14 08:41:54 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 14 Jan 2010 17:41:54 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001140826q2d014484v8807bf3b98b78f63@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
	<191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
	<191c3a031001140826q2d014484v8807bf3b98b78f63@mail.gmail.com>
Message-ID: <9853f4ff1001140841n603ddc45n972b3aba83600d01@mail.gmail.com>

Anthony,

     Thanks. I understand it's complicated. Another option is to be able to
configure whether or not to discard all other B-legs on receiving media but
on receiving an final code like 200. This way we will still get the first
media that arrived but provide an answer on the actual channel that provided
the 200.

My request comes from 2 sides, one commercial and the other personal. For
the commercial I just need to now whether it is possible by some tweaking so
the functionality can be offered.

The other, I've been involved in FS as a user from the very beginning
(although I was absent for some time), and I'd like to help in the
development of the project by letting the community know what's out there
commercially speaking.


Hope you can help.

David

On Thu, Jan 14, 2010 at 5:26 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> David,
>
> Your request is somewhat challenging because we have to make sure
> FreeSWITCH is agnostic in protocols and that aside, we also have a very
> complicated and feature-rich originate API that does not currently support
> sending audio from one leg to the A leg while it's trying to call 10 other B
> legs.  Please try to understand that this request is a very special side
> case and we would have to do many hours of work to make it possible.  I am
> not sure if you are simply asking if its possible or if you want us to
> implement it for you but I am afraid it would fall under commercial support
> to undertake that unique of a feature that only helps a very small
> percentage of our user base considering all the work we already have to do
> every day.
>
>
>
>
> On Thu, Jan 14, 2010 at 10:22 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Sergey,
>>
>> The bug you reference was closed because proxy_media mode by design send's
>> the B leg's codec to the A
>> LEG so if there is a failure condition there is no easy graceful way to
>> back out and try another call.
>> I will try to make it possible if yo post a bounty, I estimate a minimum
>> of $1000USD in consulting time.
>>
>> The other one I might look at once you have apologized.
>>
>> I can probably add some code to make the bridge exit without terminating
>> the A leg even when hangup_after_bridge=true in the case where the the B leg
>> is not answered but right now I am sort of annoyed with your attitude in
>> this thread.
>>
>>
>>
>> On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin <
>> sos at sokhapkin.dyndns.org> wrote:
>>
>>> Case 1 (bypass_media is off) is already on jira,
>>> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
>>> installation with latest trunk to reproduce case 2, when
>>> bypass_media=true
>>> and 10 seconds delay happens when 18X and then error are received on leg
>>> b.
>>>
>>> On Wednesday 13 January 2010, Michael Jerris wrote:
>>> > On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
>>> > > Critical issues are when SIP error come after 18X provisional
>>> response.
>>> > >
>>> > > - if bypass_media is false then dialplan stops and leg a is
>>> explicitly
>>> > > hang up (switch_ivr_bridge.c, line 513).
>>> >
>>> >  behavior can be modified with continue_on_fail and hangup_after_bridge
>>> > channel vars, perhaps ignore_early_media as well
>>> >
>>> > > - if bypass_media is true, then dialplan continue, but there is 10
>>> > > seconds delay before next bridge application sends INVITE to gateway
>>> (
>>> > >
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
>>> > >4.html ) I didn't track down yet why this happens looking at FS
>>> sources.
>>> >
>>> > see response on that thread
>>> >
>>> > > Why I didn't open a bug on jira? Because FS behaves according to the
>>> > > design and specs :-) But not according to real world requirements...
>>> >
>>> > Really, people are trying to help you and your going to be snarky in
>>> > response?
>>> >
>>> > > On Wednesday 13 January 2010, Brian West wrote:
>>> > >> Can you elaborate on these "Critical" issues you seem to be having?
>>>  Why
>>> > >> aren't you opening a jira for them if they are that critical to your
>>> > >> needs?
>>> >
>>> > Mike
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sos at sokhapkin.dyndns.org  Thu Jan 14 08:41:59 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Jan 2010 11:41:59 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
	<191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
Message-ID: <201001141141.59831.sos@sokhapkin.dyndns.org>

Anthony, I apology, I didn't expect you'll be annoyed. As for the issue, the 
issue is not in "proxy_media" setting, same happens when both "proxy_media" 
and "bypass_media" are not set. The code does explicit leg A hangup in 
switch_ivr_bridge.c, line 513. The condition there is "if leg A is up and leg 
A is not in answered state, then hangup leg A", this prevents dialplan to 
continue execution on early media bridge termination.

Issue when "bypass_media=true" is completely different, I'll open the issue on 
jira.

On Thursday 14 January 2010, Anthony Minessale wrote:
> Sergey,
>
> The bug you reference was closed because proxy_media mode by design send's
> the B leg's codec to the A
> LEG so if there is a failure condition there is no easy graceful way to
> back out and try another call.
> I will try to make it possible if yo post a bounty, I estimate a minimum of
> $1000USD in consulting time.
>
> The other one I might look at once you have apologized.
>
> I can probably add some code to make the bridge exit without terminating
> the A leg even when hangup_after_bridge=true in the case where the the B
> leg is not answered but right now I am sort of annoyed with your attitude
> in this thread.
>
>
> On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin
>
> <sos at sokhapkin.dyndns.org>wrote:
> > Case 1 (bypass_media is off) is already on jira,
> > http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
> > installation with latest trunk to reproduce case 2, when
> > bypass_media=true and 10 seconds delay happens when 18X and then error
> > are received on leg b.
> >
> > On Wednesday 13 January 2010, Michael Jerris wrote:
> > > On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> > > > Critical issues are when SIP error come after 18X provisional
> > > > response.
> > > >
> > > > - if bypass_media is false then dialplan stops and leg a is
> > > > explicitly hang up (switch_ivr_bridge.c, line 513).
> > >
> > >  behavior can be modified with continue_on_fail and hangup_after_bridge
> > > channel vars, perhaps ignore_early_media as well
> > >
> > > > - if bypass_media is true, then dialplan continue, but there is 10
> > > > seconds delay before next bridge application sends INVITE to gateway
> > > > (
> >
> > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
> >
> > > >4.html ) I didn't track down yet why this happens looking at FS
> > > > sources.
> > >
> > > see response on that thread
> > >
> > > > Why I didn't open a bug on jira? Because FS behaves according to the
> > > > design and specs :-) But not according to real world requirements...
> > >
> > > Really, people are trying to help you and your going to be snarky in
> > > response?
> > >
> > > > On Wednesday 13 January 2010, Brian West wrote:
> > > >> Can you elaborate on these "Critical" issues you seem to be having?
> >
> >  Why
> >
> > > >> aren't you opening a jira for them if they are that critical to your
> > > >> needs?
> > >
> > > Mike
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From anthony.minessale at gmail.com  Thu Jan 14 08:47:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 10:47:24 -0600
Subject: [Freeswitch-users] Not able to capture the custom event
In-Reply-To: <13529f9d1001140124t42e9dfe5u3e70a6976aa032a8@mail.gmail.com>
References: <13529f9d1001140124t42e9dfe5u3e70a6976aa032a8@mail.gmail.com>
Message-ID: <191c3a031001140847q1238bbcdi4d9dd84648ee2a27@mail.gmail.com>

you cant subscribe to all custom events, you have to specify a list of
subclasses
you have to say events plain custom myevent::ACDnotify

On Thu, Jan 14, 2010 at 3:24 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:

> Hi Guys,
>
> I'm testing to see whether a custom event can be triggered and captured.
> Here's the extension:
>
>     <extension name="test_ivr">
>       <condition field="destination_number" expression="^777$">
>         <action application="event" data="Event-Name=*CUSTOM*
> ,Event-Subclass=myevent::ACDnotify"/>
>         <action application="transfer" data="9999 XML default" />
>       </condition>
>     </extension>
>
> If I telnet and use "events plain all", I can get the event. However, if I
> use "events plain CUSTOM" or "event plain CUSTOM", it doesn't get captured.
> Is there anything wrong with the command?
>
> Regards,
> -Jingwei
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mrene_lists at avgs.ca  Thu Jan 14 08:47:30 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 14 Jan 2010 11:47:30 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001141141.59831.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
	<191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
	<201001141141.59831.sos@sokhapkin.dyndns.org>
Message-ID: <23BD1322-7882-4D43-80AB-27872096B4C9@avgs.ca>

One way you could fix it is set "hangup_after_bridge=false" and then  
set "execute_on_answer=set\shangup_after_bridge=true". That will make  
it continue executing the dialplan atter a failure that came in after  
183. Note that there is a side effect to this method, since audio was  
being relayed, you are forced to use the same codec on the 2nd  
carrier, best to tell them they are wrong failing a call after 183.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 14-Jan-10, at 11:41 AM, Sergey Okhapkin wrote:

> Anthony, I apology, I didn't expect you'll be annoyed. As for the  
> issue, the
> issue is not in "proxy_media" setting, same happens when both  
> "proxy_media"
> and "bypass_media" are not set. The code does explicit leg A hangup in
> switch_ivr_bridge.c, line 513. The condition there is "if leg A is  
> up and leg
> A is not in answered state, then hangup leg A", this prevents  
> dialplan to
> continue execution on early media bridge termination.
>
> Issue when "bypass_media=true" is completely different, I'll open  
> the issue on
> jira.
>
> On Thursday 14 January 2010, Anthony Minessale wrote:
>> Sergey,
>>
>> The bug you reference was closed because proxy_media mode by design  
>> send's
>> the B leg's codec to the A
>> LEG so if there is a failure condition there is no easy graceful  
>> way to
>> back out and try another call.
>> I will try to make it possible if yo post a bounty, I estimate a  
>> minimum of
>> $1000USD in consulting time.
>>
>> The other one I might look at once you have apologized.
>>
>> I can probably add some code to make the bridge exit without  
>> terminating
>> the A leg even when hangup_after_bridge=true in the case where the  
>> the B
>> leg is not answered but right now I am sort of annoyed with your  
>> attitude
>> in this thread.
>>
>>
>> On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin
>>
>> <sos at sokhapkin.dyndns.org>wrote:
>>> Case 1 (bypass_media is off) is already on jira,
>>> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
>>> installation with latest trunk to reproduce case 2, when
>>> bypass_media=true and 10 seconds delay happens when 18X and then  
>>> error
>>> are received on leg b.
>>>
>>> On Wednesday 13 January 2010, Michael Jerris wrote:
>>>> On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
>>>>> Critical issues are when SIP error come after 18X provisional
>>>>> response.
>>>>>
>>>>> - if bypass_media is false then dialplan stops and leg a is
>>>>> explicitly hang up (switch_ivr_bridge.c, line 513).
>>>>
>>>> behavior can be modified with continue_on_fail and  
>>>> hangup_after_bridge
>>>> channel vars, perhaps ignore_early_media as well
>>>>
>>>>> - if bypass_media is true, then dialplan continue, but there is 10
>>>>> seconds delay before next bridge application sends INVITE to  
>>>>> gateway
>>>>> (
>>>
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
>>>
>>>>> 4.html ) I didn't track down yet why this happens looking at FS
>>>>> sources.
>>>>
>>>> see response on that thread
>>>>
>>>>> Why I didn't open a bug on jira? Because FS behaves according to  
>>>>> the
>>>>> design and specs :-) But not according to real world  
>>>>> requirements...
>>>>
>>>> Really, people are trying to help you and your going to be snarky  
>>>> in
>>>> response?
>>>>
>>>>> On Wednesday 13 January 2010, Brian West wrote:
>>>>>> Can you elaborate on these "Critical" issues you seem to be  
>>>>>> having?
>>>
>>> Why
>>>
>>>>>> aren't you opening a jira for them if they are that critical to  
>>>>>> your
>>>>>> needs?
>>>>
>>>> Mike
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
>>>> s http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mrene_lists at avgs.ca  Thu Jan 14 08:49:06 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 14 Jan 2010 11:49:06 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001141141.59831.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
	<191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
	<201001141141.59831.sos@sokhapkin.dyndns.org>
Message-ID: <911CE0A0-5A54-4991-9642-98C236DC7678@avgs.ca>

Or try r16309, where Anthony fixed it so it doesnt hangup if the b-leg  
isnt answered yet :D

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 14-Jan-10, at 11:41 AM, Sergey Okhapkin wrote:

> Anthony, I apology, I didn't expect you'll be annoyed. As for the  
> issue, the
> issue is not in "proxy_media" setting, same happens when both  
> "proxy_media"
> and "bypass_media" are not set. The code does explicit leg A hangup in
> switch_ivr_bridge.c, line 513. The condition there is "if leg A is  
> up and leg
> A is not in answered state, then hangup leg A", this prevents  
> dialplan to
> continue execution on early media bridge termination.
>
> Issue when "bypass_media=true" is completely different, I'll open  
> the issue on
> jira.
>
> On Thursday 14 January 2010, Anthony Minessale wrote:
>> Sergey,
>>
>> The bug you reference was closed because proxy_media mode by design  
>> send's
>> the B leg's codec to the A
>> LEG so if there is a failure condition there is no easy graceful  
>> way to
>> back out and try another call.
>> I will try to make it possible if yo post a bounty, I estimate a  
>> minimum of
>> $1000USD in consulting time.
>>
>> The other one I might look at once you have apologized.
>>
>> I can probably add some code to make the bridge exit without  
>> terminating
>> the A leg even when hangup_after_bridge=true in the case where the  
>> the B
>> leg is not answered but right now I am sort of annoyed with your  
>> attitude
>> in this thread.
>>
>>
>> On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin
>>
>> <sos at sokhapkin.dyndns.org>wrote:
>>> Case 1 (bypass_media is off) is already on jira,
>>> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
>>> installation with latest trunk to reproduce case 2, when
>>> bypass_media=true and 10 seconds delay happens when 18X and then  
>>> error
>>> are received on leg b.
>>>
>>> On Wednesday 13 January 2010, Michael Jerris wrote:
>>>> On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
>>>>> Critical issues are when SIP error come after 18X provisional
>>>>> response.
>>>>>
>>>>> - if bypass_media is false then dialplan stops and leg a is
>>>>> explicitly hang up (switch_ivr_bridge.c, line 513).
>>>>
>>>> behavior can be modified with continue_on_fail and  
>>>> hangup_after_bridge
>>>> channel vars, perhaps ignore_early_media as well
>>>>
>>>>> - if bypass_media is true, then dialplan continue, but there is 10
>>>>> seconds delay before next bridge application sends INVITE to  
>>>>> gateway
>>>>> (
>>>
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
>>>
>>>>> 4.html ) I didn't track down yet why this happens looking at FS
>>>>> sources.
>>>>
>>>> see response on that thread
>>>>
>>>>> Why I didn't open a bug on jira? Because FS behaves according to  
>>>>> the
>>>>> design and specs :-) But not according to real world  
>>>>> requirements...
>>>>
>>>> Really, people are trying to help you and your going to be snarky  
>>>> in
>>>> response?
>>>>
>>>>> On Wednesday 13 January 2010, Brian West wrote:
>>>>>> Can you elaborate on these "Critical" issues you seem to be  
>>>>>> having?
>>>
>>> Why
>>>
>>>>>> aren't you opening a jira for them if they are that critical to  
>>>>>> your
>>>>>> needs?
>>>>
>>>> Mike
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
>>>> s http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From info at daccii.it  Thu Jan 14 08:50:21 2010
From: info at daccii.it (Albano Daniele Salvatore - Lavoro)
Date: Thu, 14 Jan 2010 17:50:21 +0100
Subject: [Freeswitch-users] ESL Mono/Managed
In-Reply-To: <1E6792AC-17F2-4DE9-9798-431350A5E77D@freeswitch.org>
References: <1E6792AC-17F2-4DE9-9798-431350A5E77D@freeswitch.org>
Message-ID: <4B4F4B4D.1060509@daccii.it>

Hi,

i should start to use it in short for a software i wrote in C#, so if 
you wants i can write them :)

However, i can't start before the next week, my develoment machine broke 
so i'm waiting some pieces to replace the broke stuff :\ I should be 
ready at the start of the next week.

Best Regards,
Daniele

Brian West ha scritto:
> I need someone that knows C# to write up some examples using ESL in C#
> 
> cd libs/esl
> make managedmod
> cd managed
> 
> Look at the perl example single_command.pl
> require ESL;
> 
> my $command = shift;
> my $args = join(" ", @ARGV);
> 
> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
> my $e = $con->api($command, $args);
> print $e->getBody();
> 
> 
> Need to write the same thing in C# and commit it as an example.  Also other examples exist in the perl directory those should be translated also if possible.
> 
> If you're interested please find me on IRC (bkw_)
> 
> Thanks,
> Brina
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rupa at rupa.com  Thu Jan 14 08:55:04 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 14 Jan 2010 10:55:04 -0600
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
Message-ID: <b39273531001140855i606a06a4ne55be84d825b1a74@mail.gmail.com>

I don't see an email2fax yet, but an important first step is in:

contrib/sathieu/email2pdf/email2pdf

On Thu, Jan 14, 2010 at 8:54 AM, Costa Zikalala <costa.zikalala at gmail.com>wrote:

> Hi All
>
> Has anyone worked on a email2fax script for mod_fax?
> If not how much would it cost for some genius here to quickly whip-up one?
>
> Ideally both email2fax and fax2email should come standard with mod_fax or
> is this not possible?
>
> Thanks
> Costa
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From brian at freeswitch.org  Thu Jan 14 08:57:40 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Jan 2010 10:57:40 -0600
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <191c3a031001140837m7cffcdd5w71886d6c8ba1dafe@mail.gmail.com>
References: <5d2828f1001131130s6ef166c0wc98f70023420c0fd@mail.gmail.com>
	<A2167AE7633A5C4DAE56EEBD1428ECC0F362E8@BESONE.corp.dom>
	<5d2828f1001140715m2aba7ed6n3cef077c64943e76@mail.gmail.com>
	<191c3a031001140837m7cffcdd5w71886d6c8ba1dafe@mail.gmail.com>
Message-ID: <2B0DD918-7E81-44B8-BF7F-0E9A0834A038@freeswitch.org>

And now with tcl bindings.  :P

/b

On Jan 14, 2010, at 10:37 AM, Anthony Minessale wrote:

> Don't forget that lua,perl,python,managed,java all share the same exact C++ source file with swig so the same exact api applies to all of the above.
> 
> On Thu, Jan 14, 2010 at 9:15 AM, Mike van Lammeren <mike at van.lammeren.net> wrote:
> Hi Claudio!
> 
> Thanks for the additions to the wiki! Every little bit helps.
> 
> I don't think I explained myself well, earlier. The point I was trying to make about the wiki is that rather than remove the section about "originate", it would be better to make an entry like "originate -- Does anyone know what this does?"
> 
> Mike van Lammeren

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From anthony.minessale at gmail.com  Thu Jan 14 08:58:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 10:58:24 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001141141.59831.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
	<191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
	<201001141141.59831.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031001140858i8ef1b09kf5d547914b5b8b68@mail.gmail.com>

r16310

Thank you for the apology.

On Thu, Jan 14, 2010 at 10:41 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Anthony, I apology, I didn't expect you'll be annoyed. As for the issue,
> the
> issue is not in "proxy_media" setting, same happens when both "proxy_media"
> and "bypass_media" are not set. The code does explicit leg A hangup in
> switch_ivr_bridge.c, line 513. The condition there is "if leg A is up and
> leg
> A is not in answered state, then hangup leg A", this prevents dialplan to
> continue execution on early media bridge termination.
>
> Issue when "bypass_media=true" is completely different, I'll open the issue
> on
> jira.
>
> On Thursday 14 January 2010, Anthony Minessale wrote:
> > Sergey,
> >
> > The bug you reference was closed because proxy_media mode by design
> send's
> > the B leg's codec to the A
> > LEG so if there is a failure condition there is no easy graceful way to
> > back out and try another call.
> > I will try to make it possible if yo post a bounty, I estimate a minimum
> of
> > $1000USD in consulting time.
> >
> > The other one I might look at once you have apologized.
> >
> > I can probably add some code to make the bridge exit without terminating
> > the A leg even when hangup_after_bridge=true in the case where the the B
> > leg is not answered but right now I am sort of annoyed with your attitude
> > in this thread.
> >
> >
> > On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin
> >
> > <sos at sokhapkin.dyndns.org>wrote:
> > > Case 1 (bypass_media is off) is already on jira,
> > > http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
> > > installation with latest trunk to reproduce case 2, when
> > > bypass_media=true and 10 seconds delay happens when 18X and then error
> > > are received on leg b.
> > >
> > > On Wednesday 13 January 2010, Michael Jerris wrote:
> > > > On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> > > > > Critical issues are when SIP error come after 18X provisional
> > > > > response.
> > > > >
> > > > > - if bypass_media is false then dialplan stops and leg a is
> > > > > explicitly hang up (switch_ivr_bridge.c, line 513).
> > > >
> > > >  behavior can be modified with continue_on_fail and
> hangup_after_bridge
> > > > channel vars, perhaps ignore_early_media as well
> > > >
> > > > > - if bypass_media is true, then dialplan continue, but there is 10
> > > > > seconds delay before next bridge application sends INVITE to
> gateway
> > > > > (
> > >
> > >
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
> > >
> > > > >4.html ) I didn't track down yet why this happens looking at FS
> > > > > sources.
> > > >
> > > > see response on that thread
> > > >
> > > > > Why I didn't open a bug on jira? Because FS behaves according to
> the
> > > > > design and specs :-) But not according to real world
> requirements...
> > > >
> > > > Really, people are trying to help you and your going to be snarky in
> > > > response?
> > > >
> > > > > On Wednesday 13 January 2010, Brian West wrote:
> > > > >> Can you elaborate on these "Critical" issues you seem to be
> having?
> > >
> > >  Why
> > >
> > > > >> aren't you opening a jira for them if they are that critical to
> your
> > > > >> needs?
> > > >
> > > > Mike
> > > >
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-user
> > > >s http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Thu Jan 14 09:00:09 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Jan 2010 11:00:09 -0600
Subject: [Freeswitch-users] mod_java is moving to unsupported on Jan 20th.
Message-ID: <E9807833-85C3-493C-BA11-F7C3586F3E38@freeswitch.org>

If nobody steps up to take over maintenance of mod_java by the 20th of January we'll be moving it to unsupported.  If you wish to take over maintenance please contact me off list ASAP.

Thanks,
Brian West
FreeSWITCH


From sos at sokhapkin.dyndns.org  Thu Jan 14 09:00:10 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Jan 2010 12:00:10 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <23BD1322-7882-4D43-80AB-27872096B4C9@avgs.ca>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141141.59831.sos@sokhapkin.dyndns.org>
	<23BD1322-7882-4D43-80AB-27872096B4C9@avgs.ca>
Message-ID: <201001141200.10315.sos@sokhapkin.dyndns.org>

"hangup_after_bridge=false" is the default setting.

On Thursday 14 January 2010, Mathieu Rene wrote:
> One way you could fix it is set "hangup_after_bridge=false" and then
> set "execute_on_answer=set\shangup_after_bridge=true". That will make
> it continue executing the dialplan atter a failure that came in after
> 183. Note that there is a side effect to this method, since audio was
> being relayed, you are forced to use the same codec on the 2nd
> carrier, best to tell them they are wrong failing a call after 183.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
> On 14-Jan-10, at 11:41 AM, Sergey Okhapkin wrote:
> > Anthony, I apology, I didn't expect you'll be annoyed. As for the
> > issue, the
> > issue is not in "proxy_media" setting, same happens when both
> > "proxy_media"
> > and "bypass_media" are not set. The code does explicit leg A hangup in
> > switch_ivr_bridge.c, line 513. The condition there is "if leg A is
> > up and leg
> > A is not in answered state, then hangup leg A", this prevents
> > dialplan to
> > continue execution on early media bridge termination.
> >
> > Issue when "bypass_media=true" is completely different, I'll open
> > the issue on
> > jira.
> >
> > On Thursday 14 January 2010, Anthony Minessale wrote:
> >> Sergey,
> >>
> >> The bug you reference was closed because proxy_media mode by design
> >> send's
> >> the B leg's codec to the A
> >> LEG so if there is a failure condition there is no easy graceful
> >> way to
> >> back out and try another call.
> >> I will try to make it possible if yo post a bounty, I estimate a
> >> minimum of
> >> $1000USD in consulting time.
> >>
> >> The other one I might look at once you have apologized.
> >>
> >> I can probably add some code to make the bridge exit without
> >> terminating
> >> the A leg even when hangup_after_bridge=true in the case where the
> >> the B
> >> leg is not answered but right now I am sort of annoyed with your
> >> attitude
> >> in this thread.
> >>
> >>
> >> On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin
> >>
> >> <sos at sokhapkin.dyndns.org>wrote:
> >>> Case 1 (bypass_media is off) is already on jira,
> >>> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
> >>> installation with latest trunk to reproduce case 2, when
> >>> bypass_media=true and 10 seconds delay happens when 18X and then
> >>> error
> >>> are received on leg b.
> >>>
> >>> On Wednesday 13 January 2010, Michael Jerris wrote:
> >>>> On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> >>>>> Critical issues are when SIP error come after 18X provisional
> >>>>> response.
> >>>>>
> >>>>> - if bypass_media is false then dialplan stops and leg a is
> >>>>> explicitly hang up (switch_ivr_bridge.c, line 513).
> >>>>
> >>>> behavior can be modified with continue_on_fail and
> >>>> hangup_after_bridge
> >>>> channel vars, perhaps ignore_early_media as well
> >>>>
> >>>>> - if bypass_media is true, then dialplan continue, but there is 10
> >>>>> seconds delay before next bridge application sends INVITE to
> >>>>> gateway
> >>>>> (
> >>>
> >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024
> >>>35
> >>>
> >>>>> 4.html ) I didn't track down yet why this happens looking at FS
> >>>>> sources.
> >>>>
> >>>> see response on that thread
> >>>>
> >>>>> Why I didn't open a bug on jira? Because FS behaves according to
> >>>>> the
> >>>>> design and specs :-) But not according to real world
> >>>>> requirements...
> >>>>
> >>>> Really, people are trying to help you and your going to be snarky
> >>>> in
> >>>> response?
> >>>>
> >>>>> On Wednesday 13 January 2010, Brian West wrote:
> >>>>>> Can you elaborate on these "Critical" issues you seem to be
> >>>>>> having?
> >>>
> >>> Why
> >>>
> >>>>>> aren't you opening a jira for them if they are that critical to
> >>>>>> your
> >>>>>> needs?
> >>>>
> >>>> Mike
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> >>>>r s http://www.freeswitch.org
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> >>>s http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From freeswitch at aastral.net  Thu Jan 14 09:20:23 2010
From: freeswitch at aastral.net (Bill W)
Date: Thu, 14 Jan 2010 12:20:23 -0500
Subject: [Freeswitch-users] Question about a1-hash
In-Reply-To: <5d2828f1001140827w386fe2acw478ae378b0ecfe53@mail.gmail.com>
References: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>	<4B4F430A.8080006@aastral.net>
	<5d2828f1001140827w386fe2acw478ae378b0ecfe53@mail.gmail.com>
Message-ID: <4B4F5257.6080508@aastral.net>

Hey Mike/Brian,

Okay, I'm missing something here.  Sure, you can calculate the response, 
but how are you going to validate that against what is in the database?

Thanks,
Bill






From mike at jerris.com  Thu Jan 14 09:20:31 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Jan 2010 12:20:31 -0500
Subject: [Freeswitch-users] playback and play_and_get_digits strange
	misunderstanding
In-Reply-To: <6d698be09281bb173cf0de33c1c2e44b@thom.fr.eu.org>
References: <6d698be09281bb173cf0de33c1c2e44b@thom.fr.eu.org>
Message-ID: <816ED94F-2501-46F0-ACE6-AD82011F8466@jerris.com>

Could you try this with latest trunk as well for comparison, I just changed the way the sounds base directory is set and I want to make sure I did not mess anything up.

Mike

On Jan 14, 2010, at 3:23 AM, Fran?ois Legal wrote:

> So I could (kind of) solve this by myself. There is in vars.xml the variable ${sound_prefix}. I did set it properly to my french sound path and then it worked.
> 
>  
> However, for the sake of discussion, I did try this with 1.0.5pre8 and the result was different :
> 
> with the extension
> 
>   <extension name="Disable Ringing on FXS">
>    <condition field="destination_number" expression="^#1234$">
>     <action application="set_global" data="dont_ring_fxs=true"/>
>     <action application="answer"/>
>     <action application="sleep" data="1000"/>
>     <action application="playback" data="misc/ringing_disabled.wav"/>
>     <action application="hangup" data="normal_clearing"/>
>    </condition>
>   </extension>
> 
> FS was trying to play the file ${FREESWITCH_PATH}/sounds/en/us/callie/misc/ringing_disabled.wav
> 
> with the extension
> 
>   <extension name="Disable Ringing on FXS">
>    <condition field="destination_number" expression="^#1234$">
>     <action application="set" data="default_language=fr"/>
>     <action application="set_global" data="dont_ring_fxs=true"/>
>     <action application="answer"/>
>     <action application="sleep" data="1000"/>
>     <action application="playback" data="misc/ringing_disabled.wav"/>
>     <action application="hangup" data="normal_clearing"/>
>    </condition>
>   </extension>
> 
> FS was trying to load ${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/ringing_disabled.wav
> 
> Now with the latest snapshot, with either one of the 2 mentionned extensions, FS tries to play the file in ${sound_prefix}/filepath/${codec_bit_rate}/filename whereas 1.0.5pre8 did not add the codec_bit_rate in the path but took care of the default_language variable.
> 
>  
> Fran?ois
> 
>  
> On Wed, 13 Jan 2010 16:32:16 +0100, Fran?ois Legal wrote:
> 
> Hello,
> 
>  
> trying to make so dialplan extensions that use the playback and play_and_get_digits applications, but I'm having trouble with the file name specification.
> 
>  
> The files I want to play are in the french language (fr/fr/julie as configured in lang/fr/fr.xml)
> 
> My extension is as follows :
> 
> 
>   <extension name="Disable Ringing on FXS">
>    <condition field="destination_number" expression="^#1234$">
>     <action application="set" data="default_language=fr"/>
>     <action application="set_global" data="dont_ring_fxs=true"/>
>     <action application="answer"/>
>     <action application="sleep" data="1000"/>
>     <action application="playback" data="misc/ringing_disabled.wav"/>
>     <action application="hangup" data="normal_clearing"/>
>    </condition>
>   </extension>
> 
> The channel is using a bit rate of 8000 Hz, so by the set default_language=fr I would expect freeswitch to playback the file at ${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/8000/ringing_disabled.wav whereas it tries to playback the file at ${FREESWITCH_PATH}/sounds/en/us/callie/misc/8000/ringing_disabled.wav
> 
> I have the same with play_and_get_digits application.
> 
>  
> What am I doing wrong ?
> 
>  
> Fran?ois
> 
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Thu Jan 14 09:21:35 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Jan 2010 12:21:35 -0500
Subject: [Freeswitch-users] sip trunk question: why call throughexternal
	profile is challenged?
In-Reply-To: <20100114082640.2C5D311F32@mail.nstel.ru>
References: <20100114082640.2C5D311F32@mail.nstel.ru>
Message-ID: <AD524CC0-726B-46D3-A22C-0B1F44C9B899@jerris.com>

if you look in the sample configs for the words blind and auth you will find all these settings, also you can setup acls for ip auth to not challenge.

Mike

On Jan 14, 2010, at 3:26 AM, Nikolay Kondratyev wrote:

> Mike, thanks for the reply.
>  
> Mmm? looks like I need more detailed instructions where to dig?
> Is there a way to turn off ?challenging? completely?
> I thought that <param name="auth-calls" value="false"/> should do it, but alas?
> By the way should this parameter be visible in either ?sofia status profile external? or ?sofia status gateway sipx4.lab.nstel.ru? ? I don?t see it?
>  
> I attached traces of failed and successful calls.
>  
> Thanks and regards,
> Nikolay.
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Wednesday, January 13, 2010 8:30 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] sip trunk question: why call throughexternal profile is challenged?
>  
> Look at how sipx sets up the users when they build the extensions and such for conferences, there was something special here, but I can't recall what.
>  
> Mike
>  
> On Jan 13, 2010, at 9:10 AM, Nikolay Kondratyev wrote:
> 
> 
> Hi all!
>  
> I?m brand new to FreeSwitch, but have some experience with SipX.
> Our company is evaluating FS.
> For test purposes I set up FS on a virtual machine (vmware esxi). I use CentOS.
> The FS version I use is 1.0.5-20100110-0400.
>  
> I have a question regarding sip trunk between FS and SipX.
> I created the following GW in external profile:
> [freeswitch at freeswitch external]$ cat sipx-lab.xml | grep -v '<!--'
> <include>
>   <gateway name="sipx4.lab.nstel.ru">
>   <param name="username" value="zxcv"/>
>   <param name="password" value="2007"/>
>   <param name="register" value="false"/>
>   </gateway>
> </include>
>  
> External.xml file is not modified after installation.
>  
> I see this gateway via fs_cli:
> freeswitch at internal> sofia status gateway sipx4.lab.nstel.ru
> =================================================================================================
> Name            sipx4.lab.nstel.ru
> Profile         external
> Scheme          Digest
> Realm           sipx4.lab.nstel.ru
> Username        zxcv
> Password        yes
> From            <sip:zxcv at sipx4.lab.nstel.ru;transport=udp>
> Contact         <sip:gw+sipx4.lab.nstel.ru at 172.23.22.49:5080;transport=udp;gw=sipx4.lab.nstel.ru>
> Exten           zxcv
> To              sip:zxcv at sipx4.lab.nstel.ru
> Proxy           sip:sipx4.lab.nstel.ru
> Context         public
> Expires         3600
> Freq            3600
> Ping            0
> PingFreq        0
> PingState       0/0/0
> State           NOREG
> Status          UP
> CallsIN         0
> CallsOUT        0
> =================================================================================================
>  
> I created new FS extension 2853. I registers (xlite) and I can call it.
>  
> Now I want to call FS user from sipx.
>  
> From the sipx side one can configure link to FS differently. There are two options:
> 1. Call FS directly trough sipxproxy (the core part of sipx works as sip proxy, not as b2bua)
> 2. Call trough embedded b2bua, named sipxbridge.
>  
> When a call is going trough sipxbridge, it is successfully landed at FS extension 2853.
> But when a call is going from the sipxproxy, it is challenged with Status 407 ?proxy authentication required?, and then call fails.
> (I?m not sure if sipx should handle this challenge, but this is separate question for the sipx forum).
> At the default log level I see nothing in the freeswitch.log.
>  
> So the question is why one call through external profile is being challenged while the other is not?
> I suspect that the reason is in the difference in the two Invite messages:
>  
> Here is challenged Invite:
> INVITE sip:2853 at fs.lab.nstel.ru:5080;transport=udp SIP/2.0
> Record-Route: <sip:172.23.12.104:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EOTRiZjI3MjgtYWMxNzBjZTktMTNjNC0yZjZkZWMtNjNkNmUwN2UtMmY2ZGVj.900_ntap%2Aid%7EMjIyNjAtMQ%60%60%213ba6d85f6946c4c6001bee1d3b54474f>
> From: "testphone3857"<sip:3857 at lab.nstel.ru>;tag=94bf2728-ac170ce9-13c4-2f6dec-63d6e07e-2f6dec
> To: <sip:2853 at lab.nstel.ru>
> Call-Id: 94bed2a0-ac170ce9-13c4-2f6dec-68ffdb81-2f6dec at lab.nstel.ru
> Cseq: 2 INVITE
> Via: SIP/2.0/UDP 172.23.12.104;branch=z9hG4bK-sipXecs-000eaf48a36fe4029c7cde004a6f44424847
> Via: SIP/2.0/TCP 172.23.12.104;branch=z9hG4bK-sipXecs-000bf3008cfa3f3e70470789c75232ba9499~ac7fd729330fd563f83cacd941311e75;id=22260-1
> Via: SIP/2.0/UDP 172.23.12.233:5060;branch=z9hG4bK-2f6dec-b9456412-7ad07784
> Max-Forwards: 18
> Supported: replaces
> User-Agent: LG-Nortel LIP 6812 v1.2.38sp SN/00405A18B634
> Contact: <sip:3857 at 172.23.12.233:5060;x-sipX-nonat>
> Proxy-Authorization: Digest username="3857",realm="lab.nstel.ru",nonce="e2152722611af1bbe59f3a8eb31b8eb84b4db2ca",uri="sip:2853 at lab.nstel.ru",response="a09c34fbd897783c9b9af50bd044ccaa",algorithm=MD5
> Content-Type: application/sdp
> Content-Length: 301
> Date: Wed, 13 Jan 2010 11:47:22 GMT
> Expires: 60
> X-Sipx-Handled: X172.23.12.104-81.211.30.104
>  
> v=0
> o=LGEIPP 16246 16247 IN IP4 172.23.12.233
> s=SIP Call
> c=IN IP4 172.23.12.233
> t=0 0
> m=audio 23008 RTP/AVP 0 8 18 4 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:18 annexb=no
> a=fmtp:101 0-11
> a=sendrecv
>  
>  
> And here is successful Invite:
> INVITE sip:2853 at fs.lab.nstel.ru;user=phone SIP/2.0
> Call-ID: 94bec3b8-ac170ce9-13c4-2f6d47-2fee09d0-2f6d47 at lab.nstel.ru.0
> CSeq: 1 INVITE
> From: "testphone3857" <sip:3857 at 172.23.12.104>;tag=816640414244159961
> To: <sip:2853 at fs.lab.nstel.ru;user=phone>
> Via: SIP/2.0/UDP 172.23.12.104:5080;branch=z9hG4bK885dde2c8680f5315845cd3350b8b605373534
> Max-Forwards: 70
> User-Agent: sipXecs/4.0.2 sipXecs/sipxbridge (Linux)
> P-Asserted-Identity: <sip:1004 at 172.23.12.104>
> Contact: <sip:1004 at 172.23.12.104:5080;transport=udp>
> Route: <sip:172.23.22.49:5080;transport=udp;lr>
> Session-Expires: 1800;refresher=uac
> Allow: INVITE,BYE,ACK,CANCEL,OPTIONS
> Content-Type: application/sdp
> Content-Length: 315
>  
> v=0
> o=sipxbridge 6640787141824452741 1 IN IP4 172.23.12.104
> s=SIP Call
> c=IN IP4 172.23.12.104
> t=0 0
> m=audio 30248 RTP/AVP 0 8 18 4 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:18 annexb=no
> a=fmtp:101 0-11
> a=sendrecv
>  
> Can somebody please if it is a FS configuration problem or a software problem or is it a problem on the sipx side?
>  
> Thanks in advance,
> Nikolay.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
>  
> <call_from_sipx_via_unmanaged_gw.pcap><call_from_sipx_via_siptrunk_sip-only.pcap>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Thu Jan 14 09:21:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 11:21:43 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001141200.10315.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141141.59831.sos@sokhapkin.dyndns.org>
	<23BD1322-7882-4D43-80AB-27872096B4C9@avgs.ca>
	<201001141200.10315.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031001140921n5de0a9eakeb3144aba91c78d5@mail.gmail.com>

ok both methods work now.
your jira was fixed in r16311 and the non bypass equivalent was fixed in the
previous commit

On Thu, Jan 14, 2010 at 11:00 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> "hangup_after_bridge=false" is the default setting.
>
> On Thursday 14 January 2010, Mathieu Rene wrote:
> > One way you could fix it is set "hangup_after_bridge=false" and then
> > set "execute_on_answer=set\shangup_after_bridge=true". That will make
> > it continue executing the dialplan atter a failure that came in after
> > 183. Note that there is a side effect to this method, since audio was
> > being relayed, you are forced to use the same codec on the 2nd
> > carrier, best to tell them they are wrong failing a call after 183.
> >
> > Mathieu Rene
> > Avant-Garde Solutions Inc
> > Office: + 1 (514) 664-1044 x100
> > Cell: +1 (514) 664-1044 x200
> > mrene at avgs.ca
> >
> > On 14-Jan-10, at 11:41 AM, Sergey Okhapkin wrote:
> > > Anthony, I apology, I didn't expect you'll be annoyed. As for the
> > > issue, the
> > > issue is not in "proxy_media" setting, same happens when both
> > > "proxy_media"
> > > and "bypass_media" are not set. The code does explicit leg A hangup in
> > > switch_ivr_bridge.c, line 513. The condition there is "if leg A is
> > > up and leg
> > > A is not in answered state, then hangup leg A", this prevents
> > > dialplan to
> > > continue execution on early media bridge termination.
> > >
> > > Issue when "bypass_media=true" is completely different, I'll open
> > > the issue on
> > > jira.
> > >
> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> > >> Sergey,
> > >>
> > >> The bug you reference was closed because proxy_media mode by design
> > >> send's
> > >> the B leg's codec to the A
> > >> LEG so if there is a failure condition there is no easy graceful
> > >> way to
> > >> back out and try another call.
> > >> I will try to make it possible if yo post a bounty, I estimate a
> > >> minimum of
> > >> $1000USD in consulting time.
> > >>
> > >> The other one I might look at once you have apologized.
> > >>
> > >> I can probably add some code to make the bridge exit without
> > >> terminating
> > >> the A leg even when hangup_after_bridge=true in the case where the
> > >> the B
> > >> leg is not answered but right now I am sort of annoyed with your
> > >> attitude
> > >> in this thread.
> > >>
> > >>
> > >> On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin
> > >>
> > >> <sos at sokhapkin.dyndns.org>wrote:
> > >>> Case 1 (bypass_media is off) is already on jira,
> > >>> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
> > >>> installation with latest trunk to reproduce case 2, when
> > >>> bypass_media=true and 10 seconds delay happens when 18X and then
> > >>> error
> > >>> are received on leg b.
> > >>>
> > >>> On Wednesday 13 January 2010, Michael Jerris wrote:
> > >>>> On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> > >>>>> Critical issues are when SIP error come after 18X provisional
> > >>>>> response.
> > >>>>>
> > >>>>> - if bypass_media is false then dialplan stops and leg a is
> > >>>>> explicitly hang up (switch_ivr_bridge.c, line 513).
> > >>>>
> > >>>> behavior can be modified with continue_on_fail and
> > >>>> hangup_after_bridge
> > >>>> channel vars, perhaps ignore_early_media as well
> > >>>>
> > >>>>> - if bypass_media is true, then dialplan continue, but there is 10
> > >>>>> seconds delay before next bridge application sends INVITE to
> > >>>>> gateway
> > >>>>> (
> > >>>
> > >>>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024
> > >>>35
> > >>>
> > >>>>> 4.html ) I didn't track down yet why this happens looking at FS
> > >>>>> sources.
> > >>>>
> > >>>> see response on that thread
> > >>>>
> > >>>>> Why I didn't open a bug on jira? Because FS behaves according to
> > >>>>> the
> > >>>>> design and specs :-) But not according to real world
> > >>>>> requirements...
> > >>>>
> > >>>> Really, people are trying to help you and your going to be snarky
> > >>>> in
> > >>>> response?
> > >>>>
> > >>>>> On Wednesday 13 January 2010, Brian West wrote:
> > >>>>>> Can you elaborate on these "Critical" issues you seem to be
> > >>>>>> having?
> > >>>
> > >>> Why
> > >>>
> > >>>>>> aren't you opening a jira for them if they are that critical to
> > >>>>>> your
> > >>>>>> needs?
> > >>>>
> > >>>> Mike
> > >>>>
> > >>>>
> > >>>> _______________________________________________
> > >>>> FreeSWITCH-users mailing list
> > >>>> FreeSWITCH-users at lists.freeswitch.org
> > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-use
> > >>>>r s http://www.freeswitch.org
> > >>>
> > >>> _______________________________________________
> > >>> FreeSWITCH-users mailing list
> > >>> FreeSWITCH-users at lists.freeswitch.org
> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >>>s http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Thu Jan 14 09:23:21 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 11:23:21 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001140841n603ddc45n972b3aba83600d01@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<632944D3-BDFA-46C2-AB56-21572CD28F86@jerris.com>
	<201001140719.12363.sos@sokhapkin.dyndns.org>
	<191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>
	<191c3a031001140826q2d014484v8807bf3b98b78f63@mail.gmail.com>
	<9853f4ff1001140841n603ddc45n972b3aba83600d01@mail.gmail.com>
Message-ID: <191c3a031001140923i72afe059ja18c6911c60c81e2@mail.gmail.com>

David I already understood your request.  Even what you consider a
compromise is a lot of work.
You really should post a bounty for something this exotic.


On Thu, Jan 14, 2010 at 10:41 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Anthony,
>
>      Thanks. I understand it's complicated. Another option is to be able to
> configure whether or not to discard all other B-legs on receiving media but
> on receiving an final code like 200. This way we will still get the first
> media that arrived but provide an answer on the actual channel that provided
> the 200.
>
> My request comes from 2 sides, one commercial and the other personal. For
> the commercial I just need to now whether it is possible by some tweaking so
> the functionality can be offered.
>
> The other, I've been involved in FS as a user from the very beginning
> (although I was absent for some time), and I'd like to help in the
> development of the project by letting the community know what's out there
> commercially speaking.
>
>
> Hope you can help.
>
> David
>
> On Thu, Jan 14, 2010 at 5:26 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> David,
>>
>> Your request is somewhat challenging because we have to make sure
>> FreeSWITCH is agnostic in protocols and that aside, we also have a very
>> complicated and feature-rich originate API that does not currently support
>> sending audio from one leg to the A leg while it's trying to call 10 other B
>> legs.  Please try to understand that this request is a very special side
>> case and we would have to do many hours of work to make it possible.  I am
>> not sure if you are simply asking if its possible or if you want us to
>> implement it for you but I am afraid it would fall under commercial support
>> to undertake that unique of a feature that only helps a very small
>> percentage of our user base considering all the work we already have to do
>> every day.
>>
>>
>>
>>
>> On Thu, Jan 14, 2010 at 10:22 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> Sergey,
>>>
>>> The bug you reference was closed because proxy_media mode by design
>>> send's the B leg's codec to the A
>>> LEG so if there is a failure condition there is no easy graceful way to
>>> back out and try another call.
>>> I will try to make it possible if yo post a bounty, I estimate a minimum
>>> of $1000USD in consulting time.
>>>
>>> The other one I might look at once you have apologized.
>>>
>>> I can probably add some code to make the bridge exit without terminating
>>> the A leg even when hangup_after_bridge=true in the case where the the B leg
>>> is not answered but right now I am sort of annoyed with your attitude in
>>> this thread.
>>>
>>>
>>>
>>> On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin <
>>> sos at sokhapkin.dyndns.org> wrote:
>>>
>>>> Case 1 (bypass_media is off) is already on jira,
>>>> http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
>>>> installation with latest trunk to reproduce case 2, when
>>>> bypass_media=true
>>>> and 10 seconds delay happens when 18X and then error are received on leg
>>>> b.
>>>>
>>>> On Wednesday 13 January 2010, Michael Jerris wrote:
>>>> > On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
>>>> > > Critical issues are when SIP error come after 18X provisional
>>>> response.
>>>> > >
>>>> > > - if bypass_media is false then dialplan stops and leg a is
>>>> explicitly
>>>> > > hang up (switch_ivr_bridge.c, line 513).
>>>> >
>>>> >  behavior can be modified with continue_on_fail and
>>>> hangup_after_bridge
>>>> > channel vars, perhaps ignore_early_media as well
>>>> >
>>>> > > - if bypass_media is true, then dialplan continue, but there is 10
>>>> > > seconds delay before next bridge application sends INVITE to gateway
>>>> (
>>>> > >
>>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
>>>> > >4.html ) I didn't track down yet why this happens looking at FS
>>>> sources.
>>>> >
>>>> > see response on that thread
>>>> >
>>>> > > Why I didn't open a bug on jira? Because FS behaves according to the
>>>> > > design and specs :-) But not according to real world requirements...
>>>> >
>>>> > Really, people are trying to help you and your going to be snarky in
>>>> > response?
>>>> >
>>>> > > On Wednesday 13 January 2010, Brian West wrote:
>>>> > >> Can you elaborate on these "Critical" issues you seem to be having?
>>>>  Why
>>>> > >> aren't you opening a jira for them if they are that critical to
>>>> your
>>>> > >> needs?
>>>> >
>>>> > Mike
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From Claudio.Cavalera at italtel.it  Thu Jan 14 09:25:08 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Thu, 14 Jan 2010 18:25:08 +0100
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <5d2828f1001140715m2aba7ed6n3cef077c64943e76@mail.gmail.com>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F362EA@BESONE.corp.dom>

We are telling the same thing Mike, I've also written to elaborate the
"originate" entry on mod_lua wiki.
We could take examples from here maybe:
http://wiki.freeswitch.org/wiki/Session_originate
 
ciao,
claudio


________________________________

	From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike
van Lammeren
	Sent: Thursday, January 14, 2010 4:16 PM
	To: freeswitch-users at lists.freeswitch.org
	Subject: Re: [Freeswitch-users] playing with sessions in lua
	
	
	Hi Claudio! 

	Thanks for the additions to the wiki! Every little bit helps.

	I don't think I explained myself well, earlier. The point I was
trying to make about the wiki is that rather than remove the section
about "originate", it would be better to make an entry like "originate
-- Does anyone know what this does?"

	Mike van Lammeren


	On Thu, Jan 14, 2010 at 6:23 AM, Cavalera Claudio Luigi
<Claudio.Cavalera at italtel.it> wrote:
	

		Hi Mike,
		in fact i've completed that page with the list of
available  session functions.
		 
		I've not removed "session:originate" yet, but it would
be better if someone could provide an example in order to write an
example in the wiki.
		I've added this valuable example also with the help of
rupa
	
http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.Session
		 
		I would like to write something also about api_on_answer
to use an api instead of a dialplan application.
		 
		BRs,
		Claudio


________________________________

			From:
freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike
van Lammeren
			Sent: Wednesday, January 13, 2010 8:30 PM 

			To: freeswitch-users at lists.freeswitch.org
			Subject: Re: [Freeswitch-users] playing with
sessions in lua
			

			Hello! 

			Before you remove "session:originate" from the
wiki, you should take a look at this:
	
http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocument
ed_Session_Functions.3F

	
<http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_find_useful_undocumen
ted_Session_Functions.3F> There is, in fact, a function called
"originate".

			Mike van Lammeren


			On Wed, Jan 13, 2010 at 5:22 AM, Cavalera
Claudio Luigi <Claudio.Cavalera at italtel.it> wrote:
			

				Thanks a lot Anthony,
				some comments inline (and please forgive
me for my broken email client).
				 
				
				>>  example1: Consider this simple lua
script in which i create two sessions:
				
				 >>  api = freeswitch.API();
				 >>  api:execute("originate",
"sofia/internal/1001%192.168.1.1 &park");
				 >>  api:execute("originate",
"sofia/internal/1002%192.168.1.1 &park");
				
				 >  capture the output from api:execute
the uuid is in there
				
				
				Thx a lot,
				this was one piece i was missing
although it's already on the wiki here:
	
http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls 
				


				 > because lua calls it
freeswitch.bridge
				
				 > session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
				 > session2 =
freeswitch.Session("sofia/internal/1002%192.168.1.1", session1);
				 > freeswitch.bridge(session1,
session2); 
				 
				good to now, there isn't any example of
freeswitch.bridge in the wiki and i would like to add one.
				Where I could find the full api of
				freeswitch.Session( ) ?
				because I've seen this working also
without "session1" in the second line:
				
				session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
				
				session2 =
freeswitch.Session("sofia/internal/1002%192.168.1.1");
				freeswitch.bridge(session1, session2); 
				
				 also is there any difference between
freeswitch.bridge and freeswitch.execute(uuid_bridge ...) ?
				 
				 >> example3: yet another possibility
				 >> local session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
				 >> session1:originate(session2,
"sofia/internal/1002%192.168.1.1", 1000);
				 >> but it does not work either.
				
				 > The above is gibberish try:
				 > local session1 =
freeswitch.Session("sofia/internal/1001%192.168.1.1");
				 > session1:execute("bridge",
"sofia/internal/1002%192.168.1.1"); 
				 
				Okay i will report also this bridge
example on the wiki which was missing.
				But does session:originate make sense in
some cases or not? Otherwise i'm going to remove this line on the wiki
	
http://wiki.freeswitch.org/wiki/Mod_lua#session:originate
<http://wiki.freeswitch.org/wiki/Mod_lua#session:originate> 
				 
				Thanks,
				Claudio

				 

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materia di protezione dei dati personali). Se avete ricevuto questa
comunicazione per errore, vi preghiamo di darne immediata notizia al
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_______________________________________________
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From brian at freeswitch.org  Thu Jan 14 09:30:55 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Jan 2010 11:30:55 -0600
Subject: [Freeswitch-users] Question about a1-hash
In-Reply-To: <4B4F5257.6080508@aastral.net>
References: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>	<4B4F430A.8080006@aastral.net>
	<5d2828f1001140827w386fe2acw478ae378b0ecfe53@mail.gmail.com>
	<4B4F5257.6080508@aastral.net>
Message-ID: <41473BE6-E347-4FCD-AD15-9964265C53EB@freeswitch.org>

You take what is in the data base and create an a1-hash if you have the parts to create it.. if not your SOL.

/b

On Jan 14, 2010, at 11:20 AM, Bill W wrote:

> Hey Mike/Brian,
> 
> Okay, I'm missing something here.  Sure, you can calculate the response, 
> but how are you going to validate that against what is in the database?
> 
> Thanks,
> Bill




From mike at van.lammeren.net  Thu Jan 14 09:36:04 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 14 Jan 2010 12:36:04 -0500
Subject: [Freeswitch-users] Question about a1-hash
In-Reply-To: <4B4F5257.6080508@aastral.net>
References: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>
	<4B4F430A.8080006@aastral.net>
	<5d2828f1001140827w386fe2acw478ae378b0ecfe53@mail.gmail.com>
	<4B4F5257.6080508@aastral.net>
Message-ID: <5d2828f1001140936t62dbebbdg85489d203e4f3295@mail.gmail.com>

Oh, I still have to go with my solution of using the MD5 hash of the
password for the phones. Since I cannot change the database for other
reasons, I am still unable to generate the a1-hash. Like FreeSWITCH, my
script doesn't know the password either!

All the same, I'm glad that FreeSWITCH uses digest authentication. It is a
good way to implement authentication.


Mike van Lammeren

On Thu, Jan 14, 2010 at 12:20 PM, Bill W <freeswitch at aastral.net> wrote:

> Hey Mike/Brian,
>
> Okay, I'm missing something here.  Sure, you can calculate the response,
> but how are you going to validate that against what is in the database?
>
> Thanks,
> Bill
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From djbinter at yahoo.com  Thu Jan 14 09:39:19 2010
From: djbinter at yahoo.com (DJB)
Date: Thu, 14 Jan 2010 09:39:19 -0800 (PST)
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <43692088-738A-475C-8D60-E40D296ECEB1@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
	<201001132326.41354.sos@sokhapkin.dyndns.org>
	<52E8BE9E-B6D6-411C-9080-697AF24017B4@jerris.com>
	<191c3a031001132114s18ed2baw2ff802dbd615fe01@mail.gmail.com>
	<191c3a031001132115u2d88bab7p7cbb27ab1eaad466@mail.gmail.com>
	<9853f4ff1001140226y2b035f37rfcafb03190435ec1@mail.gmail.com>
	<43692088-738A-475C-8D60-E40D296ECEB1@jerris.com>
Message-ID: <407935.47740.qm@web37505.mail.mud.yahoo.com>

What about sending Sip 183 with SDP (no 200OK), so that your customers can hear recordings?

djbinter 




________________________________
From: Michael Jerris <mike at jerris.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, January 14, 2010 8:12:59 AM
Subject: Re: [Freeswitch-users] multiple leg and multiple rtp

Okay, so that is a very interesting use case.   Seems a bizarre way to do this for the carrier, but interesting none the less.  I'd say the chances of actually muxing the early media is small, what might be possible would be something to say which b legs media to pass along to the a leg.  I have not looked at all at how complicated this is, it will be down deep somewhere in switch_ivr_originate code around where we if for ignore_early_media.  This code is pretty complex, I can't say that we will ever actually add this functionality, but at least now I won't blow it off as complete nonsense.

Mike





On Jan 14, 2010, at 5:26 AM, David Villasmil wrote:

Hello again,
>
>Using ingore early media only ignores ALL media, that's not what I need. At least in europe i've seen it many times, MNOs provide a service with which you can have a song played to the caller while the call is connecting to your cell phone. This is basically what I'm trying to achieve.
>
>The content provider is in possesions of the media and it require us to fork the call and send a SIP INVITE to the on one leg and the call to the destination number on another leg. they will only provide a progress, no answer.
>
>I CAN do it locally by playing the file as a custom ringback but that's not the standard in terms of commercial use of the content.
>
>Is there any way to modify the behaviour of fs when i receives media? Let's say i.e. it doesn't drop the other leg, but provides the first early media it receives and just wait for some channel to answer? I will not have both media but it would work.
>
>thanks
>
>David
>
>
>On Thu, Jan 14, 2010 at 6:15 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
>Perhaps best not to help him anymore without an apology for the snap judgement and comparison to asterisk clearly designed to push our buttons.
>>>>
>>We'll be here when you realize we were trying to help you but I can't promise we will still have paitence......
>>
>>
>>On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike at jerris.com> wrote:
>>>
>>>You already are running on trunk.
>>>
>>>>>>Mike
>>>
>>>>>>On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote:
>>>
>>>> I run FreeSWITCH Version 1.0.5pre10 (16012M...
>>>_______________________________________________
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists...
>>_______________________________________________
>>>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>_______________________________________________
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>



      
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From vgribanovsky at rnd.stcnet.ru  Thu Jan 14 08:58:14 2010
From: vgribanovsky at rnd.stcnet.ru (Vadim Gribanovsky)
Date: Thu, 14 Jan 2010 19:58:14 +0300
Subject: [Freeswitch-users] FS can't direct call to SBC
Message-ID: <1263488294.2918.26.camel@localhost.localdomain>

Hi all!
        
        Let me ask FS experts and users to help to resolve following
        problem:
        
        /////////////////////////////////////////////////////////////////////////////////////////////////
        1.VOIP test environment:
        =>local (10.10.10.235/24) FS(v1.0.4) installation + 
        =>local IP-Phone (GXP2020) +
        =>SBC (acts as stateless proxy) provides secure access (TLS
        +SRTP) to FS
        for +
        =>external NATed IP-PHONE (PJSIPua)
        as depicted below:
        PHONE(10.10.10.31) <-udp-> FS(10.10.10.235) <-tcp-> (Local
        10.10.10.238)SBC(Public P.U.B.LS)<-tls-> INTERNET<-tls->
        NAT(Public
        P.U.B.LN) <-tls-> PHONE (10.10.10.21)
        
        2.Problem:
        => local IP-Phone get registration to FS, -OK!
        => external NATed IP-PHONE get registration to FS via SBC, -OK!
        => FS try to send NOTIFY request to external NATed IP-PHONE
        using
        [sip_contact_host:port] record instead
        [sip_received_ip:port](SBC) -
        FAIL!
        => external NATed IP-PHONE calls local IP-Phone - OK!
        => local IP-Phone couldn't call external NATed IP-PHONE - FS
        redirect to
        voicemail - FAIL!
        
        3. Logs:
        
        =>FS CLI
        API CALL [sofia(status)] output:
                             Name          Type
        Data      State
        =================================================================================================
                     vfsp_profile       profile
        sip:mod_sofia at 10.10.10.235:5090      RUNNING (0)
                          vfsp_gw       gateway
        sip:FS_VFSP at 10.10.10.238      NOREG
                         internal       profile
        sip:mod_sofia at 10.10.10.235:5060      RUNNING (0)
                    internal-ipv6       profile
        sip:mod_sofia@[::1]:5060      RUNNING (0)
                     10.10.10.235         alias
        internal      ALIASED
        =================================================================================================
        3 profiles 1 alias
        
        API CALL [sofia(status profile internal)] output:
        =================================================================================================
        Name                    internal
        Domain Name             N/A
        DBName                  sofia_reg_internal
        Pres Hosts       
        Dialplan                XML
        Context                 public
        Challenge Realm         auto_from
        RTP-IP                  10.10.10.235
        Ext-RTP-IP              10.10.10.235
        SIP-IP                  10.10.10.235
        Ext-SIP-IP              10.10.10.235
        URL                     sip:mod_sofia at 10.10.10.235:5060
        BIND-URL                sip:mod_sofia at 10.10.10.235:5060
        HOLD-MUSIC              local_stream://moh
        OUTBOUND-PROXY          N/A
        CODECS
        G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
        TEL-EVENT               101
        DTMF-MODE               rfc2833
        CNG                     13
        SESSION-TO              0
        MAX-DIALOG              0
        NOMEDIA                 false
        LATE-NEG                false
        PROXY-MEDIA             false
        AGGRESSIVENAT           false
        STUN-ENABLED            true
        STUN-AUTO-DISABLE       false
        CALLS-IN                3
        FAILED-CALLS-IN         0
        CALLS-OUT               0
        FAILED-CALLS-OUT        0
        
        Registrations:
        =================================================================================================
        Call-ID:        db639b76d2884388 at 10.10.10.31
        User:           1010 at 10.10.10.235
        Contact:        "user"
        <sip:1010 at 10.10.10.31:5060;transport=udp;user=phone>
        Agent:          Grandstream GXP2020 1.2.1.4
        Status:         Registered(UDP)(unknown) EXP(2010-01-14
        16:00:09)
        Host:           aq
        IP:             10.10.10.31
        Port:           5060
        Auth-User:      1010
        Auth-Realm:     10.10.10.235
        
        =================================================================================================
        
        
        API CALL [sofia(status profile vfsp_profile)] output:
        =================================================================================================
        Name                    vfsp_profile
        Domain Name             N/A
        DBName                  sofia_reg_vfsp_profile
        Pres Hosts       
        Dialplan                XML
        Context                 public
        Challenge Realm         auto_to
        RTP-IP                  10.10.10.235
        Ext-RTP-IP              stun:stun.freeswitch.org
        SIP-IP                  10.10.10.235
        Ext-SIP-IP              P.U.B.L
        URL                     sip:mod_sofia at 10.10.10.235:5090
        BIND-URL                sip:mod_sofia at 10.10.10.235:5090
        HOLD-MUSIC              local_stream://moh
        OUTBOUND-PROXY          N/A
        CODECS                  PCMU,PCMA,GSM
        TEL-EVENT               101
        DTMF-MODE               rfc2833
        CNG                     13
        SESSION-TO              0
        MAX-DIALOG              0
        NOMEDIA                 false
        LATE-NEG                false
        PROXY-MEDIA             false
        AGGRESSIVENAT           true
        STUN-ENABLED            true
        STUN-AUTO-DISABLE       false
        CALLS-IN                0
        FAILED-CALLS-IN         0
        CALLS-OUT               0
        FAILED-CALLS-OUT        0
        
        Registrations:
        =================================================================================================
        Call-ID:        6197eb4f-72fa-428f-8c0e-5c5551d185e2
        User:           1007 at 10.10.10.235
        Contact:        "user" <sip:1007 at P.U.B.LN:43459;transport=tcp>
        Agent:          PJSUA v1.0.3/i686-pc-linux-gnu
        Status:         Registered(TCP)(unknown) EXP(2010-01-14
        16:04:46)
        Host:           aq
        IP:             10.10.10.238
        Port:           38782
        Auth-User:      1007
        Auth-Realm:     10.10.10.235
        
        =================================================================================================
        
        => sniffer
        pkt1. SBC -> FS
        REGISTER sip:10.10.10.235;transport=tcp SIP/2.0
        Record-Route: <sip:10.10.10.238:5060;transport=tcp;lr>
        Via: SIP/2.0/tcp
        10.10.10.238:36707;rport;branch=z9hG4bKPj66cc4b809a6bb1510af258991c1f0587
        Via: SIP/2.0/tcp
        10.10.10.21:32780;rport=43459;received=P.U.B.L;branch=z9hG4bKPj71ce8307-a505-4477-9c47-348cb947b1be
        Max-Forwards: 69
        From:
        <sip:1007 at 10.10.10.235>;tag=a702b4fb-d565-4627-9cff-5fb6bfed2922
        To: <sip:1007 at 10.10.10.235>
        Call-ID: b618fe07-a1ef-4eee-926c-ff8e8f3c9096
        CSeq: 49600 REGISTER
        User-Agent: PJSUA v1.0.3/i686-pc-linux-gnu
        Contact: <sip:1007 at P.U.B.LN:43459;transport=tcp>
        Expires: 300
        
        pkt2. FS -> SBC
        SIP/2.0 200 OK
        Via: SIP/2.0/TCP
        10.10.10.238:36707;rport=36707;branch=z9hG4bKPj66cc4b809a6bb1510af258991c1f0587
        Via: SIP/2.0/TCP
        10.10.10.21:32780;rport=43459;received=P.U.B.LN;branch=z9hG4bKPj71ce8307-a505-4477-9c47-348cb947b1be
        From:
        <sip:1007 at 10.10.10.235>;tag=a702b4fb-d565-4627-9cff-5fb6bfed2922
        To: <sip:1007 at 10.10.10.235>;tag=Brpemg1NcteDc
        Call-ID: b618fe07-a1ef-4eee-926c-ff8e8f3c9096
        CSeq: 49600 REGISTER
        Contact: <sip:1007 at P.U.B.LN:43459;transport=tcp>;expires=300
        Date: Thu, 14 Jan 2010 13:15:37 GMT
        User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
        
        pkt3. FS -> external NATed IP-PHONE (trying to send NOTIFY via
        default
        gateway instead SBC!!!)
        Internet Protocol, Src: 10.10.10.235 (10.10.10.235), Dst:
        P.U.B.LN
        (P.U.B.LN)
        Transmission Control Protocol, Src Port: 52304 (52304), Dst
        Port: 43459
        (43459), Seq: 0, Len: 0
        Flags: 0x02 (SYN)
        
        4. default gateway -> FS
        Internet Protocol, Src: P.U.B.LN (P.U.B.LN), Dst: 10.10.10.235
        (10.10.10.235)
        Transmission Control Protocol, Src Port: 43459 (43459), Dst
        Port: 52304
        (52304), Seq: 1, Ack: 1, Len: 0
        Flags: 0x14 (RST, ACK)
        ///////////////////////////////////////////////////////////////////////////////////////////////////////////
        
        The same happened when i use default FS configuration (with
        external profile).
        Can somebody push me to the right way?
        
        Thanks in advance,
        
        Vadim
        
        
        
        
        




From daletrub at gmail.com  Thu Jan 14 09:49:15 2010
From: daletrub at gmail.com (Dale Trub)
Date: Thu, 14 Jan 2010 09:49:15 -0800
Subject: [Freeswitch-users] ISP with SIP trunking for 10k+ ports?
Message-ID: <c23fe9f21001140949o372a2ae1kaaca6d43ba7a6fe1@mail.gmail.com>

We're looking for an ISP with SIP trunking and DIDs to host our FS servers,
and are having a hard time locating an ISP who meet these two requirements
a) can handle 10,000+ simultaneous inbound ports
b) returns our calls in timely manner

We have a six-figure annual budget for this service ... and so far the "big
guys" sales teams can't get it together to put together a bid.

The ideal partner would share with us the backend telcom revenue in the
neighborhood of .5c/min, but we're open to other arrangements.

Does anyone have any recommendations for ISPs with SIP trunking and DIDs
that they can share?  Specific salespeople contact info especially
appreciated!

Thanks,

Dale
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From freeswitch at aastral.net  Thu Jan 14 09:50:56 2010
From: freeswitch at aastral.net (Bill W)
Date: Thu, 14 Jan 2010 12:50:56 -0500
Subject: [Freeswitch-users] Question about a1-hash
In-Reply-To: <41473BE6-E347-4FCD-AD15-9964265C53EB@freeswitch.org>
References: <5d2828f1001140744u45029fb7mcb6106e1b1991b60@mail.gmail.com>	<4B4F430A.8080006@aastral.net>	<5d2828f1001140827w386fe2acw478ae378b0ecfe53@mail.gmail.com>	<4B4F5257.6080508@aastral.net>
	<41473BE6-E347-4FCD-AD15-9964265C53EB@freeswitch.org>
Message-ID: <4B4F5980.5000006@aastral.net>

Ah okay, it is just as I thought.  Either I store the a1-hash in the 
database, or store the password in clear text so I can calculate a1-hash 
on the fly.

Or as Mike says, have the phone password be the actual hash value of the 
password in the database, and then calculate an a1-hash with that hashed 
  password.


Thanks,
Bill W.


Brian West wrote:
> You take what is in the data base and create an a1-hash if you have the parts to create it.. if not your SOL.
> 
> /b
> 
> On Jan 14, 2010, at 11:20 AM, Bill W wrote:
> 
>



From costa.zikalala at gmail.com  Thu Jan 14 09:58:10 2010
From: costa.zikalala at gmail.com (Costa Zikalala)
Date: Thu, 14 Jan 2010 19:58:10 +0200
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <b39273531001140855i606a06a4ne55be84d825b1a74@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com> 
	<b39273531001140855i606a06a4ne55be84d825b1a74@mail.gmail.com>
Message-ID: <59daa2cd1001140958x49f64247r554d76d596d1264f@mail.gmail.com>

Thank you so much for this Rupa.



2010/1/14 Rupa Schomaker <rupa at rupa.com>

> I don't see an email2fax yet, but an important first step is in:
>
> contrib/sathieu/email2pdf/email2pdf
>
> On Thu, Jan 14, 2010 at 8:54 AM, Costa Zikalala <costa.zikalala at gmail.com>wrote:
>
>> Hi All
>>
>> Has anyone worked on a email2fax script for mod_fax?
>> If not how much would it cost for some genius here to quickly whip-up one?
>>
>> Ideally both email2fax and fax2email should come standard with mod_fax or
>> is this not possible?
>>
>> Thanks
>> Costa
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Jan 14 09:59:21 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 11:59:21 -0600
Subject: [Freeswitch-users] ISP with SIP trunking for 10k+ ports?
In-Reply-To: <c23fe9f21001140949o372a2ae1kaaca6d43ba7a6fe1@mail.gmail.com>
References: <c23fe9f21001140949o372a2ae1kaaca6d43ba7a6fe1@mail.gmail.com>
Message-ID: <191c3a031001140959n59342638r280b950ef6884da4@mail.gmail.com>

Maybe you could direct some of that budget to the FS project or sponsor this
summer's ClueCon MMX  =D
Also, If you call us at 877 742 2583 we can probably connect you some telco
resources.



On Thu, Jan 14, 2010 at 11:49 AM, Dale Trub <daletrub at gmail.com> wrote:

> We're looking for an ISP with SIP trunking and DIDs to host our FS servers,
> and are having a hard time locating an ISP who meet these two requirements
> a) can handle 10,000+ simultaneous inbound ports
> b) returns our calls in timely manner
>
> We have a six-figure annual budget for this service ... and so far the "big
> guys" sales teams can't get it together to put together a bid.
>
> The ideal partner would share with us the backend telcom revenue in the
> neighborhood of .5c/min, but we're open to other arrangements.
>
> Does anyone have any recommendations for ISPs with SIP trunking and DIDs
> that they can share?  Specific salespeople contact info especially
> appreciated!
>
> Thanks,
>
> Dale
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Thu Jan 14 10:01:19 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 12:01:19 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <407935.47740.qm@web37505.mail.mud.yahoo.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
	<201001132326.41354.sos@sokhapkin.dyndns.org>
	<52E8BE9E-B6D6-411C-9080-697AF24017B4@jerris.com>
	<191c3a031001132114s18ed2baw2ff802dbd615fe01@mail.gmail.com>
	<191c3a031001132115u2d88bab7p7cbb27ab1eaad466@mail.gmail.com>
	<9853f4ff1001140226y2b035f37rfcafb03190435ec1@mail.gmail.com>
	<43692088-738A-475C-8D60-E40D296ECEB1@jerris.com>
	<407935.47740.qm@web37505.mail.mud.yahoo.com>
Message-ID: <191c3a031001141001q3b164652s87c1561e2a2daf2c@mail.gmail.com>

he wants to call 3 people at once and let the A leg hear early media from
call #1 while call #2 and #3 still are progressing which is not simple to do
without doing thousands of dollars in development.


On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:

> What about sending Sip 183 with SDP (no 200OK), so that your customers can
> hear recordings?
>
> djbinter
>
>
> ------------------------------
> *From:* Michael Jerris <mike at jerris.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thu, January 14, 2010 8:12:59 AM
> *Subject:* Re: [Freeswitch-users] multiple leg and multiple rtp
>
> Okay, so that is a very interesting use case.   Seems a bizarre way to do
> this for the carrier, but interesting none the less.  I'd say the chances of
> actually muxing the early media is small, what might be possible would be
> something to say which b legs media to pass along to the a leg.  I have not
> looked at all at how complicated this is, it will be down deep somewhere in
> switch_ivr_originate code around where we if for ignore_early_media.  This
> code is pretty complex, I can't say that we will ever actually add this
> functionality, but at least now I won't blow it off as complete nonsense.
>
> Mike
>
>
>
> On Jan 14, 2010, at 5:26 AM, David Villasmil wrote:
>
> Hello again,
>
> Using ingore early media only ignores ALL media, that's not what I need. At
> least in europe i've seen it many times, MNOs provide a service with which
> you can have a song played to the caller while the call is connecting to
> your cell phone. This is basically what I'm trying to achieve.
>
> The content provider is in possesions of the media and it require us to
> fork the call and send a SIP INVITE to the on one leg and the call to the
> destination number on another leg. they will only provide a progress, no
> answer.
>
> I CAN do it locally by playing the file as a custom ringback but that's not
> the standard in terms of commercial use of the content.
>
> Is there any way to modify the behaviour of fs when i receives media? Let's
> say i.e. it doesn't drop the other leg, but provides the first early media
> it receives and just wait for some channel to answer? I will not have both
> media but it would work.
>
> thanks
>
> David
>
> On Thu, Jan 14, 2010 at 6:15 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Perhaps best not to help him anymore without an apology for the snap
>> judgement and comparison to asterisk clearly designed to push our buttons.
>> We'll be here when you realize we were trying to help you but I can't
>> promise we will still have paitence......
>>
>> On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike at jerris.com> wrote:
>>
>> You already are running on trunk.
>>
>> Mike
>>
>> On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote: > I run FreeSWITCH
>> Version 1.0.5pre10 (16012M...
>>
>> _______________________________________________ FreeSWITCH-users mailing
>> list FreeSWITCH-users at lists...
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mrene_lists at avgs.ca  Thu Jan 14 10:08:49 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 14 Jan 2010 13:08:49 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001141001q3b164652s87c1561e2a2daf2c@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132308.03170.sos@sokhapkin.dyndns.org>
	<5501D6D0-4B50-476A-B5B8-74F289DE520B@freeswitch.org>
	<201001132326.41354.sos@sokhapkin.dyndns.org>
	<52E8BE9E-B6D6-411C-9080-697AF24017B4@jerris.com>
	<191c3a031001132114s18ed2baw2ff802dbd615fe01@mail.gmail.com>
	<191c3a031001132115u2d88bab7p7cbb27ab1eaad466@mail.gmail.com>
	<9853f4ff1001140226y2b035f37rfcafb03190435ec1@mail.gmail.com>
	<43692088-738A-475C-8D60-E40D296ECEB1@jerris.com>
	<407935.47740.qm@web37505.mail.mud.yahoo.com>
	<191c3a031001141001q3b164652s87c1561e2a2daf2c@mail.gmail.com>
Message-ID: <C3EB94B1-BA8B-4D89-A781-0E47E4C1C8AB@avgs.ca>

For the music I guess you could use "uuid_displace" to mix it in  
before you call bridge but even then you can't fork-dial multiple  
destinations and have early media at the same time, it doesnt make  
much sense.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 14-Jan-10, at 1:01 PM, Anthony Minessale wrote:

> he wants to call 3 people at once and let the A leg hear early media  
> from call #1 while call #2 and #3 still are progressing which is not  
> simple to do without doing thousands of dollars in development.
>
>
> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
> What about sending Sip 183 with SDP (no 200OK), so that your  
> customers can hear recordings?
>
> djbinter
>
>
> From: Michael Jerris <mike at jerris.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thu, January 14, 2010 8:12:59 AM
> Subject: Re: [Freeswitch-users] multiple leg and multiple rtp
>
> Okay, so that is a very interesting use case.   Seems a bizarre way  
> to do this for the carrier, but interesting none the less.  I'd say  
> the chances of actually muxing the early media is small, what might  
> be possible would be something to say which b legs media to pass  
> along to the a leg.  I have not looked at all at how complicated  
> this is, it will be down deep somewhere in switch_ivr_originate code  
> around where we if for ignore_early_media.  This code is pretty  
> complex, I can't say that we will ever actually add this  
> functionality, but at least now I won't blow it off as complete  
> nonsense.
>
> Mike
>
>
>
> On Jan 14, 2010, at 5:26 AM, David Villasmil wrote:
>
>> Hello again,
>>
>> Using ingore early media only ignores ALL media, that's not what I  
>> need. At least in europe i've seen it many times, MNOs provide a  
>> service with which you can have a song played to the caller while  
>> the call is connecting to your cell phone. This is basically what  
>> I'm trying to achieve.
>>
>> The content provider is in possesions of the media and it require  
>> us to fork the call and send a SIP INVITE to the on one leg and the  
>> call to the destination number on another leg. they will only  
>> provide a progress, no answer.
>>
>> I CAN do it locally by playing the file as a custom ringback but  
>> that's not the standard in terms of commercial use of the content.
>>
>> Is there any way to modify the behaviour of fs when i receives  
>> media? Let's say i.e. it doesn't drop the other leg, but provides  
>> the first early media it receives and just wait for some channel to  
>> answer? I will not have both media but it would work.
>>
>> thanks
>>
>> David
>>
>> On Thu, Jan 14, 2010 at 6:15 AM, Anthony Minessale <anthony.minessale at gmail.com 
>> > wrote:
>> Perhaps best not to help him anymore without an apology for the  
>> snap judgement and comparison to asterisk clearly designed to push  
>> our buttons.
>> We'll be here when you realize we were trying to help you but I  
>> can't promise we will still have paitence......
>>
>>
>>> On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike at jerris.com> wrote:
>>>
>>> You already are running on trunk.
>>>
>>> Mike
>>> On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote: > I run  
>>> FreeSWITCH Version 1.0.5pre10 (16012M...
>>>
>>> _______________________________________________ FreeSWITCH-users  
>>> mailing list FreeSWITCH-users at lists...
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kokoska.rokoska at post.cz  Thu Jan 14 10:10:07 2010
From: kokoska.rokoska at post.cz (kokoska.rokoska)
Date: Thu, 14 Jan 2010 19:10:07 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <23BD1322-7882-4D43-80AB-27872096B4C9@avgs.ca>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>	<201001140719.12363.sos@sokhapkin.dyndns.org>	<191c3a031001140822i75b06440qa447fbaa46c883cb@mail.gmail.com>	<201001141141.59831.sos@sokhapkin.dyndns.org>
	<23BD1322-7882-4D43-80AB-27872096B4C9@avgs.ca>
Message-ID: <4B4F5DFF.6000504@post.cz>

Dne 14.1.2010 17:47, Mathieu Rene napsal(a):
> One way you could fix it is set "hangup_after_bridge=false" and then  
> set "execute_on_answer=set\shangup_after_bridge=true". That will make  
> it continue executing the dialplan atter a failure that came in after  
> 183. Note that there is a side effect to this method, since audio was  
> being relayed, you are forced to use the same codec on the 2nd  
> carrier, best to tell them they are wrong failing a call after 183.
> 
> Mathieu Rene

Hello Mathieu, are you sure, that "failing call" after 183 is wrong?
I don't think so, but may be I'm wrong, so could you, please, point me
to the docs?

BTW: I'm prety sure, that this behaviour is VERY common in EU. IMO
nearly all big carriers in case of call failure (according to ETSI
recommendations) send first call-progress with media ("The number your
are calling is unavailable" etc.) and than reject call with proper "Q"
reason. And I think it should be mapped to SIP like 183+SDP and than
4XX/5XX/6XX.

Just my 2c :-)

Best regards,

kokoska.rokoska



From sos at sokhapkin.dyndns.org  Thu Jan 14 10:18:34 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Jan 2010 13:18:34 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001141001q3b164652s87c1561e2a2daf2c@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<407935.47740.qm@web37505.mail.mud.yahoo.com>
	<191c3a031001141001q3b164652s87c1561e2a2daf2c@mail.gmail.com>
Message-ID: <201001141318.34958.sos@sokhapkin.dyndns.org>

The issue here is when "originate" routine should return and 
set "originate_status" variable. Current behavior is to return on early 
media, but what if to introduce a variable "originate_wait_for_answer" with 
default value "false" and use the variable in originate code to decide when 
to return - on 18X or "200 OK"?

On Thursday 14 January 2010, Anthony Minessale wrote:
> he wants to call 3 people at once and let the A leg hear early media from
> call #1 while call #2 and #3 still are progressing which is not simple to
> do without doing thousands of dollars in development.
>
> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
> > What about sending Sip 183 with SDP (no 200OK), so that your customers
> > can hear recordings?
> >
> > djbinter
> >
> >
> > ------------------------------
> > *From:* Michael Jerris <mike at jerris.com>
> > *To:* freeswitch-users at lists.freeswitch.org
> > *Sent:* Thu, January 14, 2010 8:12:59 AM
> > *Subject:* Re: [Freeswitch-users] multiple leg and multiple rtp
> >
> > Okay, so that is a very interesting use case.   Seems a bizarre way to do
> > this for the carrier, but interesting none the less.  I'd say the chances
> > of actually muxing the early media is small, what might be possible would
> > be something to say which b legs media to pass along to the a leg.  I
> > have not looked at all at how complicated this is, it will be down deep
> > somewhere in switch_ivr_originate code around where we if for
> > ignore_early_media.  This code is pretty complex, I can't say that we
> > will ever actually add this functionality, but at least now I won't blow
> > it off as complete nonsense.
> >
> > Mike
> >
> >
> >
> > On Jan 14, 2010, at 5:26 AM, David Villasmil wrote:
> >
> > Hello again,
> >
> > Using ingore early media only ignores ALL media, that's not what I need.
> > At least in europe i've seen it many times, MNOs provide a service with
> > which you can have a song played to the caller while the call is
> > connecting to your cell phone. This is basically what I'm trying to
> > achieve.
> >
> > The content provider is in possesions of the media and it require us to
> > fork the call and send a SIP INVITE to the on one leg and the call to the
> > destination number on another leg. they will only provide a progress, no
> > answer.
> >
> > I CAN do it locally by playing the file as a custom ringback but that's
> > not the standard in terms of commercial use of the content.
> >
> > Is there any way to modify the behaviour of fs when i receives media?
> > Let's say i.e. it doesn't drop the other leg, but provides the first
> > early media it receives and just wait for some channel to answer? I will
> > not have both media but it would work.
> >
> > thanks
> >
> > David
> >
> > On Thu, Jan 14, 2010 at 6:15 AM, Anthony Minessale <
> >
> > anthony.minessale at gmail.com> wrote:
> >> Perhaps best not to help him anymore without an apology for the snap
> >> judgement and comparison to asterisk clearly designed to push our
> >> buttons. We'll be here when you realize we were trying to help you but I
> >> can't promise we will still have paitence......
> >>
> >> On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike at jerris.com> wrote:
> >>
> >> You already are running on trunk.
> >>
> >> Mike
> >>
> >> On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote: > I run FreeSWITCH
> >> Version 1.0.5pre10 (16012M...
> >>
> >> _______________________________________________ FreeSWITCH-users mailing
> >> list FreeSWITCH-users at lists...
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From sos at sokhapkin.dyndns.org  Thu Jan 14 10:27:40 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Jan 2010 13:27:40 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001140858i8ef1b09kf5d547914b5b8b68@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141141.59831.sos@sokhapkin.dyndns.org>
	<191c3a031001140858i8ef1b09kf5d547914b5b8b68@mail.gmail.com>
Message-ID: <201001141327.40713.sos@sokhapkin.dyndns.org>

Confirmed. r16311 has the issue resolved in both bypass_media=true/false 
modes.

On Thursday 14 January 2010, Anthony Minessale wrote:
> r16310
>
> Thank you for the apology.
>
> On Thu, Jan 14, 2010 at 10:41 AM, Sergey Okhapkin
>
> <sos at sokhapkin.dyndns.org>wrote:
> > Anthony, I apology, I didn't expect you'll be annoyed. As for the issue,
> > the
> > issue is not in "proxy_media" setting, same happens when both
> > "proxy_media" and "bypass_media" are not set. The code does explicit leg
> > A hangup in switch_ivr_bridge.c, line 513. The condition there is "if leg
> > A is up and leg
> > A is not in answered state, then hangup leg A", this prevents dialplan to
> > continue execution on early media bridge termination.
> >
> > Issue when "bypass_media=true" is completely different, I'll open the
> > issue on
> > jira.
> >
> > On Thursday 14 January 2010, Anthony Minessale wrote:
> > > Sergey,
> > >
> > > The bug you reference was closed because proxy_media mode by design
> >
> > send's
> >
> > > the B leg's codec to the A
> > > LEG so if there is a failure condition there is no easy graceful way to
> > > back out and try another call.
> > > I will try to make it possible if yo post a bounty, I estimate a
> > > minimum
> >
> > of
> >
> > > $1000USD in consulting time.
> > >
> > > The other one I might look at once you have apologized.
> > >
> > > I can probably add some code to make the bridge exit without
> > > terminating the A leg even when hangup_after_bridge=true in the case
> > > where the the B leg is not answered but right now I am sort of annoyed
> > > with your attitude in this thread.
> > >
> > >
> > > On Thu, Jan 14, 2010 at 6:19 AM, Sergey Okhapkin
> > >
> > > <sos at sokhapkin.dyndns.org>wrote:
> > > > Case 1 (bypass_media is off) is already on jira,
> > > > http://jira.freeswitch.org/browse/FSCORE-257 , I will prepare a test
> > > > installation with latest trunk to reproduce case 2, when
> > > > bypass_media=true and 10 seconds delay happens when 18X and then
> > > > error are received on leg b.
> > > >
> > > > On Wednesday 13 January 2010, Michael Jerris wrote:
> > > > > On Jan 13, 2010, at 11:08 PM, Sergey Okhapkin wrote:
> > > > > > Critical issues are when SIP error come after 18X provisional
> > > > > > response.
> > > > > >
> > > > > > - if bypass_media is false then dialplan stops and leg a is
> > > > > > explicitly hang up (switch_ivr_bridge.c, line 513).
> > > > >
> > > > >  behavior can be modified with continue_on_fail and
> >
> > hangup_after_bridge
> >
> > > > > channel vars, perhaps ignore_early_media as well
> > > > >
> > > > > > - if bypass_media is true, then dialplan continue, but there is
> > > > > > 10 seconds delay before next bridge application sends INVITE to
> >
> > gateway
> >
> > > > > > (
> >
> > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/02435
> >
> > > > > >4.html ) I didn't track down yet why this happens looking at FS
> > > > > > sources.
> > > > >
> > > > > see response on that thread
> > > > >
> > > > > > Why I didn't open a bug on jira? Because FS behaves according to
> >
> > the
> >
> > > > > > design and specs :-) But not according to real world
> >
> > requirements...
> >
> > > > > Really, people are trying to help you and your going to be snarky
> > > > > in response?
> > > > >
> > > > > > On Wednesday 13 January 2010, Brian West wrote:
> > > > > >> Can you elaborate on these "Critical" issues you seem to be
> >
> > having?
> >
> > > >  Why
> > > >
> > > > > >> aren't you opening a jira for them if they are that critical to
> >
> > your
> >
> > > > > >> needs?
> > > > >
> > > > > Mike
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-user
> >
> > > > >s http://www.freeswitch.org
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From david.villasmil.work at gmail.com  Thu Jan 14 10:29:19 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 14 Jan 2010 19:29:19 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <C3EB94B1-BA8B-4D89-A781-0E47E4C1C8AB@avgs.ca>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001132326.41354.sos@sokhapkin.dyndns.org>
	<52E8BE9E-B6D6-411C-9080-697AF24017B4@jerris.com>
	<191c3a031001132114s18ed2baw2ff802dbd615fe01@mail.gmail.com>
	<191c3a031001132115u2d88bab7p7cbb27ab1eaad466@mail.gmail.com>
	<9853f4ff1001140226y2b035f37rfcafb03190435ec1@mail.gmail.com>
	<43692088-738A-475C-8D60-E40D296ECEB1@jerris.com>
	<407935.47740.qm@web37505.mail.mud.yahoo.com>
	<191c3a031001141001q3b164652s87c1561e2a2daf2c@mail.gmail.com>
	<C3EB94B1-BA8B-4D89-A781-0E47E4C1C8AB@avgs.ca>
Message-ID: <9853f4ff1001141029i4be469bek779fc2921099be8c@mail.gmail.com>

Anthony,

It's actually

2 call fork, 1 has the audio the other is a regular call, a-side
should hear audio from whichever comes first BUT not discarding both
b-legs and bridging to whichever answers.




On Thu, Jan 14, 2010 at 7:08 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>
> For the music I guess you could use "uuid_displace" to mix it in before you call bridge but even then you can't fork-dial multiple destinations and have early media at the same time, it doesnt make much sense.
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
> On 14-Jan-10, at 1:01 PM, Anthony Minessale wrote:
>
> he wants to call 3 people at once and let the A leg hear early media from call #1 while call #2 and #3 still are progressing which is not simple to do without doing thousands of dollars in development.
>
>
> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>>
>> What about sending Sip 183 with SDP (no 200OK), so that your customers?can hear recordings?
>> djbinter
>>
>>
>> ________________________________
>> From: Michael Jerris <mike at jerris.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Thu, January 14, 2010 8:12:59 AM
>> Subject: Re: [Freeswitch-users] multiple leg and multiple rtp
>>
>> Okay, so that is a very interesting use case. ? Seems a bizarre way to do this for the carrier, but interesting none the less. ?I'd say the chances of actually muxing the early media is small, what might be possible would be something to say which b legs media to pass along to the a leg. ?I have not looked at all at how complicated this is, it will be down deep somewhere in switch_ivr_originate code around where we if for ignore_early_media. ?This code is pretty complex, I can't say that we will ever actually add this functionality, but at least now I won't blow it off as complete nonsense.
>> Mike
>>
>>
>> On Jan 14, 2010, at 5:26 AM, David Villasmil wrote:
>>
>> Hello again,
>>
>> Using ingore early media only ignores ALL media, that's not what I need. At least in europe i've seen it many times, MNOs provide a service with which you can have a song played to the caller while the call is connecting to your cell phone. This is basically what I'm trying to achieve.
>>
>> The content provider is in possesions of the media and it require us to fork the call and send a SIP INVITE to the on one leg and the call to the destination number on another leg. they will only provide a progress, no answer.
>>
>> I CAN do it locally by playing the file as a custom ringback but that's not the standard in terms of commercial use of the content.
>>
>> Is there any way to modify the behaviour of fs when i receives media? Let's say i.e. it doesn't drop the other leg, but provides the first early media it receives and just wait for some channel to answer? I will not have both media but it would work.
>>
>> thanks
>>
>> David
>>
>> On Thu, Jan 14, 2010 at 6:15 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>>
>>> Perhaps best not to help him anymore without an apology for the snap judgement and comparison to asterisk clearly designed to push our buttons.
>>> We'll be here when you realize we were trying to help you but I can't promise we will still have paitence......
>>>
>>> On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike at jerris.com> wrote:
>>>
>>> You already are running on trunk.
>>>
>>> Mike
>>>
>>> On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote: > I run FreeSWITCH Version 1.0.5pre10 (16012M...
>>>
>>> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists...
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mike at jerris.com  Thu Jan 14 10:34:07 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Jan 2010 13:34:07 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001141318.34958.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<407935.47740.qm@web37505.mail.mud.yahoo.com>
	<191c3a031001141001q3b164652s87c1561e2a2daf2c@mail.gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
Message-ID: <A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>

this is exactly what ignore_early_media does now.

Mike

On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:

> The issue here is when "originate" routine should return and 
> set "originate_status" variable. Current behavior is to return on early 
> media, but what if to introduce a variable "originate_wait_for_answer" with 
> default value "false" and use the variable in originate code to decide when 
> to return - on 18X or "200 OK"?
> 
> On Thursday 14 January 2010, Anthony Minessale wrote:
>> he wants to call 3 people at once and let the A leg hear early media from
>> call #1 while call #2 and #3 still are progressing which is not simple to
>> do without doing thousands of dollars in development.
>> 
>> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>>> What about sending Sip 183 with SDP (no 200OK), so that your customers
>>> can hear recordings?




From sos at sokhapkin.dyndns.org  Thu Jan 14 10:42:02 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Jan 2010 13:42:02 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
Message-ID: <201001141342.02178.sos@sokhapkin.dyndns.org>

No, not exactly. ignore_early_media doesn't pass early media to the caller if 
bypass_media is false.

On Thursday 14 January 2010, Michael Jerris wrote:
> this is exactly what ignore_early_media does now.
>
> Mike
>
> On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> > The issue here is when "originate" routine should return and
> > set "originate_status" variable. Current behavior is to return on early
> > media, but what if to introduce a variable "originate_wait_for_answer"
> > with default value "false" and use the variable in originate code to
> > decide when to return - on 18X or "200 OK"?
> >
> > On Thursday 14 January 2010, Anthony Minessale wrote:
> >> he wants to call 3 people at once and let the A leg hear early media
> >> from call #1 while call #2 and #3 still are progressing which is not
> >> simple to do without doing thousands of dollars in development.
> >>
> >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
> >>> What about sending Sip 183 with SDP (no 200OK), so that your customers
> >>> can hear recordings?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From devel at thom.fr.eu.org  Thu Jan 14 10:57:22 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Thu, 14 Jan 2010 19:57:22 +0100
Subject: [Freeswitch-users] =?utf-8?q?Fwd=3A_Re=3A__playback_and_play=5Fan?=
 =?utf-8?q?d=5Fget=5Fdigits_strange_misunderstanding?=
Message-ID: <71bf5a5d66375b2d7c333d26b29917f1@thom.fr.eu.org>

 

I did with FreeSWITCH Version 1.0.5-20100112-0400 (hacked) shall I
download another one (I don't have svn). 

Fran?ois 

On Thu, 14 Jan 2010
12:20:31 -0500, Michael Jerris wrote: Could you try this with latest trunk
as well for comparison, I just changed the way the sounds base directory is
set and I want to make sure I did not mess anything up. Mike 
  On Jan 14,
2010, at 3:23 AM, Fran?ois Legal wrote: 

So I could (kind of) solve this
by myself. There is in vars.xml the variable ${sound_prefix}. I did set it
properly to my french sound path and then it worked.  

However, for the
sake of discussion, I did try this with 1.0.5pre8 and the result was
different : 

with the extension 

FS was trying to play the file
${FREESWITCH_PATH}/sounds/en/us/callie/misc/ringing_disabled.wav 

with the
extension 

FS was trying to load
${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/ringing_disabled.wav 

Now with
the latest snapshot, with either one of the 2 mentionned extensions, FS
tries to play the file in
${sound_prefix}/filepath/${codec_bit_rate}/filename whereas 1.0.5pre8 did
not add the codec_bit_rate in the path but took care of the
default_language variable.  

Fran?ois  

On Wed, 13 Jan 2010 16:32:16
+0100, Fran?ois Legal wrote:  

Hello,  

trying to make so dialplan
extensions that use the playback and play_and_get_digits applications, but
I'm having trouble with the file name specification.  

The files I want to
play are in the french language (fr/fr/julie as configured in
lang/fr/fr.xml) 

My extension is as follows : 

The channel is using a bit
rate of 8000 Hz, so by the set default_language=fr I would expect
freeswitch to playback the file at
${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/8000/ringing_disabled.wav
whereas it tries to playback the file at
${FREESWITCH_PATH}/sounds/en/us/callie/misc/8000/ringing_disabled.wav 

I
have the same with play_and_get_digits application.  

What am I doing
wrong ?  

Fran?ois   

_______________________________________________
FreeSWITCH-users mailing
list
FreeSWITCH-users at lists.freeswitch.org
[1]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




Links:
------
[1] mailto:FreeSWITCH-users at lists.freeswitch.org
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From mustafa.pk at gmail.com  Thu Jan 14 11:09:52 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Fri, 15 Jan 2010 00:09:52 +0500
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
Message-ID: <8213d6071001141109l1d475b1j3af35b5586708e42@mail.gmail.com>

fax2email is quiet easy you just need to keep a mapping of did's with
respective email addresses of persons in a database or something, and
an api call on hangup can trigger an mail script to convert and send
fax in a more user friendly format like pfd or doc etc.

for email2fax you would need to setup an smtp service fax.example.com,
your smtp service can pipe the email  to a custom script to
extract file attachments using a mime decoder and send them to
freeswitch. for all document format conversion things you can use
openoffice macros functionality which is callable from command line.

-m

On Thu, Jan 14, 2010 at 7:54 PM, Costa Zikalala
<costa.zikalala at gmail.com> wrote:
> Hi All
>
> Has anyone worked on a email2fax script for mod_fax?
> If not how much would it cost for some genius here to quickly whip-up one?
>
> Ideally both email2fax and fax2email should come standard with mod_fax or is
> this not possible?
>
> Thanks
> Costa
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com



From mouncifbb at gmail.com  Thu Jan 14 11:23:53 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Thu, 14 Jan 2010 14:23:53 -0500
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <8213d6071001141109l1d475b1j3af35b5586708e42@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
	<8213d6071001141109l1d475b1j3af35b5586708e42@mail.gmail.com>
Message-ID: <ade741d61001141123x393512f6x81768de82a84c214@mail.gmail.com>

Curious! How do you trigger an api call on hangup?

On Thu, Jan 14, 2010 at 2:09 PM, Ghulam Mustafa <mustafa.pk at gmail.com>wrote:

> fax2email is quiet easy you just need to keep a mapping of did's with
> respective email addresses of persons in a database or something, and
> an api call on hangup can trigger an mail script to convert and send
> fax in a more user friendly format like pfd or doc etc.
>
> for email2fax you would need to setup an smtp service fax.example.com,
> your smtp service can pipe the email  to a custom script to
> extract file attachments using a mime decoder and send them to
> freeswitch. for all document format conversion things you can use
> openoffice macros functionality which is callable from command line.
>
> -m
>
> On Thu, Jan 14, 2010 at 7:54 PM, Costa Zikalala
> <costa.zikalala at gmail.com> wrote:
> > Hi All
> >
> > Has anyone worked on a email2fax script for mod_fax?
> > If not how much would it cost for some genius here to quickly whip-up
> one?
> >
> > Ideally both email2fax and fax2email should come standard with mod_fax or
> is
> > this not possible?
> >
> > Thanks
> > Costa
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mrene_lists at avgs.ca  Thu Jan 14 11:26:30 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 14 Jan 2010 14:26:30 -0500
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <ade741d61001141123x393512f6x81768de82a84c214@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
	<8213d6071001141109l1d475b1j3af35b5586708e42@mail.gmail.com>
	<ade741d61001141123x393512f6x81768de82a84c214@mail.gmail.com>
Message-ID: <50B0D531-15AC-4718-ABD1-BBA74D6E30C1@avgs.ca>

You set the api_hangup_hook variable to what you want to run.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 14-Jan-10, at 2:23 PM, Mouncif Benniane wrote:

> Curious! How do you trigger an api call on hangup?
>
> On Thu, Jan 14, 2010 at 2:09 PM, Ghulam Mustafa  
> <mustafa.pk at gmail.com> wrote:
> fax2email is quiet easy you just need to keep a mapping of did's with
> respective email addresses of persons in a database or something, and
> an api call on hangup can trigger an mail script to convert and send
> fax in a more user friendly format like pfd or doc etc.
>
> for email2fax you would need to setup an smtp service fax.example.com,
> your smtp service can pipe the email  to a custom script to
> extract file attachments using a mime decoder and send them to
> freeswitch. for all document format conversion things you can use
> openoffice macros functionality which is callable from command line.
>
> -m
>
> On Thu, Jan 14, 2010 at 7:54 PM, Costa Zikalala
> <costa.zikalala at gmail.com> wrote:
> > Hi All
> >
> > Has anyone worked on a email2fax script for mod_fax?
> > If not how much would it cost for some genius here to quickly whip- 
> up one?
> >
> > Ideally both email2fax and fax2email should come standard with  
> mod_fax or is
> > this not possible?
> >
> > Thanks
> > Costa
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mustafa.pk at gmail.com  Thu Jan 14 11:31:27 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Fri, 15 Jan 2010 00:31:27 +0500
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <ade741d61001141123x393512f6x81768de82a84c214@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
	<8213d6071001141109l1d475b1j3af35b5586708e42@mail.gmail.com>
	<ade741d61001141123x393512f6x81768de82a84c214@mail.gmail.com>
Message-ID: <8213d6071001141131m3a6ab686xf716dc26983ade5a@mail.gmail.com>

you can do it within dialplan, or you can trigger an event or
something, there are several ways to get it done!

On Fri, Jan 15, 2010 at 12:23 AM, Mouncif Benniane <mouncifbb at gmail.com> wrote:
> Curious! How do you trigger an api call on hangup?
>
> On Thu, Jan 14, 2010 at 2:09 PM, Ghulam Mustafa <mustafa.pk at gmail.com>
> wrote:
>>
>> fax2email is quiet easy you just need to keep a mapping of did's with
>> respective email addresses of persons in a database or something, and
>> an api call on hangup can trigger an mail script to convert and send
>> fax in a more user friendly format like pfd or doc etc.
>>
>> for email2fax you would need to setup an smtp service fax.example.com,
>> your smtp service can pipe the email ?to a custom script to
>> extract file attachments using a mime decoder and send them to
>> freeswitch. for all document format conversion things you can use
>> openoffice macros functionality which is callable from command line.
>>
>> -m
>>
>> On Thu, Jan 14, 2010 at 7:54 PM, Costa Zikalala
>> <costa.zikalala at gmail.com> wrote:
>> > Hi All
>> >
>> > Has anyone worked on a email2fax script for mod_fax?
>> > If not how much would it cost for some genius here to quickly whip-up
>> > one?
>> >
>> > Ideally both email2fax and fax2email should come standard with mod_fax
>> > or is
>> > this not possible?
>> >
>> > Thanks
>> > Costa
>> >
>> >
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Ghulam Mustafa
>> cell: +92 333.611.7681
>> sip: cyrenity at ekiga.net
>> mail: mustafa.pk at gmail.com
>> web: cyrenity.wordpress.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com



From msc at freeswitch.org  Thu Jan 14 11:51:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Jan 2010 11:51:37 -0800
Subject: [Freeswitch-users] Announcement: FreeSWITCH Event Socket Library
	Supports TCL
Message-ID: <87f2f3b91001141151w361de47fgc6e1db6f9929ea06@mail.gmail.com>

Just to let everyone know, we've added TCL support for ESL. More info here:
http://www.freeswitch.org/node/228

Let us know if you do anything interesting with it and we'll add it to the
wiki.
-Michael
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From mouncifbb at gmail.com  Thu Jan 14 11:54:03 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Thu, 14 Jan 2010 14:54:03 -0500
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <8213d6071001141131m3a6ab686xf716dc26983ade5a@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
	<8213d6071001141109l1d475b1j3af35b5586708e42@mail.gmail.com>
	<ade741d61001141123x393512f6x81768de82a84c214@mail.gmail.com>
	<8213d6071001141131m3a6ab686xf716dc26983ade5a@mail.gmail.com>
Message-ID: <ade741d61001141154m75b9fcbak76e874e80d023ac0@mail.gmail.com>

thanks, how would it look in dialplan if I have to call a javascript?



On Thu, Jan 14, 2010 at 2:31 PM, Ghulam Mustafa <mustafa.pk at gmail.com>wrote:

> you can do it within dialplan, or you can trigger an event or
> something, there are several ways to get it done!
>
> On Fri, Jan 15, 2010 at 12:23 AM, Mouncif Benniane <mouncifbb at gmail.com>
> wrote:
> > Curious! How do you trigger an api call on hangup?
> >
> > On Thu, Jan 14, 2010 at 2:09 PM, Ghulam Mustafa <mustafa.pk at gmail.com>
> > wrote:
> >>
> >> fax2email is quiet easy you just need to keep a mapping of did's with
> >> respective email addresses of persons in a database or something, and
> >> an api call on hangup can trigger an mail script to convert and send
> >> fax in a more user friendly format like pfd or doc etc.
> >>
> >> for email2fax you would need to setup an smtp service fax.example.com,
> >> your smtp service can pipe the email  to a custom script to
> >> extract file attachments using a mime decoder and send them to
> >> freeswitch. for all document format conversion things you can use
> >> openoffice macros functionality which is callable from command line.
> >>
> >> -m
> >>
> >> On Thu, Jan 14, 2010 at 7:54 PM, Costa Zikalala
> >> <costa.zikalala at gmail.com> wrote:
> >> > Hi All
> >> >
> >> > Has anyone worked on a email2fax script for mod_fax?
> >> > If not how much would it cost for some genius here to quickly whip-up
> >> > one?
> >> >
> >> > Ideally both email2fax and fax2email should come standard with mod_fax
> >> > or is
> >> > this not possible?
> >> >
> >> > Thanks
> >> > Costa
> >> >
> >> >
> >> >
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Ghulam Mustafa
> >> cell: +92 333.611.7681
> >> sip: cyrenity at ekiga.net
> >> mail: mustafa.pk at gmail.com
> >> web: cyrenity.wordpress.com
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch-users at digitaldan.com  Thu Jan 14 12:30:34 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Thu, 14 Jan 2010 13:30:34 -0700 (MST)
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <32185789.01263487832057.JavaMail.root@zimbra>
Message-ID: <13392952.131263501034741.JavaMail.root@zimbra>

So I did figure it out, it ended up being the CISCO_SKIP_MARK_BIT_2833 issue mentioned on the RTP troubleshooting wiki page. I had incorrectly added this to my dialplan rather then to the external sip profile where it belonged. 


Thanks 
Dan- 

----- Original Message ----- 
From: "Dan" <freeswitch-users at digitaldan.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 1:56:16 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 


Thanks, I will try and turn up the debugging on it. I have a lua script that is calling session:getDigits(20,"#",15000) . When I watch it under fs_cli I'm getting this output 



2010-01-13 10:44:15.750329 [NOTICE] switch_cpp.cpp:1129 main: Getting digits 
2010-01-13 10:44:17.586676 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 0:400 
2010-01-13 10:44:17.790705 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 5:400 
2010-01-13 10:44:17.910737 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 1:400 
2010-01-13 10:44:18.114779 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 7:400 
2010-01-13 10:44:18.234812 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 2:400 
2010-01-13 10:44:18.462832 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 8:400 
2010-01-13 10:44:18.582852 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 9:400 
2010-01-13 10:44:23.311740 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF #:400 
2010-01-13 10:44:23.311740 [DEBUG] switch_cpp.cpp:735 getDigits dtmf_buf: 0517289 




It will drop any one or more numbers from any part of that sequence. 


Thanks 
Dan 
(sorry for the duped message) ----- Original Message ----- 
From: "Michael Jerris" <mike at jerris.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 1:41:51 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

What are you doing with these digits on the freeswitch side, what is it thinking is passed, and do you see anything in debug logs. You also could uncomment the #define DEBUG_2833 from switch_rtp.c for more info. 


Mike 




On Jan 13, 2010, at 3:12 PM, Dan wrote: 




Thanks for your reply, 


Wireshark shows all of the digits coming through cleanly, in the case below (also on pastebin) its 0951*72789, freeswitch does not seem to be processing one or more of the rtp events. Is there some specific debugging I should turn on that might give me some more info on why freeswitch is not processing some of these events? 


http://pastebin.freeswitch.org/11780 



449 12.280148 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
450 12.285108 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
451 12.290082 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
452 12.335129 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
453 12.340124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
454 12.345124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
455 12.365162 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21862, Time=359229277, Mark 
456 12.380167 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21863, Time=359229605 
461 12.385248 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
462 12.385253 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
465 12.390163 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
466 12.435211 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
467 12.440201 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
468 12.445205 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
469 12.480251 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21870, Time=359230405 
470 12.485275 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
471 12.485284 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
472 12.490251 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
475 12.535277 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
476 12.540287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
477 12.545287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
478 12.580337 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21877, Time=359231197 
484 12.585337 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
485 12.585343 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
486 12.590362 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
492 12.635393 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
493 12.640368 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
494 12.645369 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
495 12.665406 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21884, Time=359231677, Mark 
496 12.680431 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21885, Time=359232005 
497 12.680438 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
502 12.685408 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
506 12.690412 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
507 12.735450 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
508 12.740453 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
509 12.745469 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
511 12.780526 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21892, Time=359232805 
512 12.780534 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
513 12.785502 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
514 12.790492 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
515 12.835544 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
516 12.840537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
517 12.845537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
523 12.880582 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21899, Time=359233605 
524 12.885593 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
525 12.885599 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
526 12.890580 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
528 12.935618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
529 12.940618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
530 12.945618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
531 12.965656 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21906, Time=359234077, Mark 
532 12.980659 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21907, Time=359234413 
538 12.985690 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
539 12.985695 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
540 12.990657 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
541 13.035703 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
542 13.040701 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
543 13.045700 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
544 13.080762 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21914, Time=359235205 
545 13.080768 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
546 13.085736 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
547 13.090798 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
549 13.135809 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
550 13.140780 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
551 13.145793 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
554 13.180827 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21921, Time=359236005 
558 13.185847 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
559 13.185854 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
560 13.190821 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
566 13.235872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
567 13.240864 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
568 13.245872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) ----- Original Message ----- 
From: "Brian West" < brian at freeswitch.org > 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 12:39:52 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

Cisco gateways can detect DTMF at random if you have kids screaming or loud noises in the background... very unreliable. 


/b 



On Jan 13, 2010, at 1:36 PM, Dan wrote: 


Sorry, I should have made that more clear, this is using 2833. As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through). 

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From anthony.minessale at gmail.com  Thu Jan 14 13:04:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 15:04:00 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001141342.02178.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>

I added bridge_early_media=true to do the best I can do.
This is the most I will do, especially for free, nobody can take a hint that
you should be paying for all these custom requests so take it or leave it
but this thread is done.........



On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> No, not exactly. ignore_early_media doesn't pass early media to the caller
> if
> bypass_media is false.
>
> On Thursday 14 January 2010, Michael Jerris wrote:
> > this is exactly what ignore_early_media does now.
> >
> > Mike
> >
> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> > > The issue here is when "originate" routine should return and
> > > set "originate_status" variable. Current behavior is to return on early
> > > media, but what if to introduce a variable "originate_wait_for_answer"
> > > with default value "false" and use the variable in originate code to
> > > decide when to return - on 18X or "200 OK"?
> > >
> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> > >> he wants to call 3 people at once and let the A leg hear early media
> > >> from call #1 while call #2 and #3 still are progressing which is not
> > >> simple to do without doing thousands of dollars in development.
> > >>
> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
> customers
> > >>> can hear recordings?
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Thu Jan 14 14:01:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 16:01:52 -0600
Subject: [Freeswitch-users] sip trunk question: why call throughexternal
	profile is challenged?
In-Reply-To: <AD524CC0-726B-46D3-A22C-0B1F44C9B899@jerris.com>
References: <20100114082640.2C5D311F32@mail.nstel.ru>
	<AD524CC0-726B-46D3-A22C-0B1F44C9B899@jerris.com>
Message-ID: <191c3a031001141401h4cef36a7p84bac827bc94352f@mail.gmail.com>

try setting param accept-blind-auth to true in your sofia profile config
internal.xml
iirc it was made just for sipX who feels the need to send auth headers even
when nobody asked for them.
so even when auth-calls is false we will still try to parse the auth if one
is sent.


On Thu, Jan 14, 2010 at 11:21 AM, Michael Jerris <mike at jerris.com> wrote:

> if you look in the sample configs for the words blind and auth you will
> find all these settings, also you can setup acls for ip auth to not
> challenge.
>
> Mike
>
> On Jan 14, 2010, at 3:26 AM, Nikolay Kondratyev wrote:
>
>  Mike, thanks for the reply.
>
>
>
> Mmm? looks like I need more detailed instructions where to dig?
>
> Is there a way to turn off ?challenging? completely?
>
> I thought that <param name="auth-calls" value="false"/> should do it, but
> alas?
>
> By the way should this parameter be visible in either ?sofia status profile
> external? or ?sofia status gateway sipx4.lab.nstel.ru? ? I don?t see it?
>
>
>
> I attached traces of failed and successful calls.
>
>
>
> Thanks and regards,
>
> Nikolay.
>   ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Jerris
> *Sent:* Wednesday, January 13, 2010 8:30 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] sip trunk question: why call
> throughexternal profile is challenged?
>
>
>
> Look at how sipx sets up the users when they build the extensions and such
> for conferences, there was something special here, but I can't recall what.
>
>
>
> Mike
>
>
>
> On Jan 13, 2010, at 9:10 AM, Nikolay Kondratyev wrote:
>
>
>
>  Hi all!
>
>
>
> I?m brand new to FreeSwitch, but have some experience with SipX.
>
> Our company is evaluating FS.
>
> For test purposes I set up FS on a virtual machine (vmware esxi). I use
> CentOS.
>
> The FS version I use is 1.0.5-20100110-0400.
>
>
>
> I have a question regarding sip trunk between FS and SipX.
>
> I created the following GW in external profile:
>
> [freeswitch at freeswitch external]$ cat sipx-lab.xml | grep -v '<!--'
>
> <include>
>
>   <gateway name="sipx4.lab.nstel.ru">
>
>   <param name="username" value="zxcv"/>
>
>   <param name="password" value="2007"/>
>
>   <param name="register" value="false"/>
>
>   </gateway>
>
> </include>
>
>
>
> External.xml file is not modified after installation.
>
>
>
> I see this gateway via fs_cli:
>
> freeswitch at internal> sofia status gateway sipx4.lab.nstel.ru
>
>
> =================================================================================================
>
> Name            sipx4.lab.nstel.ru
>
> Profile         external
>
> Scheme          Digest
>
> Realm           sipx4.lab.nstel.ru
>
> Username        zxcv
>
> Password        yes
>
> From            <sip:zxcv at sipx4.lab.nstel.ru;transport=udp>
>
> Contact         <
> sip:gw+sipx4.lab.nstel.ru at 172.23.22.49:5080;transport=udp;gw=sipx4.lab.nstel.ru
> >
>
> Exten           zxcv
>
> To              sip:zxcv at sipx4.lab.nstel.ru
>
> Proxy           sip:sipx4.lab.nstel.ru
>
> Context         public
>
> Expires         3600
>
> Freq            3600
>
> Ping            0
>
> PingFreq        0
>
> PingState       0/0/0
>
> State           NOREG
>
> Status          UP
>
> CallsIN         0
>
> CallsOUT        0
>
>
> =================================================================================================
>
>
>
> I created new FS extension 2853. I registers (xlite) and I can call it.
>
>
>
> Now I want to call FS user from sipx.
>
>
>
> From the sipx side one can configure link to FS differently. There are two
> options:
>
> 1. Call FS directly trough sipxproxy (the core part of sipx works as sip
> proxy, not as b2bua)
>
> 2. Call trough embedded b2bua, named sipxbridge.
>
>
>
> When a call is going trough sipxbridge, it is successfully landed at FS
> extension 2853.
>
> But when a call is going from the sipxproxy, it is challenged with Status
> 407 ?proxy authentication required?, and then call fails.
>
> (I?m not sure if sipx should handle this challenge, but this is separate
> question for the sipx forum).
>
> At the default log level I see nothing in the freeswitch.log.
>
>
>
> So the question is why one call through external profile is being
> challenged while the other is not?
>
> I suspect that the reason is in the difference in the two Invite messages:
>
>
>
> Here is challenged Invite:
>
> INVITE sip:2853 at fs.lab.nstel.ru:5080;transport=udp SIP/2.0
>
> Record-Route: <
> sip:172.23.12.104:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EOTRiZjI3MjgtYWMxNzBjZTktMTNjNC0yZjZkZWMtNjNkNmUwN2UtMmY2ZGVj.900_ntap%2Aid%7EMjIyNjAtMQ%60%60%213ba6d85f6946c4c6001bee1d3b54474f
> >
>
> From: "testphone3857"<sip:3857 at lab.nstel.ru
> >;tag=94bf2728-ac170ce9-13c4-2f6dec-63d6e07e-2f6dec
>
> To: <sip:2853 at lab.nstel.ru>
>
> Call-Id: 94bed2a0-ac170ce9-13c4-2f6dec-68ffdb81-2f6dec at lab.nstel.ru
>
> Cseq: 2 INVITE
>
> Via: SIP/2.0/UDP
> 172.23.12.104;branch=z9hG4bK-sipXecs-000eaf48a36fe4029c7cde004a6f44424847
>
> Via: SIP/2.0/TCP
> 172.23.12.104;branch=z9hG4bK-sipXecs-000bf3008cfa3f3e70470789c75232ba9499~ac7fd729330fd563f83cacd941311e75;id=22260-1
>
> Via: SIP/2.0/UDP 172.23.12.233:5060
> ;branch=z9hG4bK-2f6dec-b9456412-7ad07784
>
> Max-Forwards: 18
>
> Supported: replaces
>
> User-Agent: LG-Nortel LIP 6812 v1.2.38sp SN/00405A18B634
>
> Contact: <sip:3857 at 172.23.12.233:5060;x-sipX-nonat>
>
> Proxy-Authorization: Digest username="3857",realm="lab.nstel.ru
> ",nonce="e2152722611af1bbe59f3a8eb31b8eb84b4db2ca",uri="
> sip:2853 at lab.nstel.ru
> ",response="a09c34fbd897783c9b9af50bd044ccaa",algorithm=MD5
>
> Content-Type: application/sdp
>
> Content-Length: 301
>
> Date: Wed, 13 Jan 2010 11:47:22 GMT
>
> Expires: 60
>
> X-Sipx-Handled: X172.23.12.104-81.211.30.104
>
>
>
> v=0
>
> o=LGEIPP 16246 16247 IN IP4 172.23.12.233
>
> s=SIP Call
>
> c=IN IP4 172.23.12.233
>
> t=0 0
>
> m=audio 23008 RTP/AVP 0 8 18 4 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:18 G729/8000
>
> a=rtpmap:4 G723/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:18 annexb=no
>
> a=fmtp:101 0-11
>
> a=sendrecv
>
>
>
>
>
> And here is successful Invite:
>
> INVITE sip:2853 at fs.lab.nstel.ru;user=phone SIP/2.0
>
> Call-ID: 94bec3b8-ac170ce9-13c4-2f6d47-2fee09d0-2f6d47 at lab.nstel.ru.0
>
> CSeq: 1 INVITE
>
> From: "testphone3857" <sip:3857 at 172.23.12.104>;tag=816640414244159961
>
> To: <sip:2853 at fs.lab.nstel.ru;user=phone>
>
> Via: SIP/2.0/UDP 172.23.12.104:5080
> ;branch=z9hG4bK885dde2c8680f5315845cd3350b8b605373534
>
> Max-Forwards: 70
>
> User-Agent: sipXecs/4.0.2 sipXecs/sipxbridge (Linux)
>
> P-Asserted-Identity: <sip:1004 at 172.23.12.104>
>
> Contact: <sip:1004 at 172.23.12.104:5080;transport=udp>
>
> Route: <sip:172.23.22.49:5080;transport=udp;lr>
>
> Session-Expires: 1800;refresher=uac
>
> Allow: INVITE,BYE,ACK,CANCEL,OPTIONS
>
> Content-Type: application/sdp
>
> Content-Length: 315
>
>
>
> v=0
>
> o=sipxbridge 6640787141824452741 1 IN IP4 172.23.12.104
>
> s=SIP Call
>
> c=IN IP4 172.23.12.104
>
> t=0 0
>
> m=audio 30248 RTP/AVP 0 8 18 4 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:18 G729/8000
>
> a=rtpmap:4 G723/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:18 annexb=no
>
> a=fmtp:101 0-11
>
> a=sendrecv
>
>
>
> Can somebody please if it is a FS configuration problem or a software
> problem or is it a problem on the sipx side?
>
>
>
> Thanks in advance,
>
> Nikolay.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   <call_from_sipx_via_unmanaged_gw.pcap>
> <call_from_sipx_via_siptrunk_sip-only.pcap>
> _______________________________________________
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at jerris.com  Thu Jan 14 14:22:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Jan 2010 17:22:18 -0500
Subject: [Freeswitch-users] sip trunk question: why call throughexternal
	profile is challenged?
In-Reply-To: <191c3a031001141401h4cef36a7p84bac827bc94352f@mail.gmail.com>
References: <20100114082640.2C5D311F32@mail.nstel.ru>
	<AD524CC0-726B-46D3-A22C-0B1F44C9B899@jerris.com>
	<191c3a031001141401h4cef36a7p84bac827bc94352f@mail.gmail.com>
Message-ID: <1BD8DFE1-E3DD-46C3-A383-C9627939BB65@jerris.com>

This now rings a bell, also, if you create a gateway on that profile who's gateway name matches the realm on the incoming auth header, I think that also was working.

Mike

On Jan 14, 2010, at 5:01 PM, Anthony Minessale wrote:

> try setting param accept-blind-auth to true in your sofia profile config internal.xml
> iirc it was made just for sipX who feels the need to send auth headers even when nobody asked for them.
> so even when auth-calls is false we will still try to parse the auth if one is sent.
> 
> 
> On Thu, Jan 14, 2010 at 11:21 AM, Michael Jerris <mike at jerris.com> wrote:
> if you look in the sample configs for the words blind and auth you will find all these settings, also you can setup acls for ip auth to not challenge.
> 
> Mike
> 
> On Jan 14, 2010, at 3:26 AM, Nikolay Kondratyev wrote:
> 
>> Mike, thanks for the reply.
>> 
>>  
>> Mmm? looks like I need more detailed instructions where to dig?
>> 
>> Is there a way to turn off ?challenging? completely?
>> 
>> I thought that <param name="auth-calls" value="false"/> should do it, but alas?
>> 
>> By the way should this parameter be visible in either ?sofia status profile external? or ?sofia status gateway sipx4.lab.nstel.ru? ? I don?t see it?
>> 
>>  
>> I attached traces of failed and successful calls.
>> 

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From harry at vangberg.name  Thu Jan 14 15:06:07 2010
From: harry at vangberg.name (Harry Vangberg)
Date: Fri, 15 Jan 2010 00:06:07 +0100
Subject: [Freeswitch-users] No ringback.
Message-ID: <74d41a3d1001141506oc23ccc0ka41ea0a07ef2c3ce@mail.gmail.com>

Hello.

I am controlling my calls with Freeswitcher via the event socket. When
a call comes in I set the following channel variables:

hangup_after_bridge=true
ignore_early_media=true
call_timeout=20
originate_continue_on_timeout=true
instant_ringback=true
ringback=%(2000,4000,440,480)
transfer_ringback=%(2000,4000,440,480)

Then I pre_answer and bridge to one or more endpoints. The trouble is,
there's no ringback. This is with trunk at 16305. For reference, here's
excerpt from the log:

2010-01-14 22:47:14.540747 [NOTICE] switch_channel.c:642 New Channel
sofia/external/31354228 at 129.142.224.250
[c27529c2-015e-11df-8d0d-c7205d2c96c7]
2010-01-14 22:47:14.540747 [DEBUG] sofia.c:3897 Channel
sofia/external/31354228 at 129.142.224.250 entering state [received][100]
2010-01-14 22:47:14.540747 [DEBUG] sofia.c:3908 Remote SDP:
v=0
o=root 22377 22377 IN IP4 129.142.224.250
s=session
c=IN IP4 129.142.224.250
t=0 0
m=audio 14962 RTP/AVP 8 0 3 97 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

2010-01-14 22:47:14.540747 [DEBUG] sofia_glue.c:3344 Audio Codec
Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
2010-01-14 22:47:14.540747 [DEBUG] sofia_glue.c:2171 Set Codec
sofia/external/31354228 at 129.142.224.250 PCMA/8000 20 ms 160 samples
2010-01-14 22:47:14.540747 [DEBUG] sofia_glue.c:3300 Set 2833 dtmf
payload to 101
2010-01-14 22:47:14.540747 [DEBUG] sofia.c:4056
(sofia/external/31354228 at 129.142.224.250) State Change CS_NEW ->
CS_INIT
2010-01-14 22:47:14.540747 [DEBUG] switch_core_session.c:1012 Send
signal sofia/external/31354228 at 129.142.224.250 [BREAK]
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:314
(sofia/external/31354228 at 129.142.224.250) Running State Change CS_INIT
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:338
(sofia/external/31354228 at 129.142.224.250) State INIT
2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:83
sofia/external/31354228 at 129.142.224.250 SOFIA INIT
2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:111
(sofia/external/31354228 at 129.142.224.250) State Change CS_INIT ->
CS_ROUTING
2010-01-14 22:47:14.540747 [DEBUG] switch_core_session.c:1012 Send
signal sofia/external/31354228 at 129.142.224.250 [BREAK]
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:338
(sofia/external/31354228 at 129.142.224.250) State INIT going to sleep
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:314
(sofia/external/31354228 at 129.142.224.250) Running State Change
CS_ROUTING
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:341
(sofia/external/31354228 at 129.142.224.250) State ROUTING
2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:132
sofia/external/31354228 at 129.142.224.250 SOFIA ROUTING
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:78
sofia/external/31354228 at 129.142.224.250 Standard ROUTING
2010-01-14 22:47:14.540747 [INFO] mod_dialplan_xml.c:408 Processing
31354228->77344536 in context public
Dialplan: sofia/external/31354228 at 129.142.224.250 parsing
[public->ff-phone-sound-recorder] continue=false
Dialplan: sofia/external/31354228 at 129.142.224.250 Regex (FAIL)
[ff-phone-sound-recorder] destination_number(77344536) =~ /^77348624$/
break=on-false
Dialplan: sofia/external/31354228 at 129.142.224.250 parsing
[public->ff-ivr] continue=false
Dialplan: sofia/external/31354228 at 129.142.224.250 Regex (PASS)
[ff-ivr] destination_number(77344536) =~ /^(\d+)$/ break=on-false
Dialplan: sofia/external/31354228 at 129.142.224.250 Action
socket(127.0.0.1:8084 full)
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:122
(sofia/external/31354228 at 129.142.224.250) State Change CS_ROUTING ->
CS_EXECUTE
2010-01-14 22:47:14.540747 [DEBUG] switch_core_session.c:1012 Send
signal sofia/external/31354228 at 129.142.224.250 [BREAK]
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:341
(sofia/external/31354228 at 129.142.224.250) State ROUTING going to sleep
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:314
(sofia/external/31354228 at 129.142.224.250) Running State Change
CS_EXECUTE
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:348
(sofia/external/31354228 at 129.142.224.250) State EXECUTE
2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:181
sofia/external/31354228 at 129.142.224.250 SOFIA EXECUTE
2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:159
sofia/external/31354228 at 129.142.224.250 Standard EXECUTE
EXECUTE sofia/external/31354228 at 129.142.224.250 socket(127.0.0.1:8084 full)
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
set(hangup_after_bridge=true)
EXECUTE sofia/external/31354228 at 129.142.224.250 set(hangup_after_bridge=true)
2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
sofia/external/31354228 at 129.142.224.250 SET
[hangup_after_bridge]=[true]
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
set(ignore_early_media=true)
EXECUTE sofia/external/31354228 at 129.142.224.250 set(ignore_early_media=true)
2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
sofia/external/31354228 at 129.142.224.250 SET
[ignore_early_media]=[true]
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
set(call_timeout=20)
EXECUTE sofia/external/31354228 at 129.142.224.250 set(call_timeout=20)
2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
sofia/external/31354228 at 129.142.224.250 SET [call_timeout]=[20]
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
set(originate_continue_on_timeout=true)
EXECUTE sofia/external/31354228 at 129.142.224.250
set(originate_continue_on_timeout=true)
2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
sofia/external/31354228 at 129.142.224.250 SET
[originate_continue_on_timeout]=[true]
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
set(instant_ringback=true)
EXECUTE sofia/external/31354228 at 129.142.224.250 set(instant_ringback=true)
2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
sofia/external/31354228 at 129.142.224.250 SET [instant_ringback]=[true]
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
set(ringback=%(2000,4000,440,480))
EXECUTE sofia/external/31354228 at 129.142.224.250
set(ringback=%(2000,4000,440,480))
2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
sofia/external/31354228 at 129.142.224.250 SET
[ringback]=[%(2000,4000,440,480)]
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
set(transfer_ringback=%(2000,4000,440,480))
EXECUTE sofia/external/31354228 at 129.142.224.250
set(transfer_ringback=%(2000,4000,440,480))
2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
sofia/external/31354228 at 129.142.224.250 SET
[transfer_ringback]=[%(2000,4000,440,480)]
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
bind_meta_app(6 b ai
transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
public)
EXECUTE sofia/external/31354228 at 129.142.224.250 bind_meta_app(6 b ai
transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
public)
2010-01-14 22:47:14.792781 [INFO] switch_ivr_async.c:2322 Bound B-Leg:
6 transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
public
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
bind_meta_app(7 b ai transfer::ff-transfer- XML public)
EXECUTE sofia/external/31354228 at 129.142.224.250 bind_meta_app(7 b ai
transfer::ff-transfer- XML public)
2010-01-14 22:47:14.792781 [INFO] switch_ivr_async.c:2322 Bound B-Leg:
7 transfer::ff-transfer- XML public
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
bind_meta_app(1 b ai
transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
public)
EXECUTE sofia/external/31354228 at 129.142.224.250 bind_meta_app(1 b ai
transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
public)
2010-01-14 22:47:14.792781 [INFO] switch_ivr_async.c:2322 Bound B-Leg:
1 transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
public
2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute pre_answer()
EXECUTE sofia/external/31354228 at 129.142.224.250 pre_answer()
2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:736
sofia/external/31354228 at 129.142.224.250 receive message [PROGRESS]
2010-01-14 22:47:14.792781 [INFO] mod_dptools.c:736 Sending early media
2010-01-14 22:47:14.792781 [DEBUG] sofia_glue.c:2409 AUDIO RTP
[sofia/external/31354228 at 129.142.224.250] 10.224.54.102 port 22822 ->
129.142.224.250 port 14962 codec: 8 ms: 20
2010-01-14 22:47:14.792781 [DEBUG] switch_rtp.c:1168 Starting timer
[soft] 160 bytes per 20ms
2010-01-14 22:47:14.807902 [INFO] mod_sofia.c:1779 Ring SDP:
v=0
o=FreeSWITCH 1263486412 1263486413 IN IP4 79.125.35.134
s=FreeSWITCH
c=IN IP4 79.125.35.134
t=0 0
m=audio 22822 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2010-01-14 22:47:14.807902 [NOTICE] mod_sofia.c:1782 Pre-Answer
sofia/external/31354228 at 129.142.224.250!
2010-01-14 22:47:14.807902 [DEBUG] switch_core_session.c:633 Send
signal sofia/external/31354228 at 129.142.224.250 [BREAK]
2010-01-14 22:47:14.827576 [DEBUG] sofia.c:3892 Channel
sofia/external/31354228 at 129.142.224.250 skipping state [early][183]
2010-01-14 22:47:14.827576 [DEBUG] switch_ivr.c:551
sofia/external/31354228 at 129.142.224.250 Command Execute
bridge({sip_cid_type=rpid,origination_caller_id_name=77344536,origination_caller_id_number=77344536}sofia/gateway/gw1.fonet.dk/004546934488|loopback/ff-record-77344536)
EXECUTE sofia/external/31354228 at 129.142.224.250
bridge({sip_cid_type=rpid,origination_caller_id_name=77344536,origination_caller_id_number=77344536}sofia/gateway/gw1.fonet.dk/004546934488|loopback/ff-record-77344536)
2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:1738
variable string 0 = [sip_cid_type=rpid]
2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:1738
variable string 1 = [origination_caller_id_name=77344536]
2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:1738
variable string 2 = [origination_caller_id_number=77344536]
2010-01-14 22:47:14.827576 [NOTICE] switch_channel.c:642 New Channel
sofia/external/004546934488 [c2a1095c-015e-11df-8d0e-c7205d2c96c7]
2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:3227
(sofia/external/004546934488) State Change CS_NEW -> CS_INIT
2010-01-14 22:47:14.827576 [DEBUG] switch_core_session.c:1012 Send
signal sofia/external/004546934488 [BREAK]
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:314
(sofia/external/004546934488) Running State Change CS_INIT
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:338
(sofia/external/004546934488) State INIT
2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:83
sofia/external/004546934488 SOFIA INIT
2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:111
(sofia/external/004546934488) State Change CS_INIT -> CS_ROUTING
2010-01-14 22:47:14.827576 [DEBUG] switch_core_session.c:1012 Send
signal sofia/external/004546934488 [BREAK]
2010-01-14 22:47:14.827576 [DEBUG] sofia.c:3897 Channel
sofia/external/004546934488 entering state [calling][0]
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:338
(sofia/external/004546934488) State INIT going to sleep
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:314
(sofia/external/004546934488) Running State Change CS_ROUTING
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:341
(sofia/external/004546934488) State ROUTING
2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:132
sofia/external/004546934488 SOFIA ROUTING
2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:66
(sofia/external/004546934488) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2010-01-14 22:47:14.827576 [DEBUG] switch_core_session.c:1012 Send
signal sofia/external/004546934488 [BREAK]
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:341
(sofia/external/004546934488) State ROUTING going to sleep
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:314
(sofia/external/004546934488) Running State Change CS_CONSUME_MEDIA
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:360
(sofia/external/004546934488) State CONSUME_MEDIA
2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:360
(sofia/external/004546934488) State CONSUME_MEDIA going to sleep
2010-01-14 22:47:14.876945 [DEBUG] switch_ivr_originate.c:1057 Raw
Codec Activation Success L16 at 8000hz 1 channel 20ms
2010-01-14 22:47:14.876945 [DEBUG] switch_core_codec.c:112
sofia/external/31354228 at 129.142.224.250 Push codec L16:10
2010-01-14 22:47:14.876945 [DEBUG] switch_ivr_originate.c:1119 Play
Ringback Tone [%(2000,4000,440,480)]
2010-01-14 22:47:14.876945 [DEBUG] switch_core_io.c:652
sofia/external/31354228 at 129.142.224.250 receive message
[TRANSCODING_NECESSARY]
2010-01-14 22:47:14.876945 [DEBUG] sofia.c:3897 Channel
sofia/external/004546934488 entering state [calling][0]
2010-01-14 22:47:14.912774 [DEBUG] switch_rtp.c:2022 Correct ip/port confirmed.
2010-01-14 22:47:15.668817 [INFO] sofia.c:514 Update Callee ID to
"004546934488" <004546934488>
2010-01-14 22:47:15.668817 [DEBUG] sofia.c:3897 Channel
sofia/external/004546934488 entering state [proceeding][180]
2010-01-14 22:47:15.668817 [NOTICE] sofia.c:3967 Ring-Ready
sofia/external/004546934488!
2010-01-14 22:47:15.668817 [DEBUG] sofia.c:3897 Channel
sofia/external/004546934488 entering state [proceeding][183]
2010-01-14 22:47:15.668817 [DEBUG] sofia.c:3908 Remote SDP:
v=0
o=root 22377 22377 IN IP4 129.142.224.250
s=session
c=IN IP4 129.142.224.250
t=0 0
m=audio 11676 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20

2010-01-14 22:47:15.668817 [DEBUG] sofia_glue.c:3344 Audio Codec
Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
2010-01-14 22:47:15.668817 [DEBUG] sofia_glue.c:2171 Set Codec
sofia/external/004546934488 PCMA/8000 20 ms 160 samples
2010-01-14 22:47:15.668817 [DEBUG] sofia_glue.c:2409 AUDIO RTP
[sofia/external/004546934488] 10.224.54.102 port 17204 ->
129.142.224.250 port 11676 codec: 8 ms: 20
2010-01-14 22:47:15.668817 [DEBUG] switch_rtp.c:1168 Starting timer
[soft] 160 bytes per 20ms
2010-01-14 22:47:15.668817 [NOTICE] sofia_glue.c:2937 Pre-Answer
sofia/external/004546934488!
2010-01-14 22:47:15.668817 [DEBUG] switch_channel.c:2053 Send signal
sofia/external/31354228 at 129.142.224.250 [BREAK]

-- 
Harry Vangberg  <harry at vangberg.name>  http://harry.vangberg.name



From david.villasmil.work at gmail.com  Thu Jan 14 15:06:59 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 00:06:59 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
Message-ID: <9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>

Anthony,

I did take the "hint", don't worry. We will probably ask for a bounty
but first we need to know:
1.- whether this is possible
2.- how long it would take
3.- how will it exactly work
4.- of course, what's the bounty (be gentle ;) )

We would of course give this back to the community.

in the meantime, bridge_early_media=true will discard the 2nd B-leg
and conversely "false" it will keep on trying to connect and if it
connects the other B-leg if will bridge to that one?

Thanks

David

On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I added bridge_early_media=true to do the best I can do.
> This is the most I will do, especially for free, nobody can take a hint that
> you should be paying for all these custom requests so take it or leave it
> but this thread is done.........
>
>
>
> On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
> wrote:
>>
>> No, not exactly. ignore_early_media doesn't pass early media to the caller
>> if
>> bypass_media is false.
>>
>> On Thursday 14 January 2010, Michael Jerris wrote:
>> > this is exactly what ignore_early_media does now.
>> >
>> > Mike
>> >
>> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> > > The issue here is when "originate" routine should return and
>> > > set "originate_status" variable. Current behavior is to return on
>> > > early
>> > > media, but what if to introduce a variable "originate_wait_for_answer"
>> > > with default value "false" and use the variable in originate code to
>> > > decide when to return - on 18X or "200 OK"?
>> > >
>> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> > >> he wants to call 3 people at once and let the A leg hear early media
>> > >> from call #1 while call #2 and #3 still are progressing which is not
>> > >> simple to do without doing thousands of dollars in development.
>> > >>
>> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>> > >>> customers
>> > >>> can hear recordings?
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From freeswitch-users at digitaldan.com  Thu Jan 14 15:28:47 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Thu, 14 Jan 2010 16:28:47 -0700 (MST)
Subject: [Freeswitch-users] Cisco media gateway and DTMF
In-Reply-To: <13392952.131263501034741.JavaMail.root@zimbra>
Message-ID: <13986368.261263511726829.JavaMail.root@zimbra>

Actually that was not it, I had also commented out <param name="rtp-timer-name" value="soft"/> in the external.xml sip profile which is what fixed my issue. 


My question then is what does disabling the setting do? Is there any harm in not having it? 





With it disabled, I get all the DTMF digits, but I also get the following " Failed DTMF payload check." lines in my log. 



2010-01-14 16:20:02.430026 [NOTICE] switch_cpp.cpp:1129 main: Getting digits 
2010-01-14 16:20:04.246261 [ERR] switch_rtp.c:2196 Failed DTMF payload check. 
2010-01-14 16:20:04.251201 [ERR] switch_rtp.c:2196 Failed DTMF payload check. 
2010-01-14 16:20:04.256198 [ERR] switch_rtp.c:2196 Failed DTMF payload check. 
2010-01-14 16:20:04.296267 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 0:400 
2010-01-14 16:20:04.396331 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 9:400 
2010-01-14 16:20:04.496411 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 5:400 
2010-01-14 16:20:04.596505 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 1:400 
2010-01-14 16:20:04.696581 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF *:400 
2010-01-14 16:20:04.796660 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 7:400 
2010-01-14 16:20:04.896769 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 2:400 
2010-01-14 16:20:04.996830 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 7:400 
2010-01-14 16:20:05.096906 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 8:400 
2010-01-14 16:20:05.196993 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 9:400 
2010-01-14 16:20:09.950844 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF #:400 
2010-01-14 16:20:09.951064 [DEBUG] switch_cpp.cpp:735 getDigits dtmf_buf: 0951*72789 

----- Original Message ----- 
From: "Dan" <freeswitch-users at digitaldan.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Thursday, January 14, 2010 1:30:34 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 


So I did figure it out, it ended up being the CISCO_SKIP_MARK_BIT_2833 issue mentioned on the RTP troubleshooting wiki page. I had incorrectly added this to my dialplan rather then to the external sip profile where it belonged. 


Thanks 
Dan- 

----- Original Message ----- 
From: "Dan" <freeswitch-users at digitaldan.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 1:56:16 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 


Thanks, I will try and turn up the debugging on it. I have a lua script that is calling session:getDigits(20,"#",15000) . When I watch it under fs_cli I'm getting this output 



2010-01-13 10:44:15.750329 [NOTICE] switch_cpp.cpp:1129 main: Getting digits 
2010-01-13 10:44:17.586676 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 0:400 
2010-01-13 10:44:17.790705 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 5:400 
2010-01-13 10:44:17.910737 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 1:400 
2010-01-13 10:44:18.114779 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 7:400 
2010-01-13 10:44:18.234812 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 2:400 
2010-01-13 10:44:18.462832 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 8:400 
2010-01-13 10:44:18.582852 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 9:400 
2010-01-13 10:44:23.311740 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF #:400 
2010-01-13 10:44:23.311740 [DEBUG] switch_cpp.cpp:735 getDigits dtmf_buf: 0517289 




It will drop any one or more numbers from any part of that sequence. 


Thanks 
Dan 
(sorry for the duped message) ----- Original Message ----- 
From: "Michael Jerris" <mike at jerris.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 1:41:51 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

What are you doing with these digits on the freeswitch side, what is it thinking is passed, and do you see anything in debug logs. You also could uncomment the #define DEBUG_2833 from switch_rtp.c for more info. 


Mike 




On Jan 13, 2010, at 3:12 PM, Dan wrote: 




Thanks for your reply, 


Wireshark shows all of the digits coming through cleanly, in the case below (also on pastebin) its 0951*72789, freeswitch does not seem to be processing one or more of the rtp events. Is there some specific debugging I should turn on that might give me some more info on why freeswitch is not processing some of these events? 


http://pastebin.freeswitch.org/11780 



449 12.280148 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
450 12.285108 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
451 12.290082 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 
452 12.335129 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
453 12.340124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
454 12.345124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 
455 12.365162 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21862, Time=359229277, Mark 
456 12.380167 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21863, Time=359229605 
461 12.385248 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
462 12.385253 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
465 12.390163 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
466 12.435211 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
467 12.440201 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
468 12.445205 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
469 12.480251 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21870, Time=359230405 
470 12.485275 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
471 12.485284 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
472 12.490251 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 
475 12.535277 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
476 12.540287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
477 12.545287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 
478 12.580337 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21877, Time=359231197 
484 12.585337 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
485 12.585343 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
486 12.590362 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 
492 12.635393 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
493 12.640368 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
494 12.645369 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 
495 12.665406 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21884, Time=359231677, Mark 
496 12.680431 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21885, Time=359232005 
497 12.680438 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
502 12.685408 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
506 12.690412 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 
507 12.735450 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
508 12.740453 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
509 12.745469 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 
511 12.780526 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21892, Time=359232805 
512 12.780534 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
513 12.785502 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
514 12.790492 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
515 12.835544 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
516 12.840537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
517 12.845537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
523 12.880582 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21899, Time=359233605 
524 12.885593 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
525 12.885599 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
526 12.890580 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 
528 12.935618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
529 12.940618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
530 12.945618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 
531 12.965656 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21906, Time=359234077, Mark 
532 12.980659 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21907, Time=359234413 
538 12.985690 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
539 12.985695 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
540 12.990657 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 
541 13.035703 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
542 13.040701 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
543 13.045700 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 
544 13.080762 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21914, Time=359235205 
545 13.080768 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
546 13.085736 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
547 13.090798 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 
549 13.135809 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
550 13.140780 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
551 13.145793 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 
554 13.180827 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21921, Time=359236005 
558 13.185847 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
559 13.185854 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
560 13.190821 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 
566 13.235872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
567 13.240864 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 
568 13.245872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) ----- Original Message ----- 
From: "Brian West" < brian at freeswitch.org > 
To: freeswitch-users at lists.freeswitch.org 
Sent: Wednesday, January 13, 2010 12:39:52 PM 
Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF 

Cisco gateways can detect DTMF at random if you have kids screaming or loud noises in the background... very unreliable. 


/b 



On Jan 13, 2010, at 1:36 PM, Dan wrote: 


Sorry, I should have made that more clear, this is using 2833. As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through). 

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From harry at vangberg.name  Thu Jan 14 15:44:08 2010
From: harry at vangberg.name (Harry Vangberg)
Date: Fri, 15 Jan 2010 00:44:08 +0100
Subject: [Freeswitch-users] No ringback.
In-Reply-To: <74d41a3d1001141506oc23ccc0ka41ea0a07ef2c3ce@mail.gmail.com>
References: <74d41a3d1001141506oc23ccc0ka41ea0a07ef2c3ce@mail.gmail.com>
Message-ID: <74d41a3d1001141544h496c9412s69e50fd76b70ce84@mail.gmail.com>

Oh, damn. I need answer, not pre_answer. Yay.

2010/1/15 Harry Vangberg <harry at vangberg.name>:
> Hello.
>
> I am controlling my calls with Freeswitcher via the event socket. When
> a call comes in I set the following channel variables:
>
> hangup_after_bridge=true
> ignore_early_media=true
> call_timeout=20
> originate_continue_on_timeout=true
> instant_ringback=true
> ringback=%(2000,4000,440,480)
> transfer_ringback=%(2000,4000,440,480)
>
> Then I pre_answer and bridge to one or more endpoints. The trouble is,
> there's no ringback. This is with trunk at 16305. For reference, here's
> excerpt from the log:
>
> 2010-01-14 22:47:14.540747 [NOTICE] switch_channel.c:642 New Channel
> sofia/external/31354228 at 129.142.224.250
> [c27529c2-015e-11df-8d0d-c7205d2c96c7]
> 2010-01-14 22:47:14.540747 [DEBUG] sofia.c:3897 Channel
> sofia/external/31354228 at 129.142.224.250 entering state [received][100]
> 2010-01-14 22:47:14.540747 [DEBUG] sofia.c:3908 Remote SDP:
> v=0
> o=root 22377 22377 IN IP4 129.142.224.250
> s=session
> c=IN IP4 129.142.224.250
> t=0 0
> m=audio 14962 RTP/AVP 8 0 3 97 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
> 2010-01-14 22:47:14.540747 [DEBUG] sofia_glue.c:3344 Audio Codec
> Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
> 2010-01-14 22:47:14.540747 [DEBUG] sofia_glue.c:2171 Set Codec
> sofia/external/31354228 at 129.142.224.250 PCMA/8000 20 ms 160 samples
> 2010-01-14 22:47:14.540747 [DEBUG] sofia_glue.c:3300 Set 2833 dtmf
> payload to 101
> 2010-01-14 22:47:14.540747 [DEBUG] sofia.c:4056
> (sofia/external/31354228 at 129.142.224.250) State Change CS_NEW ->
> CS_INIT
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_session.c:1012 Send
> signal sofia/external/31354228 at 129.142.224.250 [BREAK]
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/31354228 at 129.142.224.250) Running State Change CS_INIT
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:338
> (sofia/external/31354228 at 129.142.224.250) State INIT
> 2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:83
> sofia/external/31354228 at 129.142.224.250 SOFIA INIT
> 2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:111
> (sofia/external/31354228 at 129.142.224.250) State Change CS_INIT ->
> CS_ROUTING
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_session.c:1012 Send
> signal sofia/external/31354228 at 129.142.224.250 [BREAK]
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:338
> (sofia/external/31354228 at 129.142.224.250) State INIT going to sleep
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/31354228 at 129.142.224.250) Running State Change
> CS_ROUTING
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/31354228 at 129.142.224.250) State ROUTING
> 2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:132
> sofia/external/31354228 at 129.142.224.250 SOFIA ROUTING
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:78
> sofia/external/31354228 at 129.142.224.250 Standard ROUTING
> 2010-01-14 22:47:14.540747 [INFO] mod_dialplan_xml.c:408 Processing
> 31354228->77344536 in context public
> Dialplan: sofia/external/31354228 at 129.142.224.250 parsing
> [public->ff-phone-sound-recorder] continue=false
> Dialplan: sofia/external/31354228 at 129.142.224.250 Regex (FAIL)
> [ff-phone-sound-recorder] destination_number(77344536) =~ /^77348624$/
> break=on-false
> Dialplan: sofia/external/31354228 at 129.142.224.250 parsing
> [public->ff-ivr] continue=false
> Dialplan: sofia/external/31354228 at 129.142.224.250 Regex (PASS)
> [ff-ivr] destination_number(77344536) =~ /^(\d+)$/ break=on-false
> Dialplan: sofia/external/31354228 at 129.142.224.250 Action
> socket(127.0.0.1:8084 full)
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:122
> (sofia/external/31354228 at 129.142.224.250) State Change CS_ROUTING ->
> CS_EXECUTE
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_session.c:1012 Send
> signal sofia/external/31354228 at 129.142.224.250 [BREAK]
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/31354228 at 129.142.224.250) State ROUTING going to sleep
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/31354228 at 129.142.224.250) Running State Change
> CS_EXECUTE
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:348
> (sofia/external/31354228 at 129.142.224.250) State EXECUTE
> 2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:181
> sofia/external/31354228 at 129.142.224.250 SOFIA EXECUTE
> 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:159
> sofia/external/31354228 at 129.142.224.250 Standard EXECUTE
> EXECUTE sofia/external/31354228 at 129.142.224.250 socket(127.0.0.1:8084 full)
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> set(hangup_after_bridge=true)
> EXECUTE sofia/external/31354228 at 129.142.224.250 set(hangup_after_bridge=true)
> 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
> sofia/external/31354228 at 129.142.224.250 SET
> [hangup_after_bridge]=[true]
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> set(ignore_early_media=true)
> EXECUTE sofia/external/31354228 at 129.142.224.250 set(ignore_early_media=true)
> 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
> sofia/external/31354228 at 129.142.224.250 SET
> [ignore_early_media]=[true]
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> set(call_timeout=20)
> EXECUTE sofia/external/31354228 at 129.142.224.250 set(call_timeout=20)
> 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
> sofia/external/31354228 at 129.142.224.250 SET [call_timeout]=[20]
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> set(originate_continue_on_timeout=true)
> EXECUTE sofia/external/31354228 at 129.142.224.250
> set(originate_continue_on_timeout=true)
> 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
> sofia/external/31354228 at 129.142.224.250 SET
> [originate_continue_on_timeout]=[true]
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> set(instant_ringback=true)
> EXECUTE sofia/external/31354228 at 129.142.224.250 set(instant_ringback=true)
> 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
> sofia/external/31354228 at 129.142.224.250 SET [instant_ringback]=[true]
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> set(ringback=%(2000,4000,440,480))
> EXECUTE sofia/external/31354228 at 129.142.224.250
> set(ringback=%(2000,4000,440,480))
> 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
> sofia/external/31354228 at 129.142.224.250 SET
> [ringback]=[%(2000,4000,440,480)]
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> set(transfer_ringback=%(2000,4000,440,480))
> EXECUTE sofia/external/31354228 at 129.142.224.250
> set(transfer_ringback=%(2000,4000,440,480))
> 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818
> sofia/external/31354228 at 129.142.224.250 SET
> [transfer_ringback]=[%(2000,4000,440,480)]
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> bind_meta_app(6 b ai
> transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
> public)
> EXECUTE sofia/external/31354228 at 129.142.224.250 bind_meta_app(6 b ai
> transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
> public)
> 2010-01-14 22:47:14.792781 [INFO] switch_ivr_async.c:2322 Bound B-Leg:
> 6 transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
> public
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> bind_meta_app(7 b ai transfer::ff-transfer- XML public)
> EXECUTE sofia/external/31354228 at 129.142.224.250 bind_meta_app(7 b ai
> transfer::ff-transfer- XML public)
> 2010-01-14 22:47:14.792781 [INFO] switch_ivr_async.c:2322 Bound B-Leg:
> 7 transfer::ff-transfer- XML public
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> bind_meta_app(1 b ai
> transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
> public)
> EXECUTE sofia/external/31354228 at 129.142.224.250 bind_meta_app(1 b ai
> transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
> public)
> 2010-01-14 22:47:14.792781 [INFO] switch_ivr_async.c:2322 Bound B-Leg:
> 1 transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML
> public
> 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute pre_answer()
> EXECUTE sofia/external/31354228 at 129.142.224.250 pre_answer()
> 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:736
> sofia/external/31354228 at 129.142.224.250 receive message [PROGRESS]
> 2010-01-14 22:47:14.792781 [INFO] mod_dptools.c:736 Sending early media
> 2010-01-14 22:47:14.792781 [DEBUG] sofia_glue.c:2409 AUDIO RTP
> [sofia/external/31354228 at 129.142.224.250] 10.224.54.102 port 22822 ->
> 129.142.224.250 port 14962 codec: 8 ms: 20
> 2010-01-14 22:47:14.792781 [DEBUG] switch_rtp.c:1168 Starting timer
> [soft] 160 bytes per 20ms
> 2010-01-14 22:47:14.807902 [INFO] mod_sofia.c:1779 Ring SDP:
> v=0
> o=FreeSWITCH 1263486412 1263486413 IN IP4 79.125.35.134
> s=FreeSWITCH
> c=IN IP4 79.125.35.134
> t=0 0
> m=audio 22822 RTP/AVP 8
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> 2010-01-14 22:47:14.807902 [NOTICE] mod_sofia.c:1782 Pre-Answer
> sofia/external/31354228 at 129.142.224.250!
> 2010-01-14 22:47:14.807902 [DEBUG] switch_core_session.c:633 Send
> signal sofia/external/31354228 at 129.142.224.250 [BREAK]
> 2010-01-14 22:47:14.827576 [DEBUG] sofia.c:3892 Channel
> sofia/external/31354228 at 129.142.224.250 skipping state [early][183]
> 2010-01-14 22:47:14.827576 [DEBUG] switch_ivr.c:551
> sofia/external/31354228 at 129.142.224.250 Command Execute
> bridge({sip_cid_type=rpid,origination_caller_id_name=77344536,origination_caller_id_number=77344536}sofia/gateway/gw1.fonet.dk/004546934488|loopback/ff-record-77344536)
> EXECUTE sofia/external/31354228 at 129.142.224.250
> bridge({sip_cid_type=rpid,origination_caller_id_name=77344536,origination_caller_id_number=77344536}sofia/gateway/gw1.fonet.dk/004546934488|loopback/ff-record-77344536)
> 2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:1738
> variable string 0 = [sip_cid_type=rpid]
> 2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:1738
> variable string 1 = [origination_caller_id_name=77344536]
> 2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:1738
> variable string 2 = [origination_caller_id_number=77344536]
> 2010-01-14 22:47:14.827576 [NOTICE] switch_channel.c:642 New Channel
> sofia/external/004546934488 [c2a1095c-015e-11df-8d0e-c7205d2c96c7]
> 2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:3227
> (sofia/external/004546934488) State Change CS_NEW -> CS_INIT
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_session.c:1012 Send
> signal sofia/external/004546934488 [BREAK]
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/004546934488) Running State Change CS_INIT
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:338
> (sofia/external/004546934488) State INIT
> 2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:83
> sofia/external/004546934488 SOFIA INIT
> 2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:111
> (sofia/external/004546934488) State Change CS_INIT -> CS_ROUTING
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_session.c:1012 Send
> signal sofia/external/004546934488 [BREAK]
> 2010-01-14 22:47:14.827576 [DEBUG] sofia.c:3897 Channel
> sofia/external/004546934488 entering state [calling][0]
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:338
> (sofia/external/004546934488) State INIT going to sleep
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/004546934488) Running State Change CS_ROUTING
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/004546934488) State ROUTING
> 2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:132
> sofia/external/004546934488 SOFIA ROUTING
> 2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:66
> (sofia/external/004546934488) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_session.c:1012 Send
> signal sofia/external/004546934488 [BREAK]
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/004546934488) State ROUTING going to sleep
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/004546934488) Running State Change CS_CONSUME_MEDIA
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:360
> (sofia/external/004546934488) State CONSUME_MEDIA
> 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:360
> (sofia/external/004546934488) State CONSUME_MEDIA going to sleep
> 2010-01-14 22:47:14.876945 [DEBUG] switch_ivr_originate.c:1057 Raw
> Codec Activation Success L16 at 8000hz 1 channel 20ms
> 2010-01-14 22:47:14.876945 [DEBUG] switch_core_codec.c:112
> sofia/external/31354228 at 129.142.224.250 Push codec L16:10
> 2010-01-14 22:47:14.876945 [DEBUG] switch_ivr_originate.c:1119 Play
> Ringback Tone [%(2000,4000,440,480)]
> 2010-01-14 22:47:14.876945 [DEBUG] switch_core_io.c:652
> sofia/external/31354228 at 129.142.224.250 receive message
> [TRANSCODING_NECESSARY]
> 2010-01-14 22:47:14.876945 [DEBUG] sofia.c:3897 Channel
> sofia/external/004546934488 entering state [calling][0]
> 2010-01-14 22:47:14.912774 [DEBUG] switch_rtp.c:2022 Correct ip/port confirmed.
> 2010-01-14 22:47:15.668817 [INFO] sofia.c:514 Update Callee ID to
> "004546934488" <004546934488>
> 2010-01-14 22:47:15.668817 [DEBUG] sofia.c:3897 Channel
> sofia/external/004546934488 entering state [proceeding][180]
> 2010-01-14 22:47:15.668817 [NOTICE] sofia.c:3967 Ring-Ready
> sofia/external/004546934488!
> 2010-01-14 22:47:15.668817 [DEBUG] sofia.c:3897 Channel
> sofia/external/004546934488 entering state [proceeding][183]
> 2010-01-14 22:47:15.668817 [DEBUG] sofia.c:3908 Remote SDP:
> v=0
> o=root 22377 22377 IN IP4 129.142.224.250
> s=session
> c=IN IP4 129.142.224.250
> t=0 0
> m=audio 11676 RTP/AVP 8
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
>
> 2010-01-14 22:47:15.668817 [DEBUG] sofia_glue.c:3344 Audio Codec
> Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
> 2010-01-14 22:47:15.668817 [DEBUG] sofia_glue.c:2171 Set Codec
> sofia/external/004546934488 PCMA/8000 20 ms 160 samples
> 2010-01-14 22:47:15.668817 [DEBUG] sofia_glue.c:2409 AUDIO RTP
> [sofia/external/004546934488] 10.224.54.102 port 17204 ->
> 129.142.224.250 port 11676 codec: 8 ms: 20
> 2010-01-14 22:47:15.668817 [DEBUG] switch_rtp.c:1168 Starting timer
> [soft] 160 bytes per 20ms
> 2010-01-14 22:47:15.668817 [NOTICE] sofia_glue.c:2937 Pre-Answer
> sofia/external/004546934488!
> 2010-01-14 22:47:15.668817 [DEBUG] switch_channel.c:2053 Send signal
> sofia/external/31354228 at 129.142.224.250 [BREAK]
>
> --
> Harry Vangberg ?<harry at vangberg.name> ?http://harry.vangberg.name
>



-- 
Harry Vangberg  <harry at vangberg.name>  http://harry.vangberg.name



From anthony.minessale at gmail.com  Thu Jan 14 15:54:22 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 17:54:22 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
Message-ID: <191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>

{bridge_early_media=true}
in the dial string in place of ignore_early_media=true

the first b leg in the list who sends 183 will become the ringback device
for A leg it will hear the early media
for that leg while the other legs still ring.  If some other leg answers the
final call will still be bridged to the leg who answered.


I would estimate it at $500 payable on the big paypal button on
http://www.freeswitch.org
but, I already added the patch to tree earlier today so I guess it's up to
you to pay it or not.




On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Anthony,
>
> I did take the "hint", don't worry. We will probably ask for a bounty
> but first we need to know:
> 1.- whether this is possible
> 2.- how long it would take
> 3.- how will it exactly work
> 4.- of course, what's the bounty (be gentle ;) )
>
> We would of course give this back to the community.
>
> in the meantime, bridge_early_media=true will discard the 2nd B-leg
> and conversely "false" it will keep on trying to connect and if it
> connects the other B-leg if will bridge to that one?
>
> Thanks
>
> David
>
> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > I added bridge_early_media=true to do the best I can do.
> > This is the most I will do, especially for free, nobody can take a hint
> that
> > you should be paying for all these custom requests so take it or leave it
> > but this thread is done.........
> >
> >
> >
> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin <
> sos at sokhapkin.dyndns.org>
> > wrote:
> >>
> >> No, not exactly. ignore_early_media doesn't pass early media to the
> caller
> >> if
> >> bypass_media is false.
> >>
> >> On Thursday 14 January 2010, Michael Jerris wrote:
> >> > this is exactly what ignore_early_media does now.
> >> >
> >> > Mike
> >> >
> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> >> > > The issue here is when "originate" routine should return and
> >> > > set "originate_status" variable. Current behavior is to return on
> >> > > early
> >> > > media, but what if to introduce a variable
> "originate_wait_for_answer"
> >> > > with default value "false" and use the variable in originate code to
> >> > > decide when to return - on 18X or "200 OK"?
> >> > >
> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> >> > >> he wants to call 3 people at once and let the A leg hear early
> media
> >> > >> from call #1 while call #2 and #3 still are progressing which is
> not
> >> > >> simple to do without doing thousands of dollars in development.
> >> > >>
> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
> >> > >>> customers
> >> > >>> can hear recordings?
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
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From dave at 3c.co.uk  Thu Jan 14 16:50:05 2010
From: dave at 3c.co.uk (David Knell)
Date: Fri, 15 Jan 2010 00:50:05 +0000
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
Message-ID: <1263516605.11216.91.camel@local.freepabx.com>

Hi David,

Excuse me if I'm being dumb (which is, sadly, a pretty common
occurrence) but there's a bunch of cases where what you're seeking fails
badly.  For example, if one of the calls ends up on a message along the
lines of "The number you have dialled has not been recognised.  Please
check, and try again", and that's the first early media (not unlikely,
all other things being equal), then the information passed back to the
caller is worse than useless.  Isn't it?

Cheers --

Dave

> Anthony,
> 
> I did take the "hint", don't worry. We will probably ask for a bounty
> but first we need to know:
> 1.- whether this is possible
> 2.- how long it would take
> 3.- how will it exactly work
> 4.- of course, what's the bounty (be gentle ;) )
> 
> We would of course give this back to the community.
> 
> in the meantime, bridge_early_media=true will discard the 2nd B-leg
> and conversely "false" it will keep on trying to connect and if it
> connects the other B-leg if will bridge to that one?
> 
> Thanks
> 
> David
> 
> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > I added bridge_early_media=true to do the best I can do.
> > This is the most I will do, especially for free, nobody can take a hint that
> > you should be paying for all these custom requests so take it or leave it
> > but this thread is done.........
> >
> >
> >
> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
> > wrote:
> >>
> >> No, not exactly. ignore_early_media doesn't pass early media to the caller
> >> if
> >> bypass_media is false.
> >>
> >> On Thursday 14 January 2010, Michael Jerris wrote:
> >> > this is exactly what ignore_early_media does now.
> >> >
> >> > Mike
> >> >
> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> >> > > The issue here is when "originate" routine should return and
> >> > > set "originate_status" variable. Current behavior is to return on
> >> > > early
> >> > > media, but what if to introduce a variable "originate_wait_for_answer"
> >> > > with default value "false" and use the variable in originate code to
> >> > > decide when to return - on 18X or "200 OK"?
> >> > >
> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> >> > >> he wants to call 3 people at once and let the A leg hear early media
> >> > >> from call #1 while call #2 and #3 still are progressing which is not
> >> > >> simple to do without doing thousands of dollars in development.
> >> > >>
> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
> >> > >>> customers
> >> > >>> can hear recordings?
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From jingwei.yang at gmail.com  Thu Jan 14 17:09:31 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Fri, 15 Jan 2010 09:09:31 +0800
Subject: [Freeswitch-users] Not able to capture the custom event
In-Reply-To: <191c3a031001140847q1238bbcdi4d9dd84648ee2a27@mail.gmail.com>
References: <13529f9d1001140124t42e9dfe5u3e70a6976aa032a8@mail.gmail.com>
	<191c3a031001140847q1238bbcdi4d9dd84648ee2a27@mail.gmail.com>
Message-ID: <13529f9d1001141709v6ed055f7i4b99a68517f58f38@mail.gmail.com>

Yes, it is the one. Thanks a lot for the help!

On Fri, Jan 15, 2010 at 12:47 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you cant subscribe to all custom events, you have to specify a list of
> subclasses
> you have to say events plain custom myevent::ACDnotify
>
> On Thu, Jan 14, 2010 at 3:24 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>
>> Hi Guys,
>>
>> I'm testing to see whether a custom event can be triggered and captured.
>> Here's the extension:
>>
>>     <extension name="test_ivr">
>>       <condition field="destination_number" expression="^777$">
>>         <action application="event" data="Event-Name=*CUSTOM*
>> ,Event-Subclass=myevent::ACDnotify"/>
>>         <action application="transfer" data="9999 XML default" />
>>       </condition>
>>     </extension>
>>
>> If I telnet and use "events plain all", I can get the event. However, if I
>> use "events plain CUSTOM" or "event plain CUSTOM", it doesn't get captured.
>> Is there anything wrong with the command?
>>
>> Regards,
>> -Jingwei
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Jan 14 17:23:35 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Jan 2010 17:23:35 -0800
Subject: [Freeswitch-users] No hangup tone after zap channel closed,
	tones 	in general
In-Reply-To: <8b18ffe229516c90285de8a54a00e845@thom.fr.eu.org>
References: <8b18ffe229516c90285de8a54a00e845@thom.fr.eu.org>
Message-ID: <87f2f3b91001141723u4b0b12e1rd0f922db493492cd@mail.gmail.com>

On Wed, Jan 13, 2010 at 10:25 AM, Fran?ois Legal <devel at thom.fr.eu.org>wrote:

> Hello,
>
>
>
> How shall I do to get a hangup tone on an FXS port after the channel is
> closed ?
>
> Is there something specific to configure in openzap ?
>
>
>
> For instance, if I call the following extension from an FXS port :
>
>     <!-- Send a 180 and let the far end generate ringback. -->
>     <extension name="ringback_180">
>       <condition field="destination_number" expression="^9980$">
>     <action application="ring_ready"/>
>     <action application="sleep" data="20000"/>
>     <action application="answer"/>
>     <action application="sleep" data="1000"/>
>     <action application="playback" data="voicemail/vm-goodbye.wav"/>
>     <action application="hangup"/>
>       </condition>
>     </extension>
>
> I get no tone after the hangup application is called.
>
>
>
> I also wonder if there is some documentation on the tones.conf file format,
> and about the variables uk-ring, us-ring, bong-ring, sit in vars.xml (when
> are they used, what is the syntax). I could not find any info on wiki.
>
>
>
http://wiki.freeswitch.org/wiki/TGML
-MC
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From mike at jerris.com  Thu Jan 14 17:45:55 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Jan 2010 20:45:55 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <1263516605.11216.91.camel@local.freepabx.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<1263516605.11216.91.camel@local.freepabx.com>
Message-ID: <38A2A04D-1950-4195-9C81-B6A6AE386161@jerris.com>

such are the perils of tdm voip interop.  on isdn we get indication of the failure with progress indicator, when doing sip interop, we should be able to choose this at a tdm gateway, unfortunately, there is no way in sip to indicate failure with inband progress so no way we can act on that.

Mike

On Jan 14, 2010, at 7:50 PM, David Knell wrote:

> Hi David,
> 
> Excuse me if I'm being dumb (which is, sadly, a pretty common
> occurrence) but there's a bunch of cases where what you're seeking fails
> badly.  For example, if one of the calls ends up on a message along the
> lines of "The number you have dialled has not been recognised.  Please
> check, and try again", and that's the first early media (not unlikely,
> all other things being equal), then the information passed back to the
> caller is worse than useless.  Isn't it?
> 
> Cheers --
> 
> Dave
> 
>> Anthony,
>> 
>> I did take the "hint", don't worry. We will probably ask for a bounty
>> but first we need to know:
>> 1.- whether this is possible
>> 2.- how long it would take
>> 3.- how will it exactly work
>> 4.- of course, what's the bounty (be gentle ;) )
>> 
>> We would of course give this back to the community.
>> 
>> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>> and conversely "false" it will keep on trying to connect and if it
>> connects the other B-leg if will bridge to that one?
>> 
>> Thanks
>> 
>> David
>> 
>> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> I added bridge_early_media=true to do the best I can do.
>>> This is the most I will do, especially for free, nobody can take a hint that
>>> you should be paying for all these custom requests so take it or leave it
>>> but this thread is done.........
>>> 
>>> 
>>> 
>>> On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
>>> wrote:
>>>> 
>>>> No, not exactly. ignore_early_media doesn't pass early media to the caller
>>>> if
>>>> bypass_media is false.
>>>> 
>>>> On Thursday 14 January 2010, Michael Jerris wrote:
>>>>> this is exactly what ignore_early_media does now.
>>>>> 
>>>>> Mike
>>>>> 
>>>>> On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>>>>>> The issue here is when "originate" routine should return and
>>>>>> set "originate_status" variable. Current behavior is to return on
>>>>>> early
>>>>>> media, but what if to introduce a variable "originate_wait_for_answer"
>>>>>> with default value "false" and use the variable in originate code to
>>>>>> decide when to return - on 18X or "200 OK"?
>>>>>> 
>>>>>> On Thursday 14 January 2010, Anthony Minessale wrote:
>>>>>>> he wants to call 3 people at once and let the A leg hear early media
>>>>>>> from call #1 while call #2 and #3 still are progressing which is not
>>>>>>> simple to do without doing thousands of dollars in development.
>>>>>>> 
>>>>>>> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>>>>>>>> What about sending Sip 183 with SDP (no 200OK), so that your
>>>>>>>> customers
>>>>>>>> can hear recordings?
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> --
>>> Anthony Minessale II
>>> 
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>> 
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>> 
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From david.villasmil.work at gmail.com  Thu Jan 14 17:54:15 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 02:54:15 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <1263516605.11216.91.camel@local.freepabx.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<1263516605.11216.91.camel@local.freepabx.com>
Message-ID: <9853f4ff1001141754r1a642814sacaa67be2cedbf4a@mail.gmail.com>

Dave,

In that case the early media is good also, the a-leg will hear the guy
is not available.

cheers

David

On Fri, Jan 15, 2010 at 1:50 AM, David Knell <dave at 3c.co.uk> wrote:
> Hi David,
>
> Excuse me if I'm being dumb (which is, sadly, a pretty common
> occurrence) but there's a bunch of cases where what you're seeking fails
> badly. ?For example, if one of the calls ends up on a message along the
> lines of "The number you have dialled has not been recognised. ?Please
> check, and try again", and that's the first early media (not unlikely,
> all other things being equal), then the information passed back to the
> caller is worse than useless. ?Isn't it?
>
> Cheers --
>
> Dave
>
>> Anthony,
>>
>> I did take the "hint", don't worry. We will probably ask for a bounty
>> but first we need to know:
>> 1.- whether this is possible
>> 2.- how long it would take
>> 3.- how will it exactly work
>> 4.- of course, what's the bounty (be gentle ;) )
>>
>> We would of course give this back to the community.
>>
>> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>> and conversely "false" it will keep on trying to connect and if it
>> connects the other B-leg if will bridge to that one?
>>
>> Thanks
>>
>> David
>>
>> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > I added bridge_early_media=true to do the best I can do.
>> > This is the most I will do, especially for free, nobody can take a hint that
>> > you should be paying for all these custom requests so take it or leave it
>> > but this thread is done.........
>> >
>> >
>> >
>> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
>> > wrote:
>> >>
>> >> No, not exactly. ignore_early_media doesn't pass early media to the caller
>> >> if
>> >> bypass_media is false.
>> >>
>> >> On Thursday 14 January 2010, Michael Jerris wrote:
>> >> > this is exactly what ignore_early_media does now.
>> >> >
>> >> > Mike
>> >> >
>> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> >> > > The issue here is when "originate" routine should return and
>> >> > > set "originate_status" variable. Current behavior is to return on
>> >> > > early
>> >> > > media, but what if to introduce a variable "originate_wait_for_answer"
>> >> > > with default value "false" and use the variable in originate code to
>> >> > > decide when to return - on 18X or "200 OK"?
>> >> > >
>> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> >> > >> he wants to call 3 people at once and let the A leg hear early media
>> >> > >> from call #1 while call #2 and #3 still are progressing which is not
>> >> > >> simple to do without doing thousands of dollars in development.
>> >> > >>
>> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>> >> > >>> customers
>> >> > >>> can hear recordings?
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From david.villasmil.work at gmail.com  Thu Jan 14 17:55:39 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 02:55:39 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
Message-ID: <9853f4ff1001141755o7045257cn8562d17a85d52450@mail.gmail.com>

Anthony,

     What about mixing the RTPs? what's the bounty for that?

Cheers

David

On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> {bridge_early_media=true}
> in the dial string in place of ignore_early_media=true
>
> the first b leg in the list who sends 183 will become the ringback device
> for A leg it will hear the early media
> for that leg while the other legs still ring.? If some other leg answers the
> final call will still be bridged to the leg who answered.
>
>
> I would estimate it at $500 payable on the big paypal button on
> http://www.freeswitch.org
> but, I already added the patch to tree earlier today so I guess it's up to
> you to pay it or not.
>
>
>
>
> On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> Anthony,
>>
>> I did take the "hint", don't worry. We will probably ask for a bounty
>> but first we need to know:
>> 1.- whether this is possible
>> 2.- how long it would take
>> 3.- how will it exactly work
>> 4.- of course, what's the bounty (be gentle ;) )
>>
>> We would of course give this back to the community.
>>
>> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>> and conversely "false" it will keep on trying to connect and if it
>> connects the other B-leg if will bridge to that one?
>>
>> Thanks
>>
>> David
>>
>> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > I added bridge_early_media=true to do the best I can do.
>> > This is the most I will do, especially for free, nobody can take a hint
>> > that
>> > you should be paying for all these custom requests so take it or leave
>> > it
>> > but this thread is done.........
>> >
>> >
>> >
>> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>> > <sos at sokhapkin.dyndns.org>
>> > wrote:
>> >>
>> >> No, not exactly. ignore_early_media doesn't pass early media to the
>> >> caller
>> >> if
>> >> bypass_media is false.
>> >>
>> >> On Thursday 14 January 2010, Michael Jerris wrote:
>> >> > this is exactly what ignore_early_media does now.
>> >> >
>> >> > Mike
>> >> >
>> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> >> > > The issue here is when "originate" routine should return and
>> >> > > set "originate_status" variable. Current behavior is to return on
>> >> > > early
>> >> > > media, but what if to introduce a variable
>> >> > > "originate_wait_for_answer"
>> >> > > with default value "false" and use the variable in originate code
>> >> > > to
>> >> > > decide when to return - on 18X or "200 OK"?
>> >> > >
>> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> >> > >> he wants to call 3 people at once and let the A leg hear early
>> >> > >> media
>> >> > >> from call #1 while call #2 and #3 still are progressing which is
>> >> > >> not
>> >> > >> simple to do without doing thousands of dollars in development.
>> >> > >>
>> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>> >> > >>> customers
>> >> > >>> can hear recordings?
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Thu Jan 14 18:00:34 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Jan 2010 18:00:34 -0800
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <ade741d61001141154m75b9fcbak76e874e80d023ac0@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
	<8213d6071001141109l1d475b1j3af35b5586708e42@mail.gmail.com>
	<ade741d61001141123x393512f6x81768de82a84c214@mail.gmail.com>
	<8213d6071001141131m3a6ab686xf716dc26983ade5a@mail.gmail.com>
	<ade741d61001141154m75b9fcbak76e874e80d023ac0@mail.gmail.com>
Message-ID: <87f2f3b91001141800q68550eeei310351bd4f7fcdab@mail.gmail.com>

On Thu, Jan 14, 2010 at 11:54 AM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> thanks, how would it look in dialplan if I have to call a javascript?
>
>
>
<action application="set" data="api_hangup_hook=jsrun /path/to/my_script.js
arg1 arg2 arg3"/>

-MC
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From msc at freeswitch.org  Thu Jan 14 18:05:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Jan 2010 18:05:32 -0800
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001141754r1a642814sacaa67be2cedbf4a@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<1263516605.11216.91.camel@local.freepabx.com>
	<9853f4ff1001141754r1a642814sacaa67be2cedbf4a@mail.gmail.com>
Message-ID: <87f2f3b91001141805t3a50b114ib7098d0efd04f4a9@mail.gmail.com>

On Thu, Jan 14, 2010 at 5:54 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Dave,
>
> In that case the early media is good also, the a-leg will hear the guy
> is not available.
>
> But what if he is available on one of the other b-legs? What should happen
in that scenario?

-MC


> cheers
>
> David
>
> On Fri, Jan 15, 2010 at 1:50 AM, David Knell <dave at 3c.co.uk> wrote:
> > Hi David,
> >
> > Excuse me if I'm being dumb (which is, sadly, a pretty common
> > occurrence) but there's a bunch of cases where what you're seeking fails
> > badly.  For example, if one of the calls ends up on a message along the
> > lines of "The number you have dialled has not been recognised.  Please
> > check, and try again", and that's the first early media (not unlikely,
> > all other things being equal), then the information passed back to the
> > caller is worse than useless.  Isn't it?
> >
> > Cheers --
> >
> > Dave
> >
> >> Anthony,
> >>
> >> I did take the "hint", don't worry. We will probably ask for a bounty
> >> but first we need to know:
> >> 1.- whether this is possible
> >> 2.- how long it would take
> >> 3.- how will it exactly work
> >> 4.- of course, what's the bounty (be gentle ;) )
> >>
> >> We would of course give this back to the community.
> >>
> >> in the meantime, bridge_early_media=true will discard the 2nd B-leg
> >> and conversely "false" it will keep on trying to connect and if it
> >> connects the other B-leg if will bridge to that one?
> >>
> >> Thanks
> >>
> >> David
> >>
> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >> > I added bridge_early_media=true to do the best I can do.
> >> > This is the most I will do, especially for free, nobody can take a
> hint that
> >> > you should be paying for all these custom requests so take it or leave
> it
> >> > but this thread is done.........
> >> >
> >> >
> >> >
> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin <
> sos at sokhapkin.dyndns.org>
> >> > wrote:
> >> >>
> >> >> No, not exactly. ignore_early_media doesn't pass early media to the
> caller
> >> >> if
> >> >> bypass_media is false.
> >> >>
> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
> >> >> > this is exactly what ignore_early_media does now.
> >> >> >
> >> >> > Mike
> >> >> >
> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> >> >> > > The issue here is when "originate" routine should return and
> >> >> > > set "originate_status" variable. Current behavior is to return on
> >> >> > > early
> >> >> > > media, but what if to introduce a variable
> "originate_wait_for_answer"
> >> >> > > with default value "false" and use the variable in originate code
> to
> >> >> > > decide when to return - on 18X or "200 OK"?
> >> >> > >
> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> >> >> > >> he wants to call 3 people at once and let the A leg hear early
> media
> >> >> > >> from call #1 while call #2 and #3 still are progressing which is
> not
> >> >> > >> simple to do without doing thousands of dollars in development.
> >> >> > >>
> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com>
> wrote:
> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
> >> >> > >>> customers
> >> >> > >>> can hear recordings?
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >>
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.villasmil.work at gmail.com  Thu Jan 14 18:13:32 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 03:13:32 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <87f2f3b91001141805t3a50b114ib7098d0efd04f4a9@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<1263516605.11216.91.camel@local.freepabx.com>
	<9853f4ff1001141754r1a642814sacaa67be2cedbf4a@mail.gmail.com>
	<87f2f3b91001141805t3a50b114ib7098d0efd04f4a9@mail.gmail.com>
Message-ID: <9853f4ff1001141813n58c50df9q9e97d352a8d97199@mail.gmail.com>

Michael,

     At least in my case, there will only be 2 legs, 1 providing the
music/audio, the other is the terminating side.

David

On Fri, Jan 15, 2010 at 3:05 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Thu, Jan 14, 2010 at 5:54 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> Dave,
>>
>> In that case the early media is good also, the a-leg will hear the guy
>> is not available.
>>
> But what if he is available on one of the other b-legs? What should happen
> in that scenario?
>
> -MC
>
>>
>> cheers
>>
>> David
>>
>> On Fri, Jan 15, 2010 at 1:50 AM, David Knell <dave at 3c.co.uk> wrote:
>> > Hi David,
>> >
>> > Excuse me if I'm being dumb (which is, sadly, a pretty common
>> > occurrence) but there's a bunch of cases where what you're seeking fails
>> > badly. ?For example, if one of the calls ends up on a message along the
>> > lines of "The number you have dialled has not been recognised. ?Please
>> > check, and try again", and that's the first early media (not unlikely,
>> > all other things being equal), then the information passed back to the
>> > caller is worse than useless. ?Isn't it?
>> >
>> > Cheers --
>> >
>> > Dave
>> >
>> >> Anthony,
>> >>
>> >> I did take the "hint", don't worry. We will probably ask for a bounty
>> >> but first we need to know:
>> >> 1.- whether this is possible
>> >> 2.- how long it would take
>> >> 3.- how will it exactly work
>> >> 4.- of course, what's the bounty (be gentle ;) )
>> >>
>> >> We would of course give this back to the community.
>> >>
>> >> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>> >> and conversely "false" it will keep on trying to connect and if it
>> >> connects the other B-leg if will bridge to that one?
>> >>
>> >> Thanks
>> >>
>> >> David
>> >>
>> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> >> <anthony.minessale at gmail.com> wrote:
>> >> > I added bridge_early_media=true to do the best I can do.
>> >> > This is the most I will do, especially for free, nobody can take a
>> >> > hint that
>> >> > you should be paying for all these custom requests so take it or
>> >> > leave it
>> >> > but this thread is done.........
>> >> >
>> >> >
>> >> >
>> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>> >> > <sos at sokhapkin.dyndns.org>
>> >> > wrote:
>> >> >>
>> >> >> No, not exactly. ignore_early_media doesn't pass early media to the
>> >> >> caller
>> >> >> if
>> >> >> bypass_media is false.
>> >> >>
>> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
>> >> >> > this is exactly what ignore_early_media does now.
>> >> >> >
>> >> >> > Mike
>> >> >> >
>> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> >> >> > > The issue here is when "originate" routine should return and
>> >> >> > > set "originate_status" variable. Current behavior is to return
>> >> >> > > on
>> >> >> > > early
>> >> >> > > media, but what if to introduce a variable
>> >> >> > > "originate_wait_for_answer"
>> >> >> > > with default value "false" and use the variable in originate
>> >> >> > > code to
>> >> >> > > decide when to return - on 18X or "200 OK"?
>> >> >> > >
>> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> >> >> > >> he wants to call 3 people at once and let the A leg hear early
>> >> >> > >> media
>> >> >> > >> from call #1 while call #2 and #3 still are progressing which
>> >> >> > >> is not
>> >> >> > >> simple to do without doing thousands of dollars in development.
>> >> >> > >>
>> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com>
>> >> >> > >> wrote:
>> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>> >> >> > >>> customers
>> >> >> > >>> can hear recordings?
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >>
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> > Anthony Minessale II
>> >> >
>> >> > FreeSWITCH http://www.freeswitch.org/
>> >> > ClueCon http://www.cluecon.com/
>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >
>> >> > AIM: anthm
>> >> > MSN:anthony_minessale at hotmail.com
>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> > IRC: irc.freenode.net #freeswitch
>> >> >
>> >> > FreeSWITCH Developer Conference
>> >> > sip:888 at conference.freeswitch.org
>> >> > iax:guest at conference.freeswitch.org/888
>> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> > pstn:+19193869900
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Thu Jan 14 18:20:18 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Jan 2010 20:20:18 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001141813n58c50df9q9e97d352a8d97199@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<1263516605.11216.91.camel@local.freepabx.com>
	<9853f4ff1001141754r1a642814sacaa67be2cedbf4a@mail.gmail.com>
	<87f2f3b91001141805t3a50b114ib7098d0efd04f4a9@mail.gmail.com>
	<9853f4ff1001141813n58c50df9q9e97d352a8d97199@mail.gmail.com>
Message-ID: <5BA86C46-D725-457C-A6EA-6C2650EE2FAE@freeswitch.org>

Are you just trying to provide music ringback?

/b

On Jan 14, 2010, at 8:13 PM, David Villasmil wrote:

> Michael,
> 
>     At least in my case, there will only be 2 legs, 1 providing the
> music/audio, the other is the terminating side.
> 
> David




From david.villasmil.work at gmail.com  Thu Jan 14 19:04:29 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 04:04:29 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <5BA86C46-D725-457C-A6EA-6C2650EE2FAE@freeswitch.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<1263516605.11216.91.camel@local.freepabx.com>
	<9853f4ff1001141754r1a642814sacaa67be2cedbf4a@mail.gmail.com>
	<87f2f3b91001141805t3a50b114ib7098d0efd04f4a9@mail.gmail.com>
	<9853f4ff1001141813n58c50df9q9e97d352a8d97199@mail.gmail.com>
	<5BA86C46-D725-457C-A6EA-6C2650EE2FAE@freeswitch.org>
Message-ID: <9853f4ff1001141904h386bfe19kb70dcdb281c3e3b8@mail.gmail.com>

Brian,

     Coming from an external content provider via SIP, yes. That's why
I would need the rtps mixed.

Thanks

David

On Fri, Jan 15, 2010 at 3:20 AM, Brian West <brian at freeswitch.org> wrote:
> Are you just trying to provide music ringback?
>
> /b
>
> On Jan 14, 2010, at 8:13 PM, David Villasmil wrote:
>
>> Michael,
>>
>> ? ? At least in my case, there will only be 2 legs, 1 providing the
>> music/audio, the other is the terminating side.
>>
>> David
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From anthony.minessale at gmail.com  Thu Jan 14 19:17:31 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jan 2010 21:17:31 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001141917s4d679a7fqe9d410b407e27573@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
	<9853f4ff1001141755o7045257cn8562d17a85d52450@mail.gmail.com>
	<191c3a031001141917s4d679a7fqe9d410b407e27573@mail.gmail.com>
Message-ID: <191c3a031001141917o4dfe19b9h4f9b50c3ab184281@mail.gmail.com>

If you want to mix them it would cost double.

On Jan 14, 2010 8:01 PM, "David Villasmil" <david.villasmil.work at gmail.com>
wrote:

Anthony,

    What about mixing the RTPs? what's the bounty for that?

Cheers

David

On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale

<anthony.minessale at gmail.com> wrote: > {bridge_early_media=true} > in the
dial string in place of ig...
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From pete at privateconnect.com  Fri Jan 15 00:21:24 2010
From: pete at privateconnect.com (Pete Mueller)
Date: Fri, 15 Jan 2010 01:21:24 -0700
Subject: [Freeswitch-users] Eavesdrop in LUA
In-Reply-To: <4B4F33C7.6020403@laposte.net>
References: <4B4ED32E.30706@laposte.net> <4B4F33C7.6020403@laposte.net>
Message-ID: <007201ca95bb$ba673770$2f35a650$@com>

I had a similar problem.  I solved it by first making bridging the call
between A and B.
Then originate C with a LUA script, the last line of which is:

session:execute("eavesdrop", uuid_of_a_leg)

The down side here is that A and B can talk while C is ringing, but in my
case that is not a problem.
-p

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod
Sent: Thursday, January 14, 2010 8:10 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Eavesdrop in LUA

Hi all,

I have an incomplete solution for those interested.

I did it like this in dialplan:
        <action application="set" data="api_on_answer=originate 
sofia/internal/2000 at 10.10.10.10 &eavesdrop(${uuid})"/>-->
        <action application="bridge" 
data="{ecoute=${caller_id_number}}sofia/gateway/${caller_id_number}/${destin
ation_number}"/>

so when a call is setup, FS initiate a new call to 2000 and eavesdrop 
the call.
But I have a small problem, the callee receives no sound until the 
eavesdropper send a SIP reply, so there is a 2-3 seconds delay before 
caller and callee can talk each other.

rod


rod a ?crit :
> Hi all,
>
> I'm trying to do this in LUA:
>      A call B
>
> and I'd like to setup a new call to C with eavesdrop of A conversation 
> with B.
>
> I have no idea how to do this if someone can help.
> I switched to LUA cause I see no way to achieve this with dialplan 
> (snippets are welcome).
>
> regards,
> rod
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From jingwei.yang at gmail.com  Fri Jan 15 02:51:44 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Fri, 15 Jan 2010 18:51:44 +0800
Subject: [Freeswitch-users]  Questions about mod_fifo
Message-ID: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>

Hi Guys,

I'm implementing an ACD system using ESL and mod_fifo. Based on what Anthony
suggested in this post:
http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html

*You can make an event socket application that listens for FIFO events and
keeps track of what FIFOs are currently busy and when there are people
waiting you can have that script generate a call to a group of SIP phones so
when the first one answers, it sends them in as an agent where they can
field the calls.
*

1. How should I handle the concurrent issues? If I bridge a user to two
agents and both of them answers, how does FS take care of this situation?
Will a slower agent get a busy tone automatically?

2. If the socket application is brought up after some users have called in,
what command should I use to check the busy queues? fifo list?

3. Am I using fifo list and fifo count correctly?

here's the testing dialplan:

    <extension name="in_fifo">
      <condition field="destination_number" expression="^779$">
        <action application="set" data="fifo_music=$${hold_music}"/>
        <action application="answer"/>
        <action application="fifo" data="*myq *in"/>
      </condition>
    </extension>

when a call comes in and gets queued, these are the results of some commands
I tried.

freeswitch at localhost.localdomain> fifo list
API CALL [fifo(list)] output:
<fifo_report>
  <fifo name="cool_fifo at 192.168.1.104" consumer_count="0" caller_count="0"
waiting_count="0" importance="0">
    <callers></callers>
    <consumers></consumers>
  </fifo>
</fifo_report>


freeswitch at localhost.localdomain> fifo list myq
API CALL [fifo(list myq)] output:
<fifo_report></fifo_report>


freeswitch at localhost.localdomain> fifo count myq
API CALL [fifo(count myq)] output:
none

It seems *myq* doesn't get created at all? Please enlighten.

Thanks and best regards,
-Jingwei
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From devel at thom.fr.eu.org  Fri Jan 15 02:53:48 2010
From: devel at thom.fr.eu.org (devel at thom.fr.eu.org)
Date: Fri, 15 Jan 2010 11:53:48 +0100
Subject: [Freeswitch-users] No hangup tone after zap channel closed,
	tones 	in general
In-Reply-To: <87f2f3b91001141723u4b0b12e1rd0f922db493492cd@mail.gmail.com>
References: <8b18ffe229516c90285de8a54a00e845@thom.fr.eu.org>
	<87f2f3b91001141723u4b0b12e1rd0f922db493492cd@mail.gmail.com>
Message-ID: <001201ca95d1$045eb6e0$0d1c24a0$@fr.eu.org>

Thank you for the link. I googled through but could not find anything
relevant.

 

So then with my FXS port, do I have to, when a call is over, bridge the
channel (which is either A or B leg depending on the cases) to an extension
with for instance

<action application="playback" data="tone_stream://%(500,500,440)"/>

 

Thanks

 

Fran?ois

 

 

De : freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael
Collins
Envoy? : vendredi 15 janvier 2010 02:24
? : freeswitch-users at lists.freeswitch.org
Objet : Re: [Freeswitch-users] No hangup tone after zap channel closed,
tones in general

 

 

On Wed, Jan 13, 2010 at 10:25 AM, Fran?ois Legal <devel at thom.fr.eu.org>
wrote:

Hello,

 

How shall I do to get a hangup tone on an FXS port after the channel is
closed ?

Is there something specific to configure in openzap ?

 

For instance, if I call the following extension from an FXS port :

    <!-- Send a 180 and let the far end generate ringback. -->
    <extension name="ringback_180">
      <condition field="destination_number" expression="^9980$">
    <action application="ring_ready"/>
    <action application="sleep" data="20000"/>
    <action application="answer"/>
    <action application="sleep" data="1000"/>
    <action application="playback" data="voicemail/vm-goodbye.wav"/>
    <action application="hangup"/>
      </condition>
    </extension>

I get no tone after the hangup application is called.

 

I also wonder if there is some documentation on the tones.conf file format,
and about the variables uk-ring, us-ring, bong-ring, sit in vars.xml (when
are they used, what is the syntax). I could not find any info on wiki.

 

http://wiki.freeswitch.org/wiki/TGML
-MC

 

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From david.villasmil.work at gmail.com  Fri Jan 15 03:14:42 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 12:14:42 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001141917o4dfe19b9h4f9b50c3ab184281@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
	<9853f4ff1001141755o7045257cn8562d17a85d52450@mail.gmail.com>
	<191c3a031001141917s4d679a7fqe9d410b407e27573@mail.gmail.com>
	<191c3a031001141917o4dfe19b9h4f9b50c3ab184281@mail.gmail.com>
Message-ID: <9853f4ff1001150314s4ab3b34dya96ce14fd7bac43a@mail.gmail.com>

Anthony,

Thanks a lot, I believe you can count on it. Be aware this will not be
done immediately, though. It will be in 1-2 months.

David

On Fri, Jan 15, 2010 at 4:17 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> If you want to mix them it would cost double.
>
> On Jan 14, 2010 8:01 PM, "David Villasmil" <david.villasmil.work at gmail.com>
> wrote:
>
> Anthony,
>
> ? ? What about mixing the RTPs? what's the bounty for that?
>
> Cheers
>
> David
>
> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>
> <anthony.minessale at gmail.com> wrote: > {bridge_early_media=true} > in the
> dial string in place of ig...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From david.villasmil.work at gmail.com  Fri Jan 15 03:21:16 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 12:21:16 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
Message-ID: <9853f4ff1001150321v1fb0a793s9836016aa8eb39e@mail.gmail.com>

Hello again Anthony,

I just tested it, and although functionality does not, first incoming
audio is coming in all garbled... do you know why?

David

On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> {bridge_early_media=true}
> in the dial string in place of ignore_early_media=true
>
> the first b leg in the list who sends 183 will become the ringback device
> for A leg it will hear the early media
> for that leg while the other legs still ring.? If some other leg answers the
> final call will still be bridged to the leg who answered.
>
>
> I would estimate it at $500 payable on the big paypal button on
> http://www.freeswitch.org
> but, I already added the patch to tree earlier today so I guess it's up to
> you to pay it or not.
>
>
>
>
> On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> Anthony,
>>
>> I did take the "hint", don't worry. We will probably ask for a bounty
>> but first we need to know:
>> 1.- whether this is possible
>> 2.- how long it would take
>> 3.- how will it exactly work
>> 4.- of course, what's the bounty (be gentle ;) )
>>
>> We would of course give this back to the community.
>>
>> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>> and conversely "false" it will keep on trying to connect and if it
>> connects the other B-leg if will bridge to that one?
>>
>> Thanks
>>
>> David
>>
>> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > I added bridge_early_media=true to do the best I can do.
>> > This is the most I will do, especially for free, nobody can take a hint
>> > that
>> > you should be paying for all these custom requests so take it or leave
>> > it
>> > but this thread is done.........
>> >
>> >
>> >
>> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>> > <sos at sokhapkin.dyndns.org>
>> > wrote:
>> >>
>> >> No, not exactly. ignore_early_media doesn't pass early media to the
>> >> caller
>> >> if
>> >> bypass_media is false.
>> >>
>> >> On Thursday 14 January 2010, Michael Jerris wrote:
>> >> > this is exactly what ignore_early_media does now.
>> >> >
>> >> > Mike
>> >> >
>> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> >> > > The issue here is when "originate" routine should return and
>> >> > > set "originate_status" variable. Current behavior is to return on
>> >> > > early
>> >> > > media, but what if to introduce a variable
>> >> > > "originate_wait_for_answer"
>> >> > > with default value "false" and use the variable in originate code
>> >> > > to
>> >> > > decide when to return - on 18X or "200 OK"?
>> >> > >
>> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> >> > >> he wants to call 3 people at once and let the A leg hear early
>> >> > >> media
>> >> > >> from call #1 while call #2 and #3 still are progressing which is
>> >> > >> not
>> >> > >> simple to do without doing thousands of dollars in development.
>> >> > >>
>> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>> >> > >>> customers
>> >> > >>> can hear recordings?
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From sos at sokhapkin.dyndns.org  Fri Jan 15 03:51:05 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 15 Jan 2010 06:51:05 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001150321v1fb0a793s9836016aa8eb39e@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
	<9853f4ff1001150321v1fb0a793s9836016aa8eb39e@mail.gmail.com>
Message-ID: <201001150651.05983.sos@sokhapkin.dyndns.org>

Which audio? Early media or after 200 OK?

On Friday 15 January 2010, David Villasmil wrote:
> Hello again Anthony,
>
> I just tested it, and although functionality does not, first incoming
> audio is coming in all garbled... do you know why?
>
> David
>
> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>
> <anthony.minessale at gmail.com> wrote:
> > {bridge_early_media=true}
> > in the dial string in place of ignore_early_media=true
> >
> > the first b leg in the list who sends 183 will become the ringback device
> > for A leg it will hear the early media
> > for that leg while the other legs still ring.? If some other leg answers
> > the final call will still be bridged to the leg who answered.
> >
> >
> > I would estimate it at $500 payable on the big paypal button on
> > http://www.freeswitch.org
> > but, I already added the patch to tree earlier today so I guess it's up
> > to you to pay it or not.
> >
> >
> >
> >
> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> >
> > <david.villasmil.work at gmail.com> wrote:
> >> Anthony,
> >>
> >> I did take the "hint", don't worry. We will probably ask for a bounty
> >> but first we need to know:
> >> 1.- whether this is possible
> >> 2.- how long it would take
> >> 3.- how will it exactly work
> >> 4.- of course, what's the bounty (be gentle ;) )
> >>
> >> We would of course give this back to the community.
> >>
> >> in the meantime, bridge_early_media=true will discard the 2nd B-leg
> >> and conversely "false" it will keep on trying to connect and if it
> >> connects the other B-leg if will bridge to that one?
> >>
> >> Thanks
> >>
> >> David
> >>
> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> >>
> >> <anthony.minessale at gmail.com> wrote:
> >> > I added bridge_early_media=true to do the best I can do.
> >> > This is the most I will do, especially for free, nobody can take a
> >> > hint that
> >> > you should be paying for all these custom requests so take it or leave
> >> > it
> >> > but this thread is done.........
> >> >
> >> >
> >> >
> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
> >> > <sos at sokhapkin.dyndns.org>
> >> >
> >> > wrote:
> >> >> No, not exactly. ignore_early_media doesn't pass early media to the
> >> >> caller
> >> >> if
> >> >> bypass_media is false.
> >> >>
> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
> >> >> > this is exactly what ignore_early_media does now.
> >> >> >
> >> >> > Mike
> >> >> >
> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> >> >> > > The issue here is when "originate" routine should return and
> >> >> > > set "originate_status" variable. Current behavior is to return on
> >> >> > > early
> >> >> > > media, but what if to introduce a variable
> >> >> > > "originate_wait_for_answer"
> >> >> > > with default value "false" and use the variable in originate code
> >> >> > > to
> >> >> > > decide when to return - on 18X or "200 OK"?
> >> >> > >
> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> >> >> > >> he wants to call 3 people at once and let the A leg hear early
> >> >> > >> media
> >> >> > >> from call #1 while call #2 and #3 still are progressing which is
> >> >> > >> not
> >> >> > >> simple to do without doing thousands of dollars in development.
> >> >> > >>
> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
> >> >> > >>> customers
> >> >> > >>> can hear recordings?
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> >> >users http://www.freeswitch.org
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
> >> >>ers http://www.freeswitch.org
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> >> >rs http://www.freeswitch.org
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From david.villasmil.work at gmail.com  Fri Jan 15 04:15:46 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 13:15:46 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001150651.05983.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
	<9853f4ff1001150321v1fb0a793s9836016aa8eb39e@mail.gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
Message-ID: <9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>

Yeah, sorry. Early media. Audio after 200 is fine. Early media was ok
before the change.

David

On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> Which audio? Early media or after 200 OK?
>
> On Friday 15 January 2010, David Villasmil wrote:
>> Hello again Anthony,
>>
>> I just tested it, and although functionality does not, first incoming
>> audio is coming in all garbled... do you know why?
>>
>> David
>>
>> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>>
>> <anthony.minessale at gmail.com> wrote:
>> > {bridge_early_media=true}
>> > in the dial string in place of ignore_early_media=true
>> >
>> > the first b leg in the list who sends 183 will become the ringback device
>> > for A leg it will hear the early media
>> > for that leg while the other legs still ring.? If some other leg answers
>> > the final call will still be bridged to the leg who answered.
>> >
>> >
>> > I would estimate it at $500 payable on the big paypal button on
>> > http://www.freeswitch.org
>> > but, I already added the patch to tree earlier today so I guess it's up
>> > to you to pay it or not.
>> >
>> >
>> >
>> >
>> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
>> >
>> > <david.villasmil.work at gmail.com> wrote:
>> >> Anthony,
>> >>
>> >> I did take the "hint", don't worry. We will probably ask for a bounty
>> >> but first we need to know:
>> >> 1.- whether this is possible
>> >> 2.- how long it would take
>> >> 3.- how will it exactly work
>> >> 4.- of course, what's the bounty (be gentle ;) )
>> >>
>> >> We would of course give this back to the community.
>> >>
>> >> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>> >> and conversely "false" it will keep on trying to connect and if it
>> >> connects the other B-leg if will bridge to that one?
>> >>
>> >> Thanks
>> >>
>> >> David
>> >>
>> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> >>
>> >> <anthony.minessale at gmail.com> wrote:
>> >> > I added bridge_early_media=true to do the best I can do.
>> >> > This is the most I will do, especially for free, nobody can take a
>> >> > hint that
>> >> > you should be paying for all these custom requests so take it or leave
>> >> > it
>> >> > but this thread is done.........
>> >> >
>> >> >
>> >> >
>> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>> >> > <sos at sokhapkin.dyndns.org>
>> >> >
>> >> > wrote:
>> >> >> No, not exactly. ignore_early_media doesn't pass early media to the
>> >> >> caller
>> >> >> if
>> >> >> bypass_media is false.
>> >> >>
>> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
>> >> >> > this is exactly what ignore_early_media does now.
>> >> >> >
>> >> >> > Mike
>> >> >> >
>> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> >> >> > > The issue here is when "originate" routine should return and
>> >> >> > > set "originate_status" variable. Current behavior is to return on
>> >> >> > > early
>> >> >> > > media, but what if to introduce a variable
>> >> >> > > "originate_wait_for_answer"
>> >> >> > > with default value "false" and use the variable in originate code
>> >> >> > > to
>> >> >> > > decide when to return - on 18X or "200 OK"?
>> >> >> > >
>> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> >> >> > >> he wants to call 3 people at once and let the A leg hear early
>> >> >> > >> media
>> >> >> > >> from call #1 while call #2 and #3 still are progressing which is
>> >> >> > >> not
>> >> >> > >> simple to do without doing thousands of dollars in development.
>> >> >> > >>
>> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>> >> >> > >>> customers
>> >> >> > >>> can hear recordings?
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >> >> >users http://www.freeswitch.org
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> >> >>ers http://www.freeswitch.org
>> >> >
>> >> > --
>> >> > Anthony Minessale II
>> >> >
>> >> > FreeSWITCH http://www.freeswitch.org/
>> >> > ClueCon http://www.cluecon.com/
>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >
>> >> > AIM: anthm
>> >> > MSN:anthony_minessale at hotmail.com
>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> > IRC: irc.freenode.net #freeswitch
>> >> >
>> >> > FreeSWITCH Developer Conference
>> >> > sip:888 at conference.freeswitch.org
>> >> > iax:guest at conference.freeswitch.org/888
>> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> > pstn:+19193869900
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> >> >rs http://www.freeswitch.org
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From sos at sokhapkin.dyndns.org  Fri Jan 15 04:26:17 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 15 Jan 2010 07:26:17 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
Message-ID: <201001150726.17430.sos@sokhapkin.dyndns.org>

Is bypass_media on or off?

On Friday 15 January 2010, David Villasmil wrote:
> Yeah, sorry. Early media. Audio after 200 is fine. Early media was ok
> before the change.
>
> David
>
> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
>
> <sos at sokhapkin.dyndns.org> wrote:
> > Which audio? Early media or after 200 OK?
> >
> > On Friday 15 January 2010, David Villasmil wrote:
> >> Hello again Anthony,
> >>
> >> I just tested it, and although functionality does not, first incoming
> >> audio is coming in all garbled... do you know why?
> >>
> >> David
> >>
> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
> >>
> >> <anthony.minessale at gmail.com> wrote:
> >> > {bridge_early_media=true}
> >> > in the dial string in place of ignore_early_media=true
> >> >
> >> > the first b leg in the list who sends 183 will become the ringback
> >> > device for A leg it will hear the early media
> >> > for that leg while the other legs still ring.? If some other leg
> >> > answers the final call will still be bridged to the leg who answered.
> >> >
> >> >
> >> > I would estimate it at $500 payable on the big paypal button on
> >> > http://www.freeswitch.org
> >> > but, I already added the patch to tree earlier today so I guess it's
> >> > up to you to pay it or not.
> >> >
> >> >
> >> >
> >> >
> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> >> >
> >> > <david.villasmil.work at gmail.com> wrote:
> >> >> Anthony,
> >> >>
> >> >> I did take the "hint", don't worry. We will probably ask for a bounty
> >> >> but first we need to know:
> >> >> 1.- whether this is possible
> >> >> 2.- how long it would take
> >> >> 3.- how will it exactly work
> >> >> 4.- of course, what's the bounty (be gentle ;) )
> >> >>
> >> >> We would of course give this back to the community.
> >> >>
> >> >> in the meantime, bridge_early_media=true will discard the 2nd B-leg
> >> >> and conversely "false" it will keep on trying to connect and if it
> >> >> connects the other B-leg if will bridge to that one?
> >> >>
> >> >> Thanks
> >> >>
> >> >> David
> >> >>
> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> >> >>
> >> >> <anthony.minessale at gmail.com> wrote:
> >> >> > I added bridge_early_media=true to do the best I can do.
> >> >> > This is the most I will do, especially for free, nobody can take a
> >> >> > hint that
> >> >> > you should be paying for all these custom requests so take it or
> >> >> > leave it
> >> >> > but this thread is done.........
> >> >> >
> >> >> >
> >> >> >
> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
> >> >> > <sos at sokhapkin.dyndns.org>
> >> >> >
> >> >> > wrote:
> >> >> >> No, not exactly. ignore_early_media doesn't pass early media to
> >> >> >> the caller
> >> >> >> if
> >> >> >> bypass_media is false.
> >> >> >>
> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
> >> >> >> > this is exactly what ignore_early_media does now.
> >> >> >> >
> >> >> >> > Mike
> >> >> >> >
> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> >> >> >> > > The issue here is when "originate" routine should return and
> >> >> >> > > set "originate_status" variable. Current behavior is to return
> >> >> >> > > on early
> >> >> >> > > media, but what if to introduce a variable
> >> >> >> > > "originate_wait_for_answer"
> >> >> >> > > with default value "false" and use the variable in originate
> >> >> >> > > code to
> >> >> >> > > decide when to return - on 18X or "200 OK"?
> >> >> >> > >
> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> >> >> >> > >> he wants to call 3 people at once and let the A leg hear
> >> >> >> > >> early media
> >> >> >> > >> from call #1 while call #2 and #3 still are progressing which
> >> >> >> > >> is not
> >> >> >> > >> simple to do without doing thousands of dollars in
> >> >> >> > >> development.
> >> >> >> > >>
> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> 
wrote:
> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
> >> >> >> > >>> customers
> >> >> >> > >>> can hear recordings?
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswit
> >> >> >> >ch- users http://www.freeswitch.org
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch
> >> >> >>-us ers http://www.freeswitch.org
> >> >> >
> >> >> > --
> >> >> > Anthony Minessale II
> >> >> >
> >> >> > FreeSWITCH http://www.freeswitch.org/
> >> >> > ClueCon http://www.cluecon.com/
> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >> >
> >> >> > AIM: anthm
> >> >> > MSN:anthony_minessale at hotmail.com
> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> >> > IRC: irc.freenode.net #freeswitch
> >> >> >
> >> >> > FreeSWITCH Developer Conference
> >> >> > sip:888 at conference.freeswitch.org
> >> >> > iax:guest at conference.freeswitch.org/888
> >> >> > googletalk:conf+888 at conference.freeswitch.org
> >> >> > pstn:+19193869900
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> >> >use rs http://www.freeswitch.org
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
> >> >>ers http://www.freeswitch.org
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> >> >rs http://www.freeswitch.org
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From david.villasmil.work at gmail.com  Fri Jan 15 04:38:36 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 13:38:36 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001150726.17430.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
	<201001150726.17430.sos@sokhapkin.dyndns.org>
Message-ID: <9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>

Default, haven't touched it i suppose it's off, i haven't set it anywhere

On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> Is bypass_media on or off?
>
> On Friday 15 January 2010, David Villasmil wrote:
>> Yeah, sorry. Early media. Audio after 200 is fine. Early media was ok
>> before the change.
>>
>> David
>>
>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
>>
>> <sos at sokhapkin.dyndns.org> wrote:
>> > Which audio? Early media or after 200 OK?
>> >
>> > On Friday 15 January 2010, David Villasmil wrote:
>> >> Hello again Anthony,
>> >>
>> >> I just tested it, and although functionality does not, first incoming
>> >> audio is coming in all garbled... do you know why?
>> >>
>> >> David
>> >>
>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>> >>
>> >> <anthony.minessale at gmail.com> wrote:
>> >> > {bridge_early_media=true}
>> >> > in the dial string in place of ignore_early_media=true
>> >> >
>> >> > the first b leg in the list who sends 183 will become the ringback
>> >> > device for A leg it will hear the early media
>> >> > for that leg while the other legs still ring.? If some other leg
>> >> > answers the final call will still be bridged to the leg who answered.
>> >> >
>> >> >
>> >> > I would estimate it at $500 payable on the big paypal button on
>> >> > http://www.freeswitch.org
>> >> > but, I already added the patch to tree earlier today so I guess it's
>> >> > up to you to pay it or not.
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
>> >> >
>> >> > <david.villasmil.work at gmail.com> wrote:
>> >> >> Anthony,
>> >> >>
>> >> >> I did take the "hint", don't worry. We will probably ask for a bounty
>> >> >> but first we need to know:
>> >> >> 1.- whether this is possible
>> >> >> 2.- how long it would take
>> >> >> 3.- how will it exactly work
>> >> >> 4.- of course, what's the bounty (be gentle ;) )
>> >> >>
>> >> >> We would of course give this back to the community.
>> >> >>
>> >> >> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>> >> >> and conversely "false" it will keep on trying to connect and if it
>> >> >> connects the other B-leg if will bridge to that one?
>> >> >>
>> >> >> Thanks
>> >> >>
>> >> >> David
>> >> >>
>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> >> >>
>> >> >> <anthony.minessale at gmail.com> wrote:
>> >> >> > I added bridge_early_media=true to do the best I can do.
>> >> >> > This is the most I will do, especially for free, nobody can take a
>> >> >> > hint that
>> >> >> > you should be paying for all these custom requests so take it or
>> >> >> > leave it
>> >> >> > but this thread is done.........
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>> >> >> > <sos at sokhapkin.dyndns.org>
>> >> >> >
>> >> >> > wrote:
>> >> >> >> No, not exactly. ignore_early_media doesn't pass early media to
>> >> >> >> the caller
>> >> >> >> if
>> >> >> >> bypass_media is false.
>> >> >> >>
>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
>> >> >> >> > this is exactly what ignore_early_media does now.
>> >> >> >> >
>> >> >> >> > Mike
>> >> >> >> >
>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> >> >> >> > > The issue here is when "originate" routine should return and
>> >> >> >> > > set "originate_status" variable. Current behavior is to return
>> >> >> >> > > on early
>> >> >> >> > > media, but what if to introduce a variable
>> >> >> >> > > "originate_wait_for_answer"
>> >> >> >> > > with default value "false" and use the variable in originate
>> >> >> >> > > code to
>> >> >> >> > > decide when to return - on 18X or "200 OK"?
>> >> >> >> > >
>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> >> >> >> > >> he wants to call 3 people at once and let the A leg hear
>> >> >> >> > >> early media
>> >> >> >> > >> from call #1 while call #2 and #3 still are progressing which
>> >> >> >> > >> is not
>> >> >> >> > >> simple to do without doing thousands of dollars in
>> >> >> >> > >> development.
>> >> >> >> > >>
>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com>
> wrote:
>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>> >> >> >> > >>> customers
>> >> >> >> > >>> can hear recordings?
>> >> >> >> >
>> >> >> >> > _______________________________________________
>> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >
>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswit
>> >> >> >> >ch- users http://www.freeswitch.org
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch
>> >> >> >>-us ers http://www.freeswitch.org
>> >> >> >
>> >> >> > --
>> >> >> > Anthony Minessale II
>> >> >> >
>> >> >> > FreeSWITCH http://www.freeswitch.org/
>> >> >> > ClueCon http://www.cluecon.com/
>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >> >
>> >> >> > AIM: anthm
>> >> >> > MSN:anthony_minessale at hotmail.com
>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> >> > IRC: irc.freenode.net #freeswitch
>> >> >> >
>> >> >> > FreeSWITCH Developer Conference
>> >> >> > sip:888 at conference.freeswitch.org
>> >> >> > iax:guest at conference.freeswitch.org/888
>> >> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> >> > pstn:+19193869900
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >> >> >use rs http://www.freeswitch.org
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> >> >>ers http://www.freeswitch.org
>> >> >
>> >> > --
>> >> > Anthony Minessale II
>> >> >
>> >> > FreeSWITCH http://www.freeswitch.org/
>> >> > ClueCon http://www.cluecon.com/
>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >
>> >> > AIM: anthm
>> >> > MSN:anthony_minessale at hotmail.com
>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> > IRC: irc.freenode.net #freeswitch
>> >> >
>> >> > FreeSWITCH Developer Conference
>> >> > sip:888 at conference.freeswitch.org
>> >> > iax:guest at conference.freeswitch.org/888
>> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> > pstn:+19193869900
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> >> >rs http://www.freeswitch.org
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From david.villasmil.work at gmail.com  Fri Jan 15 04:42:44 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 13:42:44 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001150321v1fb0a793s9836016aa8eb39e@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001141318.34958.sos@sokhapkin.dyndns.org>
	<A3B19842-9856-4236-917B-3C13AB9F3D33@jerris.com>
	<201001141342.02178.sos@sokhapkin.dyndns.org>
	<191c3a031001141304m3fca5dcfp7dc2864e142c32b3@mail.gmail.com>
	<9853f4ff1001141506q26ee8d28ja963eb2810720ea7@mail.gmail.com>
	<191c3a031001141554l795328e7wa011c60264eb9544@mail.gmail.com>
	<9853f4ff1001150321v1fb0a793s9836016aa8eb39e@mail.gmail.com>
Message-ID: <9853f4ff1001150442v1644e9e8jd512a64d902ae78c@mail.gmail.com>

If I set bridge_early_media to "false", early audio comes in OK, if I
set it to true it's garbled.

David

On Fri, Jan 15, 2010 at 12:21 PM, David Villasmil
<david.villasmil.work at gmail.com> wrote:
> Hello again Anthony,
>
> I just tested it, and although functionality does not, first incoming
> audio is coming in all garbled... do you know why?
>
> David
>
> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> {bridge_early_media=true}
>> in the dial string in place of ignore_early_media=true
>>
>> the first b leg in the list who sends 183 will become the ringback device
>> for A leg it will hear the early media
>> for that leg while the other legs still ring.? If some other leg answers the
>> final call will still be bridged to the leg who answered.
>>
>>
>> I would estimate it at $500 payable on the big paypal button on
>> http://www.freeswitch.org
>> but, I already added the patch to tree earlier today so I guess it's up to
>> you to pay it or not.
>>
>>
>>
>>
>> On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
>> <david.villasmil.work at gmail.com> wrote:
>>>
>>> Anthony,
>>>
>>> I did take the "hint", don't worry. We will probably ask for a bounty
>>> but first we need to know:
>>> 1.- whether this is possible
>>> 2.- how long it would take
>>> 3.- how will it exactly work
>>> 4.- of course, what's the bounty (be gentle ;) )
>>>
>>> We would of course give this back to the community.
>>>
>>> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>>> and conversely "false" it will keep on trying to connect and if it
>>> connects the other B-leg if will bridge to that one?
>>>
>>> Thanks
>>>
>>> David
>>>
>>> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>> > I added bridge_early_media=true to do the best I can do.
>>> > This is the most I will do, especially for free, nobody can take a hint
>>> > that
>>> > you should be paying for all these custom requests so take it or leave
>>> > it
>>> > but this thread is done.........
>>> >
>>> >
>>> >
>>> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>>> > <sos at sokhapkin.dyndns.org>
>>> > wrote:
>>> >>
>>> >> No, not exactly. ignore_early_media doesn't pass early media to the
>>> >> caller
>>> >> if
>>> >> bypass_media is false.
>>> >>
>>> >> On Thursday 14 January 2010, Michael Jerris wrote:
>>> >> > this is exactly what ignore_early_media does now.
>>> >> >
>>> >> > Mike
>>> >> >
>>> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>>> >> > > The issue here is when "originate" routine should return and
>>> >> > > set "originate_status" variable. Current behavior is to return on
>>> >> > > early
>>> >> > > media, but what if to introduce a variable
>>> >> > > "originate_wait_for_answer"
>>> >> > > with default value "false" and use the variable in originate code
>>> >> > > to
>>> >> > > decide when to return - on 18X or "200 OK"?
>>> >> > >
>>> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>>> >> > >> he wants to call 3 people at once and let the A leg hear early
>>> >> > >> media
>>> >> > >> from call #1 while call #2 and #3 still are progressing which is
>>> >> > >> not
>>> >> > >> simple to do without doing thousands of dollars in development.
>>> >> > >>
>>> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com> wrote:
>>> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>>> >> > >>> customers
>>> >> > >>> can hear recordings?
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >
>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com
>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org
>>> > iax:guest at conference.freeswitch.org/888
>>> > googletalk:conf+888 at conference.freeswitch.org
>>> > pstn:+19193869900
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>



From david.villasmil.work at gmail.com  Fri Jan 15 04:43:28 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 13:43:28 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
	<201001150726.17430.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>
Message-ID: <9853f4ff1001150443t5a18cc6bxa5032f391ab6f0ed@mail.gmail.com>

I set it to "off" just in case, same thing.

On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
<david.villasmil.work at gmail.com> wrote:
> Default, haven't touched it i suppose it's off, i haven't set it anywhere
>
> On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
> <sos at sokhapkin.dyndns.org> wrote:
>> Is bypass_media on or off?
>>
>> On Friday 15 January 2010, David Villasmil wrote:
>>> Yeah, sorry. Early media. Audio after 200 is fine. Early media was ok
>>> before the change.
>>>
>>> David
>>>
>>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
>>>
>>> <sos at sokhapkin.dyndns.org> wrote:
>>> > Which audio? Early media or after 200 OK?
>>> >
>>> > On Friday 15 January 2010, David Villasmil wrote:
>>> >> Hello again Anthony,
>>> >>
>>> >> I just tested it, and although functionality does not, first incoming
>>> >> audio is coming in all garbled... do you know why?
>>> >>
>>> >> David
>>> >>
>>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>>> >>
>>> >> <anthony.minessale at gmail.com> wrote:
>>> >> > {bridge_early_media=true}
>>> >> > in the dial string in place of ignore_early_media=true
>>> >> >
>>> >> > the first b leg in the list who sends 183 will become the ringback
>>> >> > device for A leg it will hear the early media
>>> >> > for that leg while the other legs still ring.? If some other leg
>>> >> > answers the final call will still be bridged to the leg who answered.
>>> >> >
>>> >> >
>>> >> > I would estimate it at $500 payable on the big paypal button on
>>> >> > http://www.freeswitch.org
>>> >> > but, I already added the patch to tree earlier today so I guess it's
>>> >> > up to you to pay it or not.
>>> >> >
>>> >> >
>>> >> >
>>> >> >
>>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
>>> >> >
>>> >> > <david.villasmil.work at gmail.com> wrote:
>>> >> >> Anthony,
>>> >> >>
>>> >> >> I did take the "hint", don't worry. We will probably ask for a bounty
>>> >> >> but first we need to know:
>>> >> >> 1.- whether this is possible
>>> >> >> 2.- how long it would take
>>> >> >> 3.- how will it exactly work
>>> >> >> 4.- of course, what's the bounty (be gentle ;) )
>>> >> >>
>>> >> >> We would of course give this back to the community.
>>> >> >>
>>> >> >> in the meantime, bridge_early_media=true will discard the 2nd B-leg
>>> >> >> and conversely "false" it will keep on trying to connect and if it
>>> >> >> connects the other B-leg if will bridge to that one?
>>> >> >>
>>> >> >> Thanks
>>> >> >>
>>> >> >> David
>>> >> >>
>>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>>> >> >>
>>> >> >> <anthony.minessale at gmail.com> wrote:
>>> >> >> > I added bridge_early_media=true to do the best I can do.
>>> >> >> > This is the most I will do, especially for free, nobody can take a
>>> >> >> > hint that
>>> >> >> > you should be paying for all these custom requests so take it or
>>> >> >> > leave it
>>> >> >> > but this thread is done.........
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>>> >> >> > <sos at sokhapkin.dyndns.org>
>>> >> >> >
>>> >> >> > wrote:
>>> >> >> >> No, not exactly. ignore_early_media doesn't pass early media to
>>> >> >> >> the caller
>>> >> >> >> if
>>> >> >> >> bypass_media is false.
>>> >> >> >>
>>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
>>> >> >> >> > this is exactly what ignore_early_media does now.
>>> >> >> >> >
>>> >> >> >> > Mike
>>> >> >> >> >
>>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>>> >> >> >> > > The issue here is when "originate" routine should return and
>>> >> >> >> > > set "originate_status" variable. Current behavior is to return
>>> >> >> >> > > on early
>>> >> >> >> > > media, but what if to introduce a variable
>>> >> >> >> > > "originate_wait_for_answer"
>>> >> >> >> > > with default value "false" and use the variable in originate
>>> >> >> >> > > code to
>>> >> >> >> > > decide when to return - on 18X or "200 OK"?
>>> >> >> >> > >
>>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>>> >> >> >> > >> he wants to call 3 people at once and let the A leg hear
>>> >> >> >> > >> early media
>>> >> >> >> > >> from call #1 while call #2 and #3 still are progressing which
>>> >> >> >> > >> is not
>>> >> >> >> > >> simple to do without doing thousands of dollars in
>>> >> >> >> > >> development.
>>> >> >> >> > >>
>>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <djbinter at yahoo.com>
>> wrote:
>>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that your
>>> >> >> >> > >>> customers
>>> >> >> >> > >>> can hear recordings?
>>> >> >> >> >
>>> >> >> >> > _______________________________________________
>>> >> >> >> > FreeSWITCH-users mailing list
>>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >> >
>>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswit
>>> >> >> >> >ch- users http://www.freeswitch.org
>>> >> >> >>
>>> >> >> >> _______________________________________________
>>> >> >> >> FreeSWITCH-users mailing list
>>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >>
>>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch
>>> >> >> >>-us ers http://www.freeswitch.org
>>> >> >> >
>>> >> >> > --
>>> >> >> > Anthony Minessale II
>>> >> >> >
>>> >> >> > FreeSWITCH http://www.freeswitch.org/
>>> >> >> > ClueCon http://www.cluecon.com/
>>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >> >> >
>>> >> >> > AIM: anthm
>>> >> >> > MSN:anthony_minessale at hotmail.com
>>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> >> >> > IRC: irc.freenode.net #freeswitch
>>> >> >> >
>>> >> >> > FreeSWITCH Developer Conference
>>> >> >> > sip:888 at conference.freeswitch.org
>>> >> >> > iax:guest at conference.freeswitch.org/888
>>> >> >> > googletalk:conf+888 at conference.freeswitch.org
>>> >> >> > pstn:+19193869900
>>> >> >> >
>>> >> >> > _______________________________________________
>>> >> >> > FreeSWITCH-users mailing list
>>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> >> >> >use rs http://www.freeswitch.org
>>> >> >>
>>> >> >> _______________________________________________
>>> >> >> FreeSWITCH-users mailing list
>>> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>>> >> >>ers http://www.freeswitch.org
>>> >> >
>>> >> > --
>>> >> > Anthony Minessale II
>>> >> >
>>> >> > FreeSWITCH http://www.freeswitch.org/
>>> >> > ClueCon http://www.cluecon.com/
>>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >> >
>>> >> > AIM: anthm
>>> >> > MSN:anthony_minessale at hotmail.com
>>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> >> > IRC: irc.freenode.net #freeswitch
>>> >> >
>>> >> > FreeSWITCH Developer Conference
>>> >> > sip:888 at conference.freeswitch.org
>>> >> > iax:guest at conference.freeswitch.org/888
>>> >> > googletalk:conf+888 at conference.freeswitch.org
>>> >> > pstn:+19193869900
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>> >> >rs http://www.freeswitch.org
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>



From kond at nstel.ru  Fri Jan 15 06:33:39 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Fri, 15 Jan 2010 17:33:39 +0300
Subject: [Freeswitch-users] eavesdrop problem?
Message-ID: <20100115143339.ED51B11A9D@mail.nstel.ru>

Hi all,

 

I want to use eavesdrop application.

Playing with it I found that when one tries to eavesdrop caller the feature
works ok.

But when trying to eavesdrop callee eavesdrop attempt failes.

I just updated to the latest version from http://latest.freeswitch.org
<http://latest.freeswitch.org/>  

[freeswitch at freeswitch log]$ fs_cli -x version

FreeSWITCH Version 1.0.5-20100115-0400 (16318M)

 

My setup is as following:

I have 3 internal extensions: 2853, 2840, 2879 (all are xlite).

1. 2879 calls 2840. Then 2853 dials 882879 - eavesdrop worked as expected.

2. 2840 calls 2879. Then 2853 dials 882879 - eavesdrop failed. 

 

I attached logs for both cases.

 

I don't believe it's intended behavior. 

 

Can anybody please advise if it is a configuration or a software problem?

 

Thanks and regards,

Nikolay.

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From brian at freeswitch.org  Fri Jan 15 06:47:18 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 15 Jan 2010 08:47:18 -0600
Subject: [Freeswitch-users] eavesdrop problem?
In-Reply-To: <20100115143339.ED51B11A9D@mail.nstel.ru>
References: <20100115143339.ED51B11A9D@mail.nstel.ru>
Message-ID: <452BA845-A9A9-4465-ACF3-EF34AFBA159D@freeswitch.org>

Bugs do not belong on the mailing list.  http://jira.freeswitch.org, also do not attach zip files gz files or anything that will require us to download unpack and view them locally.  Doing this will delay attention to your issue.

/b

On Jan 15, 2010, at 8:33 AM, Nikolay Kondratyev wrote:

> Hi all,
>  
> I want to use eavesdrop application.
> Playing with it I found that when one tries to eavesdrop caller the feature works ok.
> But when trying to eavesdrop callee eavesdrop attempt failes.
> I just updated to the latest version from http://latest.freeswitch.org
> [freeswitch at freeswitch log]$ fs_cli -x version
> FreeSWITCH Version 1.0.5-20100115-0400 (16318M)
>  
> My setup is as following:
> I have 3 internal extensions: 2853, 2840, 2879 (all are xlite).
> 1. 2879 calls 2840. Then 2853 dials 882879 ? eavesdrop worked as expected.
> 2. 2840 calls 2879. Then 2853 dials 882879 ? eavesdrop failed.
>  
> I attached logs for both cases.
>  
> I don?t believe it?s intended behavior.
>  
> Can anybody please advise if it is a configuration or a software problem?
>  
> Thanks and regards,
> Nikolay.
> <successful_eavesdrop.log.gz><failed_eavesdrop.log.gz>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kond at nstel.ru  Fri Jan 15 06:57:43 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Fri, 15 Jan 2010 17:57:43 +0300
Subject: [Freeswitch-users] sip trunk question: why call
	throughexternalprofile is challenged?
In-Reply-To: <1BD8DFE1-E3DD-46C3-A383-C9627939BB65@jerris.com>
Message-ID: <20100115145743.6291912003@mail.nstel.ru>

Mike, Anthony, thanks for the advice.

I set accept-blind-auth in my external profile and FS does not challenge the
invite any more.

Did I understand right that without accept-blind-auth FS challenged incoming
Invite because of the presence of Proxy-Authorization header in the Invite?

Do I understand right that if I place accept-blind-auth inside
<gateway></gateway> it will work for that gateway only?

Thank and regards,

Nikolay.

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Friday, January 15, 2010 1:22 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] sip trunk question: why call
throughexternalprofile is challenged?

 

This now rings a bell, also, if you create a gateway on that profile who's
gateway name matches the realm on the incoming auth header, I think that
also was working.

 

Mike

 

On Jan 14, 2010, at 5:01 PM, Anthony Minessale wrote:





try setting param accept-blind-auth to true in your sofia profile config
internal.xml
iirc it was made just for sipX who feels the need to send auth headers even
when nobody asked for them.
so even when auth-calls is false we will still try to parse the auth if one
is sent.



On Thu, Jan 14, 2010 at 11:21 AM, Michael Jerris <mike at jerris.com> wrote:



if you look in the sample configs for the words blind and auth you will find
all these settings, also you can setup acls for ip auth to not challenge.

 

Mike

 

On Jan 14, 2010, at 3:26 AM, Nikolay Kondratyev wrote:

 

Mike, thanks for the reply.

 

Mmm. looks like I need more detailed instructions where to dig.

Is there a way to turn off "challenging" completely?

I thought that <param name="auth-calls" value="false"/> should do it, but
alas.

By the way should this parameter be visible in either "sofia status profile
external" or "sofia status gateway sipx4.lab.nstel.ru
<http://sipx4.lab.nstel.ru/> " ? I don't see it.

 

I attached traces of failed and successful calls.

 

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From null at invalid.name  Fri Jan 15 06:58:36 2010
From: null at invalid.name (Dan Lane)
Date: Fri, 15 Jan 2010 14:58:36 +0000
Subject: [Freeswitch-users] Nibble_total_billed variable missing when using
	nibblebill on b-leg
Message-ID: <b042b23b1001150658j26891262qa396e761c54ba2c8@mail.gmail.com>

 I'm doing something similar to the example below in order to bill on
the b-leg. Billing is working but the variable nibble_total_billed
isn't being set once the call is finished.

<action application="bridge"
data="${sofia_contact(internal/user@$${domain})},[enable_heartbeat_events=60,nibble_account=1,nibble_rate=0.01]sofia/gateway/blah/1234"/>

I see that a few others have experienced this issue (including Jira
MODAPP-385) so has anyone found a work-around to coerce this into
working?

If not, what will it take to get the issue resolved?



From kond at nstel.ru  Fri Jan 15 07:18:32 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Fri, 15 Jan 2010 18:18:32 +0300
Subject: [Freeswitch-users] eavesdrop problem?
In-Reply-To: <452BA845-A9A9-4465-ACF3-EF34AFBA159D@freeswitch.org>
Message-ID: <20100115151832.E848811F55@mail.nstel.ru>

Brian,

 

Should I open an issue in the jira (since it's a bug)?

 

In general: I'm new to FS. When I'm not sure if my problem is a bug or
misconfiguration, I think, I should better first discuss the problem in the
list before opening an issue.

Because if everybody will write everything into jira, jira will turn into
mail list, while it is not intended for that purpose.

 

Regarding attaching gzipped files: what is the size limit for attachments in
the list?

 

Thanks and regards,

Nikolay. 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Friday, January 15, 2010 5:47 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] eavesdrop problem?

 

Bugs do not belong on the mailing list.  http://jira.freeswitch.org, also do
not attach zip files gz files or anything that will require us to download
unpack and view them locally.  Doing this will delay attention to your
issue.

 

/b

 

On Jan 15, 2010, at 8:33 AM, Nikolay Kondratyev wrote:





Hi all,

 

I want to use eavesdrop application.

Playing with it I found that when one tries to eavesdrop caller the feature
works ok.

But when trying to eavesdrop callee eavesdrop attempt failes.

I just updated to the latest version from http://latest.freeswitch.org
<http://latest.freeswitch.org/> 

[freeswitch at freeswitch log]$ fs_cli -x version

FreeSWITCH Version 1.0.5-20100115-0400 (16318M)

 

My setup is as following:

I have 3 internal extensions: 2853, 2840, 2879 (all are xlite).

1. 2879 calls 2840. Then 2853 dials 882879 - eavesdrop worked as expected.

2. 2840 calls 2879. Then 2853 dials 882879 - eavesdrop failed.

 

I attached logs for both cases.

 

I don't believe it's intended behavior.

 

Can anybody please advise if it is a configuration or a software problem?

 

Thanks and regards,

Nikolay.

<successful_eavesdrop.log.gz><failed_eavesdrop.log.gz>______________________
_________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From lawwton at gmail.com  Fri Jan 15 07:00:17 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 10:00:17 -0500
Subject: [Freeswitch-users] Conference Questions
Message-ID: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>

Hello:

I've been using asterisk for a little bit over three years now. A
couple of months ago I found out about freeswitch, took a look at it,
thought it was interesting and moved on. A few weeks ago, I started
looking at a project I've been wanting to work on for quite a while
using conferences and started exploring systems and different
approaches. Based on the requirements I have, I decided to use
freeswitch. It seemed like it had the best support for conferencing so
I went for it. According to some documentation I found it also seems
to allow for more concurrent calls than asterisk which is an added
bonus.

I got a server ready, installed FC8 on it which is what I have in
production now, unpacked freeswitch there and so far it's running
beautifully. Very painless process really to get it installed, I was
happy to see that. Configuration seems a bit different since it's XML;
but being a developer myself I can see many advantages to having done
that in the future as the system scales and grows in complexity.

Sorry for the long introduction, getting to my question now. So ...
What I want to be able to do is the following:

Create and control conferences via the HTTP API. I've been reading a
bit for the past two days the documentation and I am becoming more
familiar now with how things are done using ESL, the support for PHP,
perl and I believe others.

a) It seemed to me like the way to setup the moderator of the
conference is by setting a parameter in the DialPlan and specifying
based on a condition who the moderator is, say for instance the
destination number. That's fine and it makes sense, however, say that
I am creating a new conference and I want to have 3 participants where
one of them is the moderator. What would I have to do to specify that
person A dialing for example number xxx-xxx-xxxx is the moderator (via
HTTP)? Would I have to create my own call to the system and add say an
entry to DialPlan with the right parameter for the moderator, then
create the conference?

b) When a conference is created, or when I go to create a new
conference via HTTP using the API, does it allow for example for all
numbers that will be added to be dialed at once? Or should the process
be dial each participant, sending say 3 http requests via the API? The
API command "conference dial" seems to only take one argument for
destination number; but I am asking just in case I missed something.

Thanks in advance for the help and I apologize for the long email.

Alfredo



From mike at jerris.com  Fri Jan 15 07:44:15 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 Jan 2010 10:44:15 -0500
Subject: [Freeswitch-users] sip trunk question: why call
	throughexternalprofile is challenged?
In-Reply-To: <20100115145743.6291912003@mail.nstel.ru>
References: <20100115145743.6291912003@mail.nstel.ru>
Message-ID: <24747A07-BB64-4331-955D-09051591804D@jerris.com>

I am pretty sure that does not work, as we have not matched the gateway yet.

Mike

On Jan 15, 2010, at 9:57 AM, Nikolay Kondratyev wrote:

> Mike, Anthony, thanks for the advice.
> I set accept-blind-auth in my external profile and FS does not challenge the invite any more.
> Did I understand right that without accept-blind-auth FS challenged incoming Invite because of the presence of Proxy-Authorization header in the Invite?
> Do I understand right that if I place accept-blind-auth inside <gateway></gateway> it will work for that gateway only?
> Thank and regards,
> Nikolay.
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Friday, January 15, 2010 1:22 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] sip trunk question: why call throughexternalprofile is challenged?
>  
> This now rings a bell, also, if you create a gateway on that profile who's gateway name matches the realm on the incoming auth header, I think that also was working.
>  
> Mike
>  
> On Jan 14, 2010, at 5:01 PM, Anthony Minessale wrote:
> 
> 
> try setting param accept-blind-auth to true in your sofia profile config internal.xml
> iirc it was made just for sipX who feels the need to send auth headers even when nobody asked for them.
> so even when auth-calls is false we will still try to parse the auth if one is sent.
> 
> 
> On Thu, Jan 14, 2010 at 11:21 AM, Michael Jerris <mike at jerris.com> wrote:
> 
> if you look in the sample configs for the words blind and auth you will find all these settings, also you can setup acls for ip auth to not challenge.
>  
> Mike
>  
> On Jan 14, 2010, at 3:26 AM, Nikolay Kondratyev wrote:
>  
>> Mike, thanks for the reply.
>>  
>> Mmm? looks like I need more detailed instructions where to dig?
>> Is there a way to turn off ?challenging? completely?
>> I thought that <param name="auth-calls" value="false"/> should do it, but alas?
>> By the way should this parameter be visible in either ?sofia status profile external? or ?sofia status gateway sipx4.lab.nstel.ru? ? I don?t see it?
>>  
>> I attached traces of failed and successful calls.
> 
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Fri Jan 15 08:05:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 10:05:03 -0600
Subject: [Freeswitch-users] eavesdrop problem?
In-Reply-To: <20100115143339.ED51B11A9D@mail.nstel.ru>
References: <20100115143339.ED51B11A9D@mail.nstel.ru>
Message-ID: <191c3a031001150805y2c6f0bd0pcea8a18d9d866b4b@mail.gmail.com>

don't bother,

only inbound legs are added to the db that is used to lookup for eavesdrop
because the action is in the dialplan.
The extensions to eavesdrop you are using are just a demo to show you how to
work it.
you need to know the uuid of the channel you are trying to eavesdrop on
before you can do what you want.




On Fri, Jan 15, 2010 at 8:33 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

>  Hi all,
>
>
>
> I want to use eavesdrop application.
>
> Playing with it I found that when one tries to eavesdrop caller the feature
> works ok.
>
> But when trying to eavesdrop callee eavesdrop attempt failes.
>
> I just updated to the latest version from http://latest.freeswitch.org
>
> [freeswitch at freeswitch log]$ fs_cli -x version
>
> FreeSWITCH Version 1.0.5-20100115-0400 (16318M)
>
>
>
> My setup is as following:
>
> I have 3 internal extensions: 2853, 2840, 2879 (all are xlite).
>
> 1. 2879 calls 2840. Then 2853 dials 882879 ? eavesdrop worked as expected.
>
> 2. 2840 calls 2879. Then 2853 dials 882879 ? eavesdrop failed.
>
>
>
> I attached logs for both cases.
>
>
>
> I don?t believe it?s intended behavior.
>
>
>
> Can anybody please advise if it is a configuration or a software problem?
>
>
>
> Thanks and regards,
>
> Nikolay.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Jan 15 08:08:53 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 10:08:53 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001150443t5a18cc6bxa5032f391ab6f0ed@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
	<201001150726.17430.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>
	<9853f4ff1001150443t5a18cc6bxa5032f391ab6f0ed@mail.gmail.com>
Message-ID: <191c3a031001150808i69b3937boe0e903057f984e96@mail.gmail.com>

one of the many reasons its a bad idea.
Probably the leg with the bad audio is a different ptime.
Now the amount of work I have to do escalates I would prefer you commit to
commercial support by emailing me at consulting at freeswitch.org to continue
with this.


On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> I set it to "off" just in case, same thing.
>
> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
> > Default, haven't touched it i suppose it's off, i haven't set it anywhere
> >
> > On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
> > <sos at sokhapkin.dyndns.org> wrote:
> >> Is bypass_media on or off?
> >>
> >> On Friday 15 January 2010, David Villasmil wrote:
> >>> Yeah, sorry. Early media. Audio after 200 is fine. Early media was ok
> >>> before the change.
> >>>
> >>> David
> >>>
> >>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
> >>>
> >>> <sos at sokhapkin.dyndns.org> wrote:
> >>> > Which audio? Early media or after 200 OK?
> >>> >
> >>> > On Friday 15 January 2010, David Villasmil wrote:
> >>> >> Hello again Anthony,
> >>> >>
> >>> >> I just tested it, and although functionality does not, first
> incoming
> >>> >> audio is coming in all garbled... do you know why?
> >>> >>
> >>> >> David
> >>> >>
> >>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
> >>> >>
> >>> >> <anthony.minessale at gmail.com> wrote:
> >>> >> > {bridge_early_media=true}
> >>> >> > in the dial string in place of ignore_early_media=true
> >>> >> >
> >>> >> > the first b leg in the list who sends 183 will become the ringback
> >>> >> > device for A leg it will hear the early media
> >>> >> > for that leg while the other legs still ring.  If some other leg
> >>> >> > answers the final call will still be bridged to the leg who
> answered.
> >>> >> >
> >>> >> >
> >>> >> > I would estimate it at $500 payable on the big paypal button on
> >>> >> > http://www.freeswitch.org
> >>> >> > but, I already added the patch to tree earlier today so I guess
> it's
> >>> >> > up to you to pay it or not.
> >>> >> >
> >>> >> >
> >>> >> >
> >>> >> >
> >>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> >>> >> >
> >>> >> > <david.villasmil.work at gmail.com> wrote:
> >>> >> >> Anthony,
> >>> >> >>
> >>> >> >> I did take the "hint", don't worry. We will probably ask for a
> bounty
> >>> >> >> but first we need to know:
> >>> >> >> 1.- whether this is possible
> >>> >> >> 2.- how long it would take
> >>> >> >> 3.- how will it exactly work
> >>> >> >> 4.- of course, what's the bounty (be gentle ;) )
> >>> >> >>
> >>> >> >> We would of course give this back to the community.
> >>> >> >>
> >>> >> >> in the meantime, bridge_early_media=true will discard the 2nd
> B-leg
> >>> >> >> and conversely "false" it will keep on trying to connect and if
> it
> >>> >> >> connects the other B-leg if will bridge to that one?
> >>> >> >>
> >>> >> >> Thanks
> >>> >> >>
> >>> >> >> David
> >>> >> >>
> >>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> >>> >> >>
> >>> >> >> <anthony.minessale at gmail.com> wrote:
> >>> >> >> > I added bridge_early_media=true to do the best I can do.
> >>> >> >> > This is the most I will do, especially for free, nobody can
> take a
> >>> >> >> > hint that
> >>> >> >> > you should be paying for all these custom requests so take it
> or
> >>> >> >> > leave it
> >>> >> >> > but this thread is done.........
> >>> >> >> >
> >>> >> >> >
> >>> >> >> >
> >>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
> >>> >> >> > <sos at sokhapkin.dyndns.org>
> >>> >> >> >
> >>> >> >> > wrote:
> >>> >> >> >> No, not exactly. ignore_early_media doesn't pass early media
> to
> >>> >> >> >> the caller
> >>> >> >> >> if
> >>> >> >> >> bypass_media is false.
> >>> >> >> >>
> >>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
> >>> >> >> >> > this is exactly what ignore_early_media does now.
> >>> >> >> >> >
> >>> >> >> >> > Mike
> >>> >> >> >> >
> >>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> >>> >> >> >> > > The issue here is when "originate" routine should return
> and
> >>> >> >> >> > > set "originate_status" variable. Current behavior is to
> return
> >>> >> >> >> > > on early
> >>> >> >> >> > > media, but what if to introduce a variable
> >>> >> >> >> > > "originate_wait_for_answer"
> >>> >> >> >> > > with default value "false" and use the variable in
> originate
> >>> >> >> >> > > code to
> >>> >> >> >> > > decide when to return - on 18X or "200 OK"?
> >>> >> >> >> > >
> >>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> >>> >> >> >> > >> he wants to call 3 people at once and let the A leg hear
> >>> >> >> >> > >> early media
> >>> >> >> >> > >> from call #1 while call #2 and #3 still are progressing
> which
> >>> >> >> >> > >> is not
> >>> >> >> >> > >> simple to do without doing thousands of dollars in
> >>> >> >> >> > >> development.
> >>> >> >> >> > >>
> >>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <
> djbinter at yahoo.com>
> >> wrote:
> >>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that
> your
> >>> >> >> >> > >>> customers
> >>> >> >> >> > >>> can hear recordings?
> >>> >> >> >> >
> >>> >> >> >> > _______________________________________________
> >>> >> >> >> > FreeSWITCH-users mailing list
> >>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >> >> >
> >>> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswit
> >>> >> >> >> >ch- users http://www.freeswitch.org
> >>> >> >> >>
> >>> >> >> >> _______________________________________________
> >>> >> >> >> FreeSWITCH-users mailing list
> >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >> >>
> >>> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch
> >>> >> >> >>-us ers http://www.freeswitch.org
> >>> >> >> >
> >>> >> >> > --
> >>> >> >> > Anthony Minessale II
> >>> >> >> >
> >>> >> >> > FreeSWITCH http://www.freeswitch.org/
> >>> >> >> > ClueCon http://www.cluecon.com/
> >>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>> >> >> >
> >>> >> >> > AIM: anthm
> >>> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> >> >> > IRC: irc.freenode.net #freeswitch
> >>> >> >> >
> >>> >> >> > FreeSWITCH Developer Conference
> >>> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> >> >> > iax:guest at conference.freeswitch.org/888
> >>> >> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> >> >> > pstn:+19193869900
> >>> >> >> >
> >>> >> >> > _______________________________________________
> >>> >> >> > FreeSWITCH-users mailing list
> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-
> >>> >> >> >use rs http://www.freeswitch.org
> >>> >> >>
> >>> >> >> _______________________________________________
> >>> >> >> FreeSWITCH-users mailing list
> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-us
> >>> >> >>ers http://www.freeswitch.org
> >>> >> >
> >>> >> > --
> >>> >> > Anthony Minessale II
> >>> >> >
> >>> >> > FreeSWITCH http://www.freeswitch.org/
> >>> >> > ClueCon http://www.cluecon.com/
> >>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>> >> >
> >>> >> > AIM: anthm
> >>> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> >> > IRC: irc.freenode.net #freeswitch
> >>> >> >
> >>> >> > FreeSWITCH Developer Conference
> >>> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> >> > iax:guest at conference.freeswitch.org/888
> >>> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> >> > pstn:+19193869900
> >>> >> >
> >>> >> > _______________________________________________
> >>> >> > FreeSWITCH-users mailing list
> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-use
> >>> >> >rs http://www.freeswitch.org
> >>> >>
> >>> >> _______________________________________________
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Jan 15 08:13:28 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 10:13:28 -0600
Subject: [Freeswitch-users] Conference Questions
In-Reply-To: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>
References: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>
Message-ID: <191c3a031001150813g69a86fb9w3a1601d06df5517e@mail.gmail.com>

look at the "mad boss" extension in the default dialplan
conf/dialplan/default.xml to see how to craft an all-hands conference.
otherwise individual calls to originate to send people to the conference is
also ok.


On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil <lawwton at gmail.com
> wrote:

> Hello:
>
> I've been using asterisk for a little bit over three years now. A
> couple of months ago I found out about freeswitch, took a look at it,
> thought it was interesting and moved on. A few weeks ago, I started
> looking at a project I've been wanting to work on for quite a while
> using conferences and started exploring systems and different
> approaches. Based on the requirements I have, I decided to use
> freeswitch. It seemed like it had the best support for conferencing so
> I went for it. According to some documentation I found it also seems
> to allow for more concurrent calls than asterisk which is an added
> bonus.
>
> I got a server ready, installed FC8 on it which is what I have in
> production now, unpacked freeswitch there and so far it's running
> beautifully. Very painless process really to get it installed, I was
> happy to see that. Configuration seems a bit different since it's XML;
> but being a developer myself I can see many advantages to having done
> that in the future as the system scales and grows in complexity.
>
> Sorry for the long introduction, getting to my question now. So ...
> What I want to be able to do is the following:
>
> Create and control conferences via the HTTP API. I've been reading a
> bit for the past two days the documentation and I am becoming more
> familiar now with how things are done using ESL, the support for PHP,
> perl and I believe others.
>
> a) It seemed to me like the way to setup the moderator of the
> conference is by setting a parameter in the DialPlan and specifying
> based on a condition who the moderator is, say for instance the
> destination number. That's fine and it makes sense, however, say that
> I am creating a new conference and I want to have 3 participants where
> one of them is the moderator. What would I have to do to specify that
> person A dialing for example number xxx-xxx-xxxx is the moderator (via
> HTTP)? Would I have to create my own call to the system and add say an
> entry to DialPlan with the right parameter for the moderator, then
> create the conference?
>
> b) When a conference is created, or when I go to create a new
> conference via HTTP using the API, does it allow for example for all
> numbers that will be added to be dialed at once? Or should the process
> be dial each participant, sending say 3 http requests via the API? The
> API command "conference dial" seems to only take one argument for
> destination number; but I am asking just in case I missed something.
>
> Thanks in advance for the help and I apologize for the long email.
>
> Alfredo
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From null at invalid.name  Fri Jan 15 08:13:30 2010
From: null at invalid.name (Dan Lane)
Date: Fri, 15 Jan 2010 16:13:30 +0000
Subject: [Freeswitch-users] Nibble_total_billed variable missing when
	using nibblebill on b-leg
In-Reply-To: <b042b23b1001150658j26891262qa396e761c54ba2c8@mail.gmail.com>
References: <b042b23b1001150658j26891262qa396e761c54ba2c8@mail.gmail.com>
Message-ID: <b042b23b1001150813r4950e607w111afb727746ce5c@mail.gmail.com>

On Fri, Jan 15, 2010 at 2:58 PM, Dan Lane <null at invalid.name> wrote:
> ?I'm doing something similar to the example below in order to bill on
> the b-leg. Billing is working but the variable nibble_total_billed
> isn't being set once the call is finished.
>
> <action application="bridge"
> data="${sofia_contact(internal/user@$${domain})},[enable_heartbeat_events=60,nibble_account=1,nibble_rate=0.01]sofia/gateway/blah/1234"/>
>
> I see that a few others have experienced this issue (including Jira
> MODAPP-385) so has anyone found a work-around to coerce this into
> working?
>
> If not, what will it take to get the issue resolved?
>

Of course, if I set log-b-leg=true in mod_xml_cdr then I can see the
variable because it's only going to be set on the b-leg! *slaps
forehead*

Carry on, nothing to see here.



From lawwton at gmail.com  Fri Jan 15 08:42:15 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 11:42:15 -0500
Subject: [Freeswitch-users] Conference Questions
In-Reply-To: <191c3a031001150813g69a86fb9w3a1601d06df5517e@mail.gmail.com>
References: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>
	<191c3a031001150813g69a86fb9w3a1601d06df5517e@mail.gmail.com>
Message-ID: <5fe6fa8f1001150842v71c2eadeie7ff38231c88789d@mail.gmail.com>

Appreciate the fast response Anthony.

Response or ideas on how to implement a) ?

a) It seemed to me like the way to setup the moderator of the
conference is by setting a parameter in the DialPlan and specifying
based on a condition who the moderator is, say for instance the
destination number. That's fine and it makes sense, however, say that
I am creating a new conference and I want to have 3 participants where
one of them is the moderator. What would I have to do to specify that
person A dialing for example number xxx-xxx-xxxx is the moderator (via
HTTP)? Would I have to create my own call to the system and add say an
entry to DialPlan with the right parameter for the moderator, then
create the conference?

Thanks in advance,

Alfredo Q-V

On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> look at the "mad boss" extension in the default dialplan
> conf/dialplan/default.xml to see how to craft an all-hands conference.
> otherwise individual calls to originate to send people to the conference is
> also ok.
>
>
> On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> Hello:
>>
>> I've been using asterisk for a little bit over three years now. A
>> couple of months ago I found out about freeswitch, took a look at it,
>> thought it was interesting and moved on. A few weeks ago, I started
>> looking at a project I've been wanting to work on for quite a while
>> using conferences and started exploring systems and different
>> approaches. Based on the requirements I have, I decided to use
>> freeswitch. It seemed like it had the best support for conferencing so
>> I went for it. According to some documentation I found it also seems
>> to allow for more concurrent calls than asterisk which is an added
>> bonus.
>>
>> I got a server ready, installed FC8 on it which is what I have in
>> production now, unpacked freeswitch there and so far it's running
>> beautifully. Very painless process really to get it installed, I was
>> happy to see that. Configuration seems a bit different since it's XML;
>> but being a developer myself I can see many advantages to having done
>> that in the future as the system scales and grows in complexity.
>>
>> Sorry for the long introduction, getting to my question now. So ...
>> What I want to be able to do is the following:
>>
>> Create and control conferences via the HTTP API. I've been reading a
>> bit for the past two days the documentation and I am becoming more
>> familiar now with how things are done using ESL, the support for PHP,
>> perl and I believe others.
>>
>> a) It seemed to me like the way to setup the moderator of the
>> conference is by setting a parameter in the DialPlan and specifying
>> based on a condition who the moderator is, say for instance the
>> destination number. That's fine and it makes sense, however, say that
>> I am creating a new conference and I want to have 3 participants where
>> one of them is the moderator. What would I have to do to specify that
>> person A dialing for example number xxx-xxx-xxxx is the moderator (via
>> HTTP)? Would I have to create my own call to the system and add say an
>> entry to DialPlan with the right parameter for the moderator, then
>> create the conference?
>>
>> b) When a conference is created, or when I go to create a new
>> conference via HTTP using the API, does it allow for example for all
>> numbers that will be added to be dialed at once? Or should the process
>> be dial each participant, sending say 3 http requests via the API? The
>> API command "conference dial" seems to only take one argument for
>> destination number; but I am asking just in case I missed something.
>>
>> Thanks in advance for the help and I apologize for the long email.
>>
>> Alfredo
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Fri Jan 15 08:57:05 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Jan 2010 08:57:05 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly!
Message-ID: <87f2f3b91001150857r5d1ad490ga8d29c3f8bf856b9@mail.gmail.com>

The agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_05

Come join us and let's talk about FreeSWITCH, VoIP, and all things
telephonic!

-Michael
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From jbr at consiglia.dk  Fri Jan 15 09:02:29 2010
From: jbr at consiglia.dk (Jon Bruel)
Date: Fri, 15 Jan 2010 18:02:29 +0100
Subject: [Freeswitch-users] RTCP information
Message-ID: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE09AC568494@SBS2008SERVER.consiglia.local>

In a real setup with 5-20 VoIP calls a day, every now and then there are some problems with sound quality, and I need some tools to investigate the cause.
The phones support RTCP, and I would like to hear if I can get the FS to relay those packets to some kind of logger, including the signalling information?
/Jon

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From anthony.minessale at gmail.com  Fri Jan 15 09:02:09 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 11:02:09 -0600
Subject: [Freeswitch-users] Conference Questions
In-Reply-To: <5fe6fa8f1001150842v71c2eadeie7ff38231c88789d@mail.gmail.com>
References: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>
	<191c3a031001150813g69a86fb9w3a1601d06df5517e@mail.gmail.com>
	<5fe6fa8f1001150842v71c2eadeie7ff38231c88789d@mail.gmail.com>
Message-ID: <191c3a031001150902l26298d7aq80d284a9e04f59fc@mail.gmail.com>

the flags are set as part of the dial string so you can easily choose that,
int the example I told you to look at notice the +flags{} bit at the end of
some of the dial strings.


On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil <
lawwton at gmail.com> wrote:

> Appreciate the fast response Anthony.
>
> Response or ideas on how to implement a) ?
>
> a) It seemed to me like the way to setup the moderator of the
> conference is by setting a parameter in the DialPlan and specifying
> based on a condition who the moderator is, say for instance the
> destination number. That's fine and it makes sense, however, say that
> I am creating a new conference and I want to have 3 participants where
> one of them is the moderator. What would I have to do to specify that
> person A dialing for example number xxx-xxx-xxxx is the moderator (via
> HTTP)? Would I have to create my own call to the system and add say an
> entry to DialPlan with the right parameter for the moderator, then
> create the conference?
>
> Thanks in advance,
>
> Alfredo Q-V
>
> On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > look at the "mad boss" extension in the default dialplan
> > conf/dialplan/default.xml to see how to craft an all-hands conference.
> > otherwise individual calls to originate to send people to the conference
> is
> > also ok.
> >
> >
> > On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil
> > <lawwton at gmail.com> wrote:
> >>
> >> Hello:
> >>
> >> I've been using asterisk for a little bit over three years now. A
> >> couple of months ago I found out about freeswitch, took a look at it,
> >> thought it was interesting and moved on. A few weeks ago, I started
> >> looking at a project I've been wanting to work on for quite a while
> >> using conferences and started exploring systems and different
> >> approaches. Based on the requirements I have, I decided to use
> >> freeswitch. It seemed like it had the best support for conferencing so
> >> I went for it. According to some documentation I found it also seems
> >> to allow for more concurrent calls than asterisk which is an added
> >> bonus.
> >>
> >> I got a server ready, installed FC8 on it which is what I have in
> >> production now, unpacked freeswitch there and so far it's running
> >> beautifully. Very painless process really to get it installed, I was
> >> happy to see that. Configuration seems a bit different since it's XML;
> >> but being a developer myself I can see many advantages to having done
> >> that in the future as the system scales and grows in complexity.
> >>
> >> Sorry for the long introduction, getting to my question now. So ...
> >> What I want to be able to do is the following:
> >>
> >> Create and control conferences via the HTTP API. I've been reading a
> >> bit for the past two days the documentation and I am becoming more
> >> familiar now with how things are done using ESL, the support for PHP,
> >> perl and I believe others.
> >>
> >> a) It seemed to me like the way to setup the moderator of the
> >> conference is by setting a parameter in the DialPlan and specifying
> >> based on a condition who the moderator is, say for instance the
> >> destination number. That's fine and it makes sense, however, say that
> >> I am creating a new conference and I want to have 3 participants where
> >> one of them is the moderator. What would I have to do to specify that
> >> person A dialing for example number xxx-xxx-xxxx is the moderator (via
> >> HTTP)? Would I have to create my own call to the system and add say an
> >> entry to DialPlan with the right parameter for the moderator, then
> >> create the conference?
> >>
> >> b) When a conference is created, or when I go to create a new
> >> conference via HTTP using the API, does it allow for example for all
> >> numbers that will be added to be dialed at once? Or should the process
> >> be dial each participant, sending say 3 http requests via the API? The
> >> API command "conference dial" seems to only take one argument for
> >> destination number; but I am asking just in case I missed something.
> >>
> >> Thanks in advance for the help and I apologize for the long email.
> >>
> >> Alfredo
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From kond at nstel.ru  Fri Jan 15 09:41:47 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Fri, 15 Jan 2010 20:41:47 +0300
Subject: [Freeswitch-users] eavesdrop problem?
In-Reply-To: <191c3a031001150805y2c6f0bd0pcea8a18d9d866b4b@mail.gmail.com>
Message-ID: <20100115174147.30E6A11F5A@mail.nstel.ru>

Anthony,

Thanks for the reply.

Can you please point me to the document where I could read about it?

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop does not say
anything about it.

But let me guess: I should add 

<action application="hash"
data="insert/${domain_name}-spymap/${destination_number}/${uuid}"/>

Into <extension name="global" continue="true"> in the dialplan.

Am I close?

Thanks and regards,

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Friday, January 15, 2010 7:05 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] eavesdrop problem?

 

don't bother,

only inbound legs are added to the db that is used to lookup for eavesdrop
because the action is in the dialplan.
The extensions to eavesdrop you are using are just a demo to show you how to
work it.
you need to know the uuid of the channel you are trying to eavesdrop on
before you can do what you want.





On Fri, Jan 15, 2010 at 8:33 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Hi all,

 

I want to use eavesdrop application.

Playing with it I found that when one tries to eavesdrop caller the feature
works ok.

But when trying to eavesdrop callee eavesdrop attempt failes.

I just updated to the latest version from http://latest.freeswitch.org
<http://latest.freeswitch.org/>  

[freeswitch at freeswitch log]$ fs_cli -x version

FreeSWITCH Version 1.0.5-20100115-0400 (16318M)

 

My setup is as following:

I have 3 internal extensions: 2853, 2840, 2879 (all are xlite).

1. 2879 calls 2840. Then 2853 dials 882879 - eavesdrop worked as expected.

2. 2840 calls 2879. Then 2853 dials 882879 - eavesdrop failed. 

 

I attached logs for both cases.

 

I don't believe it's intended behavior. 

 

Can anybody please advise if it is a configuration or a software problem?

 

Thanks and regards,

Nikolay.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From lawwton at gmail.com  Fri Jan 15 09:53:48 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 12:53:48 -0500
Subject: [Freeswitch-users] Conference Questions
In-Reply-To: <191c3a031001150902l26298d7aq80d284a9e04f59fc@mail.gmail.com>
References: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>
	<191c3a031001150813g69a86fb9w3a1601d06df5517e@mail.gmail.com>
	<5fe6fa8f1001150842v71c2eadeie7ff38231c88789d@mail.gmail.com>
	<191c3a031001150902l26298d7aq80d284a9e04f59fc@mail.gmail.com>
Message-ID: <5fe6fa8f1001150953s15320d34od960a6ad828dc501@mail.gmail.com>

Awesome! That's even nicer.

Appreciate it.

Alfredo Q-V

On Fri, Jan 15, 2010 at 12:02 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> the flags are set as part of the dial string so you can easily choose that,
> int the example I told you to look at notice the +flags{} bit at the end of
> some of the dial strings.
>
>
> On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> Appreciate the fast response Anthony.
>>
>> Response or ideas on how to implement a) ?
>>
>> a) It seemed to me like the way to setup the moderator of the
>> conference is by setting a parameter in the DialPlan and specifying
>> based on a condition who the moderator is, say for instance the
>> destination number. That's fine and it makes sense, however, say that
>> I am creating a new conference and I want to have 3 participants where
>> one of them is the moderator. What would I have to do to specify that
>> person A dialing for example number xxx-xxx-xxxx is the moderator (via
>> HTTP)? Would I have to create my own call to the system and add say an
>> entry to DialPlan with the right parameter for the moderator, then
>> create the conference?
>>
>> Thanks in advance,
>>
>> Alfredo Q-V
>>
>> On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > look at the "mad boss" extension in the default dialplan
>> > conf/dialplan/default.xml to see how to craft an all-hands conference.
>> > otherwise individual calls to originate to send people to the conference
>> > is
>> > also ok.
>> >
>> >
>> > On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil
>> > <lawwton at gmail.com> wrote:
>> >>
>> >> Hello:
>> >>
>> >> I've been using asterisk for a little bit over three years now. A
>> >> couple of months ago I found out about freeswitch, took a look at it,
>> >> thought it was interesting and moved on. A few weeks ago, I started
>> >> looking at a project I've been wanting to work on for quite a while
>> >> using conferences and started exploring systems and different
>> >> approaches. Based on the requirements I have, I decided to use
>> >> freeswitch. It seemed like it had the best support for conferencing so
>> >> I went for it. According to some documentation I found it also seems
>> >> to allow for more concurrent calls than asterisk which is an added
>> >> bonus.
>> >>
>> >> I got a server ready, installed FC8 on it which is what I have in
>> >> production now, unpacked freeswitch there and so far it's running
>> >> beautifully. Very painless process really to get it installed, I was
>> >> happy to see that. Configuration seems a bit different since it's XML;
>> >> but being a developer myself I can see many advantages to having done
>> >> that in the future as the system scales and grows in complexity.
>> >>
>> >> Sorry for the long introduction, getting to my question now. So ...
>> >> What I want to be able to do is the following:
>> >>
>> >> Create and control conferences via the HTTP API. I've been reading a
>> >> bit for the past two days the documentation and I am becoming more
>> >> familiar now with how things are done using ESL, the support for PHP,
>> >> perl and I believe others.
>> >>
>> >> a) It seemed to me like the way to setup the moderator of the
>> >> conference is by setting a parameter in the DialPlan and specifying
>> >> based on a condition who the moderator is, say for instance the
>> >> destination number. That's fine and it makes sense, however, say that
>> >> I am creating a new conference and I want to have 3 participants where
>> >> one of them is the moderator. What would I have to do to specify that
>> >> person A dialing for example number xxx-xxx-xxxx is the moderator (via
>> >> HTTP)? Would I have to create my own call to the system and add say an
>> >> entry to DialPlan with the right parameter for the moderator, then
>> >> create the conference?
>> >>
>> >> b) When a conference is created, or when I go to create a new
>> >> conference via HTTP using the API, does it allow for example for all
>> >> numbers that will be added to be dialed at once? Or should the process
>> >> be dial each participant, sending say 3 http requests via the API? The
>> >> API command "conference dial" seems to only take one argument for
>> >> destination number; but I am asking just in case I missed something.
>> >>
>> >> Thanks in advance for the help and I apologize for the long email.
>> >>
>> >> Alfredo
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From mailinglist at fribert.dk  Fri Jan 15 09:56:42 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Fri, 15 Jan 2010 18:56:42 +0100
Subject: [Freeswitch-users] TimeOfDay for company phone
Message-ID: <4B50BA6A020000E1000003BB@mail.fribert.dk>

Hi Guys
 
Still working on my pfsense based freeswitch.
So far it's actually working really good, thanks to all the help on the mailinglist and the excellent tutorials, thankyou very much to all!
 
I'm trying to have one of the phonenumbers react on the TimeOfDay. I followed the guide in the wiki.
 
In public I have set it up to dial the local extension 8203 when the company phone rings.
So I put all this in the default.xml to have it process it:
 
 
   <extension name="Company-x8203">
     <condition field="destination_number" expression="^8203$">
       <action application="transfer" data="CompanyIn"/>
     </condition>
   </extension>
 
   <extension name="CompanyIn" continue="true">
     <condition field="destination_number" expression="^CompanyIn$"/>
     <condition wday="2-6"/>
     <condition minute-of-day="540-1020">
       <action application="set" data="Company_open=true"/>
       <action application="transfer" data="xfer-to-company"/>
       <anti-action application="set" data="Company_open=false"/>
       <anti-action application="transfer" data="xfer-to-company"/>
     </condition>
   </extension>
 
   <extension name="xfer-to-company">
     <condition field="destination_number" expression="^xfer-to-company$"/>
     <condition field="${Company_open}" expression="^true$">
       <action application="bridge" data="group/company@${domain_name}"/ ( 'mailto:group/company@$domain_name"/' )>
       <action application="answer"/>
       <action application="sleep" data="2000"/>
       <action application="voicemail" data="default ${domain_name} 1011"/>
       <anti-action application="voicemail" data="default ${domain_name} 1011"/>
     </condition>
   </extension>
But it doesn't react on the time. I'm wondering if wday and minute-of-day is working on the pfsense package, as far as I know it's version 0.9.5?
 
Any input on this?
 
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From jerry.richards at teotech.com  Fri Jan 15 10:37:06 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 15 Jan 2010 10:37:06 -0800
Subject: [Freeswitch-users] INVITE From Caller Spawned 2 INVITEs to Callee
Message-ID: <4187A3F7A21540DBAE8367C24919EBC9@greyhawk.tonecommander.com>


I posted a trace to http://pastebin.freeswitch.org/11810 that shows one
INVITE spawning 2 INVITEs with two different Call-IDs to the same Callee.

I can't tell from the trace why FS created two calls?

Best Regards,
Jerry




From brian at freeswitch.org  Fri Jan 15 10:43:46 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 15 Jan 2010 12:43:46 -0600
Subject: [Freeswitch-users] INVITE From Caller Spawned 2 INVITEs to
	Callee
In-Reply-To: <4187A3F7A21540DBAE8367C24919EBC9@greyhawk.tonecommander.com>
References: <4187A3F7A21540DBAE8367C24919EBC9@greyhawk.tonecommander.com>
Message-ID: <59FD1F5B-3A75-4710-AF6D-FF479B4EBDC5@freeswitch.org>

You have two registrations in the sip registration table for the same endpoint... I suspect one is stale .

/b

On Jan 15, 2010, at 12:37 PM, Jerry Richards wrote:

> 
> I posted a trace to http://pastebin.freeswitch.org/11810 that shows one
> INVITE spawning 2 INVITEs with two different Call-IDs to the same Callee.
> 
> I can't tell from the trace why FS created two calls?
> 
> Best Regards,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From robert.hadley at teotech.com  Fri Jan 15 11:11:37 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 15 Jan 2010 11:11:37 -0800
Subject: [Freeswitch-users] How do I communicate with FS when running as
	daemon?
Message-ID: <0067A4C2F6794E2A92339A7431B7EFA3@greyhawk.tonecommander.com>

Hi,

 

I have Freeswitch running as a daemon on CentOS 5.3

 

Other than the service freeswitch start|stop|status|reload|restart commands,
how do I communicate with FS when running as daemon?

 

Also, in the example file the reload command is commented out, how would I
tell FS to reloadxml?

 

Thanks,

Robert

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From lawwton at gmail.com  Fri Jan 15 11:18:27 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 14:18:27 -0500
Subject: [Freeswitch-users] Conference Questions
In-Reply-To: <5fe6fa8f1001150953s15320d34od960a6ad828dc501@mail.gmail.com>
References: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>
	<191c3a031001150813g69a86fb9w3a1601d06df5517e@mail.gmail.com>
	<5fe6fa8f1001150842v71c2eadeie7ff38231c88789d@mail.gmail.com>
	<191c3a031001150902l26298d7aq80d284a9e04f59fc@mail.gmail.com>
	<5fe6fa8f1001150953s15320d34od960a6ad828dc501@mail.gmail.com>
Message-ID: <5fe6fa8f1001151118x748970a4y49acfcf845be8739@mail.gmail.com>

Anthony:

I think I spoke too soon. I looked at the example; but I was under the
impression based on your previous comment that I would be able to
invoke the conference dial api command over HTTP and specify the flags
as part of the dial string. The wiki page has the following for the
API call I am thinking:

dial

Dial a destination via a specific endpoint (ie. call mom from the conference).

Usage: conference <confname> dial [{dial string
options}]<endpoint_module_name>/<destination> [<callerid_number>
[<callerid_name>]]

I would like to specify the privilege in this api call. Is that
doable? If not how could I accomplish it? Would I be able to pass the
flags in {dial string options}?

Thanks in advance for the help,

Alfredo

On Fri, Jan 15, 2010 at 12:53 PM, Alfredo Quiroga-Villamil
<lawwton at gmail.com> wrote:
> Awesome! That's even nicer.
>
> Appreciate it.
>
> Alfredo Q-V
>
> On Fri, Jan 15, 2010 at 12:02 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> the flags are set as part of the dial string so you can easily choose that,
>> int the example I told you to look at notice the +flags{} bit at the end of
>> some of the dial strings.
>>
>>
>> On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil
>> <lawwton at gmail.com> wrote:
>>>
>>> Appreciate the fast response Anthony.
>>>
>>> Response or ideas on how to implement a) ?
>>>
>>> a) It seemed to me like the way to setup the moderator of the
>>> conference is by setting a parameter in the DialPlan and specifying
>>> based on a condition who the moderator is, say for instance the
>>> destination number. That's fine and it makes sense, however, say that
>>> I am creating a new conference and I want to have 3 participants where
>>> one of them is the moderator. What would I have to do to specify that
>>> person A dialing for example number xxx-xxx-xxxx is the moderator (via
>>> HTTP)? Would I have to create my own call to the system and add say an
>>> entry to DialPlan with the right parameter for the moderator, then
>>> create the conference?
>>>
>>> Thanks in advance,
>>>
>>> Alfredo Q-V
>>>
>>> On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>> > look at the "mad boss" extension in the default dialplan
>>> > conf/dialplan/default.xml to see how to craft an all-hands conference.
>>> > otherwise individual calls to originate to send people to the conference
>>> > is
>>> > also ok.
>>> >
>>> >
>>> > On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil
>>> > <lawwton at gmail.com> wrote:
>>> >>
>>> >> Hello:
>>> >>
>>> >> I've been using asterisk for a little bit over three years now. A
>>> >> couple of months ago I found out about freeswitch, took a look at it,
>>> >> thought it was interesting and moved on. A few weeks ago, I started
>>> >> looking at a project I've been wanting to work on for quite a while
>>> >> using conferences and started exploring systems and different
>>> >> approaches. Based on the requirements I have, I decided to use
>>> >> freeswitch. It seemed like it had the best support for conferencing so
>>> >> I went for it. According to some documentation I found it also seems
>>> >> to allow for more concurrent calls than asterisk which is an added
>>> >> bonus.
>>> >>
>>> >> I got a server ready, installed FC8 on it which is what I have in
>>> >> production now, unpacked freeswitch there and so far it's running
>>> >> beautifully. Very painless process really to get it installed, I was
>>> >> happy to see that. Configuration seems a bit different since it's XML;
>>> >> but being a developer myself I can see many advantages to having done
>>> >> that in the future as the system scales and grows in complexity.
>>> >>
>>> >> Sorry for the long introduction, getting to my question now. So ...
>>> >> What I want to be able to do is the following:
>>> >>
>>> >> Create and control conferences via the HTTP API. I've been reading a
>>> >> bit for the past two days the documentation and I am becoming more
>>> >> familiar now with how things are done using ESL, the support for PHP,
>>> >> perl and I believe others.
>>> >>
>>> >> a) It seemed to me like the way to setup the moderator of the
>>> >> conference is by setting a parameter in the DialPlan and specifying
>>> >> based on a condition who the moderator is, say for instance the
>>> >> destination number. That's fine and it makes sense, however, say that
>>> >> I am creating a new conference and I want to have 3 participants where
>>> >> one of them is the moderator. What would I have to do to specify that
>>> >> person A dialing for example number xxx-xxx-xxxx is the moderator (via
>>> >> HTTP)? Would I have to create my own call to the system and add say an
>>> >> entry to DialPlan with the right parameter for the moderator, then
>>> >> create the conference?
>>> >>
>>> >> b) When a conference is created, or when I go to create a new
>>> >> conference via HTTP using the API, does it allow for example for all
>>> >> numbers that will be added to be dialed at once? Or should the process
>>> >> be dial each participant, sending say 3 http requests via the API? The
>>> >> API command "conference dial" seems to only take one argument for
>>> >> destination number; but I am asking just in case I missed something.
>>> >>
>>> >> Thanks in advance for the help and I apologize for the long email.
>>> >>
>>> >> Alfredo
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com
>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org
>>> > iax:guest at conference.freeswitch.org/888
>>> > googletalk:conf+888 at conference.freeswitch.org
>>> > pstn:+19193869900
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>



From gmaruzz at celliax.org  Fri Jan 15 11:22:03 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 15 Jan 2010 20:22:03 +0100
Subject: [Freeswitch-users] How do I communicate with FS when running as
	daemon?
In-Reply-To: <0067A4C2F6794E2A92339A7431B7EFA3@greyhawk.tonecommander.com>
References: <0067A4C2F6794E2A92339A7431B7EFA3@greyhawk.tonecommander.com>
Message-ID: <7b197bef1001151122h10aa87aavcb8d59d8659d7477@mail.gmail.com>

/usr/local/freeswitch/bin/fs_cli

On Fri, Jan 15, 2010 at 8:11 PM, Robert Hadley
<robert.hadley at teotech.com> wrote:
> Hi,
>
>
>
> I have Freeswitch running as a daemon on CentOS 5.3
>
>
>
> Other than the service freeswitch start|stop|status|reload|restart commands,
> how do I communicate with FS when running as daemon?
>
>
>
> Also, in the example file the reload command is commented out, how would I
> tell FS to reloadxml?
>
>
>
> Thanks,
>
> Robert
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From msc at freeswitch.org  Fri Jan 15 11:24:06 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Jan 2010 11:24:06 -0800
Subject: [Freeswitch-users] No hangup tone after zap channel closed,
	tones 	in general
In-Reply-To: <001201ca95d1$045eb6e0$0d1c24a0$@fr.eu.org>
References: <8b18ffe229516c90285de8a54a00e845@thom.fr.eu.org>
	<87f2f3b91001141723u4b0b12e1rd0f922db493492cd@mail.gmail.com>
	<001201ca95d1$045eb6e0$0d1c24a0$@fr.eu.org>
Message-ID: <87f2f3b91001151124r596418abta126ebdaae10465@mail.gmail.com>

On Fri, Jan 15, 2010 at 2:53 AM, <devel at thom.fr.eu.org> wrote:

>  Thank you for the link. I googled through but could not find anything
> relevant.
>
>
>
> So then with my FXS port, do I have to, when a call is over, bridge the
> channel (which is either A or B leg depending on the cases) to an extension
> with for instance
>
> <action application="playback" data="tone_stream://%(500,500,440)"/>
>
if you're just trying to manually send out that tone then yes, you can just
add the line in your dialplan. You can then hangup after playing the tone.
The other end will have to decide what to do on its own.
-MC
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From msc at freeswitch.org  Fri Jan 15 11:49:42 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Jan 2010 11:49:42 -0800
Subject: [Freeswitch-users] Nibble_total_billed variable missing when
	using nibblebill on b-leg
In-Reply-To: <b042b23b1001150813r4950e607w111afb727746ce5c@mail.gmail.com>
References: <b042b23b1001150658j26891262qa396e761c54ba2c8@mail.gmail.com>
	<b042b23b1001150813r4950e607w111afb727746ce5c@mail.gmail.com>
Message-ID: <87f2f3b91001151149w1fe3c74focea7753e1df38919@mail.gmail.com>

On Fri, Jan 15, 2010 at 8:13 AM, Dan Lane <null at invalid.name> wrote:

> On Fri, Jan 15, 2010 at 2:58 PM, Dan Lane <null at invalid.name> wrote:
> >  I'm doing something similar to the example below in order to bill on
> > the b-leg. Billing is working but the variable nibble_total_billed
> > isn't being set once the call is finished.
> >
> > <action application="bridge"
> > data="${sofia_contact(internal/user@
> $${domain})},[enable_heartbeat_events=60,nibble_account=1,nibble_rate=0.01]sofia/gateway/blah/1234"/>
> >
> > I see that a few others have experienced this issue (including Jira
> > MODAPP-385) so has anyone found a work-around to coerce this into
> > working?
> >
> > If not, what will it take to get the issue resolved?
> >
>
> Of course, if I set log-b-leg=true in mod_xml_cdr then I can see the
> variable because it's only going to be set on the b-leg! *slaps
> forehead*
>
> You answered your own question nicely. You're hired! Now you can answer all
the other questions on the list. :)
-MC
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From Prometheus001 at gmx.net  Fri Jan 15 11:51:51 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 15 Jan 2010 20:51:51 +0100
Subject: [Freeswitch-users] Multiple registrations: Only calling
 the	first element in the list in this mode
In-Reply-To: <b39273531001130450k16af27d2v209a300f441d39d8@mail.gmail.com>
References: <4B4DADD5.3010507@gmx.net>
	<b39273531001130450k16af27d2v209a300f441d39d8@mail.gmail.com>
Message-ID: <4B50C757.3050901@gmx.net>

Thanks Rupa,

this worked. I have documented this in the wiki:
    http://wiki.freeswitch.org/wiki/Ring_group

Best regards
Peter

Rupa Schomaker schrieb:
> Try:
>
> bridge user/100 at domain:_:user/101 at domain:_:user/102 at domain
>
> Then document it up if it works.
>
> On Wed, Jan 13, 2010 at 5:26 AM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     Hello,
>
>     I habe the following behaviour
>
>     when I call a user which is registered twice with 2 phones via
>        bridge user/100 at domain
>     both phones are ringing. This is correct as I allow multiple
>     registrations in a profile
>
>     However when I call multiple endpoints via
>        bridge user/100 at domain,user/101 at domain,user/102 at domain
>     only one phone with number100 is ringing.
>
>     Console log shows "Only calling the first element in the list in this
>     mode.":
>     2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734
>     variable
>     string 0 = [presence_id=100 at domain]
>     2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734
>     variable
>     string 1 = [transfer_fallback_extension=100]
>     2010-01-12 19:52:18.236361 [WARNING] switch_ivr_originate.c:2048 Only
>     calling the first element in the list in this mode.
>     2010-01-12 19:52:18.236361 [NOTICE] switch_channel.c:613 New Channel
>     sofia/internal/sip:100 at 10.11.12.203:2048
>     <http://sip:100 at 10.11.12.203:2048>
>     [9b95fcdc-ffab-11de-9ba2-13b2daa7ce61]
>
>     Is there any way to work around this? I need all phones to be
>     ringing in
>     this scenario.
>
>     Best regards
>     Peter
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> -Rupa
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From msc at freeswitch.org  Fri Jan 15 12:09:06 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Jan 2010 12:09:06 -0800
Subject: [Freeswitch-users] How do I communicate with FS when running as
	daemon?
In-Reply-To: <7b197bef1001151122h10aa87aavcb8d59d8659d7477@mail.gmail.com>
References: <0067A4C2F6794E2A92339A7431B7EFA3@greyhawk.tonecommander.com>
	<7b197bef1001151122h10aa87aavcb8d59d8659d7477@mail.gmail.com>
Message-ID: <87f2f3b91001151209y5d07e75ge0a9487788f7c152@mail.gmail.com>

On Fri, Jan 15, 2010 at 11:22 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org>wrote:

> /usr/local/freeswitch/bin/fs_cli
>

fs_cli lets you connect kinda like asterisk -r on steroids.

you can also do stuff like fs_cli -x "show channels" | grep my_extension

Have fun!
-MC
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From fvillarroel at yahoo.com  Fri Jan 15 12:26:13 2010
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Fri, 15 Jan 2010 12:26:13 -0800 (PST)
Subject: [Freeswitch-users] Domains.
Message-ID: <2083.99622.qm@web34302.mail.mud.yahoo.com>

Dear.

I installed FS FreeSWITCH Version 1.0.trunk (16144)

I have a problem when i send traffic from a external gateway, the calls are rejected:

2010-01-05 16:52:37.254194 [DEBUG] sofia.c:5359 0 acls to check for proxy
2010-01-05 16:52:37.254194 [DEBUG] sofia.c:5377 network ip is a proxy [0]
2010-01-05 16:52:37.254194 [DEBUG] sofia.c:5405 IP XXX.XXX.XX.125 Rejected by acl "domains". Falling back to Digest auth.


Anyone could me explain like i can do.

Regards.

Fernando


      



From anthony.minessale at gmail.com  Fri Jan 15 12:28:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 14:28:07 -0600
Subject: [Freeswitch-users] Conference Questions
In-Reply-To: <5fe6fa8f1001151118x748970a4y49acfcf845be8739@mail.gmail.com>
References: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>
	<191c3a031001150813g69a86fb9w3a1601d06df5517e@mail.gmail.com>
	<5fe6fa8f1001150842v71c2eadeie7ff38231c88789d@mail.gmail.com>
	<191c3a031001150902l26298d7aq80d284a9e04f59fc@mail.gmail.com>
	<5fe6fa8f1001150953s15320d34od960a6ad828dc501@mail.gmail.com>
	<5fe6fa8f1001151118x748970a4y49acfcf845be8739@mail.gmail.com>
Message-ID: <191c3a031001151228p406600d3h342529e4ac98bf68@mail.gmail.com>

look up the originate api instead of conference dial

originate /dial/string conference:myconf+flags{foo} inline


On Fri, Jan 15, 2010 at 1:18 PM, Alfredo Quiroga-Villamil <lawwton at gmail.com
> wrote:

> Anthony:
>
> I think I spoke too soon. I looked at the example; but I was under the
> impression based on your previous comment that I would be able to
> invoke the conference dial api command over HTTP and specify the flags
> as part of the dial string. The wiki page has the following for the
> API call I am thinking:
>
> dial
>
> Dial a destination via a specific endpoint (ie. call mom from the
> conference).
>
> Usage: conference <confname> dial [{dial string
> options}]<endpoint_module_name>/<destination> [<callerid_number>
> [<callerid_name>]]
>
> I would like to specify the privilege in this api call. Is that
> doable? If not how could I accomplish it? Would I be able to pass the
> flags in {dial string options}?
>
> Thanks in advance for the help,
>
> Alfredo
>
> On Fri, Jan 15, 2010 at 12:53 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
> > Awesome! That's even nicer.
> >
> > Appreciate it.
> >
> > Alfredo Q-V
> >
> > On Fri, Jan 15, 2010 at 12:02 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >> the flags are set as part of the dial string so you can easily choose
> that,
> >> int the example I told you to look at notice the +flags{} bit at the end
> of
> >> some of the dial strings.
> >>
> >>
> >> On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil
> >> <lawwton at gmail.com> wrote:
> >>>
> >>> Appreciate the fast response Anthony.
> >>>
> >>> Response or ideas on how to implement a) ?
> >>>
> >>> a) It seemed to me like the way to setup the moderator of the
> >>> conference is by setting a parameter in the DialPlan and specifying
> >>> based on a condition who the moderator is, say for instance the
> >>> destination number. That's fine and it makes sense, however, say that
> >>> I am creating a new conference and I want to have 3 participants where
> >>> one of them is the moderator. What would I have to do to specify that
> >>> person A dialing for example number xxx-xxx-xxxx is the moderator (via
> >>> HTTP)? Would I have to create my own call to the system and add say an
> >>> entry to DialPlan with the right parameter for the moderator, then
> >>> create the conference?
> >>>
> >>> Thanks in advance,
> >>>
> >>> Alfredo Q-V
> >>>
> >>> On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale
> >>> <anthony.minessale at gmail.com> wrote:
> >>> > look at the "mad boss" extension in the default dialplan
> >>> > conf/dialplan/default.xml to see how to craft an all-hands
> conference.
> >>> > otherwise individual calls to originate to send people to the
> conference
> >>> > is
> >>> > also ok.
> >>> >
> >>> >
> >>> > On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil
> >>> > <lawwton at gmail.com> wrote:
> >>> >>
> >>> >> Hello:
> >>> >>
> >>> >> I've been using asterisk for a little bit over three years now. A
> >>> >> couple of months ago I found out about freeswitch, took a look at
> it,
> >>> >> thought it was interesting and moved on. A few weeks ago, I started
> >>> >> looking at a project I've been wanting to work on for quite a while
> >>> >> using conferences and started exploring systems and different
> >>> >> approaches. Based on the requirements I have, I decided to use
> >>> >> freeswitch. It seemed like it had the best support for conferencing
> so
> >>> >> I went for it. According to some documentation I found it also seems
> >>> >> to allow for more concurrent calls than asterisk which is an added
> >>> >> bonus.
> >>> >>
> >>> >> I got a server ready, installed FC8 on it which is what I have in
> >>> >> production now, unpacked freeswitch there and so far it's running
> >>> >> beautifully. Very painless process really to get it installed, I was
> >>> >> happy to see that. Configuration seems a bit different since it's
> XML;
> >>> >> but being a developer myself I can see many advantages to having
> done
> >>> >> that in the future as the system scales and grows in complexity.
> >>> >>
> >>> >> Sorry for the long introduction, getting to my question now. So ...
> >>> >> What I want to be able to do is the following:
> >>> >>
> >>> >> Create and control conferences via the HTTP API. I've been reading a
> >>> >> bit for the past two days the documentation and I am becoming more
> >>> >> familiar now with how things are done using ESL, the support for
> PHP,
> >>> >> perl and I believe others.
> >>> >>
> >>> >> a) It seemed to me like the way to setup the moderator of the
> >>> >> conference is by setting a parameter in the DialPlan and specifying
> >>> >> based on a condition who the moderator is, say for instance the
> >>> >> destination number. That's fine and it makes sense, however, say
> that
> >>> >> I am creating a new conference and I want to have 3 participants
> where
> >>> >> one of them is the moderator. What would I have to do to specify
> that
> >>> >> person A dialing for example number xxx-xxx-xxxx is the moderator
> (via
> >>> >> HTTP)? Would I have to create my own call to the system and add say
> an
> >>> >> entry to DialPlan with the right parameter for the moderator, then
> >>> >> create the conference?
> >>> >>
> >>> >> b) When a conference is created, or when I go to create a new
> >>> >> conference via HTTP using the API, does it allow for example for all
> >>> >> numbers that will be added to be dialed at once? Or should the
> process
> >>> >> be dial each participant, sending say 3 http requests via the API?
> The
> >>> >> API command "conference dial" seems to only take one argument for
> >>> >> destination number; but I am asking just in case I missed something.
> >>> >>
> >>> >> Thanks in advance for the help and I apologize for the long email.
> >>> >>
> >>> >> Alfredo
> >>> >>
> >>> >> _______________________________________________
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>
> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >
> >>> >
> >>> >
> >>> > --
> >>> > Anthony Minessale II
> >>> >
> >>> > FreeSWITCH http://www.freeswitch.org/
> >>> > ClueCon http://www.cluecon.com/
> >>> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>> >
> >>> > AIM: anthm
> >>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> > IRC: irc.freenode.net #freeswitch
> >>> >
> >>> > FreeSWITCH Developer Conference
> >>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> > iax:guest at conference.freeswitch.org/888
> >>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> > pstn:+19193869900
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From kristian.kielhofner at gmail.com  Fri Jan 15 12:33:52 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 15 Jan 2010 15:33:52 -0500
Subject: [Freeswitch-users] Changing the username portion of the RURI with
	registered devices
Message-ID: <2d9149cd1001151233h6a468169yd213b1379940e1ae@mail.gmail.com>

Hello everyone,

  Is there (or can we get) a way to set the username portion of the
request URI (along with To:) arbitrarily?  Obviously this can be done
when specifying a bridge string but I'm wondering if it would be
possible with registered contacts.  So...

  Let's say I have another FS/Asterisk/etc system registered with the
username "gw".  Let's say I want to direct multiple DIDs to that
registered endpoint.  I'd bridge to:

sofia/internal/gw%domain.local

(or whatever my domain was)

  The INVITE will go out with a destination URI of the registered
contact.  Probably something like:

gw at 192.168.1.10 (assuming that's where I was registered)

  I'd like to be able to do something like:

{user_uri=9415551212}sofia/internal/gw%local.domain

  Before sending the INVITE to the device, FS/Sofia would replace the
username portion in the RURI/To: with 9415551212, something like:

9415551212 at 192.168.1.10

  Is this currently possible?

  Basically I don't want to have to independently register each DID in
this scenario.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From brian at freeswitch.org  Fri Jan 15 12:40:04 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 15 Jan 2010 14:40:04 -0600
Subject: [Freeswitch-users] Changing the username portion of the RURI
	with registered devices
In-Reply-To: <2d9149cd1001151233h6a468169yd213b1379940e1ae@mail.gmail.com>
References: <2d9149cd1001151233h6a468169yd213b1379940e1ae@mail.gmail.com>
Message-ID: <5BFE0182-1CC6-43BA-A591-6B6F2A7A0C31@freeswitch.org>

you can set the sip-force-user param on the user in the directory to force it.

or

    <!--                                                                                                                                                        
        This is an example of how to override the RURI on an outgoing invite to a registered contact.                                                           
    -->                                                                                                                                                         

    <extension name="ruri">                                                                                                                                     
      <condition field="destination_number" expression="^ruri$">                                                                                                
        <action application="bridge" data="sofia/${ruri_profile}/${ruri_user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/> 
      </condition>                                                                                                                                              
    </extension>                                                                                                                                                
                                                                                                                                                                
    <extension name="7004">                                                                                                                                     
      <condition field="destination_number" expression="^7004$">                                                                                                
        <action application="set" data="ruri_profile=default"/>                                                                                                 
        <action application="set" data="ruri_user=2000"/>                                                                                                       
        <action application="set" data="ruri_contact=1001@${domain_name}"/>                                                                                     
        <action application="execute_extension" data="ruri"/>                                                                                                   
      </condition>                                                                                                                                              
    </extension>                                                                                                                                                

/b


On Jan 15, 2010, at 2:33 PM, Kristian Kielhofner wrote:

> Hello everyone,
> 
>  Is there (or can we get) a way to set the username portion of the
> request URI (along with To:) arbitrarily?  Obviously this can be done
> when specifying a bridge string but I'm wondering if it would be
> possible with registered contacts.  So...
> 
>  Let's say I have another FS/Asterisk/etc system registered with the
> username "gw".  Let's say I want to direct multiple DIDs to that
> registered endpoint.  I'd bridge to:
> 
> sofia/internal/gw%domain.local
> 
> (or whatever my domain was)
> 
>  The INVITE will go out with a destination URI of the registered
> contact.  Probably something like:
> 
> gw at 192.168.1.10 (assuming that's where I was registered)
> 
>  I'd like to be able to do something like:
> 
> {user_uri=9415551212}sofia/internal/gw%local.domain
> 
>  Before sending the INVITE to the device, FS/Sofia would replace the
> username portion in the RURI/To: with 9415551212, something like:
> 
> 9415551212 at 192.168.1.10
> 
>  Is this currently possible?
> 
>  Basically I don't want to have to independently register each DID in
> this scenario.
> 
> -- 
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From Mailings at kh-dev.de  Fri Jan 15 12:57:57 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Fri, 15 Jan 2010 21:57:57 +0100
Subject: [Freeswitch-users] proxy_media seems to be broken
In-Reply-To: <194CB7A1-D382-4F98-AB9A-21AFFABFFD6B@freeswitch.org>
References: <A2F1C0E212BBBA4E8C0E7BE568361F0403B145DC1941@srv02.khdev.corp>
	<608345.82340.qm@web37502.mail.mud.yahoo.com>
	<194CB7A1-D382-4F98-AB9A-21AFFABFFD6B@freeswitch.org>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403B145DC1948@srv02.khdev.corp>

I saw the discussion, but the discussions were about bypass_media and I wasn't sure if that also applied to proxy_media.
That's why I was asking.


Ok, but now I got another problem with proxy_media/bypass_media in combination with T38 (latest tarball).

When using proxy_media FreeSWITCH hangs up immediately after my fax took the call with: [CS_EXCHANGE_MEDIA] [INCOMPATIBLE_DESTINATION]

In the log of my Cisco SPA2102 there's the message:
Peer Confirm T38
Peer Confirm T38
No T38 in SDP
No T38 in SDP


Then I tried bypass_media instead of proxy_media. There FS doesn't hang up, but the same log entries in my Cisco.
Maybe the SDP isn't copied after the reinvite for T38???

Thanks, Klaus



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Tuesday, January 12, 2010 4:53 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] proxy_media seems to be broken

And the tarball is updated already automatically too.  Please update to the latest FreeSWITCH... report any issues to jira if you have them in the future.

In the future please read thru the mailing list as this was discussed in two different threads yesterday with the details and the rev where it was fixed.

Thanks,
/b

On Jan 11, 2010, at 9:34 PM, DJB wrote:


Yes, it has been fixed in 16250.


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From lawwton at gmail.com  Fri Jan 15 13:07:33 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 16:07:33 -0500
Subject: [Freeswitch-users] Conference Questions
In-Reply-To: <191c3a031001151228p406600d3h342529e4ac98bf68@mail.gmail.com>
References: <5fe6fa8f1001150700l5d210ccene39260f3297f24c6@mail.gmail.com>
	<191c3a031001150813g69a86fb9w3a1601d06df5517e@mail.gmail.com>
	<5fe6fa8f1001150842v71c2eadeie7ff38231c88789d@mail.gmail.com>
	<191c3a031001150902l26298d7aq80d284a9e04f59fc@mail.gmail.com>
	<5fe6fa8f1001150953s15320d34od960a6ad828dc501@mail.gmail.com>
	<5fe6fa8f1001151118x748970a4y49acfcf845be8739@mail.gmail.com>
	<191c3a031001151228p406600d3h342529e4ac98bf68@mail.gmail.com>
Message-ID: <5fe6fa8f1001151307l4f1f680fn71896f4be0373887@mail.gmail.com>

Great!

Thanks Anthony, really appreciate the help.

Alfredo Q-V

On Fri, Jan 15, 2010 at 3:28 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> look up the originate api instead of conference dial
>
> originate /dial/string conference:myconf+flags{foo} inline
>
>
> On Fri, Jan 15, 2010 at 1:18 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> Anthony:
>>
>> I think I spoke too soon. I looked at the example; but I was under the
>> impression based on your previous comment that I would be able to
>> invoke the conference dial api command over HTTP and specify the flags
>> as part of the dial string. The wiki page has the following for the
>> API call I am thinking:
>>
>> dial
>>
>> Dial a destination via a specific endpoint (ie. call mom from the
>> conference).
>>
>> Usage: conference <confname> dial [{dial string
>> options}]<endpoint_module_name>/<destination> [<callerid_number>
>> [<callerid_name>]]
>>
>> I would like to specify the privilege in this api call. Is that
>> doable? If not how could I accomplish it? Would I be able to pass the
>> flags in {dial string options}?
>>
>> Thanks in advance for the help,
>>
>> Alfredo
>>
>> On Fri, Jan 15, 2010 at 12:53 PM, Alfredo Quiroga-Villamil
>> <lawwton at gmail.com> wrote:
>> > Awesome! That's even nicer.
>> >
>> > Appreciate it.
>> >
>> > Alfredo Q-V
>> >
>> > On Fri, Jan 15, 2010 at 12:02 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >> the flags are set as part of the dial string so you can easily choose
>> >> that,
>> >> int the example I told you to look at notice the +flags{} bit at the
>> >> end of
>> >> some of the dial strings.
>> >>
>> >>
>> >> On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil
>> >> <lawwton at gmail.com> wrote:
>> >>>
>> >>> Appreciate the fast response Anthony.
>> >>>
>> >>> Response or ideas on how to implement a) ?
>> >>>
>> >>> a) It seemed to me like the way to setup the moderator of the
>> >>> conference is by setting a parameter in the DialPlan and specifying
>> >>> based on a condition who the moderator is, say for instance the
>> >>> destination number. That's fine and it makes sense, however, say that
>> >>> I am creating a new conference and I want to have 3 participants where
>> >>> one of them is the moderator. What would I have to do to specify that
>> >>> person A dialing for example number xxx-xxx-xxxx is the moderator (via
>> >>> HTTP)? Would I have to create my own call to the system and add say an
>> >>> entry to DialPlan with the right parameter for the moderator, then
>> >>> create the conference?
>> >>>
>> >>> Thanks in advance,
>> >>>
>> >>> Alfredo Q-V
>> >>>
>> >>> On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale
>> >>> <anthony.minessale at gmail.com> wrote:
>> >>> > look at the "mad boss" extension in the default dialplan
>> >>> > conf/dialplan/default.xml to see how to craft an all-hands
>> >>> > conference.
>> >>> > otherwise individual calls to originate to send people to the
>> >>> > conference
>> >>> > is
>> >>> > also ok.
>> >>> >
>> >>> >
>> >>> > On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil
>> >>> > <lawwton at gmail.com> wrote:
>> >>> >>
>> >>> >> Hello:
>> >>> >>
>> >>> >> I've been using asterisk for a little bit over three years now. A
>> >>> >> couple of months ago I found out about freeswitch, took a look at
>> >>> >> it,
>> >>> >> thought it was interesting and moved on. A few weeks ago, I started
>> >>> >> looking at a project I've been wanting to work on for quite a while
>> >>> >> using conferences and started exploring systems and different
>> >>> >> approaches. Based on the requirements I have, I decided to use
>> >>> >> freeswitch. It seemed like it had the best support for conferencing
>> >>> >> so
>> >>> >> I went for it. According to some documentation I found it also
>> >>> >> seems
>> >>> >> to allow for more concurrent calls than asterisk which is an added
>> >>> >> bonus.
>> >>> >>
>> >>> >> I got a server ready, installed FC8 on it which is what I have in
>> >>> >> production now, unpacked freeswitch there and so far it's running
>> >>> >> beautifully. Very painless process really to get it installed, I
>> >>> >> was
>> >>> >> happy to see that. Configuration seems a bit different since it's
>> >>> >> XML;
>> >>> >> but being a developer myself I can see many advantages to having
>> >>> >> done
>> >>> >> that in the future as the system scales and grows in complexity.
>> >>> >>
>> >>> >> Sorry for the long introduction, getting to my question now. So ...
>> >>> >> What I want to be able to do is the following:
>> >>> >>
>> >>> >> Create and control conferences via the HTTP API. I've been reading
>> >>> >> a
>> >>> >> bit for the past two days the documentation and I am becoming more
>> >>> >> familiar now with how things are done using ESL, the support for
>> >>> >> PHP,
>> >>> >> perl and I believe others.
>> >>> >>
>> >>> >> a) It seemed to me like the way to setup the moderator of the
>> >>> >> conference is by setting a parameter in the DialPlan and specifying
>> >>> >> based on a condition who the moderator is, say for instance the
>> >>> >> destination number. That's fine and it makes sense, however, say
>> >>> >> that
>> >>> >> I am creating a new conference and I want to have 3 participants
>> >>> >> where
>> >>> >> one of them is the moderator. What would I have to do to specify
>> >>> >> that
>> >>> >> person A dialing for example number xxx-xxx-xxxx is the moderator
>> >>> >> (via
>> >>> >> HTTP)? Would I have to create my own call to the system and add say
>> >>> >> an
>> >>> >> entry to DialPlan with the right parameter for the moderator, then
>> >>> >> create the conference?
>> >>> >>
>> >>> >> b) When a conference is created, or when I go to create a new
>> >>> >> conference via HTTP using the API, does it allow for example for
>> >>> >> all
>> >>> >> numbers that will be added to be dialed at once? Or should the
>> >>> >> process
>> >>> >> be dial each participant, sending say 3 http requests via the API?
>> >>> >> The
>> >>> >> API command "conference dial" seems to only take one argument for
>> >>> >> destination number; but I am asking just in case I missed
>> >>> >> something.
>> >>> >>
>> >>> >> Thanks in advance for the help and I apologize for the long email.
>> >>> >>
>> >>> >> Alfredo
>> >>> >>
>> >>> >> _______________________________________________
>> >>> >> FreeSWITCH-users mailing list
>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>
>> >>> >>
>> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>> >
>> >>> > --
>> >>> > Anthony Minessale II
>> >>> >
>> >>> > FreeSWITCH http://www.freeswitch.org/
>> >>> > ClueCon http://www.cluecon.com/
>> >>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>> >
>> >>> > AIM: anthm
>> >>> > MSN:anthony_minessale at hotmail.com
>> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>> > IRC: irc.freenode.net #freeswitch
>> >>> >
>> >>> > FreeSWITCH Developer Conference
>> >>> > sip:888 at conference.freeswitch.org
>> >>> > iax:guest at conference.freeswitch.org/888
>> >>> > googletalk:conf+888 at conference.freeswitch.org
>> >>> > pstn:+19193869900
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> iax:guest at conference.freeswitch.org/888
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From robert.hadley at teotech.com  Fri Jan 15 13:07:43 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 15 Jan 2010 13:07:43 -0800
Subject: [Freeswitch-users] xset warning message starting FS as daemon
Message-ID: <10273ADF13B842FE8527E8FF93BD6C28@greyhawk.tonecommander.com>

Hi,

 

Running trunk on CentOS 5.3, I don't get any warning messages starting FS
manually with the -nc option.

 

I get this message when I start FS as a daemon.

 

[root at roberth-c53 fstrkbld]# service freeswitch start

Starting freeswitch: xset:  unable to open display ""

xset:  unable to open display ""

                                                           [  OK  ]

 

 

Is this due to something I am doing or is it FS?

 

Thanks,

Robert

 

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From jcasale at activenetwerx.com  Fri Jan 15 13:16:17 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 15 Jan 2010 21:16:17 +0000
Subject: [Freeswitch-users] OT: Voip presentation
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E907D1A2@prato.activenetwerx.local>

I have to do a voip presentation to a group and need something elaborating on the
whole process, where does a call go once it leaves it the building on the wire, how
it re-enters the pstn, wtf is a pstn clec etc...

Rather than make something from scratch (although prolly not hard as the level of
detail needed is low) I thought I would ask if anyone knew of something in the public
domain I could snag and start with?

Hopefully the next project is dropping 1/2 or a dozen ports into a Nortel Meridian
dinosaur with fs behind it...

Thanks!
jlc



From anthony.minessale at gmail.com  Fri Jan 15 13:17:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 15:17:52 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001150808i69b3937boe0e903057f984e96@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
	<201001150726.17430.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>
	<9853f4ff1001150443t5a18cc6bxa5032f391ab6f0ed@mail.gmail.com>
	<191c3a031001150808i69b3937boe0e903057f984e96@mail.gmail.com>
Message-ID: <191c3a031001151317j408794a3pa2a0a5d21d63aa62@mail.gmail.com>

Try latest trunk,

you should have exactly what you want with the same parameter, again my
paypal addr is cleary displayed as a big button on the website.


On Fri, Jan 15, 2010 at 10:08 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> one of the many reasons its a bad idea.
> Probably the leg with the bad audio is a different ptime.
> Now the amount of work I have to do escalates I would prefer you commit to
> commercial support by emailing me at consulting at freeswitch.org to continue
> with this.
>
>
>
> On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> I set it to "off" just in case, same thing.
>>
>> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
>> <david.villasmil.work at gmail.com> wrote:
>> > Default, haven't touched it i suppose it's off, i haven't set it
>> anywhere
>> >
>> > On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
>> > <sos at sokhapkin.dyndns.org> wrote:
>> >> Is bypass_media on or off?
>> >>
>> >> On Friday 15 January 2010, David Villasmil wrote:
>> >>> Yeah, sorry. Early media. Audio after 200 is fine. Early media was ok
>> >>> before the change.
>> >>>
>> >>> David
>> >>>
>> >>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
>> >>>
>> >>> <sos at sokhapkin.dyndns.org> wrote:
>> >>> > Which audio? Early media or after 200 OK?
>> >>> >
>> >>> > On Friday 15 January 2010, David Villasmil wrote:
>> >>> >> Hello again Anthony,
>> >>> >>
>> >>> >> I just tested it, and although functionality does not, first
>> incoming
>> >>> >> audio is coming in all garbled... do you know why?
>> >>> >>
>> >>> >> David
>> >>> >>
>> >>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>> >>> >>
>> >>> >> <anthony.minessale at gmail.com> wrote:
>> >>> >> > {bridge_early_media=true}
>> >>> >> > in the dial string in place of ignore_early_media=true
>> >>> >> >
>> >>> >> > the first b leg in the list who sends 183 will become the
>> ringback
>> >>> >> > device for A leg it will hear the early media
>> >>> >> > for that leg while the other legs still ring.  If some other leg
>> >>> >> > answers the final call will still be bridged to the leg who
>> answered.
>> >>> >> >
>> >>> >> >
>> >>> >> > I would estimate it at $500 payable on the big paypal button on
>> >>> >> > http://www.freeswitch.org
>> >>> >> > but, I already added the patch to tree earlier today so I guess
>> it's
>> >>> >> > up to you to pay it or not.
>> >>> >> >
>> >>> >> >
>> >>> >> >
>> >>> >> >
>> >>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
>> >>> >> >
>> >>> >> > <david.villasmil.work at gmail.com> wrote:
>> >>> >> >> Anthony,
>> >>> >> >>
>> >>> >> >> I did take the "hint", don't worry. We will probably ask for a
>> bounty
>> >>> >> >> but first we need to know:
>> >>> >> >> 1.- whether this is possible
>> >>> >> >> 2.- how long it would take
>> >>> >> >> 3.- how will it exactly work
>> >>> >> >> 4.- of course, what's the bounty (be gentle ;) )
>> >>> >> >>
>> >>> >> >> We would of course give this back to the community.
>> >>> >> >>
>> >>> >> >> in the meantime, bridge_early_media=true will discard the 2nd
>> B-leg
>> >>> >> >> and conversely "false" it will keep on trying to connect and if
>> it
>> >>> >> >> connects the other B-leg if will bridge to that one?
>> >>> >> >>
>> >>> >> >> Thanks
>> >>> >> >>
>> >>> >> >> David
>> >>> >> >>
>> >>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> >>> >> >>
>> >>> >> >> <anthony.minessale at gmail.com> wrote:
>> >>> >> >> > I added bridge_early_media=true to do the best I can do.
>> >>> >> >> > This is the most I will do, especially for free, nobody can
>> take a
>> >>> >> >> > hint that
>> >>> >> >> > you should be paying for all these custom requests so take it
>> or
>> >>> >> >> > leave it
>> >>> >> >> > but this thread is done.........
>> >>> >> >> >
>> >>> >> >> >
>> >>> >> >> >
>> >>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>> >>> >> >> > <sos at sokhapkin.dyndns.org>
>> >>> >> >> >
>> >>> >> >> > wrote:
>> >>> >> >> >> No, not exactly. ignore_early_media doesn't pass early media
>> to
>> >>> >> >> >> the caller
>> >>> >> >> >> if
>> >>> >> >> >> bypass_media is false.
>> >>> >> >> >>
>> >>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
>> >>> >> >> >> > this is exactly what ignore_early_media does now.
>> >>> >> >> >> >
>> >>> >> >> >> > Mike
>> >>> >> >> >> >
>> >>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> >>> >> >> >> > > The issue here is when "originate" routine should return
>> and
>> >>> >> >> >> > > set "originate_status" variable. Current behavior is to
>> return
>> >>> >> >> >> > > on early
>> >>> >> >> >> > > media, but what if to introduce a variable
>> >>> >> >> >> > > "originate_wait_for_answer"
>> >>> >> >> >> > > with default value "false" and use the variable in
>> originate
>> >>> >> >> >> > > code to
>> >>> >> >> >> > > decide when to return - on 18X or "200 OK"?
>> >>> >> >> >> > >
>> >>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> >>> >> >> >> > >> he wants to call 3 people at once and let the A leg hear
>> >>> >> >> >> > >> early media
>> >>> >> >> >> > >> from call #1 while call #2 and #3 still are progressing
>> which
>> >>> >> >> >> > >> is not
>> >>> >> >> >> > >> simple to do without doing thousands of dollars in
>> >>> >> >> >> > >> development.
>> >>> >> >> >> > >>
>> >>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB <
>> djbinter at yahoo.com>
>> >> wrote:
>> >>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so that
>> your
>> >>> >> >> >> > >>> customers
>> >>> >> >> >> > >>> can hear recordings?
>> >>> >> >> >> >
>> >>> >> >> >> > _______________________________________________
>> >>> >> >> >> > FreeSWITCH-users mailing list
>> >>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >> >> >> >
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> >> >> >
>> >>> >> >> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswit
>> >>> >> >> >> >ch- users http://www.freeswitch.org
>> >>> >> >> >>
>> >>> >> >> >> _______________________________________________
>> >>> >> >> >> FreeSWITCH-users mailing list
>> >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> >> >>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> >> >>
>> >>> >> >> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch
>> >>> >> >> >>-us ers http://www.freeswitch.org
>> >>> >> >> >
>> >>> >> >> > --
>> >>> >> >> > Anthony Minessale II
>> >>> >> >> >
>> >>> >> >> > FreeSWITCH http://www.freeswitch.org/
>> >>> >> >> > ClueCon http://www.cluecon.com/
>> >>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>> >> >> >
>> >>> >> >> > AIM: anthm
>> >>> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >>> >> >> > IRC: irc.freenode.net #freeswitch
>> >>> >> >> >
>> >>> >> >> > FreeSWITCH Developer Conference
>> >>> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >>> >> >> > iax:guest at conference.freeswitch.org/888
>> >>> >> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >>> >> >> > pstn:+19193869900
>> >>> >> >> >
>> >>> >> >> > _______________________________________________
>> >>> >> >> > FreeSWITCH-users mailing list
>> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-
>> >>> >> >> >use rs http://www.freeswitch.org
>> >>> >> >>
>> >>> >> >> _______________________________________________
>> >>> >> >> FreeSWITCH-users mailing list
>> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-us
>> >>> >> >>ers http://www.freeswitch.org
>> >>> >> >
>> >>> >> > --
>> >>> >> > Anthony Minessale II
>> >>> >> >
>> >>> >> > FreeSWITCH http://www.freeswitch.org/
>> >>> >> > ClueCon http://www.cluecon.com/
>> >>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>> >> >
>> >>> >> > AIM: anthm
>> >>> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >>> >> > IRC: irc.freenode.net #freeswitch
>> >>> >> >
>> >>> >> > FreeSWITCH Developer Conference
>> >>> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >>> >> > iax:guest at conference.freeswitch.org/888
>> >>> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >>> >> > pstn:+19193869900
>> >>> >> >
>> >>> >> > _______________________________________________
>> >>> >> > FreeSWITCH-users mailing list
>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-use
>> >>> >> >rs http://www.freeswitch.org
>> >>> >>
>> >>> >> _______________________________________________
>> >>> >> FreeSWITCH-users mailing list
>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> http://www.freeswitch.org
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at jerris.com  Fri Jan 15 13:24:04 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 Jan 2010 16:24:04 -0500
Subject: [Freeswitch-users] xset warning message starting FS as daemon
In-Reply-To: <10273ADF13B842FE8527E8FF93BD6C28@greyhawk.tonecommander.com>
References: <10273ADF13B842FE8527E8FF93BD6C28@greyhawk.tonecommander.com>
Message-ID: <E1A9E241-30F9-4470-BEAC-869E265341D6@jerris.com>

Its something in the init script.  are you using one from the freeswitch tree or your own?

Mike

On Jan 15, 2010, at 4:07 PM, Robert Hadley wrote:

> Hi,
>  
> Running trunk on CentOS 5.3, I don?t get any warning messages starting FS manually with the ?nc option.
>  
> I get this message when I start FS as a daemon.
>  
> [root at roberth-c53 fstrkbld]# service freeswitch start
> Starting freeswitch: xset:  unable to open display ""
> xset:  unable to open display ""
>                                                            [  OK  ]
>  
>  
> Is this due to something I am doing or is it FS?
>  
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From lawwton at gmail.com  Fri Jan 15 14:37:24 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 17:37:24 -0500
Subject: [Freeswitch-users] Issue Compiling ESL (PHP)
Message-ID: <5fe6fa8f1001151437x491dade6q8b316f312556ff07@mail.gmail.com>

All:

System:

Linux usnc-rtp-01-fs 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT
2007 i686 i686 i386 GNU/Linux

I am trying to compile ESL, following the following steps:

1- cd to my libs/esl directory as the wiki page indicates.
2- run make

I then get right away the following:

cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o
cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o
src/esl_event.o
cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o
src/esl_threadmutex.o
cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o
src/esl_config.o
g++  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
-Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o
ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o
src/esl_config.o src/esl_oop.o
ranlib libesl.a
cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L.
-L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit
cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient
-L. -lncurses -lpthread -lesl
cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver
-L. -lncurses -lpthread -lesl
cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes ivrd.c -o ivrd -L. -lncurses
-lpthread -lesl

3- I try typing then:

make phpmod and get the following:

make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
CFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
-ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
-Wstrict-prototypes -Wmissing-prototypes"
CXXFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
-Wno-unused-variable" CXX_CFLAGS="" -C php
make[1]: Entering directory
`/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
g++  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
-Wno-unused-variable -I/usr/include/php -I/usr/include/php/main
-I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext
-I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function
-c esl_wrap.cpp -o esl_wrap.o
g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/kerberos/lib
-lcrypt -lcrypt -lncurses -laspell -lpspell -lcurl -lresolv -lm -ldl
-lnsl -lm -ldl -lcurl -ldl -lm -lcrypt -lm -lm -lcrypt -lpthread -o
ESL.so -L.
make[1]: Leaving directory
`/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'

I've installed I think all the -dev dependencies listed in the wiki. Any ideas?

Thanks in advance,

Alfredo Q-V



From anthony.minessale at gmail.com  Fri Jan 15 14:45:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 16:45:52 -0600
Subject: [Freeswitch-users] eavesdrop problem?
In-Reply-To: <20100115174147.30E6A11F5A@mail.nstel.ru>
References: <191c3a031001150805y2c6f0bd0pcea8a18d9d866b4b@mail.gmail.com>
	<20100115174147.30E6A11F5A@mail.nstel.ru>
Message-ID: <191c3a031001151445n51ba1514rb387179bb837c558@mail.gmail.com>

yes, exactly.
That is a demo of how you could possibly store a uuid by inserting them into
the db keyed from your user extension in the caller id.

if you do show channel and you see the uuid for each leg that is the
argument eavesdrop takes.
you can also do "all" in place of a uuid so you can cycle all the calls with
DTMF



On Fri, Jan 15, 2010 at 11:41 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

>  Anthony,
>
> Thanks for the reply.
>
> Can you please point me to the document where I could read about it?
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop does not
> say anything about it?
>
> But let me guess: I should add
>
> <action application="hash"
> data="insert/${domain_name}-spymap/${destination_number}/${uuid}"/>
>
> Into <extension name="global" continue="true"> in the dialplan.
>
> Am I close?
>
> Thanks and regards,
>
> Nikolay.
>
>
>   ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Friday, January 15, 2010 7:05 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] eavesdrop problem?
>
>
>
> don't bother,
>
>
> only inbound legs are added to the db that is used to lookup for eavesdrop
> because the action is in the dialplan.
> The extensions to eavesdrop you are using are just a demo to show you how
> to work it.
> you need to know the uuid of the channel you are trying to eavesdrop on
> before you can do what you want.
>
>
>
>  On Fri, Jan 15, 2010 at 8:33 AM, Nikolay Kondratyev <kond at nstel.ru>
> wrote:
>
> Hi all,
>
>
>
> I want to use eavesdrop application.
>
> Playing with it I found that when one tries to eavesdrop caller the feature
> works ok.
>
> But when trying to eavesdrop callee eavesdrop attempt failes.
>
> I just updated to the latest version from http://latest.freeswitch.org
>
> [freeswitch at freeswitch log]$ fs_cli -x version
>
> FreeSWITCH Version 1.0.5-20100115-0400 (16318M)
>
>
>
> My setup is as following:
>
> I have 3 internal extensions: 2853, 2840, 2879 (all are xlite).
>
> 1. 2879 calls 2840. Then 2853 dials 882879 ? eavesdrop worked as expected.
>
> 2. 2840 calls 2879. Then 2853 dials 882879 ? eavesdrop failed.
>
>
>
> I attached logs for both cases.
>
>
>
> I don?t believe it?s intended behavior.
>
>
>
> Can anybody please advise if it is a configuration or a software problem?
>
>
>
> Thanks and regards,
>
> Nikolay.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From david.villasmil.work at gmail.com  Fri Jan 15 14:45:59 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 15 Jan 2010 23:45:59 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001151317j408794a3pa2a0a5d21d63aa62@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
	<201001150726.17430.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>
	<9853f4ff1001150443t5a18cc6bxa5032f391ab6f0ed@mail.gmail.com>
	<191c3a031001150808i69b3937boe0e903057f984e96@mail.gmail.com>
	<191c3a031001151317j408794a3pa2a0a5d21d63aa62@mail.gmail.com>
Message-ID: <9853f4ff1001151445s6a4dff0qf8c9e17c702b7f86@mail.gmail.com>

Anthony,

     Trying, Thanks. Is there anyway we can communicate directly?


David

On Fri, Jan 15, 2010 at 10:17 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Try latest trunk,
>
> you should have exactly what you want with the same parameter, again my
> paypal addr is cleary displayed as a big button on the website.
>
>
> On Fri, Jan 15, 2010 at 10:08 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> one of the many reasons its a bad idea.
>> Probably the leg with the bad audio is a different ptime.
>> Now the amount of work I have to do escalates I would prefer you commit to
>> commercial support by emailing me at consulting at freeswitch.org to continue
>> with this.
>>
>>
>> On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil
>> <david.villasmil.work at gmail.com> wrote:
>>>
>>> I set it to "off" just in case, same thing.
>>>
>>> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
>>> <david.villasmil.work at gmail.com> wrote:
>>> > Default, haven't touched it i suppose it's off, i haven't set it
>>> > anywhere
>>> >
>>> > On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
>>> > <sos at sokhapkin.dyndns.org> wrote:
>>> >> Is bypass_media on or off?
>>> >>
>>> >> On Friday 15 January 2010, David Villasmil wrote:
>>> >>> Yeah, sorry. Early media. Audio after 200 is fine. Early media was ok
>>> >>> before the change.
>>> >>>
>>> >>> David
>>> >>>
>>> >>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
>>> >>>
>>> >>> <sos at sokhapkin.dyndns.org> wrote:
>>> >>> > Which audio? Early media or after 200 OK?
>>> >>> >
>>> >>> > On Friday 15 January 2010, David Villasmil wrote:
>>> >>> >> Hello again Anthony,
>>> >>> >>
>>> >>> >> I just tested it, and although functionality does not, first
>>> >>> >> incoming
>>> >>> >> audio is coming in all garbled... do you know why?
>>> >>> >>
>>> >>> >> David
>>> >>> >>
>>> >>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>>> >>> >>
>>> >>> >> <anthony.minessale at gmail.com> wrote:
>>> >>> >> > {bridge_early_media=true}
>>> >>> >> > in the dial string in place of ignore_early_media=true
>>> >>> >> >
>>> >>> >> > the first b leg in the list who sends 183 will become the
>>> >>> >> > ringback
>>> >>> >> > device for A leg it will hear the early media
>>> >>> >> > for that leg while the other legs still ring.? If some other leg
>>> >>> >> > answers the final call will still be bridged to the leg who
>>> >>> >> > answered.
>>> >>> >> >
>>> >>> >> >
>>> >>> >> > I would estimate it at $500 payable on the big paypal button on
>>> >>> >> > http://www.freeswitch.org
>>> >>> >> > but, I already added the patch to tree earlier today so I guess
>>> >>> >> > it's
>>> >>> >> > up to you to pay it or not.
>>> >>> >> >
>>> >>> >> >
>>> >>> >> >
>>> >>> >> >
>>> >>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
>>> >>> >> >
>>> >>> >> > <david.villasmil.work at gmail.com> wrote:
>>> >>> >> >> Anthony,
>>> >>> >> >>
>>> >>> >> >> I did take the "hint", don't worry. We will probably ask for a
>>> >>> >> >> bounty
>>> >>> >> >> but first we need to know:
>>> >>> >> >> 1.- whether this is possible
>>> >>> >> >> 2.- how long it would take
>>> >>> >> >> 3.- how will it exactly work
>>> >>> >> >> 4.- of course, what's the bounty (be gentle ;) )
>>> >>> >> >>
>>> >>> >> >> We would of course give this back to the community.
>>> >>> >> >>
>>> >>> >> >> in the meantime, bridge_early_media=true will discard the 2nd
>>> >>> >> >> B-leg
>>> >>> >> >> and conversely "false" it will keep on trying to connect and if
>>> >>> >> >> it
>>> >>> >> >> connects the other B-leg if will bridge to that one?
>>> >>> >> >>
>>> >>> >> >> Thanks
>>> >>> >> >>
>>> >>> >> >> David
>>> >>> >> >>
>>> >>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>>> >>> >> >>
>>> >>> >> >> <anthony.minessale at gmail.com> wrote:
>>> >>> >> >> > I added bridge_early_media=true to do the best I can do.
>>> >>> >> >> > This is the most I will do, especially for free, nobody can
>>> >>> >> >> > take a
>>> >>> >> >> > hint that
>>> >>> >> >> > you should be paying for all these custom requests so take it
>>> >>> >> >> > or
>>> >>> >> >> > leave it
>>> >>> >> >> > but this thread is done.........
>>> >>> >> >> >
>>> >>> >> >> >
>>> >>> >> >> >
>>> >>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>>> >>> >> >> > <sos at sokhapkin.dyndns.org>
>>> >>> >> >> >
>>> >>> >> >> > wrote:
>>> >>> >> >> >> No, not exactly. ignore_early_media doesn't pass early media
>>> >>> >> >> >> to
>>> >>> >> >> >> the caller
>>> >>> >> >> >> if
>>> >>> >> >> >> bypass_media is false.
>>> >>> >> >> >>
>>> >>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
>>> >>> >> >> >> > this is exactly what ignore_early_media does now.
>>> >>> >> >> >> >
>>> >>> >> >> >> > Mike
>>> >>> >> >> >> >
>>> >>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>>> >>> >> >> >> > > The issue here is when "originate" routine should return
>>> >>> >> >> >> > > and
>>> >>> >> >> >> > > set "originate_status" variable. Current behavior is to
>>> >>> >> >> >> > > return
>>> >>> >> >> >> > > on early
>>> >>> >> >> >> > > media, but what if to introduce a variable
>>> >>> >> >> >> > > "originate_wait_for_answer"
>>> >>> >> >> >> > > with default value "false" and use the variable in
>>> >>> >> >> >> > > originate
>>> >>> >> >> >> > > code to
>>> >>> >> >> >> > > decide when to return - on 18X or "200 OK"?
>>> >>> >> >> >> > >
>>> >>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>>> >>> >> >> >> > >> he wants to call 3 people at once and let the A leg
>>> >>> >> >> >> > >> hear
>>> >>> >> >> >> > >> early media
>>> >>> >> >> >> > >> from call #1 while call #2 and #3 still are progressing
>>> >>> >> >> >> > >> which
>>> >>> >> >> >> > >> is not
>>> >>> >> >> >> > >> simple to do without doing thousands of dollars in
>>> >>> >> >> >> > >> development.
>>> >>> >> >> >> > >>
>>> >>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB
>>> >>> >> >> >> > >> <djbinter at yahoo.com>
>>> >> wrote:
>>> >>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so
>>> >>> >> >> >> > >>> that your
>>> >>> >> >> >> > >>> customers
>>> >>> >> >> >> > >>> can hear recordings?
>>> >>> >> >> >> >
>>> >>> >> >> >> > _______________________________________________
>>> >>> >> >> >> > FreeSWITCH-users mailing list
>>> >>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >>> >> >> >> >
>>> >>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >> >> >> >
>>> >>> >> >> >> >
>>> >>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswit
>>> >>> >> >> >> >ch- users http://www.freeswitch.org
>>> >>> >> >> >>
>>> >>> >> >> >> _______________________________________________
>>> >>> >> >> >> FreeSWITCH-users mailing list
>>> >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >>> >> >> >>
>>> >>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >> >> >>
>>> >>> >> >> >>
>>> >>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch
>>> >>> >> >> >>-us ers http://www.freeswitch.org
>>> >>> >> >> >
>>> >>> >> >> > --
>>> >>> >> >> > Anthony Minessale II
>>> >>> >> >> >
>>> >>> >> >> > FreeSWITCH http://www.freeswitch.org/
>>> >>> >> >> > ClueCon http://www.cluecon.com/
>>> >>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >>> >> >> >
>>> >>> >> >> > AIM: anthm
>>> >>> >> >> > MSN:anthony_minessale at hotmail.com
>>> >>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> >>> >> >> > IRC: irc.freenode.net #freeswitch
>>> >>> >> >> >
>>> >>> >> >> > FreeSWITCH Developer Conference
>>> >>> >> >> > sip:888 at conference.freeswitch.org
>>> >>> >> >> > iax:guest at conference.freeswitch.org/888
>>> >>> >> >> > googletalk:conf+888 at conference.freeswitch.org
>>> >>> >> >> > pstn:+19193869900
>>> >>> >> >> >
>>> >>> >> >> > _______________________________________________
>>> >>> >> >> > FreeSWITCH-users mailing list
>>> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >> >> >
>>> >>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> >>> >> >> >use rs http://www.freeswitch.org
>>> >>> >> >>
>>> >>> >> >> _______________________________________________
>>> >>> >> >> FreeSWITCH-users mailing list
>>> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >> >>
>>> >>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>>> >>> >> >>ers http://www.freeswitch.org
>>> >>> >> >
>>> >>> >> > --
>>> >>> >> > Anthony Minessale II
>>> >>> >> >
>>> >>> >> > FreeSWITCH http://www.freeswitch.org/
>>> >>> >> > ClueCon http://www.cluecon.com/
>>> >>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >>> >> >
>>> >>> >> > AIM: anthm
>>> >>> >> > MSN:anthony_minessale at hotmail.com
>>> >>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> >>> >> > IRC: irc.freenode.net #freeswitch
>>> >>> >> >
>>> >>> >> > FreeSWITCH Developer Conference
>>> >>> >> > sip:888 at conference.freeswitch.org
>>> >>> >> > iax:guest at conference.freeswitch.org/888
>>> >>> >> > googletalk:conf+888 at conference.freeswitch.org
>>> >>> >> > pstn:+19193869900
>>> >>> >> >
>>> >>> >> > _______________________________________________
>>> >>> >> > FreeSWITCH-users mailing list
>>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >> >
>>> >>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>> >>> >> >rs http://www.freeswitch.org
>>> >>> >>
>>> >>> >> _______________________________________________
>>> >>> >> FreeSWITCH-users mailing list
>>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >>
>>> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> >> http://www.freeswitch.org
>>> >>> >
>>> >>> > _______________________________________________
>>> >>> > FreeSWITCH-users mailing list
>>> >>> > FreeSWITCH-users at lists.freeswitch.org
>>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >
>>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> > http://www.freeswitch.org
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>
>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From lawwton at gmail.com  Fri Jan 15 14:54:42 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 17:54:42 -0500
Subject: [Freeswitch-users] Issue Compiling ESL (PHP)
In-Reply-To: <5fe6fa8f1001151437x491dade6q8b316f312556ff07@mail.gmail.com>
References: <5fe6fa8f1001151437x491dade6q8b316f312556ff07@mail.gmail.com>
Message-ID: <5fe6fa8f1001151454m92d17c9le4b811c897ae8482@mail.gmail.com>

Looking over at this, perhaps it even worked.

I see now under libs/esl/php/... the following two new files:

esl_wrap.o
ESL.o

Is there a way to verify that FS has support after running make and
make phpmod for php?

Thanks in advance,

Alfredo

On Fri, Jan 15, 2010 at 5:37 PM, Alfredo Quiroga-Villamil
<lawwton at gmail.com> wrote:
> All:
>
> System:
>
> Linux usnc-rtp-01-fs 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT
> 2007 i686 i686 i386 GNU/Linux
>
> I am trying to compile ESL, following the following steps:
>
> 1- cd to my libs/esl directory as the wiki page indicates.
> 2- run make
>
> I then get right away the following:
>
> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o
> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o
> src/esl_event.o
> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o
> src/esl_threadmutex.o
> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o
> src/esl_config.o
> g++ ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
> -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o
> ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o
> src/esl_config.o src/esl_oop.o
> ranlib libesl.a
> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L.
> -L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit
> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient
> -L. -lncurses -lpthread -lesl
> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver
> -L. -lncurses -lpthread -lesl
> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes ivrd.c -o ivrd -L. -lncurses
> -lpthread -lesl
>
> 3- I try typing then:
>
> make phpmod and get the following:
>
> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
> CFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> -Wstrict-prototypes -Wmissing-prototypes"
> CXXFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
> -Wno-unused-variable" CXX_CFLAGS="" -C php
> make[1]: Entering directory
> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
> g++ ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
> -Wno-unused-variable -I/usr/include/php -I/usr/include/php/main
> -I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext
> -I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function
> -c esl_wrap.cpp -o esl_wrap.o
> g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/kerberos/lib
> -lcrypt -lcrypt -lncurses -laspell -lpspell -lcurl -lresolv -lm -ldl
> -lnsl -lm -ldl -lcurl -ldl -lm -lcrypt -lm -lm -lcrypt -lpthread -o
> ESL.so -L.
> make[1]: Leaving directory
> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>
> I've installed I think all the -dev dependencies listed in the wiki. Any ideas?
>
> Thanks in advance,
>
> Alfredo Q-V
>



From anthony.minessale at gmail.com  Fri Jan 15 14:54:25 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 16:54:25 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001151445s6a4dff0qf8c9e17c702b7f86@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
	<201001150726.17430.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>
	<9853f4ff1001150443t5a18cc6bxa5032f391ab6f0ed@mail.gmail.com>
	<191c3a031001150808i69b3937boe0e903057f984e96@mail.gmail.com>
	<191c3a031001151317j408794a3pa2a0a5d21d63aa62@mail.gmail.com>
	<9853f4ff1001151445s6a4dff0qf8c9e17c702b7f86@mail.gmail.com>
Message-ID: <191c3a031001151454w101ae8eh978e4ced42004fb3@mail.gmail.com>

you can email me privately at this addr.


On Fri, Jan 15, 2010 at 4:45 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Anthony,
>
>     Trying, Thanks. Is there anyway we can communicate directly?
>
>
> David
>
> On Fri, Jan 15, 2010 at 10:17 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > Try latest trunk,
> >
> > you should have exactly what you want with the same parameter, again my
> > paypal addr is cleary displayed as a big button on the website.
> >
> >
> > On Fri, Jan 15, 2010 at 10:08 AM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> one of the many reasons its a bad idea.
> >> Probably the leg with the bad audio is a different ptime.
> >> Now the amount of work I have to do escalates I would prefer you commit
> to
> >> commercial support by emailing me at consulting at freeswitch.org to
> continue
> >> with this.
> >>
> >>
> >> On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil
> >> <david.villasmil.work at gmail.com> wrote:
> >>>
> >>> I set it to "off" just in case, same thing.
> >>>
> >>> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
> >>> <david.villasmil.work at gmail.com> wrote:
> >>> > Default, haven't touched it i suppose it's off, i haven't set it
> >>> > anywhere
> >>> >
> >>> > On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
> >>> > <sos at sokhapkin.dyndns.org> wrote:
> >>> >> Is bypass_media on or off?
> >>> >>
> >>> >> On Friday 15 January 2010, David Villasmil wrote:
> >>> >>> Yeah, sorry. Early media. Audio after 200 is fine. Early media was
> ok
> >>> >>> before the change.
> >>> >>>
> >>> >>> David
> >>> >>>
> >>> >>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
> >>> >>>
> >>> >>> <sos at sokhapkin.dyndns.org> wrote:
> >>> >>> > Which audio? Early media or after 200 OK?
> >>> >>> >
> >>> >>> > On Friday 15 January 2010, David Villasmil wrote:
> >>> >>> >> Hello again Anthony,
> >>> >>> >>
> >>> >>> >> I just tested it, and although functionality does not, first
> >>> >>> >> incoming
> >>> >>> >> audio is coming in all garbled... do you know why?
> >>> >>> >>
> >>> >>> >> David
> >>> >>> >>
> >>> >>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
> >>> >>> >>
> >>> >>> >> <anthony.minessale at gmail.com> wrote:
> >>> >>> >> > {bridge_early_media=true}
> >>> >>> >> > in the dial string in place of ignore_early_media=true
> >>> >>> >> >
> >>> >>> >> > the first b leg in the list who sends 183 will become the
> >>> >>> >> > ringback
> >>> >>> >> > device for A leg it will hear the early media
> >>> >>> >> > for that leg while the other legs still ring.  If some other
> leg
> >>> >>> >> > answers the final call will still be bridged to the leg who
> >>> >>> >> > answered.
> >>> >>> >> >
> >>> >>> >> >
> >>> >>> >> > I would estimate it at $500 payable on the big paypal button
> on
> >>> >>> >> > http://www.freeswitch.org
> >>> >>> >> > but, I already added the patch to tree earlier today so I
> guess
> >>> >>> >> > it's
> >>> >>> >> > up to you to pay it or not.
> >>> >>> >> >
> >>> >>> >> >
> >>> >>> >> >
> >>> >>> >> >
> >>> >>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> >>> >>> >> >
> >>> >>> >> > <david.villasmil.work at gmail.com> wrote:
> >>> >>> >> >> Anthony,
> >>> >>> >> >>
> >>> >>> >> >> I did take the "hint", don't worry. We will probably ask for
> a
> >>> >>> >> >> bounty
> >>> >>> >> >> but first we need to know:
> >>> >>> >> >> 1.- whether this is possible
> >>> >>> >> >> 2.- how long it would take
> >>> >>> >> >> 3.- how will it exactly work
> >>> >>> >> >> 4.- of course, what's the bounty (be gentle ;) )
> >>> >>> >> >>
> >>> >>> >> >> We would of course give this back to the community.
> >>> >>> >> >>
> >>> >>> >> >> in the meantime, bridge_early_media=true will discard the 2nd
> >>> >>> >> >> B-leg
> >>> >>> >> >> and conversely "false" it will keep on trying to connect and
> if
> >>> >>> >> >> it
> >>> >>> >> >> connects the other B-leg if will bridge to that one?
> >>> >>> >> >>
> >>> >>> >> >> Thanks
> >>> >>> >> >>
> >>> >>> >> >> David
> >>> >>> >> >>
> >>> >>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> >>> >>> >> >>
> >>> >>> >> >> <anthony.minessale at gmail.com> wrote:
> >>> >>> >> >> > I added bridge_early_media=true to do the best I can do.
> >>> >>> >> >> > This is the most I will do, especially for free, nobody can
> >>> >>> >> >> > take a
> >>> >>> >> >> > hint that
> >>> >>> >> >> > you should be paying for all these custom requests so take
> it
> >>> >>> >> >> > or
> >>> >>> >> >> > leave it
> >>> >>> >> >> > but this thread is done.........
> >>> >>> >> >> >
> >>> >>> >> >> >
> >>> >>> >> >> >
> >>> >>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
> >>> >>> >> >> > <sos at sokhapkin.dyndns.org>
> >>> >>> >> >> >
> >>> >>> >> >> > wrote:
> >>> >>> >> >> >> No, not exactly. ignore_early_media doesn't pass early
> media
> >>> >>> >> >> >> to
> >>> >>> >> >> >> the caller
> >>> >>> >> >> >> if
> >>> >>> >> >> >> bypass_media is false.
> >>> >>> >> >> >>
> >>> >>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
> >>> >>> >> >> >> > this is exactly what ignore_early_media does now.
> >>> >>> >> >> >> >
> >>> >>> >> >> >> > Mike
> >>> >>> >> >> >> >
> >>> >>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> >>> >>> >> >> >> > > The issue here is when "originate" routine should
> return
> >>> >>> >> >> >> > > and
> >>> >>> >> >> >> > > set "originate_status" variable. Current behavior is
> to
> >>> >>> >> >> >> > > return
> >>> >>> >> >> >> > > on early
> >>> >>> >> >> >> > > media, but what if to introduce a variable
> >>> >>> >> >> >> > > "originate_wait_for_answer"
> >>> >>> >> >> >> > > with default value "false" and use the variable in
> >>> >>> >> >> >> > > originate
> >>> >>> >> >> >> > > code to
> >>> >>> >> >> >> > > decide when to return - on 18X or "200 OK"?
> >>> >>> >> >> >> > >
> >>> >>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
> >>> >>> >> >> >> > >> he wants to call 3 people at once and let the A leg
> >>> >>> >> >> >> > >> hear
> >>> >>> >> >> >> > >> early media
> >>> >>> >> >> >> > >> from call #1 while call #2 and #3 still are
> progressing
> >>> >>> >> >> >> > >> which
> >>> >>> >> >> >> > >> is not
> >>> >>> >> >> >> > >> simple to do without doing thousands of dollars in
> >>> >>> >> >> >> > >> development.
> >>> >>> >> >> >> > >>
> >>> >>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB
> >>> >>> >> >> >> > >> <djbinter at yahoo.com>
> >>> >> wrote:
> >>> >>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so
> >>> >>> >> >> >> > >>> that your
> >>> >>> >> >> >> > >>> customers
> >>> >>> >> >> >> > >>> can hear recordings?
> >>> >>> >> >> >> >
> >>> >>> >> >> >> > _______________________________________________
> >>> >>> >> >> >> > FreeSWITCH-users mailing list
> >>> >>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >>> >> >> >> >
> >>> >>> >> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>> >> >> >> >
> >>> >>> >> >> >> >
> >>> >>> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswit
> >>> >>> >> >> >> >ch- users http://www.freeswitch.org
> >>> >>> >> >> >>
> >>> >>> >> >> >> _______________________________________________
> >>> >>> >> >> >> FreeSWITCH-users mailing list
> >>> >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >>> >> >> >>
> >>> >>> >> >> >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>> >> >> >>
> >>> >>> >> >> >>
> >>> >>> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch
> >>> >>> >> >> >>-us ers http://www.freeswitch.org
> >>> >>> >> >> >
> >>> >>> >> >> > --
> >>> >>> >> >> > Anthony Minessale II
> >>> >>> >> >> >
> >>> >>> >> >> > FreeSWITCH http://www.freeswitch.org/
> >>> >>> >> >> > ClueCon http://www.cluecon.com/
> >>> >>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>> >>> >> >> >
> >>> >>> >> >> > AIM: anthm
> >>> >>> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> >>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> >>> >> >> > IRC: irc.freenode.net #freeswitch
> >>> >>> >> >> >
> >>> >>> >> >> > FreeSWITCH Developer Conference
> >>> >>> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> >>> >> >> > iax:guest at conference.freeswitch.org/888
> >>> >>> >> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> >>> >> >> > pstn:+19193869900
> >>> >>> >> >> >
> >>> >>> >> >> > _______________________________________________
> >>> >>> >> >> > FreeSWITCH-users mailing list
> >>> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >>> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>> >> >> >
> >>> >>> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-
> >>> >>> >> >> >use rs http://www.freeswitch.org
> >>> >>> >> >>
> >>> >>> >> >> _______________________________________________
> >>> >>> >> >> FreeSWITCH-users mailing list
> >>> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >>> >> >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>> >> >>
> >>> >>> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-us
> >>> >>> >> >>ers http://www.freeswitch.org
> >>> >>> >> >
> >>> >>> >> > --
> >>> >>> >> > Anthony Minessale II
> >>> >>> >> >
> >>> >>> >> > FreeSWITCH http://www.freeswitch.org/
> >>> >>> >> > ClueCon http://www.cluecon.com/
> >>> >>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>> >>> >> >
> >>> >>> >> > AIM: anthm
> >>> >>> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> >>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> >>> >> > IRC: irc.freenode.net #freeswitch
> >>> >>> >> >
> >>> >>> >> > FreeSWITCH Developer Conference
> >>> >>> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> >>> >> > iax:guest at conference.freeswitch.org/888
> >>> >>> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> >>> >> > pstn:+19193869900
> >>> >>> >> >
> >>> >>> >> > _______________________________________________
> >>> >>> >> > FreeSWITCH-users mailing list
> >>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>> >> >
> >>> >>> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-use
> >>> >>> >> >rs http://www.freeswitch.org
> >>> >>> >>
> >>> >>> >> _______________________________________________
> >>> >>> >> FreeSWITCH-users mailing list
> >>> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>> >>
> >>> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >>> >> http://www.freeswitch.org
> >>> >>> >
> >>> >>> > _______________________________________________
> >>> >>> > FreeSWITCH-users mailing list
> >>> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>> >
> >>> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >>> > http://www.freeswitch.org
> >>> >>>
> >>> >>> _______________________________________________
> >>> >>> FreeSWITCH-users mailing list
> >>> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>>
> >>> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >>> http://www.freeswitch.org
> >>> >>
> >>> >>
> >>> >>
> >>> >> _______________________________________________
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>
> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >>
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From robert.hadley at teotech.com  Fri Jan 15 15:01:04 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 15 Jan 2010 15:01:04 -0800
Subject: [Freeswitch-users] xset warning message starting FS as daemon
In-Reply-To: <E1A9E241-30F9-4470-BEAC-869E265341D6@jerris.com>
References: <10273ADF13B842FE8527E8FF93BD6C28@greyhawk.tonecommander.com>
	<E1A9E241-30F9-4470-BEAC-869E265341D6@jerris.com>
Message-ID: <96752801704D4597954A9ED5E71E2399@greyhawk.tonecommander.com>

Hi Mike,

 

I took the one from FS build/freeswitch.init.redhat and modified it for my
paths in the /opt folder and set the FS-user to root for now (was going to
change user to freeswitch next).

 

Thanks,

Robert

 

  _____  

From: Michael Jerris [mailto:mike at jerris.com] 
Sent: Friday, January 15, 2010 1:24 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] xset warning message starting FS as daemon

 

Its something in the init script.  are you using one from the freeswitch
tree or your own?

 

Mike

 

On Jan 15, 2010, at 4:07 PM, Robert Hadley wrote:





Hi,

 

Running trunk on CentOS 5.3, I don't get any warning messages starting FS
manually with the -nc option.

 

I get this message when I start FS as a daemon.

 

[root at roberth-c53 fstrkbld]# service freeswitch start

Starting freeswitch: xset:  unable to open display ""

xset:  unable to open display ""

                                                           [  OK  ]

 

 

Is this due to something I am doing or is it FS?

 

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From anthony.minessale at gmail.com  Fri Jan 15 15:05:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 17:05:14 -0600
Subject: [Freeswitch-users] Issue Compiling ESL (PHP)
In-Reply-To: <5fe6fa8f1001151454m92d17c9le4b811c897ae8482@mail.gmail.com>
References: <5fe6fa8f1001151437x491dade6q8b316f312556ff07@mail.gmail.com>
	<5fe6fa8f1001151454m92d17c9le4b811c897ae8482@mail.gmail.com>
Message-ID: <191c3a031001151505y28e8e759sf6f3239f61d4b6b7@mail.gmail.com>

i think there is a .php and a .so you could install into your php lib dir?


On Fri, Jan 15, 2010 at 4:54 PM, Alfredo Quiroga-Villamil <lawwton at gmail.com
> wrote:

> Looking over at this, perhaps it even worked.
>
> I see now under libs/esl/php/... the following two new files:
>
> esl_wrap.o
> ESL.o
>
> Is there a way to verify that FS has support after running make and
> make phpmod for php?
>
> Thanks in advance,
>
> Alfredo
>
> On Fri, Jan 15, 2010 at 5:37 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
> > All:
> >
> > System:
> >
> > Linux usnc-rtp-01-fs 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT
> > 2007 i686 i686 i386 GNU/Linux
> >
> > I am trying to compile ESL, following the following steps:
> >
> > 1- cd to my libs/esl directory as the wiki page indicates.
> > 2- run make
> >
> > I then get right away the following:
> >
> > cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o
> > cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o
> > src/esl_event.o
> > cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o
> > src/esl_threadmutex.o
> > cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o
> > src/esl_config.o
> > g++  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
> > -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o
> > ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o
> > src/esl_config.o src/esl_oop.o
> > ranlib libesl.a
> > cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L.
> > -L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit
> > cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient
> > -L. -lncurses -lpthread -lesl
> > cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver
> > -L. -lncurses -lpthread -lesl
> > cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes ivrd.c -o ivrd -L. -lncurses
> > -lpthread -lesl
> >
> > 3- I try typing then:
> >
> > make phpmod and get the following:
> >
> > make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
> > CFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
> > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
> > -Wstrict-prototypes -Wmissing-prototypes"
> > CXXFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
> > -Wno-unused-variable" CXX_CFLAGS="" -C php
> > make[1]: Entering directory
> > `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
> > g++  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
> > -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
> > -Wno-unused-variable -I/usr/include/php -I/usr/include/php/main
> > -I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext
> > -I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function
> > -c esl_wrap.cpp -o esl_wrap.o
> > g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/kerberos/lib
> > -lcrypt -lcrypt -lncurses -laspell -lpspell -lcurl -lresolv -lm -ldl
> > -lnsl -lm -ldl -lcurl -ldl -lm -lcrypt -lm -lm -lcrypt -lpthread -o
> > ESL.so -L.
> > make[1]: Leaving directory
> > `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
> >
> > I've installed I think all the -dev dependencies listed in the wiki. Any
> ideas?
> >
> > Thanks in advance,
> >
> > Alfredo Q-V
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From david.villasmil.work at gmail.com  Fri Jan 15 15:07:55 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 16 Jan 2010 00:07:55 +0100
Subject: [Freeswitch-users] Issue Compiling ESL (PHP)
In-Reply-To: <5fe6fa8f1001151454m92d17c9le4b811c897ae8482@mail.gmail.com>
References: <5fe6fa8f1001151437x491dade6q8b316f312556ff07@mail.gmail.com>
	<5fe6fa8f1001151454m92d17c9le4b811c897ae8482@mail.gmail.com>
Message-ID: <9853f4ff1001151507oa1fa8adt5101956739d8c948@mail.gmail.com>

make a script ;)

it's very easy, try it!

David

On Fri, Jan 15, 2010 at 11:54 PM, Alfredo Quiroga-Villamil
<lawwton at gmail.com> wrote:
> Looking over at this, perhaps it even worked.
>
> I see now under libs/esl/php/... the following two new files:
>
> esl_wrap.o
> ESL.o
>
> Is there a way to verify that FS has support after running make and
> make phpmod for php?
>
> Thanks in advance,
>
> Alfredo
>
> On Fri, Jan 15, 2010 at 5:37 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>> All:
>>
>> System:
>>
>> Linux usnc-rtp-01-fs 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT
>> 2007 i686 i686 i386 GNU/Linux
>>
>> I am trying to compile ESL, following the following steps:
>>
>> 1- cd to my libs/esl directory as the wiki page indicates.
>> 2- run make
>>
>> I then get right away the following:
>>
>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o
>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o
>> src/esl_event.o
>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o
>> src/esl_threadmutex.o
>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o
>> src/esl_config.o
>> g++ ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>> -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o
>> ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o
>> src/esl_config.o src/esl_oop.o
>> ranlib libesl.a
>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L.
>> -L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit
>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient
>> -L. -lncurses -lpthread -lesl
>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver
>> -L. -lncurses -lpthread -lesl
>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes ivrd.c -o ivrd -L. -lncurses
>> -lpthread -lesl
>>
>> 3- I try typing then:
>>
>> make phpmod and get the following:
>>
>> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
>> CFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>> -Wstrict-prototypes -Wmissing-prototypes"
>> CXXFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>> -Wno-unused-variable" CXX_CFLAGS="" -C php
>> make[1]: Entering directory
>> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>> g++ ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>> -Wno-unused-variable -I/usr/include/php -I/usr/include/php/main
>> -I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext
>> -I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function
>> -c esl_wrap.cpp -o esl_wrap.o
>> g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/kerberos/lib
>> -lcrypt -lcrypt -lncurses -laspell -lpspell -lcurl -lresolv -lm -ldl
>> -lnsl -lm -ldl -lcurl -ldl -lm -lcrypt -lm -lm -lcrypt -lpthread -o
>> ESL.so -L.
>> make[1]: Leaving directory
>> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>>
>> I've installed I think all the -dev dependencies listed in the wiki. Any ideas?
>>
>> Thanks in advance,
>>
>> Alfredo Q-V
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From lawwton at gmail.com  Fri Jan 15 15:21:22 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 18:21:22 -0500
Subject: [Freeswitch-users] Issue Compiling ESL (PHP)
In-Reply-To: <9853f4ff1001151507oa1fa8adt5101956739d8c948@mail.gmail.com>
References: <5fe6fa8f1001151437x491dade6q8b316f312556ff07@mail.gmail.com>
	<5fe6fa8f1001151454m92d17c9le4b811c897ae8482@mail.gmail.com>
	<9853f4ff1001151507oa1fa8adt5101956739d8c948@mail.gmail.com>
Message-ID: <5fe6fa8f1001151521u3ef71f9g49491118bf0371ac@mail.gmail.com>

Thanks Anthony/David.

So it seems like the build worked then. I take from the previous
emails and somewhere where I think I read that I can then take:

ESL.so and ESL.php and put them on a remote system under my say for
instance 3rdParty directory and create scripts using the ESL.php
library which probably internally uses ESL.so.

Is that statement correct?

Thanks in advance,

Alfredo

On Fri, Jan 15, 2010 at 6:07 PM, David Villasmil
<david.villasmil.work at gmail.com> wrote:
> make a script ;)
>
> it's very easy, try it!
>
> David
>
> On Fri, Jan 15, 2010 at 11:54 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>> Looking over at this, perhaps it even worked.
>>
>> I see now under libs/esl/php/... the following two new files:
>>
>> esl_wrap.o
>> ESL.o
>>
>> Is there a way to verify that FS has support after running make and
>> make phpmod for php?
>>
>> Thanks in advance,
>>
>> Alfredo
>>
>> On Fri, Jan 15, 2010 at 5:37 PM, Alfredo Quiroga-Villamil
>> <lawwton at gmail.com> wrote:
>>> All:
>>>
>>> System:
>>>
>>> Linux usnc-rtp-01-fs 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT
>>> 2007 i686 i686 i386 GNU/Linux
>>>
>>> I am trying to compile ESL, following the following steps:
>>>
>>> 1- cd to my libs/esl directory as the wiki page indicates.
>>> 2- run make
>>>
>>> I then get right away the following:
>>>
>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o
>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o
>>> src/esl_event.o
>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o
>>> src/esl_threadmutex.o
>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o
>>> src/esl_config.o
>>> g++ ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>> -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o
>>> ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o
>>> src/esl_config.o src/esl_oop.o
>>> ranlib libesl.a
>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L.
>>> -L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit
>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient
>>> -L. -lncurses -lpthread -lesl
>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver
>>> -L. -lncurses -lpthread -lesl
>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes ivrd.c -o ivrd -L. -lncurses
>>> -lpthread -lesl
>>>
>>> 3- I try typing then:
>>>
>>> make phpmod and get the following:
>>>
>>> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
>>> CFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>> -Wstrict-prototypes -Wmissing-prototypes"
>>> CXXFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>> -Wno-unused-variable" CXX_CFLAGS="" -C php
>>> make[1]: Entering directory
>>> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>>> g++ ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>> -Wno-unused-variable -I/usr/include/php -I/usr/include/php/main
>>> -I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext
>>> -I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function
>>> -c esl_wrap.cpp -o esl_wrap.o
>>> g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/kerberos/lib
>>> -lcrypt -lcrypt -lncurses -laspell -lpspell -lcurl -lresolv -lm -ldl
>>> -lnsl -lm -ldl -lcurl -ldl -lm -lcrypt -lm -lm -lcrypt -lpthread -o
>>> ESL.so -L.
>>> make[1]: Leaving directory
>>> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>>>
>>> I've installed I think all the -dev dependencies listed in the wiki. Any ideas?
>>>
>>> Thanks in advance,
>>>
>>> Alfredo Q-V
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From david.villasmil.work at gmail.com  Fri Jan 15 15:25:10 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 16 Jan 2010 00:25:10 +0100
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001151454w101ae8eh978e4ced42004fb3@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<201001150651.05983.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
	<201001150726.17430.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>
	<9853f4ff1001150443t5a18cc6bxa5032f391ab6f0ed@mail.gmail.com>
	<191c3a031001150808i69b3937boe0e903057f984e96@mail.gmail.com>
	<191c3a031001151317j408794a3pa2a0a5d21d63aa62@mail.gmail.com>
	<9853f4ff1001151445s6a4dff0qf8c9e17c702b7f86@mail.gmail.com>
	<191c3a031001151454w101ae8eh978e4ced42004fb3@mail.gmail.com>
Message-ID: <9853f4ff1001151525o341cf2c7k2935538cca210815@mail.gmail.com>

Anthony,

LOL, and mounting and mounting... It does work when there is answer...
but if B(2)-side rejects or times out or any other that 200 OK,
B(1)-side stays indefinitely...


On Fri, Jan 15, 2010 at 11:54 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> you can email me privately at this addr.
>
>
> On Fri, Jan 15, 2010 at 4:45 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> Anthony,
>>
>> ? ? Trying, Thanks. Is there anyway we can communicate directly?
>>
>>
>> David
>>
>> On Fri, Jan 15, 2010 at 10:17 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > Try latest trunk,
>> >
>> > you should have exactly what you want with the same parameter, again my
>> > paypal addr is cleary displayed as a big button on the website.
>> >
>> >
>> > On Fri, Jan 15, 2010 at 10:08 AM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> one of the many reasons its a bad idea.
>> >> Probably the leg with the bad audio is a different ptime.
>> >> Now the amount of work I have to do escalates I would prefer you commit
>> >> to
>> >> commercial support by emailing me at consulting at freeswitch.org to
>> >> continue
>> >> with this.
>> >>
>> >>
>> >> On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil
>> >> <david.villasmil.work at gmail.com> wrote:
>> >>>
>> >>> I set it to "off" just in case, same thing.
>> >>>
>> >>> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
>> >>> <david.villasmil.work at gmail.com> wrote:
>> >>> > Default, haven't touched it i suppose it's off, i haven't set it
>> >>> > anywhere
>> >>> >
>> >>> > On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
>> >>> > <sos at sokhapkin.dyndns.org> wrote:
>> >>> >> Is bypass_media on or off?
>> >>> >>
>> >>> >> On Friday 15 January 2010, David Villasmil wrote:
>> >>> >>> Yeah, sorry. Early media. Audio after 200 is fine. Early media was
>> >>> >>> ok
>> >>> >>> before the change.
>> >>> >>>
>> >>> >>> David
>> >>> >>>
>> >>> >>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
>> >>> >>>
>> >>> >>> <sos at sokhapkin.dyndns.org> wrote:
>> >>> >>> > Which audio? Early media or after 200 OK?
>> >>> >>> >
>> >>> >>> > On Friday 15 January 2010, David Villasmil wrote:
>> >>> >>> >> Hello again Anthony,
>> >>> >>> >>
>> >>> >>> >> I just tested it, and although functionality does not, first
>> >>> >>> >> incoming
>> >>> >>> >> audio is coming in all garbled... do you know why?
>> >>> >>> >>
>> >>> >>> >> David
>> >>> >>> >>
>> >>> >>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>> >>> >>> >>
>> >>> >>> >> <anthony.minessale at gmail.com> wrote:
>> >>> >>> >> > {bridge_early_media=true}
>> >>> >>> >> > in the dial string in place of ignore_early_media=true
>> >>> >>> >> >
>> >>> >>> >> > the first b leg in the list who sends 183 will become the
>> >>> >>> >> > ringback
>> >>> >>> >> > device for A leg it will hear the early media
>> >>> >>> >> > for that leg while the other legs still ring.? If some other
>> >>> >>> >> > leg
>> >>> >>> >> > answers the final call will still be bridged to the leg who
>> >>> >>> >> > answered.
>> >>> >>> >> >
>> >>> >>> >> >
>> >>> >>> >> > I would estimate it at $500 payable on the big paypal button
>> >>> >>> >> > on
>> >>> >>> >> > http://www.freeswitch.org
>> >>> >>> >> > but, I already added the patch to tree earlier today so I
>> >>> >>> >> > guess
>> >>> >>> >> > it's
>> >>> >>> >> > up to you to pay it or not.
>> >>> >>> >> >
>> >>> >>> >> >
>> >>> >>> >> >
>> >>> >>> >> >
>> >>> >>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
>> >>> >>> >> >
>> >>> >>> >> > <david.villasmil.work at gmail.com> wrote:
>> >>> >>> >> >> Anthony,
>> >>> >>> >> >>
>> >>> >>> >> >> I did take the "hint", don't worry. We will probably ask for
>> >>> >>> >> >> a
>> >>> >>> >> >> bounty
>> >>> >>> >> >> but first we need to know:
>> >>> >>> >> >> 1.- whether this is possible
>> >>> >>> >> >> 2.- how long it would take
>> >>> >>> >> >> 3.- how will it exactly work
>> >>> >>> >> >> 4.- of course, what's the bounty (be gentle ;) )
>> >>> >>> >> >>
>> >>> >>> >> >> We would of course give this back to the community.
>> >>> >>> >> >>
>> >>> >>> >> >> in the meantime, bridge_early_media=true will discard the
>> >>> >>> >> >> 2nd
>> >>> >>> >> >> B-leg
>> >>> >>> >> >> and conversely "false" it will keep on trying to connect and
>> >>> >>> >> >> if
>> >>> >>> >> >> it
>> >>> >>> >> >> connects the other B-leg if will bridge to that one?
>> >>> >>> >> >>
>> >>> >>> >> >> Thanks
>> >>> >>> >> >>
>> >>> >>> >> >> David
>> >>> >>> >> >>
>> >>> >>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>> >>> >>> >> >>
>> >>> >>> >> >> <anthony.minessale at gmail.com> wrote:
>> >>> >>> >> >> > I added bridge_early_media=true to do the best I can do.
>> >>> >>> >> >> > This is the most I will do, especially for free, nobody
>> >>> >>> >> >> > can
>> >>> >>> >> >> > take a
>> >>> >>> >> >> > hint that
>> >>> >>> >> >> > you should be paying for all these custom requests so take
>> >>> >>> >> >> > it
>> >>> >>> >> >> > or
>> >>> >>> >> >> > leave it
>> >>> >>> >> >> > but this thread is done.........
>> >>> >>> >> >> >
>> >>> >>> >> >> >
>> >>> >>> >> >> >
>> >>> >>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>> >>> >>> >> >> > <sos at sokhapkin.dyndns.org>
>> >>> >>> >> >> >
>> >>> >>> >> >> > wrote:
>> >>> >>> >> >> >> No, not exactly. ignore_early_media doesn't pass early
>> >>> >>> >> >> >> media
>> >>> >>> >> >> >> to
>> >>> >>> >> >> >> the caller
>> >>> >>> >> >> >> if
>> >>> >>> >> >> >> bypass_media is false.
>> >>> >>> >> >> >>
>> >>> >>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
>> >>> >>> >> >> >> > this is exactly what ignore_early_media does now.
>> >>> >>> >> >> >> >
>> >>> >>> >> >> >> > Mike
>> >>> >>> >> >> >> >
>> >>> >>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
>> >>> >>> >> >> >> > > The issue here is when "originate" routine should
>> >>> >>> >> >> >> > > return
>> >>> >>> >> >> >> > > and
>> >>> >>> >> >> >> > > set "originate_status" variable. Current behavior is
>> >>> >>> >> >> >> > > to
>> >>> >>> >> >> >> > > return
>> >>> >>> >> >> >> > > on early
>> >>> >>> >> >> >> > > media, but what if to introduce a variable
>> >>> >>> >> >> >> > > "originate_wait_for_answer"
>> >>> >>> >> >> >> > > with default value "false" and use the variable in
>> >>> >>> >> >> >> > > originate
>> >>> >>> >> >> >> > > code to
>> >>> >>> >> >> >> > > decide when to return - on 18X or "200 OK"?
>> >>> >>> >> >> >> > >
>> >>> >>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale wrote:
>> >>> >>> >> >> >> > >> he wants to call 3 people at once and let the A leg
>> >>> >>> >> >> >> > >> hear
>> >>> >>> >> >> >> > >> early media
>> >>> >>> >> >> >> > >> from call #1 while call #2 and #3 still are
>> >>> >>> >> >> >> > >> progressing
>> >>> >>> >> >> >> > >> which
>> >>> >>> >> >> >> > >> is not
>> >>> >>> >> >> >> > >> simple to do without doing thousands of dollars in
>> >>> >>> >> >> >> > >> development.
>> >>> >>> >> >> >> > >>
>> >>> >>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB
>> >>> >>> >> >> >> > >> <djbinter at yahoo.com>
>> >>> >> wrote:
>> >>> >>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK), so
>> >>> >>> >> >> >> > >>> that your
>> >>> >>> >> >> >> > >>> customers
>> >>> >>> >> >> >> > >>> can hear recordings?
>> >>> >>> >> >> >> >
>> >>> >>> >> >> >> > _______________________________________________
>> >>> >>> >> >> >> > FreeSWITCH-users mailing list
>> >>> >>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> >> >> >> >
>> >>> >>> >> >> >> >
>> >>> >>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>> >> >> >> >
>> >>> >>> >> >> >> >
>> >>> >>> >> >> >> >
>> >>> >>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswit
>> >>> >>> >> >> >> >ch- users http://www.freeswitch.org
>> >>> >>> >> >> >>
>> >>> >>> >> >> >> _______________________________________________
>> >>> >>> >> >> >> FreeSWITCH-users mailing list
>> >>> >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> >> >> >>
>> >>> >>> >> >> >>
>> >>> >>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>> >> >> >>
>> >>> >>> >> >> >>
>> >>> >>> >> >> >>
>> >>> >>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch
>> >>> >>> >> >> >>-us ers http://www.freeswitch.org
>> >>> >>> >> >> >
>> >>> >>> >> >> > --
>> >>> >>> >> >> > Anthony Minessale II
>> >>> >>> >> >> >
>> >>> >>> >> >> > FreeSWITCH http://www.freeswitch.org/
>> >>> >>> >> >> > ClueCon http://www.cluecon.com/
>> >>> >>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>> >>> >> >> >
>> >>> >>> >> >> > AIM: anthm
>> >>> >>> >> >> > MSN:anthony_minessale at hotmail.com
>> >>> >>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>> >>> >> >> > IRC: irc.freenode.net #freeswitch
>> >>> >>> >> >> >
>> >>> >>> >> >> > FreeSWITCH Developer Conference
>> >>> >>> >> >> > sip:888 at conference.freeswitch.org
>> >>> >>> >> >> > iax:guest at conference.freeswitch.org/888
>> >>> >>> >> >> > googletalk:conf+888 at conference.freeswitch.org
>> >>> >>> >> >> > pstn:+19193869900
>> >>> >>> >> >> >
>> >>> >>> >> >> > _______________________________________________
>> >>> >>> >> >> > FreeSWITCH-users mailing list
>> >>> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> >> >> >
>> >>> >>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>> >> >> >
>> >>> >>> >> >> >
>> >>> >>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >>> >>> >> >> >use rs http://www.freeswitch.org
>> >>> >>> >> >>
>> >>> >>> >> >> _______________________________________________
>> >>> >>> >> >> FreeSWITCH-users mailing list
>> >>> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> >> >>
>> >>> >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>> >> >>
>> >>> >>> >> >>
>> >>> >>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> >>> >>> >> >>ers http://www.freeswitch.org
>> >>> >>> >> >
>> >>> >>> >> > --
>> >>> >>> >> > Anthony Minessale II
>> >>> >>> >> >
>> >>> >>> >> > FreeSWITCH http://www.freeswitch.org/
>> >>> >>> >> > ClueCon http://www.cluecon.com/
>> >>> >>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>> >>> >> >
>> >>> >>> >> > AIM: anthm
>> >>> >>> >> > MSN:anthony_minessale at hotmail.com
>> >>> >>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>> >>> >> > IRC: irc.freenode.net #freeswitch
>> >>> >>> >> >
>> >>> >>> >> > FreeSWITCH Developer Conference
>> >>> >>> >> > sip:888 at conference.freeswitch.org
>> >>> >>> >> > iax:guest at conference.freeswitch.org/888
>> >>> >>> >> > googletalk:conf+888 at conference.freeswitch.org
>> >>> >>> >> > pstn:+19193869900
>> >>> >>> >> >
>> >>> >>> >> > _______________________________________________
>> >>> >>> >> > FreeSWITCH-users mailing list
>> >>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>> >> >
>> >>> >>> >> >
>> >>> >>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> >>> >>> >> >rs http://www.freeswitch.org
>> >>> >>> >>
>> >>> >>> >> _______________________________________________
>> >>> >>> >> FreeSWITCH-users mailing list
>> >>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>> >>
>> >>> >>> >>
>> >>> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >>> >> http://www.freeswitch.org
>> >>> >>> >
>> >>> >>> > _______________________________________________
>> >>> >>> > FreeSWITCH-users mailing list
>> >>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>> >
>> >>> >>> >
>> >>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >>> > http://www.freeswitch.org
>> >>> >>>
>> >>> >>> _______________________________________________
>> >>> >>> FreeSWITCH-users mailing list
>> >>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>>
>> >>> >>>
>> >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >>> http://www.freeswitch.org
>> >>> >>
>> >>> >>
>> >>> >>
>> >>> >> _______________________________________________
>> >>> >> FreeSWITCH-users mailing list
>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>
>> >>> >>
>> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> http://www.freeswitch.org
>> >>> >>
>> >>> >
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> iax:guest at conference.freeswitch.org/888
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From mike at jerris.com  Fri Jan 15 15:35:49 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 Jan 2010 18:35:49 -0500
Subject: [Freeswitch-users] Issue Compiling ESL (PHP)
In-Reply-To: <5fe6fa8f1001151521u3ef71f9g49491118bf0371ac@mail.gmail.com>
References: <5fe6fa8f1001151437x491dade6q8b316f312556ff07@mail.gmail.com>
	<5fe6fa8f1001151454m92d17c9le4b811c897ae8482@mail.gmail.com>
	<9853f4ff1001151507oa1fa8adt5101956739d8c948@mail.gmail.com>
	<5fe6fa8f1001151521u3ef71f9g49491118bf0371ac@mail.gmail.com>
Message-ID: <F84AD964-A9DC-48D8-BA0D-22E97863CACE@jerris.com>

Sounds right to me.

Mike

On Jan 15, 2010, at 6:21 PM, Alfredo Quiroga-Villamil wrote:

> Thanks Anthony/David.
> 
> So it seems like the build worked then. I take from the previous
> emails and somewhere where I think I read that I can then take:
> 
> ESL.so and ESL.php and put them on a remote system under my say for
> instance 3rdParty directory and create scripts using the ESL.php
> library which probably internally uses ESL.so.
> 
> Is that statement correct?
> 
> Thanks in advance,
> 
> Alfredo
> 
> On Fri, Jan 15, 2010 at 6:07 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>> make a script ;)
>> 
>> it's very easy, try it!
>> 
>> David
>> 
>> On Fri, Jan 15, 2010 at 11:54 PM, Alfredo Quiroga-Villamil
>> <lawwton at gmail.com> wrote:
>>> Looking over at this, perhaps it even worked.
>>> 
>>> I see now under libs/esl/php/... the following two new files:
>>> 
>>> esl_wrap.o
>>> ESL.o
>>> 
>>> Is there a way to verify that FS has support after running make and
>>> make phpmod for php?
>>> 
>>> Thanks in advance,
>>> 
>>> Alfredo
>>> 
>>> On Fri, Jan 15, 2010 at 5:37 PM, Alfredo Quiroga-Villamil
>>> <lawwton at gmail.com> wrote:
>>>> All:
>>>> 
>>>> System:
>>>> 
>>>> Linux usnc-rtp-01-fs 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT
>>>> 2007 i686 i686 i386 GNU/Linux
>>>> 
>>>> I am trying to compile ESL, following the following steps:
>>>> 
>>>> 1- cd to my libs/esl directory as the wiki page indicates.
>>>> 2- run make
>>>> 
>>>> I then get right away the following:
>>>> 
>>>> cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o
>>>> cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o
>>>> src/esl_event.o
>>>> cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o
>>>> src/esl_threadmutex.o
>>>> cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o
>>>> src/esl_config.o
>>>> g++  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>>> -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o
>>>> ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o
>>>> src/esl_config.o src/esl_oop.o
>>>> ranlib libesl.a
>>>> cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L.
>>>> -L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit
>>>> cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient
>>>> -L. -lncurses -lpthread -lesl
>>>> cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver
>>>> -L. -lncurses -lpthread -lesl
>>>> cc  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes ivrd.c -o ivrd -L. -lncurses
>>>> -lpthread -lesl
>>>> 
>>>> 3- I try typing then:
>>>> 
>>>> make phpmod and get the following:
>>>> 
>>>> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
>>>> CFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>> -Wstrict-prototypes -Wmissing-prototypes"
>>>> CXXFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>>> -Wno-unused-variable" CXX_CFLAGS="" -C php
>>>> make[1]: Entering directory
>>>> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>>>> g++  -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>>> -Wno-unused-variable -I/usr/include/php -I/usr/include/php/main
>>>> -I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext
>>>> -I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function
>>>> -c esl_wrap.cpp -o esl_wrap.o
>>>> g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/kerberos/lib
>>>> -lcrypt -lcrypt -lncurses -laspell -lpspell -lcurl -lresolv -lm -ldl
>>>> -lnsl -lm -ldl -lcurl -ldl -lm -lcrypt -lm -lm -lcrypt -lpthread -o
>>>> ESL.so -L.
>>>> make[1]: Leaving directory
>>>> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>>>> 
>>>> I've installed I think all the -dev dependencies listed in the wiki. Any ideas?
>>>> 
>>>> Thanks in advance,
>>>> 
>>>> Alfredo Q-V
>>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From stevesteffler at shaw.ca  Fri Jan 15 15:43:12 2010
From: stevesteffler at shaw.ca (Steve Steffler)
Date: Fri, 15 Jan 2010 16:43:12 -0700
Subject: [Freeswitch-users] FIFO Originate caller ID
Message-ID: <29507085-AE04-4A43-A41A-6D279810CCEC@shaw.ca>

Hello,

I found an archived conversation on this list regarding FIFO origination caller ID, and how to modify it.  

http://old.nabble.com/FIFO-Orgination_caller_id-td26487628.html

There seems to be no easy way to customize the caller ID on originated calls from the FIFO to on-hook agents who were registered dynamically.  Anthony states a method to do it using static entries in the fifo conf file, and the general rationale is that SCREEN POPS be the preferred method, with the added nudge that good SIP phones can change the caller ID when the bridge is complete as well, which is all well and good.

The problem is for my application, all on-hook agents are using cellular phones, and they register dynamically.  Also, none of my agents are in front of a computer, so a SIP display update on the phone or screen pop on the computer in front of them is not really an option, and the only way they can identify calls from my FIFO right now is because they are the ones with NO CALLER ID (in other words, their mobile phones do not display the name, and the number is not recognized because it is set by FreeSWITCH to be "fifo+fifoname" instead of being numeric.  This is far from ideal.

I am wondering if there is anyone on the list who knows how to configure the origination_caller_id_number/name variables for dynamically registered on-hook agents so that the caller ID from the FIFO customer's incoming call is displayed to them instead of the above mangled caller ID?

I'm not disagreeing that it is an old-skewl way of thought, but in actuality it is just a way to interface with old-school telephony devices (i.e. non-Smartphone mobile phones) and I am not sure how to accomplish this.  Any help/input would be appreciated.

Regards,
Steve


From mike at jerris.com  Fri Jan 15 16:01:16 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 Jan 2010 19:01:16 -0500
Subject: [Freeswitch-users] FIFO Originate caller ID
In-Reply-To: <29507085-AE04-4A43-A41A-6D279810CCEC@shaw.ca>
References: <29507085-AE04-4A43-A41A-6D279810CCEC@shaw.ca>
Message-ID: <88D85720-C24A-43B8-A6E8-7F9179D71625@jerris.com>

At the time a call goes out to the agents, there is no specific caller they are matched too, therefore there is no way to know the caller id at this time.  When the originated call to the agent is answered, we THEN go and pick off the next caller to connect them with.  All you can do is set a caller id for the queue.

Mike

On Jan 15, 2010, at 6:43 PM, Steve Steffler wrote:

> Hello,
> 
> I found an archived conversation on this list regarding FIFO origination caller ID, and how to modify it.  
> 
> http://old.nabble.com/FIFO-Orgination_caller_id-td26487628.html
> 
> There seems to be no easy way to customize the caller ID on originated calls from the FIFO to on-hook agents who were registered dynamically.  Anthony states a method to do it using static entries in the fifo conf file, and the general rationale is that SCREEN POPS be the preferred method, with the added nudge that good SIP phones can change the caller ID when the bridge is complete as well, which is all well and good.
> 
> The problem is for my application, all on-hook agents are using cellular phones, and they register dynamically.  Also, none of my agents are in front of a computer, so a SIP display update on the phone or screen pop on the computer in front of them is not really an option, and the only way they can identify calls from my FIFO right now is because they are the ones with NO CALLER ID (in other words, their mobile phones do not display the name, and the number is not recognized because it is set by FreeSWITCH to be "fifo+fifoname" instead of being numeric.  This is far from ideal.
> 
> I am wondering if there is anyone on the list who knows how to configure the origination_caller_id_number/name variables for dynamically registered on-hook agents so that the caller ID from the FIFO customer's incoming call is displayed to them instead of the above mangled caller ID?
> 
> I'm not disagreeing that it is an old-skewl way of thought, but in actuality it is just a way to interface with old-school telephony devices (i.e. non-Smartphone mobile phones) and I am not sure how to accomplish this.  Any help/input would be appreciated.




From lawwton at gmail.com  Fri Jan 15 16:11:00 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Fri, 15 Jan 2010 19:11:00 -0500
Subject: [Freeswitch-users] Issue Compiling ESL (PHP)
In-Reply-To: <F84AD964-A9DC-48D8-BA0D-22E97863CACE@jerris.com>
References: <5fe6fa8f1001151437x491dade6q8b316f312556ff07@mail.gmail.com>
	<5fe6fa8f1001151454m92d17c9le4b811c897ae8482@mail.gmail.com>
	<9853f4ff1001151507oa1fa8adt5101956739d8c948@mail.gmail.com>
	<5fe6fa8f1001151521u3ef71f9g49491118bf0371ac@mail.gmail.com>
	<F84AD964-A9DC-48D8-BA0D-22E97863CACE@jerris.com>
Message-ID: <5fe6fa8f1001151611t506c0046u708cd49fe03eb34b@mail.gmail.com>

Thanks all, appreciate the help.

On Fri, Jan 15, 2010 at 6:35 PM, Michael Jerris <mike at jerris.com> wrote:
> Sounds right to me.
>
> Mike
>
> On Jan 15, 2010, at 6:21 PM, Alfredo Quiroga-Villamil wrote:
>
>> Thanks Anthony/David.
>>
>> So it seems like the build worked then. I take from the previous
>> emails and somewhere where I think I read that I can then take:
>>
>> ESL.so and ESL.php and put them on a remote system under my say for
>> instance 3rdParty directory and create scripts using the ESL.php
>> library which probably internally uses ESL.so.
>>
>> Is that statement correct?
>>
>> Thanks in advance,
>>
>> Alfredo
>>
>> On Fri, Jan 15, 2010 at 6:07 PM, David Villasmil
>> <david.villasmil.work at gmail.com> wrote:
>>> make a script ;)
>>>
>>> it's very easy, try it!
>>>
>>> David
>>>
>>> On Fri, Jan 15, 2010 at 11:54 PM, Alfredo Quiroga-Villamil
>>> <lawwton at gmail.com> wrote:
>>>> Looking over at this, perhaps it even worked.
>>>>
>>>> I see now under libs/esl/php/... the following two new files:
>>>>
>>>> esl_wrap.o
>>>> ESL.o
>>>>
>>>> Is there a way to verify that FS has support after running make and
>>>> make phpmod for php?
>>>>
>>>> Thanks in advance,
>>>>
>>>> Alfredo
>>>>
>>>> On Fri, Jan 15, 2010 at 5:37 PM, Alfredo Quiroga-Villamil
>>>> <lawwton at gmail.com> wrote:
>>>>> All:
>>>>>
>>>>> System:
>>>>>
>>>>> Linux usnc-rtp-01-fs 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT
>>>>> 2007 i686 i686 i386 GNU/Linux
>>>>>
>>>>> I am trying to compile ESL, following the following steps:
>>>>>
>>>>> 1- cd to my libs/esl directory as the wiki page indicates.
>>>>> 2- run make
>>>>>
>>>>> I then get right away the following:
>>>>>
>>>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o
>>>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o
>>>>> src/esl_event.o
>>>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o
>>>>> src/esl_threadmutex.o
>>>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o
>>>>> src/esl_config.o
>>>>> g++ ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>>>> -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o
>>>>> ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o
>>>>> src/esl_config.o src/esl_oop.o
>>>>> ranlib libesl.a
>>>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L.
>>>>> -L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit
>>>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient
>>>>> -L. -lncurses -lpthread -lesl
>>>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver
>>>>> -L. -lncurses -lpthread -lesl
>>>>> cc ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes ivrd.c -o ivrd -L. -lncurses
>>>>> -lpthread -lesl
>>>>>
>>>>> 3- I try typing then:
>>>>>
>>>>> make phpmod and get the following:
>>>>>
>>>>> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
>>>>> CFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2
>>>>> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings
>>>>> -Wstrict-prototypes -Wmissing-prototypes"
>>>>> CXXFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>>>> -Wno-unused-variable" CXX_CFLAGS="" -C php
>>>>> make[1]: Entering directory
>>>>> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>>>>> g++ ?-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include
>>>>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
>>>>> -Wno-unused-variable -I/usr/include/php -I/usr/include/php/main
>>>>> -I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext
>>>>> -I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function
>>>>> -c esl_wrap.cpp -o esl_wrap.o
>>>>> g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/kerberos/lib
>>>>> -lcrypt -lcrypt -lncurses -laspell -lpspell -lcurl -lresolv -lm -ldl
>>>>> -lnsl -lm -ldl -lcurl -ldl -lm -lcrypt -lm -lm -lcrypt -lpthread -o
>>>>> ESL.so -L.
>>>>> make[1]: Leaving directory
>>>>> `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php'
>>>>>
>>>>> I've installed I think all the -dev dependencies listed in the wiki. Any ideas?
>>>>>
>>>>> Thanks in advance,
>>>>>
>>>>> Alfredo Q-V
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mastermind202 at gmail.com  Fri Jan 15 16:12:01 2010
From: mastermind202 at gmail.com (mm_202)
Date: Fri, 15 Jan 2010 19:12:01 -0500
Subject: [Freeswitch-users] Domains.
In-Reply-To: <2083.99622.qm@web34302.mail.mud.yahoo.com>
References: <2083.99622.qm@web34302.mail.mud.yahoo.com>
Message-ID: <63de75711001151612k514900f4w599c39f3bb2b70e4@mail.gmail.com>

On Fri, Jan 15, 2010 at 3:26 PM, FERNANDO VILLARROEL
<fvillarroel at yahoo.com>wrote:

> Dear.
>
> I installed FS FreeSWITCH Version 1.0.trunk (16144)
>
> I have a problem when i send traffic from a external gateway, the calls are
> rejected:
>
> 2010-01-05 16:52:37.254194 [DEBUG] sofia.c:5359 0 acls to check for proxy
> 2010-01-05 16:52:37.254194 [DEBUG] sofia.c:5377 network ip is a proxy [0]
> 2010-01-05 16:52:37.254194 [DEBUG] sofia.c:5405 IP XXX.XXX.XX.125 Rejected
> by acl "domains". Falling back to Digest auth.
>
>
> Anyone could me explain like i can do.
>
> Regards.
>
> Fernando
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


Edit your conf/autoload_configs/acl.conf.xml file and add that IP into the
domains list.
Then in the FS cli run 'reloadxml' and 'reloadacl'.

Read http://wiki.freeswitch.org/wiki/Acl.conf.xml for more info.

-- mm_202.
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From anthony.minessale at gmail.com  Fri Jan 15 16:44:59 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jan 2010 18:44:59 -0600
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <9853f4ff1001151525o341cf2c7k2935538cca210815@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<9853f4ff1001150415o7d50cb1evbb83675f68cfb359@mail.gmail.com>
	<201001150726.17430.sos@sokhapkin.dyndns.org>
	<9853f4ff1001150438j14357ae9ue64928a42a77d69b@mail.gmail.com>
	<9853f4ff1001150443t5a18cc6bxa5032f391ab6f0ed@mail.gmail.com>
	<191c3a031001150808i69b3937boe0e903057f984e96@mail.gmail.com>
	<191c3a031001151317j408794a3pa2a0a5d21d63aa62@mail.gmail.com>
	<9853f4ff1001151445s6a4dff0qf8c9e17c702b7f86@mail.gmail.com>
	<191c3a031001151454w101ae8eh978e4ced42004fb3@mail.gmail.com>
	<9853f4ff1001151525o341cf2c7k2935538cca210815@mail.gmail.com>
Message-ID: <191c3a031001151644m6a838b6bp311f2a173fc65af3@mail.gmail.com>

Now we need a new feature

[leg_required=true]

set this on any legs required for the originate to proceed, if it's hungup,
the cause will be passed to any existing legs and fail the entire originate.

so use

{bridge_early_media=true}[leg_required=true]sofia/internal/real_call at foo.com
,sofia/internal/moh_call at foo.com

the leg_required will only be set on the 1st leg because of the [] vs {}
if that leg is then hungup, it will kill the other channels in the list.

please try latest trunk.



On Fri, Jan 15, 2010 at 5:25 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Anthony,
>
> LOL, and mounting and mounting... It does work when there is answer...
> but if B(2)-side rejects or times out or any other that 200 OK,
> B(1)-side stays indefinitely...
>
>
> On Fri, Jan 15, 2010 at 11:54 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > you can email me privately at this addr.
> >
> >
> > On Fri, Jan 15, 2010 at 4:45 PM, David Villasmil
> > <david.villasmil.work at gmail.com> wrote:
> >>
> >> Anthony,
> >>
> >>     Trying, Thanks. Is there anyway we can communicate directly?
> >>
> >>
> >> David
> >>
> >> On Fri, Jan 15, 2010 at 10:17 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >> > Try latest trunk,
> >> >
> >> > you should have exactly what you want with the same parameter, again
> my
> >> > paypal addr is cleary displayed as a big button on the website.
> >> >
> >> >
> >> > On Fri, Jan 15, 2010 at 10:08 AM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> one of the many reasons its a bad idea.
> >> >> Probably the leg with the bad audio is a different ptime.
> >> >> Now the amount of work I have to do escalates I would prefer you
> commit
> >> >> to
> >> >> commercial support by emailing me at consulting at freeswitch.org to
> >> >> continue
> >> >> with this.
> >> >>
> >> >>
> >> >> On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil
> >> >> <david.villasmil.work at gmail.com> wrote:
> >> >>>
> >> >>> I set it to "off" just in case, same thing.
> >> >>>
> >> >>> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
> >> >>> <david.villasmil.work at gmail.com> wrote:
> >> >>> > Default, haven't touched it i suppose it's off, i haven't set it
> >> >>> > anywhere
> >> >>> >
> >> >>> > On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
> >> >>> > <sos at sokhapkin.dyndns.org> wrote:
> >> >>> >> Is bypass_media on or off?
> >> >>> >>
> >> >>> >> On Friday 15 January 2010, David Villasmil wrote:
> >> >>> >>> Yeah, sorry. Early media. Audio after 200 is fine. Early media
> was
> >> >>> >>> ok
> >> >>> >>> before the change.
> >> >>> >>>
> >> >>> >>> David
> >> >>> >>>
> >> >>> >>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
> >> >>> >>>
> >> >>> >>> <sos at sokhapkin.dyndns.org> wrote:
> >> >>> >>> > Which audio? Early media or after 200 OK?
> >> >>> >>> >
> >> >>> >>> > On Friday 15 January 2010, David Villasmil wrote:
> >> >>> >>> >> Hello again Anthony,
> >> >>> >>> >>
> >> >>> >>> >> I just tested it, and although functionality does not, first
> >> >>> >>> >> incoming
> >> >>> >>> >> audio is coming in all garbled... do you know why?
> >> >>> >>> >>
> >> >>> >>> >> David
> >> >>> >>> >>
> >> >>> >>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
> >> >>> >>> >>
> >> >>> >>> >> <anthony.minessale at gmail.com> wrote:
> >> >>> >>> >> > {bridge_early_media=true}
> >> >>> >>> >> > in the dial string in place of ignore_early_media=true
> >> >>> >>> >> >
> >> >>> >>> >> > the first b leg in the list who sends 183 will become the
> >> >>> >>> >> > ringback
> >> >>> >>> >> > device for A leg it will hear the early media
> >> >>> >>> >> > for that leg while the other legs still ring.  If some
> other
> >> >>> >>> >> > leg
> >> >>> >>> >> > answers the final call will still be bridged to the leg who
> >> >>> >>> >> > answered.
> >> >>> >>> >> >
> >> >>> >>> >> >
> >> >>> >>> >> > I would estimate it at $500 payable on the big paypal
> button
> >> >>> >>> >> > on
> >> >>> >>> >> > http://www.freeswitch.org
> >> >>> >>> >> > but, I already added the patch to tree earlier today so I
> >> >>> >>> >> > guess
> >> >>> >>> >> > it's
> >> >>> >>> >> > up to you to pay it or not.
> >> >>> >>> >> >
> >> >>> >>> >> >
> >> >>> >>> >> >
> >> >>> >>> >> >
> >> >>> >>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> >> >>> >>> >> >
> >> >>> >>> >> > <david.villasmil.work at gmail.com> wrote:
> >> >>> >>> >> >> Anthony,
> >> >>> >>> >> >>
> >> >>> >>> >> >> I did take the "hint", don't worry. We will probably ask
> for
> >> >>> >>> >> >> a
> >> >>> >>> >> >> bounty
> >> >>> >>> >> >> but first we need to know:
> >> >>> >>> >> >> 1.- whether this is possible
> >> >>> >>> >> >> 2.- how long it would take
> >> >>> >>> >> >> 3.- how will it exactly work
> >> >>> >>> >> >> 4.- of course, what's the bounty (be gentle ;) )
> >> >>> >>> >> >>
> >> >>> >>> >> >> We would of course give this back to the community.
> >> >>> >>> >> >>
> >> >>> >>> >> >> in the meantime, bridge_early_media=true will discard the
> >> >>> >>> >> >> 2nd
> >> >>> >>> >> >> B-leg
> >> >>> >>> >> >> and conversely "false" it will keep on trying to connect
> and
> >> >>> >>> >> >> if
> >> >>> >>> >> >> it
> >> >>> >>> >> >> connects the other B-leg if will bridge to that one?
> >> >>> >>> >> >>
> >> >>> >>> >> >> Thanks
> >> >>> >>> >> >>
> >> >>> >>> >> >> David
> >> >>> >>> >> >>
> >> >>> >>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> >> >>> >>> >> >>
> >> >>> >>> >> >> <anthony.minessale at gmail.com> wrote:
> >> >>> >>> >> >> > I added bridge_early_media=true to do the best I can do.
> >> >>> >>> >> >> > This is the most I will do, especially for free, nobody
> >> >>> >>> >> >> > can
> >> >>> >>> >> >> > take a
> >> >>> >>> >> >> > hint that
> >> >>> >>> >> >> > you should be paying for all these custom requests so
> take
> >> >>> >>> >> >> > it
> >> >>> >>> >> >> > or
> >> >>> >>> >> >> > leave it
> >> >>> >>> >> >> > but this thread is done.........
> >> >>> >>> >> >> >
> >> >>> >>> >> >> >
> >> >>> >>> >> >> >
> >> >>> >>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
> >> >>> >>> >> >> > <sos at sokhapkin.dyndns.org>
> >> >>> >>> >> >> >
> >> >>> >>> >> >> > wrote:
> >> >>> >>> >> >> >> No, not exactly. ignore_early_media doesn't pass early
> >> >>> >>> >> >> >> media
> >> >>> >>> >> >> >> to
> >> >>> >>> >> >> >> the caller
> >> >>> >>> >> >> >> if
> >> >>> >>> >> >> >> bypass_media is false.
> >> >>> >>> >> >> >>
> >> >>> >>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
> >> >>> >>> >> >> >> > this is exactly what ignore_early_media does now.
> >> >>> >>> >> >> >> >
> >> >>> >>> >> >> >> > Mike
> >> >>> >>> >> >> >> >
> >> >>> >>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> >> >>> >>> >> >> >> > > The issue here is when "originate" routine should
> >> >>> >>> >> >> >> > > return
> >> >>> >>> >> >> >> > > and
> >> >>> >>> >> >> >> > > set "originate_status" variable. Current behavior
> is
> >> >>> >>> >> >> >> > > to
> >> >>> >>> >> >> >> > > return
> >> >>> >>> >> >> >> > > on early
> >> >>> >>> >> >> >> > > media, but what if to introduce a variable
> >> >>> >>> >> >> >> > > "originate_wait_for_answer"
> >> >>> >>> >> >> >> > > with default value "false" and use the variable in
> >> >>> >>> >> >> >> > > originate
> >> >>> >>> >> >> >> > > code to
> >> >>> >>> >> >> >> > > decide when to return - on 18X or "200 OK"?
> >> >>> >>> >> >> >> > >
> >> >>> >>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale
> wrote:
> >> >>> >>> >> >> >> > >> he wants to call 3 people at once and let the A
> leg
> >> >>> >>> >> >> >> > >> hear
> >> >>> >>> >> >> >> > >> early media
> >> >>> >>> >> >> >> > >> from call #1 while call #2 and #3 still are
> >> >>> >>> >> >> >> > >> progressing
> >> >>> >>> >> >> >> > >> which
> >> >>> >>> >> >> >> > >> is not
> >> >>> >>> >> >> >> > >> simple to do without doing thousands of dollars in
> >> >>> >>> >> >> >> > >> development.
> >> >>> >>> >> >> >> > >>
> >> >>> >>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB
> >> >>> >>> >> >> >> > >> <djbinter at yahoo.com>
> >> >>> >> wrote:
> >> >>> >>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK),
> so
> >> >>> >>> >> >> >> > >>> that your
> >> >>> >>> >> >> >> > >>> customers
> >> >>> >>> >> >> >> > >>> can hear recordings?
> >> >>> >>> >> >> >> >
> >> >>> >>> >> >> >> > _______________________________________________
> >> >>> >>> >> >> >> > FreeSWITCH-users mailing list
> >> >>> >>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> >> >> >> >
> >> >>> >>> >> >> >> >
> >> >>> >>> >> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>> >> >> >> >
> >> >>> >>> >> >> >> >
> >> >>> >>> >> >> >> >
> >> >>> >>> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswit
> >> >>> >>> >> >> >> >ch- users http://www.freeswitch.org
> >> >>> >>> >> >> >>
> >> >>> >>> >> >> >> _______________________________________________
> >> >>> >>> >> >> >> FreeSWITCH-users mailing list
> >> >>> >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> >> >> >>
> >> >>> >>> >> >> >>
> >> >>> >>> >> >> >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>> >> >> >>
> >> >>> >>> >> >> >>
> >> >>> >>> >> >> >>
> >> >>> >>> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch
> >> >>> >>> >> >> >>-us ers http://www.freeswitch.org
> >> >>> >>> >> >> >
> >> >>> >>> >> >> > --
> >> >>> >>> >> >> > Anthony Minessale II
> >> >>> >>> >> >> >
> >> >>> >>> >> >> > FreeSWITCH http://www.freeswitch.org/
> >> >>> >>> >> >> > ClueCon http://www.cluecon.com/
> >> >>> >>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >>> >>> >> >> >
> >> >>> >>> >> >> > AIM: anthm
> >> >>> >>> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >>> >>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >>> >>> >> >> > IRC: irc.freenode.net #freeswitch
> >> >>> >>> >> >> >
> >> >>> >>> >> >> > FreeSWITCH Developer Conference
> >> >>> >>> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >>> >>> >> >> > iax:guest at conference.freeswitch.org/888
> >> >>> >>> >> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >>> >>> >> >> > pstn:+19193869900
> >> >>> >>> >> >> >
> >> >>> >>> >> >> > _______________________________________________
> >> >>> >>> >> >> > FreeSWITCH-users mailing list
> >> >>> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> >> >> >
> >> >>> >>> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>> >> >> >
> >> >>> >>> >> >> >
> >> >>> >>> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-
> >> >>> >>> >> >> >use rs http://www.freeswitch.org
> >> >>> >>> >> >>
> >> >>> >>> >> >> _______________________________________________
> >> >>> >>> >> >> FreeSWITCH-users mailing list
> >> >>> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> >> >>
> >> >>> >>> >> >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>> >> >>
> >> >>> >>> >> >>
> >> >>> >>> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-us
> >> >>> >>> >> >>ers http://www.freeswitch.org
> >> >>> >>> >> >
> >> >>> >>> >> > --
> >> >>> >>> >> > Anthony Minessale II
> >> >>> >>> >> >
> >> >>> >>> >> > FreeSWITCH http://www.freeswitch.org/
> >> >>> >>> >> > ClueCon http://www.cluecon.com/
> >> >>> >>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >>> >>> >> >
> >> >>> >>> >> > AIM: anthm
> >> >>> >>> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >>> >>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >>> >>> >> > IRC: irc.freenode.net #freeswitch
> >> >>> >>> >> >
> >> >>> >>> >> > FreeSWITCH Developer Conference
> >> >>> >>> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >>> >>> >> > iax:guest at conference.freeswitch.org/888
> >> >>> >>> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >>> >>> >> > pstn:+19193869900
> >> >>> >>> >> >
> >> >>> >>> >> > _______________________________________________
> >> >>> >>> >> > FreeSWITCH-users mailing list
> >> >>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>> >> >
> >> >>> >>> >> >
> >> >>> >>> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-use
> >> >>> >>> >> >rs http://www.freeswitch.org
> >> >>> >>> >>
> >> >>> >>> >> _______________________________________________
> >> >>> >>> >> FreeSWITCH-users mailing list
> >> >>> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>> >>
> >> >>> >>> >>
> >> >>> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> >>> >> http://www.freeswitch.org
> >> >>> >>> >
> >> >>> >>> > _______________________________________________
> >> >>> >>> > FreeSWITCH-users mailing list
> >> >>> >>> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>> >
> >> >>> >>> >
> >> >>> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> >>> > http://www.freeswitch.org
> >> >>> >>>
> >> >>> >>> _______________________________________________
> >> >>> >>> FreeSWITCH-users mailing list
> >> >>> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>>
> >> >>> >>>
> >> >>> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> >>> http://www.freeswitch.org
> >> >>> >>
> >> >>> >>
> >> >>> >>
> >> >>> >> _______________________________________________
> >> >>> >> FreeSWITCH-users mailing list
> >> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>
> >> >>> >>
> >> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> >> http://www.freeswitch.org
> >> >>> >>
> >> >>> >
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>
> >> >>
> >> >>
> >> >> --
> >> >> Anthony Minessale II
> >> >>
> >> >> FreeSWITCH http://www.freeswitch.org/
> >> >> ClueCon http://www.cluecon.com/
> >> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >> >>
> >> >> AIM: anthm
> >> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> IRC: irc.freenode.net #freeswitch
> >> >>
> >> >> FreeSWITCH Developer Conference
> >> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> iax:guest at conference.freeswitch.org/888
> >> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> pstn:+19193869900
> >> >
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From sos at sokhapkin.dyndns.org  Fri Jan 15 17:04:25 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 15 Jan 2010 20:04:25 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <191c3a031001151644m6a838b6bp311f2a173fc65af3@mail.gmail.com>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<9853f4ff1001151525o341cf2c7k2935538cca210815@mail.gmail.com>
	<191c3a031001151644m6a838b6bp311f2a173fc65af3@mail.gmail.com>
Message-ID: <201001152004.25949.sos@sokhapkin.dyndns.org>

Sorry for the dumb question, is there a way to find out in dialplan (some 
variable?) which one of comma separated b-legs listed in originate command 
answered the call?

originate sofia/g1/number,sofia/g2/number,sofia/g3/number

Which gateway answered? g1, g2 or g3?

On Friday 15 January 2010, Anthony Minessale wrote:
> Now we need a new feature
>
> [leg_required=true]
>
> set this on any legs required for the originate to proceed, if it's hungup,
> the cause will be passed to any existing legs and fail the entire
> originate.
>
> so use
>
> {bridge_early_media=true}[leg_required=true]sofia/internal/real_call at foo.co
>m ,sofia/internal/moh_call at foo.com
>
> the leg_required will only be set on the 1st leg because of the [] vs {}
> if that leg is then hungup, it will kill the other channels in the list.
>
> please try latest trunk.
>
>
>
> On Fri, Jan 15, 2010 at 5:25 PM, David Villasmil <
>
> david.villasmil.work at gmail.com> wrote:
> > Anthony,
> >
> > LOL, and mounting and mounting... It does work when there is answer...
> > but if B(2)-side rejects or times out or any other that 200 OK,
> > B(1)-side stays indefinitely...
> >
> >
> > On Fri, Jan 15, 2010 at 11:54 PM, Anthony Minessale
> >
> > <anthony.minessale at gmail.com> wrote:
> > > you can email me privately at this addr.
> > >
> > >
> > > On Fri, Jan 15, 2010 at 4:45 PM, David Villasmil
> > >
> > > <david.villasmil.work at gmail.com> wrote:
> > >> Anthony,
> > >>
> > >>     Trying, Thanks. Is there anyway we can communicate directly?
> > >>
> > >>
> > >> David
> > >>
> > >> On Fri, Jan 15, 2010 at 10:17 PM, Anthony Minessale
> > >>
> > >> <anthony.minessale at gmail.com> wrote:
> > >> > Try latest trunk,
> > >> >
> > >> > you should have exactly what you want with the same parameter, again
> >
> > my
> >
> > >> > paypal addr is cleary displayed as a big button on the website.
> > >> >
> > >> >
> > >> > On Fri, Jan 15, 2010 at 10:08 AM, Anthony Minessale
> > >> >
> > >> > <anthony.minessale at gmail.com> wrote:
> > >> >> one of the many reasons its a bad idea.
> > >> >> Probably the leg with the bad audio is a different ptime.
> > >> >> Now the amount of work I have to do escalates I would prefer you
> >
> > commit
> >
> > >> >> to
> > >> >> commercial support by emailing me at consulting at freeswitch.org to
> > >> >> continue
> > >> >> with this.
> > >> >>
> > >> >>
> > >> >> On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil
> > >> >>
> > >> >> <david.villasmil.work at gmail.com> wrote:
> > >> >>> I set it to "off" just in case, same thing.
> > >> >>>
> > >> >>> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
> > >> >>>
> > >> >>> <david.villasmil.work at gmail.com> wrote:
> > >> >>> > Default, haven't touched it i suppose it's off, i haven't set it
> > >> >>> > anywhere
> > >> >>> >
> > >> >>> > On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
> > >> >>> >
> > >> >>> > <sos at sokhapkin.dyndns.org> wrote:
> > >> >>> >> Is bypass_media on or off?
> > >> >>> >>
> > >> >>> >> On Friday 15 January 2010, David Villasmil wrote:
> > >> >>> >>> Yeah, sorry. Early media. Audio after 200 is fine. Early media
> >
> > was
> >
> > >> >>> >>> ok
> > >> >>> >>> before the change.
> > >> >>> >>>
> > >> >>> >>> David
> > >> >>> >>>
> > >> >>> >>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
> > >> >>> >>>
> > >> >>> >>> <sos at sokhapkin.dyndns.org> wrote:
> > >> >>> >>> > Which audio? Early media or after 200 OK?
> > >> >>> >>> >
> > >> >>> >>> > On Friday 15 January 2010, David Villasmil wrote:
> > >> >>> >>> >> Hello again Anthony,
> > >> >>> >>> >>
> > >> >>> >>> >> I just tested it, and although functionality does not,
> > >> >>> >>> >> first incoming
> > >> >>> >>> >> audio is coming in all garbled... do you know why?
> > >> >>> >>> >>
> > >> >>> >>> >> David
> > >> >>> >>> >>
> > >> >>> >>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
> > >> >>> >>> >>
> > >> >>> >>> >> <anthony.minessale at gmail.com> wrote:
> > >> >>> >>> >> > {bridge_early_media=true}
> > >> >>> >>> >> > in the dial string in place of ignore_early_media=true
> > >> >>> >>> >> >
> > >> >>> >>> >> > the first b leg in the list who sends 183 will become the
> > >> >>> >>> >> > ringback
> > >> >>> >>> >> > device for A leg it will hear the early media
> > >> >>> >>> >> > for that leg while the other legs still ring.  If some
> >
> > other
> >
> > >> >>> >>> >> > leg
> > >> >>> >>> >> > answers the final call will still be bridged to the leg
> > >> >>> >>> >> > who answered.
> > >> >>> >>> >> >
> > >> >>> >>> >> >
> > >> >>> >>> >> > I would estimate it at $500 payable on the big paypal
> >
> > button
> >
> > >> >>> >>> >> > on
> > >> >>> >>> >> > http://www.freeswitch.org
> > >> >>> >>> >> > but, I already added the patch to tree earlier today so I
> > >> >>> >>> >> > guess
> > >> >>> >>> >> > it's
> > >> >>> >>> >> > up to you to pay it or not.
> > >> >>> >>> >> >
> > >> >>> >>> >> >
> > >> >>> >>> >> >
> > >> >>> >>> >> >
> > >> >>> >>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> > >> >>> >>> >> >
> > >> >>> >>> >> > <david.villasmil.work at gmail.com> wrote:
> > >> >>> >>> >> >> Anthony,
> > >> >>> >>> >> >>
> > >> >>> >>> >> >> I did take the "hint", don't worry. We will probably ask
> >
> > for
> >
> > >> >>> >>> >> >> a
> > >> >>> >>> >> >> bounty
> > >> >>> >>> >> >> but first we need to know:
> > >> >>> >>> >> >> 1.- whether this is possible
> > >> >>> >>> >> >> 2.- how long it would take
> > >> >>> >>> >> >> 3.- how will it exactly work
> > >> >>> >>> >> >> 4.- of course, what's the bounty (be gentle ;) )
> > >> >>> >>> >> >>
> > >> >>> >>> >> >> We would of course give this back to the community.
> > >> >>> >>> >> >>
> > >> >>> >>> >> >> in the meantime, bridge_early_media=true will discard
> > >> >>> >>> >> >> the 2nd
> > >> >>> >>> >> >> B-leg
> > >> >>> >>> >> >> and conversely "false" it will keep on trying to connect
> >
> > and
> >
> > >> >>> >>> >> >> if
> > >> >>> >>> >> >> it
> > >> >>> >>> >> >> connects the other B-leg if will bridge to that one?
> > >> >>> >>> >> >>
> > >> >>> >>> >> >> Thanks
> > >> >>> >>> >> >>
> > >> >>> >>> >> >> David
> > >> >>> >>> >> >>
> > >> >>> >>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> > >> >>> >>> >> >>
> > >> >>> >>> >> >> <anthony.minessale at gmail.com> wrote:
> > >> >>> >>> >> >> > I added bridge_early_media=true to do the best I can
> > >> >>> >>> >> >> > do. This is the most I will do, especially for free,
> > >> >>> >>> >> >> > nobody can
> > >> >>> >>> >> >> > take a
> > >> >>> >>> >> >> > hint that
> > >> >>> >>> >> >> > you should be paying for all these custom requests so
> >
> > take
> >
> > >> >>> >>> >> >> > it
> > >> >>> >>> >> >> > or
> > >> >>> >>> >> >> > leave it
> > >> >>> >>> >> >> > but this thread is done.........
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
> > >> >>> >>> >> >> > <sos at sokhapkin.dyndns.org>
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> > wrote:
> > >> >>> >>> >> >> >> No, not exactly. ignore_early_media doesn't pass
> > >> >>> >>> >> >> >> early media
> > >> >>> >>> >> >> >> to
> > >> >>> >>> >> >> >> the caller
> > >> >>> >>> >> >> >> if
> > >> >>> >>> >> >> >> bypass_media is false.
> > >> >>> >>> >> >> >>
> > >> >>> >>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
> > >> >>> >>> >> >> >> > this is exactly what ignore_early_media does now.
> > >> >>> >>> >> >> >> >
> > >> >>> >>> >> >> >> > Mike
> > >> >>> >>> >> >> >> >
> > >> >>> >>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin wrote:
> > >> >>> >>> >> >> >> > > The issue here is when "originate" routine should
> > >> >>> >>> >> >> >> > > return
> > >> >>> >>> >> >> >> > > and
> > >> >>> >>> >> >> >> > > set "originate_status" variable. Current behavior
> >
> > is
> >
> > >> >>> >>> >> >> >> > > to
> > >> >>> >>> >> >> >> > > return
> > >> >>> >>> >> >> >> > > on early
> > >> >>> >>> >> >> >> > > media, but what if to introduce a variable
> > >> >>> >>> >> >> >> > > "originate_wait_for_answer"
> > >> >>> >>> >> >> >> > > with default value "false" and use the variable
> > >> >>> >>> >> >> >> > > in originate
> > >> >>> >>> >> >> >> > > code to
> > >> >>> >>> >> >> >> > > decide when to return - on 18X or "200 OK"?
> > >> >>> >>> >> >> >> > >
> > >> >>> >>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale
> >
> > wrote:
> > >> >>> >>> >> >> >> > >> he wants to call 3 people at once and let the A
> >
> > leg
> >
> > >> >>> >>> >> >> >> > >> hear
> > >> >>> >>> >> >> >> > >> early media
> > >> >>> >>> >> >> >> > >> from call #1 while call #2 and #3 still are
> > >> >>> >>> >> >> >> > >> progressing
> > >> >>> >>> >> >> >> > >> which
> > >> >>> >>> >> >> >> > >> is not
> > >> >>> >>> >> >> >> > >> simple to do without doing thousands of dollars
> > >> >>> >>> >> >> >> > >> in development.
> > >> >>> >>> >> >> >> > >>
> > >> >>> >>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB
> > >> >>> >>> >> >> >> > >> <djbinter at yahoo.com>
> > >> >>> >>
> > >> >>> >> wrote:
> > >> >>> >>> >> >> >> > >>> What about sending Sip 183 with SDP (no 200OK),
> >
> > so
> >
> > >> >>> >>> >> >> >> > >>> that your
> > >> >>> >>> >> >> >> > >>> customers
> > >> >>> >>> >> >> >> > >>> can hear recordings?
> > >> >>> >>> >> >> >> >
> > >> >>> >>> >> >> >> > _______________________________________________
> > >> >>> >>> >> >> >> > FreeSWITCH-users mailing list
> > >> >>> >>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >> >>> >>> >> >> >> > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswit
> >
> > >> >>> >>> >> >> >> >ch- users http://www.freeswitch.org
> > >> >>> >>> >> >> >>
> > >> >>> >>> >> >> >> _______________________________________________
> > >> >>> >>> >> >> >> FreeSWITCH-users mailing list
> > >> >>> >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >> >>> >>> >> >> >> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch
> >
> > >> >>> >>> >> >> >>-us ers http://www.freeswitch.org
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> > --
> > >> >>> >>> >> >> > Anthony Minessale II
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> > FreeSWITCH http://www.freeswitch.org/
> > >> >>> >>> >> >> > ClueCon http://www.cluecon.com/
> > >> >>> >>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> > AIM: anthm
> > >> >>> >>> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_miness
> > >> >>> >>> >> >> >ale at hotmail.com>
> > >> >>> >>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL
> > >> >>> >>> >> >> >%3Aanthony.minessale at gmail.com> IRC: irc.freenode.net
> > >> >>> >>> >> >> > #freeswitch
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> > FreeSWITCH Developer Conference
> > >> >>> >>> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference
> > >> >>> >>> >> >> >.freeswitch.org>
> > >> >>> >>> >> >> > iax:guest at conference.freeswitch.org/888
> > >> >>> >>> >> >> > googletalk:conf+888 at conference.freeswitch.org<googleta
> > >> >>> >>> >> >> >lk%3Aconf%2B888 at conference.freeswitch.org>
> > >> >>> >>> >> >> > pstn:+19193869900
> > >> >>> >>> >> >> >
> > >> >>> >>> >> >> > _______________________________________________
> > >> >>> >>> >> >> > FreeSWITCH-users mailing list
> > >> >>> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >> >>> >>> >> >> > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-
> >
> > >> >>> >>> >> >> >use rs http://www.freeswitch.org
> > >> >>> >>> >> >>
> > >> >>> >>> >> >> _______________________________________________
> > >> >>> >>> >> >> FreeSWITCH-users mailing list
> > >> >>> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >> >>> >>> >> >> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-us
> >
> > >> >>> >>> >> >>ers http://www.freeswitch.org
> > >> >>> >>> >> >
> > >> >>> >>> >> > --
> > >> >>> >>> >> > Anthony Minessale II
> > >> >>> >>> >> >
> > >> >>> >>> >> > FreeSWITCH http://www.freeswitch.org/
> > >> >>> >>> >> > ClueCon http://www.cluecon.com/
> > >> >>> >>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> > >> >>> >>> >> >
> > >> >>> >>> >> > AIM: anthm
> > >> >>> >>> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale
> > >> >>> >>> >> >@hotmail.com>
> > >> >>> >>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3A
> > >> >>> >>> >> >anthony.minessale at gmail.com> IRC: irc.freenode.net
> > >> >>> >>> >> > #freeswitch
> > >> >>> >>> >> >
> > >> >>> >>> >> > FreeSWITCH Developer Conference
> > >> >>> >>> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.fr
> > >> >>> >>> >> >eeswitch.org> iax:guest at conference.freeswitch.org/888
> > >> >>> >>> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%
> > >> >>> >>> >> >3Aconf%2B888 at conference.freeswitch.org> pstn:+19193869900
> > >> >>> >>> >> >
> > >> >>> >>> >> > _______________________________________________
> > >> >>> >>> >> > FreeSWITCH-users mailing list
> > >> >>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >> >>> >>> >> > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-use
> >
> > >> >>> >>> >> >rs http://www.freeswitch.org
> > >> >>> >>> >>
> > >> >>> >>> >> _______________________________________________
> > >> >>> >>> >> FreeSWITCH-users mailing list
> > >> >>> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >> >>> >>> >> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> >>> >>> >> http://www.freeswitch.org
> > >> >>> >>> >
> > >> >>> >>> > _______________________________________________
> > >> >>> >>> > FreeSWITCH-users mailing list
> > >> >>> >>> > FreeSWITCH-users at lists.freeswitch.org
> > >> >>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-user
> > >> >>> >>> >s
> > >> >>> >>> >
> > >> >>> >>> >
> > >> >>> >>> > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> >>> >>> > http://www.freeswitch.org
> > >> >>> >>>
> > >> >>> >>> _______________________________________________
> > >> >>> >>> FreeSWITCH-users mailing list
> > >> >>> >>> FreeSWITCH-users at lists.freeswitch.org
> > >> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> >>> >>>
> > >> >>> >>>
> > >> >>> >>> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> >>> >>> http://www.freeswitch.org
> > >> >>> >>
> > >> >>> >> _______________________________________________
> > >> >>> >> FreeSWITCH-users mailing list
> > >> >>> >> FreeSWITCH-users at lists.freeswitch.org
> > >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> >>> >>
> > >> >>> >>
> > >> >>> >> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> >>> >> http://www.freeswitch.org
> > >> >>>
> > >> >>> _______________________________________________
> > >> >>> FreeSWITCH-users mailing list
> > >> >>> FreeSWITCH-users at lists.freeswitch.org
> > >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> >>>
> > >> >>> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> >>> http://www.freeswitch.org
> > >> >>
> > >> >> --
> > >> >> Anthony Minessale II
> > >> >>
> > >> >> FreeSWITCH http://www.freeswitch.org/
> > >> >> ClueCon http://www.cluecon.com/
> > >> >> Twitter: http://twitter.com/FreeSWITCH_wire
> > >> >>
> > >> >> AIM: anthm
> > >> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.c
> > >> >>om>
> > >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.mi
> > >> >>nessale at gmail.com> IRC: irc.freenode.net #freeswitch
> > >> >>
> > >> >> FreeSWITCH Developer Conference
> > >> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.o
> > >> >>rg> iax:guest at conference.freeswitch.org/888
> > >> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B8
> > >> >>88 at conference.freeswitch.org> pstn:+19193869900
> > >> >
> > >> > --
> > >> > Anthony Minessale II
> > >> >
> > >> > FreeSWITCH http://www.freeswitch.org/
> > >> > ClueCon http://www.cluecon.com/
> > >> > Twitter: http://twitter.com/FreeSWITCH_wire
> > >> >
> > >> > AIM: anthm
> > >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.co
> > >> >m>
> > >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.min
> > >> >essale at gmail.com> IRC: irc.freenode.net #freeswitch
> > >> >
> > >> > FreeSWITCH Developer Conference
> > >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.or
> > >> >g> iax:guest at conference.freeswitch.org/888
> > >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B88
> > >> >8 at conference.freeswitch.org> pstn:+19193869900
> > >> >
> > >> > _______________________________________________
> > >> > FreeSWITCH-users mailing list
> > >> > FreeSWITCH-users at lists.freeswitch.org
> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> > http://www.freeswitch.org
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> http://www.freeswitch.org
> > >
> > > --
> > > Anthony Minessale II
> > >
> > > FreeSWITCH http://www.freeswitch.org/
> > > ClueCon http://www.cluecon.com/
> > > Twitter: http://twitter.com/FreeSWITCH_wire
> > >
> > > AIM: anthm
> > > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.miness
> > >ale at gmail.com> IRC: irc.freenode.net #freeswitch
> > >
> > > FreeSWITCH Developer Conference
> > > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > > iax:guest at conference.freeswitch.org/888
> > > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at c
> > >onference.freeswitch.org> pstn:+19193869900
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From lists at redbonez.net  Fri Jan 15 17:28:22 2010
From: lists at redbonez.net (Adam Ford)
Date: Fri, 15 Jan 2010 18:28:22 -0700
Subject: [Freeswitch-users] Polycom configuration for FreeSWITCH
Message-ID: <003701ca964b$3241b100$96c51300$@net>

Does anyone who has successfully implemented Polycom IP301/501s, with
FreeSWITCH, have a configuration they can share with me as an example? I
have read the Polycom Admin Guide several times and understand what the
settings are/do, I am just not sure which FreeSWITCH supports, which it
doesn't, and which need special configuration to work with FreeSWITCH.

 

Thanks in advance,

-Adam

 

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From sos at sokhapkin.dyndns.org  Fri Jan 15 17:27:07 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 15 Jan 2010 20:27:07 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001152004.25949.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<191c3a031001151644m6a838b6bp311f2a173fc65af3@mail.gmail.com>
	<201001152004.25949.sos@sokhapkin.dyndns.org>
Message-ID: <201001152027.07209.sos@sokhapkin.dyndns.org>

Same question about 

originate aaa|bbb|ccc

syntax :-)


On Friday 15 January 2010, Sergey Okhapkin wrote:
> Sorry for the dumb question, is there a way to find out in dialplan (some
> variable?) which one of comma separated b-legs listed in originate command
> answered the call?
>
> originate sofia/g1/number,sofia/g2/number,sofia/g3/number
>
> Which gateway answered? g1, g2 or g3?
>
> On Friday 15 January 2010, Anthony Minessale wrote:
> > Now we need a new feature
> >
> > [leg_required=true]
> >
> > set this on any legs required for the originate to proceed, if it's
> > hungup, the cause will be passed to any existing legs and fail the entire
> > originate.
> >
> > so use
> >
> > {bridge_early_media=true}[leg_required=true]sofia/internal/real_call at foo.
> >co m ,sofia/internal/moh_call at foo.com
> >
> > the leg_required will only be set on the 1st leg because of the [] vs {}
> > if that leg is then hungup, it will kill the other channels in the list.
> >
> > please try latest trunk.
> >
> >
> >
> > On Fri, Jan 15, 2010 at 5:25 PM, David Villasmil <
> >
> > david.villasmil.work at gmail.com> wrote:
> > > Anthony,
> > >
> > > LOL, and mounting and mounting... It does work when there is answer...
> > > but if B(2)-side rejects or times out or any other that 200 OK,
> > > B(1)-side stays indefinitely...
> > >
> > >
> > > On Fri, Jan 15, 2010 at 11:54 PM, Anthony Minessale
> > >
> > > <anthony.minessale at gmail.com> wrote:
> > > > you can email me privately at this addr.
> > > >
> > > >
> > > > On Fri, Jan 15, 2010 at 4:45 PM, David Villasmil
> > > >
> > > > <david.villasmil.work at gmail.com> wrote:
> > > >> Anthony,
> > > >>
> > > >>     Trying, Thanks. Is there anyway we can communicate directly?
> > > >>
> > > >>
> > > >> David
> > > >>
> > > >> On Fri, Jan 15, 2010 at 10:17 PM, Anthony Minessale
> > > >>
> > > >> <anthony.minessale at gmail.com> wrote:
> > > >> > Try latest trunk,
> > > >> >
> > > >> > you should have exactly what you want with the same parameter,
> > > >> > again
> > >
> > > my
> > >
> > > >> > paypal addr is cleary displayed as a big button on the website.
> > > >> >
> > > >> >
> > > >> > On Fri, Jan 15, 2010 at 10:08 AM, Anthony Minessale
> > > >> >
> > > >> > <anthony.minessale at gmail.com> wrote:
> > > >> >> one of the many reasons its a bad idea.
> > > >> >> Probably the leg with the bad audio is a different ptime.
> > > >> >> Now the amount of work I have to do escalates I would prefer you
> > >
> > > commit
> > >
> > > >> >> to
> > > >> >> commercial support by emailing me at consulting at freeswitch.org to
> > > >> >> continue
> > > >> >> with this.
> > > >> >>
> > > >> >>
> > > >> >> On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil
> > > >> >>
> > > >> >> <david.villasmil.work at gmail.com> wrote:
> > > >> >>> I set it to "off" just in case, same thing.
> > > >> >>>
> > > >> >>> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
> > > >> >>>
> > > >> >>> <david.villasmil.work at gmail.com> wrote:
> > > >> >>> > Default, haven't touched it i suppose it's off, i haven't set
> > > >> >>> > it anywhere
> > > >> >>> >
> > > >> >>> > On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
> > > >> >>> >
> > > >> >>> > <sos at sokhapkin.dyndns.org> wrote:
> > > >> >>> >> Is bypass_media on or off?
> > > >> >>> >>
> > > >> >>> >> On Friday 15 January 2010, David Villasmil wrote:
> > > >> >>> >>> Yeah, sorry. Early media. Audio after 200 is fine. Early
> > > >> >>> >>> media
> > >
> > > was
> > >
> > > >> >>> >>> ok
> > > >> >>> >>> before the change.
> > > >> >>> >>>
> > > >> >>> >>> David
> > > >> >>> >>>
> > > >> >>> >>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
> > > >> >>> >>>
> > > >> >>> >>> <sos at sokhapkin.dyndns.org> wrote:
> > > >> >>> >>> > Which audio? Early media or after 200 OK?
> > > >> >>> >>> >
> > > >> >>> >>> > On Friday 15 January 2010, David Villasmil wrote:
> > > >> >>> >>> >> Hello again Anthony,
> > > >> >>> >>> >>
> > > >> >>> >>> >> I just tested it, and although functionality does not,
> > > >> >>> >>> >> first incoming
> > > >> >>> >>> >> audio is coming in all garbled... do you know why?
> > > >> >>> >>> >>
> > > >> >>> >>> >> David
> > > >> >>> >>> >>
> > > >> >>> >>> >> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
> > > >> >>> >>> >>
> > > >> >>> >>> >> <anthony.minessale at gmail.com> wrote:
> > > >> >>> >>> >> > {bridge_early_media=true}
> > > >> >>> >>> >> > in the dial string in place of ignore_early_media=true
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > the first b leg in the list who sends 183 will become
> > > >> >>> >>> >> > the ringback
> > > >> >>> >>> >> > device for A leg it will hear the early media
> > > >> >>> >>> >> > for that leg while the other legs still ring.  If some
> > >
> > > other
> > >
> > > >> >>> >>> >> > leg
> > > >> >>> >>> >> > answers the final call will still be bridged to the leg
> > > >> >>> >>> >> > who answered.
> > > >> >>> >>> >> >
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > I would estimate it at $500 payable on the big paypal
> > >
> > > button
> > >
> > > >> >>> >>> >> > on
> > > >> >>> >>> >> > http://www.freeswitch.org
> > > >> >>> >>> >> > but, I already added the patch to tree earlier today so
> > > >> >>> >>> >> > I guess
> > > >> >>> >>> >> > it's
> > > >> >>> >>> >> > up to you to pay it or not.
> > > >> >>> >>> >> >
> > > >> >>> >>> >> >
> > > >> >>> >>> >> >
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > <david.villasmil.work at gmail.com> wrote:
> > > >> >>> >>> >> >> Anthony,
> > > >> >>> >>> >> >>
> > > >> >>> >>> >> >> I did take the "hint", don't worry. We will probably
> > > >> >>> >>> >> >> ask
> > >
> > > for
> > >
> > > >> >>> >>> >> >> a
> > > >> >>> >>> >> >> bounty
> > > >> >>> >>> >> >> but first we need to know:
> > > >> >>> >>> >> >> 1.- whether this is possible
> > > >> >>> >>> >> >> 2.- how long it would take
> > > >> >>> >>> >> >> 3.- how will it exactly work
> > > >> >>> >>> >> >> 4.- of course, what's the bounty (be gentle ;) )
> > > >> >>> >>> >> >>
> > > >> >>> >>> >> >> We would of course give this back to the community.
> > > >> >>> >>> >> >>
> > > >> >>> >>> >> >> in the meantime, bridge_early_media=true will discard
> > > >> >>> >>> >> >> the 2nd
> > > >> >>> >>> >> >> B-leg
> > > >> >>> >>> >> >> and conversely "false" it will keep on trying to
> > > >> >>> >>> >> >> connect
> > >
> > > and
> > >
> > > >> >>> >>> >> >> if
> > > >> >>> >>> >> >> it
> > > >> >>> >>> >> >> connects the other B-leg if will bridge to that one?
> > > >> >>> >>> >> >>
> > > >> >>> >>> >> >> Thanks
> > > >> >>> >>> >> >>
> > > >> >>> >>> >> >> David
> > > >> >>> >>> >> >>
> > > >> >>> >>> >> >> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
> > > >> >>> >>> >> >>
> > > >> >>> >>> >> >> <anthony.minessale at gmail.com> wrote:
> > > >> >>> >>> >> >> > I added bridge_early_media=true to do the best I can
> > > >> >>> >>> >> >> > do. This is the most I will do, especially for free,
> > > >> >>> >>> >> >> > nobody can
> > > >> >>> >>> >> >> > take a
> > > >> >>> >>> >> >> > hint that
> > > >> >>> >>> >> >> > you should be paying for all these custom requests
> > > >> >>> >>> >> >> > so
> > >
> > > take
> > >
> > > >> >>> >>> >> >> > it
> > > >> >>> >>> >> >> > or
> > > >> >>> >>> >> >> > leave it
> > > >> >>> >>> >> >> > but this thread is done.........
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> > On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
> > > >> >>> >>> >> >> > <sos at sokhapkin.dyndns.org>
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> > wrote:
> > > >> >>> >>> >> >> >> No, not exactly. ignore_early_media doesn't pass
> > > >> >>> >>> >> >> >> early media
> > > >> >>> >>> >> >> >> to
> > > >> >>> >>> >> >> >> the caller
> > > >> >>> >>> >> >> >> if
> > > >> >>> >>> >> >> >> bypass_media is false.
> > > >> >>> >>> >> >> >>
> > > >> >>> >>> >> >> >> On Thursday 14 January 2010, Michael Jerris wrote:
> > > >> >>> >>> >> >> >> > this is exactly what ignore_early_media does now.
> > > >> >>> >>> >> >> >> >
> > > >> >>> >>> >> >> >> > Mike
> > > >> >>> >>> >> >> >> >
> > > >> >>> >>> >> >> >> > On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin 
wrote:
> > > >> >>> >>> >> >> >> > > The issue here is when "originate" routine
> > > >> >>> >>> >> >> >> > > should return
> > > >> >>> >>> >> >> >> > > and
> > > >> >>> >>> >> >> >> > > set "originate_status" variable. Current
> > > >> >>> >>> >> >> >> > > behavior
> > >
> > > is
> > >
> > > >> >>> >>> >> >> >> > > to
> > > >> >>> >>> >> >> >> > > return
> > > >> >>> >>> >> >> >> > > on early
> > > >> >>> >>> >> >> >> > > media, but what if to introduce a variable
> > > >> >>> >>> >> >> >> > > "originate_wait_for_answer"
> > > >> >>> >>> >> >> >> > > with default value "false" and use the variable
> > > >> >>> >>> >> >> >> > > in originate
> > > >> >>> >>> >> >> >> > > code to
> > > >> >>> >>> >> >> >> > > decide when to return - on 18X or "200 OK"?
> > > >> >>> >>> >> >> >> > >
> > > >> >>> >>> >> >> >> > > On Thursday 14 January 2010, Anthony Minessale
> > >
> > > wrote:
> > > >> >>> >>> >> >> >> > >> he wants to call 3 people at once and let the
> > > >> >>> >>> >> >> >> > >> A
> > >
> > > leg
> > >
> > > >> >>> >>> >> >> >> > >> hear
> > > >> >>> >>> >> >> >> > >> early media
> > > >> >>> >>> >> >> >> > >> from call #1 while call #2 and #3 still are
> > > >> >>> >>> >> >> >> > >> progressing
> > > >> >>> >>> >> >> >> > >> which
> > > >> >>> >>> >> >> >> > >> is not
> > > >> >>> >>> >> >> >> > >> simple to do without doing thousands of
> > > >> >>> >>> >> >> >> > >> dollars in development.
> > > >> >>> >>> >> >> >> > >>
> > > >> >>> >>> >> >> >> > >> On Thu, Jan 14, 2010 at 11:39 AM, DJB
> > > >> >>> >>> >> >> >> > >> <djbinter at yahoo.com>
> > > >> >>> >>
> > > >> >>> >> wrote:
> > > >> >>> >>> >> >> >> > >>> What about sending Sip 183 with SDP (no
> > > >> >>> >>> >> >> >> > >>> 200OK),
> > >
> > > so
> > >
> > > >> >>> >>> >> >> >> > >>> that your
> > > >> >>> >>> >> >> >> > >>> customers
> > > >> >>> >>> >> >> >> > >>> can hear recordings?
> > > >> >>> >>> >> >> >> >
> > > >> >>> >>> >> >> >> > _______________________________________________
> > > >> >>> >>> >> >> >> > FreeSWITCH-users mailing list
> > > >> >>> >>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >> >>> >>> >> >> >> > UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswit
> > >
> > > >> >>> >>> >> >> >> >ch- users http://www.freeswitch.org
> > > >> >>> >>> >> >> >>
> > > >> >>> >>> >> >> >> _______________________________________________
> > > >> >>> >>> >> >> >> FreeSWITCH-users mailing list
> > > >> >>> >>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >> >>> >>> >> >> >> UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch
> > >
> > > >> >>> >>> >> >> >>-us ers http://www.freeswitch.org
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> > --
> > > >> >>> >>> >> >> > Anthony Minessale II
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> > FreeSWITCH http://www.freeswitch.org/
> > > >> >>> >>> >> >> > ClueCon http://www.cluecon.com/
> > > >> >>> >>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> > AIM: anthm
> > > >> >>> >>> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_mine
> > > >> >>> >>> >> >> >ss ale at hotmail.com>
> > > >> >>> >>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYP
> > > >> >>> >>> >> >> >AL %3Aanthony.minessale at gmail.com> IRC:
> > > >> >>> >>> >> >> > irc.freenode.net #freeswitch
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> > FreeSWITCH Developer Conference
> > > >> >>> >>> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conferen
> > > >> >>> >>> >> >> >ce .freeswitch.org>
> > > >> >>> >>> >> >> > iax:guest at conference.freeswitch.org/888
> > > >> >>> >>> >> >> > googletalk:conf+888 at conference.freeswitch.org<google
> > > >> >>> >>> >> >> >ta lk%3Aconf%2B888 at conference.freeswitch.org>
> > > >> >>> >>> >> >> > pstn:+19193869900
> > > >> >>> >>> >> >> >
> > > >> >>> >>> >> >> > _______________________________________________
> > > >> >>> >>> >> >> > FreeSWITCH-users mailing list
> > > >> >>> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
> > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >> >>> >>> >> >> > UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-
> > >
> > > >> >>> >>> >> >> >use rs http://www.freeswitch.org
> > > >> >>> >>> >> >>
> > > >> >>> >>> >> >> _______________________________________________
> > > >> >>> >>> >> >> FreeSWITCH-users mailing list
> > > >> >>> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
> > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >> >>> >>> >> >> UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-us
> > >
> > > >> >>> >>> >> >>ers http://www.freeswitch.org
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > --
> > > >> >>> >>> >> > Anthony Minessale II
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > FreeSWITCH http://www.freeswitch.org/
> > > >> >>> >>> >> > ClueCon http://www.cluecon.com/
> > > >> >>> >>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > AIM: anthm
> > > >> >>> >>> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessa
> > > >> >>> >>> >> >le @hotmail.com>
> > > >> >>> >>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%
> > > >> >>> >>> >> >3A anthony.minessale at gmail.com> IRC: irc.freenode.net
> > > >> >>> >>> >> > #freeswitch
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > FreeSWITCH Developer Conference
> > > >> >>> >>> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.
> > > >> >>> >>> >> >fr eeswitch.org> iax:guest at conference.freeswitch.org/888
> > > >> >>> >>> >> > googletalk:conf+888 at conference.freeswitch.org<googletal
> > > >> >>> >>> >> >k% 3Aconf%2B888 at conference.freeswitch.org>
> > > >> >>> >>> >> > pstn:+19193869900
> > > >> >>> >>> >> >
> > > >> >>> >>> >> > _______________________________________________
> > > >> >>> >>> >> > FreeSWITCH-users mailing list
> > > >> >>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >> >>> >>> >> > UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-use
> > >
> > > >> >>> >>> >> >rs http://www.freeswitch.org
> > > >> >>> >>> >>
> > > >> >>> >>> >> _______________________________________________
> > > >> >>> >>> >> FreeSWITCH-users mailing list
> > > >> >>> >>> >> FreeSWITCH-users at lists.freeswitch.org
> > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > >> >>> >>> >> UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> >>> >>> >> http://www.freeswitch.org
> > > >> >>> >>> >
> > > >> >>> >>> > _______________________________________________
> > > >> >>> >>> > FreeSWITCH-users mailing list
> > > >> >>> >>> > FreeSWITCH-users at lists.freeswitch.org
> > > >> >>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-us
> > > >> >>> >>> >er s
> > > >> >>> >>> >
> > > >> >>> >>> >
> > > >> >>> >>> > UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> >>> >>> > http://www.freeswitch.org
> > > >> >>> >>>
> > > >> >>> >>> _______________________________________________
> > > >> >>> >>> FreeSWITCH-users mailing list
> > > >> >>> >>> FreeSWITCH-users at lists.freeswitch.org
> > > >> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-user
> > > >> >>> >>>s
> > > >> >>> >>>
> > > >> >>> >>>
> > > >> >>> >>> UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> >>> >>> http://www.freeswitch.org
> > > >> >>> >>
> > > >> >>> >> _______________________________________________
> > > >> >>> >> FreeSWITCH-users mailing list
> > > >> >>> >> FreeSWITCH-users at lists.freeswitch.org
> > > >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >> >>> >>
> > > >> >>> >>
> > > >> >>> >> UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> >>> >> http://www.freeswitch.org
> > > >> >>>
> > > >> >>> _______________________________________________
> > > >> >>> FreeSWITCH-users mailing list
> > > >> >>> FreeSWITCH-users at lists.freeswitch.org
> > > >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >> >>>
> > > >> >>> UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> >>> http://www.freeswitch.org
> > > >> >>
> > > >> >> --
> > > >> >> Anthony Minessale II
> > > >> >>
> > > >> >> FreeSWITCH http://www.freeswitch.org/
> > > >> >> ClueCon http://www.cluecon.com/
> > > >> >> Twitter: http://twitter.com/FreeSWITCH_wire
> > > >> >>
> > > >> >> AIM: anthm
> > > >> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail
> > > >> >>.c om>
> > > >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.
> > > >> >>mi nessale at gmail.com> IRC: irc.freenode.net #freeswitch
> > > >> >>
> > > >> >> FreeSWITCH Developer Conference
> > > >> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch
> > > >> >>.o rg> iax:guest at conference.freeswitch.org/888
> > > >> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2
> > > >> >>B8 88 at conference.freeswitch.org> pstn:+19193869900
> > > >> >
> > > >> > --
> > > >> > Anthony Minessale II
> > > >> >
> > > >> > FreeSWITCH http://www.freeswitch.org/
> > > >> > ClueCon http://www.cluecon.com/
> > > >> > Twitter: http://twitter.com/FreeSWITCH_wire
> > > >> >
> > > >> > AIM: anthm
> > > >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.
> > > >> >co m>
> > > >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.m
> > > >> >in essale at gmail.com> IRC: irc.freenode.net #freeswitch
> > > >> >
> > > >> > FreeSWITCH Developer Conference
> > > >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.
> > > >> >or g> iax:guest at conference.freeswitch.org/888
> > > >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B
> > > >> >88 8 at conference.freeswitch.org> pstn:+19193869900
> > > >> >
> > > >> > _______________________________________________
> > > >> > FreeSWITCH-users mailing list
> > > >> > FreeSWITCH-users at lists.freeswitch.org
> > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >> > UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> > http://www.freeswitch.org
> > > >>
> > > >> _______________________________________________
> > > >> FreeSWITCH-users mailing list
> > > >> FreeSWITCH-users at lists.freeswitch.org
> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >> UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> http://www.freeswitch.org
> > > >
> > > > --
> > > > Anthony Minessale II
> > > >
> > > > FreeSWITCH http://www.freeswitch.org/
> > > > ClueCon http://www.cluecon.com/
> > > > Twitter: http://twitter.com/FreeSWITCH_wire
> > > >
> > > > AIM: anthm
> > > > MSN:anthony_minessale at hotmail.com
> > > > <MSN%3Aanthony_minessale at hotmail.com>
> > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.mine
> > > >ss ale at gmail.com> IRC: irc.freenode.net #freeswitch
> > > >
> > > > FreeSWITCH Developer Conference
> > > > sip:888 at conference.freeswitch.org
> > > > <sip%3A888 at conference.freeswitch.org>
> > > > iax:guest at conference.freeswitch.org/888
> > > > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888
> > > >@c onference.freeswitch.org> pstn:+19193869900
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
> > > >er s http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From stevesteffler at shaw.ca  Fri Jan 15 20:36:03 2010
From: stevesteffler at shaw.ca (Steve Steffler)
Date: Fri, 15 Jan 2010 21:36:03 -0700
Subject: [Freeswitch-users] FIFO Originate caller ID
In-Reply-To: <88D85720-C24A-43B8-A6E8-7F9179D71625@jerris.com>
References: <29507085-AE04-4A43-A41A-6D279810CCEC@shaw.ca>
	<88D85720-C24A-43B8-A6E8-7F9179D71625@jerris.com>
Message-ID: <D85EDE43-C1D0-438C-A273-0016856CA723@shaw.ca>


Mike,

That explanation makes sense - thanks.

Perhaps a n00b question, but where can I configure the caller ID for the outbound queue calls?

Thanks,
Steve

On Jan 15, 2010, at 5:01 PM, Michael Jerris wrote:

> At the time a call goes out to the agents, there is no specific caller they are matched too, therefore there is no way to know the caller id at this time.  When the originated call to the agent is answered, we THEN go and pick off the next caller to connect them with.  All you can do is set a caller id for the queue.
> 
> Mike
> 
> On Jan 15, 2010, at 6:43 PM, Steve Steffler wrote:
> 
>> Hello,
>> 
>> I found an archived conversation on this list regarding FIFO origination caller ID, and how to modify it.  
>> 
>> http://old.nabble.com/FIFO-Orgination_caller_id-td26487628.html
>> 
>> There seems to be no easy way to customize the caller ID on originated calls from the FIFO to on-hook agents who were registered dynamically.  Anthony states a method to do it using static entries in the fifo conf file, and the general rationale is that SCREEN POPS be the preferred method, with the added nudge that good SIP phones can change the caller ID when the bridge is complete as well, which is all well and good.
>> 
>> The problem is for my application, all on-hook agents are using cellular phones, and they register dynamically.  Also, none of my agents are in front of a computer, so a SIP display update on the phone or screen pop on the computer in front of them is not really an option, and the only way they can identify calls from my FIFO right now is because they are the ones with NO CALLER ID (in other words, their mobile phones do not display the name, and the number is not recognized because it is set by FreeSWITCH to be "fifo+fifoname" instead of being numeric.  This is far from ideal.
>> 
>> I am wondering if there is anyone on the list who knows how to configure the origination_caller_id_number/name variables for dynamically registered on-hook agents so that the caller ID from the FIFO customer's incoming call is displayed to them instead of the above mangled caller ID?
>> 
>> I'm not disagreeing that it is an old-skewl way of thought, but in actuality it is just a way to interface with old-school telephony devices (i.e. non-Smartphone mobile phones) and I am not sure how to accomplish this.  Any help/input would be appreciated.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Fri Jan 15 20:56:10 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 Jan 2010 23:56:10 -0500
Subject: [Freeswitch-users] multiple leg and multiple rtp
In-Reply-To: <201001152027.07209.sos@sokhapkin.dyndns.org>
References: <40909765-9D35-423E-BAB0-1C61D65C3CA5@gmail.com>
	<191c3a031001151644m6a838b6bp311f2a173fc65af3@mail.gmail.com>
	<201001152004.25949.sos@sokhapkin.dyndns.org>
	<201001152027.07209.sos@sokhapkin.dyndns.org>
Message-ID: <84089307-723F-4910-9874-CFE09147B13E@jerris.com>

Well, in the dialplan, typically your not going to get past the bridge  
line in your dialplan because the call is done when the bridge is  
over.  But as Anthony noted you can set vars in [. ] that only apply  
to the b legs.  These you can get in the cdr or in an api hangup hook  
or otherwise in the reporting state.

Mike

On Jan 15, 2010, at 8:27 PM, Sergey Okhapkin  
<sos at sokhapkin.dyndns.org> wrote:

> Same question about
>
> originate aaa|bbb|ccc
>
> syntax :-)
>
>
> On Friday 15 January 2010, Sergey Okhapkin wrote:
>> Sorry for the dumb question, is there a way to find out in dialplan  
>> (some
>> variable?) which one of comma separated b-legs listed in originate  
>> command
>> answered the call?
>>
>> originate sofia/g1/number,sofia/g2/number,sofia/g3/number
>>
>> Which gateway answered? g1, g2 or g3?
>>
>> On Friday 15 January 2010, Anthony Minessale wrote:
>>> Now we need a new feature
>>>
>>> [leg_required=true]
>>>
>>> set this on any legs required for the originate to proceed, if it's
>>> hungup, the cause will be passed to any existing legs and fail the  
>>> entire
>>> originate.
>>>
>>> so use
>>>
>>> {bridge_early_media=true}[leg_required=true]sofia/internal/ 
>>> real_call at foo.
>>> co m ,sofia/internal/moh_call at foo.com
>>>
>>> the leg_required will only be set on the 1st leg because of the []  
>>> vs {}
>>> if that leg is then hungup, it will kill the other channels in the  
>>> list.
>>>
>>> please try latest trunk.
>>>
>>>
>>>
>>> On Fri, Jan 15, 2010 at 5:25 PM, David Villasmil <
>>>
>>> david.villasmil.work at gmail.com> wrote:
>>>> Anthony,
>>>>
>>>> LOL, and mounting and mounting... It does work when there is  
>>>> answer...
>>>> but if B(2)-side rejects or times out or any other that 200 OK,
>>>> B(1)-side stays indefinitely...
>>>>
>>>>
>>>> On Fri, Jan 15, 2010 at 11:54 PM, Anthony Minessale
>>>>
>>>> <anthony.minessale at gmail.com> wrote:
>>>>> you can email me privately at this addr.
>>>>>
>>>>>
>>>>> On Fri, Jan 15, 2010 at 4:45 PM, David Villasmil
>>>>>
>>>>> <david.villasmil.work at gmail.com> wrote:
>>>>>> Anthony,
>>>>>>
>>>>>>    Trying, Thanks. Is there anyway we can communicate directly?
>>>>>>
>>>>>>
>>>>>> David
>>>>>>
>>>>>> On Fri, Jan 15, 2010 at 10:17 PM, Anthony Minessale
>>>>>>
>>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>>> Try latest trunk,
>>>>>>>
>>>>>>> you should have exactly what you want with the same parameter,
>>>>>>> again
>>>>
>>>> my
>>>>
>>>>>>> paypal addr is cleary displayed as a big button on the website.
>>>>>>>
>>>>>>>
>>>>>>> On Fri, Jan 15, 2010 at 10:08 AM, Anthony Minessale
>>>>>>>
>>>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>>>> one of the many reasons its a bad idea.
>>>>>>>> Probably the leg with the bad audio is a different ptime.
>>>>>>>> Now the amount of work I have to do escalates I would prefer  
>>>>>>>> you
>>>>
>>>> commit
>>>>
>>>>>>>> to
>>>>>>>> commercial support by emailing me at  
>>>>>>>> consulting at freeswitch.org to
>>>>>>>> continue
>>>>>>>> with this.
>>>>>>>>
>>>>>>>>
>>>>>>>> On Fri, Jan 15, 2010 at 6:43 AM, David Villasmil
>>>>>>>>
>>>>>>>> <david.villasmil.work at gmail.com> wrote:
>>>>>>>>> I set it to "off" just in case, same thing.
>>>>>>>>>
>>>>>>>>> On Fri, Jan 15, 2010 at 1:38 PM, David Villasmil
>>>>>>>>>
>>>>>>>>> <david.villasmil.work at gmail.com> wrote:
>>>>>>>>>> Default, haven't touched it i suppose it's off, i haven't set
>>>>>>>>>> it anywhere
>>>>>>>>>>
>>>>>>>>>> On Fri, Jan 15, 2010 at 1:26 PM, Sergey Okhapkin
>>>>>>>>>>
>>>>>>>>>> <sos at sokhapkin.dyndns.org> wrote:
>>>>>>>>>>> Is bypass_media on or off?
>>>>>>>>>>>
>>>>>>>>>>> On Friday 15 January 2010, David Villasmil wrote:
>>>>>>>>>>>> Yeah, sorry. Early media. Audio after 200 is fine. Early
>>>>>>>>>>>> media
>>>>
>>>> was
>>>>
>>>>>>>>>>>> ok
>>>>>>>>>>>> before the change.
>>>>>>>>>>>>
>>>>>>>>>>>> David
>>>>>>>>>>>>
>>>>>>>>>>>> On Fri, Jan 15, 2010 at 12:51 PM, Sergey Okhapkin
>>>>>>>>>>>>
>>>>>>>>>>>> <sos at sokhapkin.dyndns.org> wrote:
>>>>>>>>>>>>> Which audio? Early media or after 200 OK?
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Friday 15 January 2010, David Villasmil wrote:
>>>>>>>>>>>>>> Hello again Anthony,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I just tested it, and although functionality does not,
>>>>>>>>>>>>>> first incoming
>>>>>>>>>>>>>> audio is coming in all garbled... do you know why?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> David
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>>>>>>>>>>> {bridge_early_media=true}
>>>>>>>>>>>>>>> in the dial string in place of ignore_early_media=true
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> the first b leg in the list who sends 183 will become
>>>>>>>>>>>>>>> the ringback
>>>>>>>>>>>>>>> device for A leg it will hear the early media
>>>>>>>>>>>>>>> for that leg while the other legs still ring.  If some
>>>>
>>>> other
>>>>
>>>>>>>>>>>>>>> leg
>>>>>>>>>>>>>>> answers the final call will still be bridged to the leg
>>>>>>>>>>>>>>> who answered.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> I would estimate it at $500 payable on the big paypal
>>>>
>>>> button
>>>>
>>>>>>>>>>>>>>> on
>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>> but, I already added the patch to tree earlier today so
>>>>>>>>>>>>>>> I guess
>>>>>>>>>>>>>>> it's
>>>>>>>>>>>>>>> up to you to pay it or not.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Thu, Jan 14, 2010 at 5:06 PM, David Villasmil
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> <david.villasmil.work at gmail.com> wrote:
>>>>>>>>>>>>>>>> Anthony,
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> I did take the "hint", don't worry. We will probably
>>>>>>>>>>>>>>>> ask
>>>>
>>>> for
>>>>
>>>>>>>>>>>>>>>> a
>>>>>>>>>>>>>>>> bounty
>>>>>>>>>>>>>>>> but first we need to know:
>>>>>>>>>>>>>>>> 1.- whether this is possible
>>>>>>>>>>>>>>>> 2.- how long it would take
>>>>>>>>>>>>>>>> 3.- how will it exactly work
>>>>>>>>>>>>>>>> 4.- of course, what's the bounty (be gentle ;) )
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> We would of course give this back to the community.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> in the meantime, bridge_early_media=true will discard
>>>>>>>>>>>>>>>> the 2nd
>>>>>>>>>>>>>>>> B-leg
>>>>>>>>>>>>>>>> and conversely "false" it will keep on trying to
>>>>>>>>>>>>>>>> connect
>>>>
>>>> and
>>>>
>>>>>>>>>>>>>>>> if
>>>>>>>>>>>>>>>> it
>>>>>>>>>>>>>>>> connects the other B-leg if will bridge to that one?
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Thanks
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> David
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> On Thu, Jan 14, 2010 at 10:04 PM, Anthony Minessale
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>>>>>>>>>>>>> I added bridge_early_media=true to do the best I can
>>>>>>>>>>>>>>>>> do. This is the most I will do, especially for free,
>>>>>>>>>>>>>>>>> nobody can
>>>>>>>>>>>>>>>>> take a
>>>>>>>>>>>>>>>>> hint that
>>>>>>>>>>>>>>>>> you should be paying for all these custom requests
>>>>>>>>>>>>>>>>> so
>>>>
>>>> take
>>>>
>>>>>>>>>>>>>>>>> it
>>>>>>>>>>>>>>>>> or
>>>>>>>>>>>>>>>>> leave it
>>>>>>>>>>>>>>>>> but this thread is done.........
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> On Thu, Jan 14, 2010 at 12:42 PM, Sergey Okhapkin
>>>>>>>>>>>>>>>>> <sos at sokhapkin.dyndns.org>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> wrote:
>>>>>>>>>>>>>>>>>> No, not exactly. ignore_early_media doesn't pass
>>>>>>>>>>>>>>>>>> early media
>>>>>>>>>>>>>>>>>> to
>>>>>>>>>>>>>>>>>> the caller
>>>>>>>>>>>>>>>>>> if
>>>>>>>>>>>>>>>>>> bypass_media is false.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> On Thursday 14 January 2010, Michael Jerris wrote:
>>>>>>>>>>>>>>>>>>> this is exactly what ignore_early_media does now.
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> Mike
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> On Jan 14, 2010, at 1:18 PM, Sergey Okhapkin
> wrote:
>>>>>>>>>>>>>>>>>>>> The issue here is when "originate" routine
>>>>>>>>>>>>>>>>>>>> should return
>>>>>>>>>>>>>>>>>>>> and
>>>>>>>>>>>>>>>>>>>> set "originate_status" variable. Current
>>>>>>>>>>>>>>>>>>>> behavior
>>>>
>>>> is
>>>>
>>>>>>>>>>>>>>>>>>>> to
>>>>>>>>>>>>>>>>>>>> return
>>>>>>>>>>>>>>>>>>>> on early
>>>>>>>>>>>>>>>>>>>> media, but what if to introduce a variable
>>>>>>>>>>>>>>>>>>>> "originate_wait_for_answer"
>>>>>>>>>>>>>>>>>>>> with default value "false" and use the variable
>>>>>>>>>>>>>>>>>>>> in originate
>>>>>>>>>>>>>>>>>>>> code to
>>>>>>>>>>>>>>>>>>>> decide when to return - on 18X or "200 OK"?
>>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>> On Thursday 14 January 2010, Anthony Minessale
>>>>
>>>> wrote:
>>>>>>>>>>>>>>>>>>>>> he wants to call 3 people at once and let the
>>>>>>>>>>>>>>>>>>>>> A
>>>>
>>>> leg
>>>>
>>>>>>>>>>>>>>>>>>>>> hear
>>>>>>>>>>>>>>>>>>>>> early media
>>>>>>>>>>>>>>>>>>>>> from call #1 while call #2 and #3 still are
>>>>>>>>>>>>>>>>>>>>> progressing
>>>>>>>>>>>>>>>>>>>>> which
>>>>>>>>>>>>>>>>>>>>> is not
>>>>>>>>>>>>>>>>>>>>> simple to do without doing thousands of
>>>>>>>>>>>>>>>>>>>>> dollars in development.
>>>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>>> On Thu, Jan 14, 2010 at 11:39 AM, DJB
>>>>>>>>>>>>>>>>>>>>> <djbinter at yahoo.com>
>>>>>>>>>>>
>>>>>>>>>>> wrote:
>>>>>>>>>>>>>>>>>>>>>> What about sending Sip 183 with SDP (no
>>>>>>>>>>>>>>>>>>>>>> 200OK),
>>>>
>>>> so
>>>>
>>>>>>>>>>>>>>>>>>>>>> that your
>>>>>>>>>>>>>>>>>>>>>> customers
>>>>>>>>>>>>>>>>>>>>>> can hear recordings?
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>>>>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswit
>>>>
>>>>>>>>>>>>>>>>>>> ch- users http://www.freeswitch.org
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>>>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch
>>>>
>>>>>>>>>>>>>>>>>> -us ers http://www.freeswitch.org
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>>>> Anthony Minessale II
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>>>>>>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>>>>>>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> AIM: anthm
>>>>>>>>>>>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_mine
>>>>>>>>>>>>>>>>> ss ale at hotmail.com>
>>>>>>>>>>>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYP
>>>>>>>>>>>>>>>>> AL %3Aanthony.minessale at gmail.com> IRC:
>>>>>>>>>>>>>>>>> irc.freenode.net #freeswitch
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> FreeSWITCH Developer Conference
>>>>>>>>>>>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conferen
>>>>>>>>>>>>>>>>> ce .freeswitch.org>
>>>>>>>>>>>>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>>>>>>>>>>>>> googletalk:conf+888 at conference.freeswitch.org<google
>>>>>>>>>>>>>>>>> ta lk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>>>>>>>>>>> pstn:+19193869900
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>
>>>>>>>>>>>>>>>>> use rs http://www.freeswitch.org
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-us
>>>>
>>>>>>>>>>>>>>>> ers http://www.freeswitch.org
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>> Anthony Minessale II
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>>>>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>>>>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> AIM: anthm
>>>>>>>>>>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessa
>>>>>>>>>>>>>>> le @hotmail.com>
>>>>>>>>>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%
>>>>>>>>>>>>>>> 3A anthony.minessale at gmail.com> IRC: irc.freenode.net
>>>>>>>>>>>>>>> #freeswitch
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> FreeSWITCH Developer Conference
>>>>>>>>>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.
>>>>>>>>>>>>>>> fr eeswitch.org> iax:guest at conference.freeswitch.org/888
>>>>>>>>>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletal
>>>>>>>>>>>>>>> k% 3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>>>>>>>>> pstn:+19193869900
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-use
>>>>
>>>>>>>>>>>>>>> rs http://www.freeswitch.org
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-us
>>>>>>>>>>>>> er s
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch- 
>>>>>>>>>>>> user
>>>>>>>>>>>> s
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch- 
>>>>>>>>>>> users
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>
>>>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>> --
>>>>>>>> Anthony Minessale II
>>>>>>>>
>>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>>
>>>>>>>> AIM: anthm
>>>>>>>> MSN:anthony_minessale at hotmail.com<MSN 
>>>>>>>> %3Aanthony_minessale at hotmail
>>>>>>>> .c om>
>>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL 
>>>>>>>> %3Aanthony.
>>>>>>>> mi nessale at gmail.com> IRC: irc.freenode.net #freeswitch
>>>>>>>>
>>>>>>>> FreeSWITCH Developer Conference
>>>>>>>> sip:888 at conference.freeswitch.org<sip 
>>>>>>>> %3A888 at conference.freeswitch
>>>>>>>> .o rg> iax:guest at conference.freeswitch.org/888
>>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk 
>>>>>>>> %3Aconf%2
>>>>>>>> B8 88 at conference.freeswitch.org> pstn:+19193869900
>>>>>>>
>>>>>>> --
>>>>>>> Anthony Minessale II
>>>>>>>
>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>>
>>>>>>> AIM: anthm
>>>>>>> MSN:anthony_minessale at hotmail.com<MSN 
>>>>>>> %3Aanthony_minessale at hotmail.
>>>>>>> co m>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL 
>>>>>>> %3Aanthony.m
>>>>>>> in essale at gmail.com> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>> sip:888 at conference.freeswitch.org<sip 
>>>>>>> %3A888 at conference.freeswitch.
>>>>>>> or g> iax:guest at conference.freeswitch.org/888
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf 
>>>>>>> %2B
>>>>>>> 88 8 at conference.freeswitch.org> pstn:+19193869900
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>
>>>>>> http://www.freeswitch.org
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>> <MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL 
>>>>> %3Aanthony.mine
>>>>> ss ale at gmail.com> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> <sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf 
>>>>> %2B888
>>>>> @c onference.freeswitch.org> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>>> freeswitch-us
>>>>> er s http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>> freeswitch-user
>>>> s http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From mike at jerris.com  Fri Jan 15 21:10:08 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 16 Jan 2010 00:10:08 -0500
Subject: [Freeswitch-users] FIFO Originate caller ID
In-Reply-To: <D85EDE43-C1D0-438C-A273-0016856CA723@shaw.ca>
References: <29507085-AE04-4A43-A41A-6D279810CCEC@shaw.ca>
	<88D85720-C24A-43B8-A6E8-7F9179D71625@jerris.com>
	<D85EDE43-C1D0-438C-A273-0016856CA723@shaw.ca>
Message-ID: <2CCD724A-65E4-4EB6-A813-50F26CC0642A@jerris.com>

You can put vars for manipulating callerid right in the originate  
string either in config or using the add for the on hook agents.

Mike

On Jan 15, 2010, at 11:36 PM, Steve Steffler <stevesteffler at shaw.ca>  
wrote:

>
> Mike,
>
> That explanation makes sense - thanks.
>
> Perhaps a n00b question, but where can I configure the caller ID for  
> the outbound queue calls?
>
> Thanks,
> Steve
>
> On Jan 15, 2010, at 5:01 PM, Michael Jerris wrote:
>
>> At the time a call goes out to the agents, there is no specific  
>> caller they are matched too, therefore there is no way to know the  
>> caller id at this time.  When the originated call to the agent is  
>> answered, we THEN go and pick off the next caller to connect them  
>> with.  All you can do is set a caller id for the queue.
>>
>> Mike
>>
>> On Jan 15, 2010, at 6:43 PM, Steve Steffler wrote:
>>
>>> Hello,
>>>
>>> I found an archived conversation on this list regarding FIFO  
>>> origination caller ID, and how to modify it.
>>>
>>> http://old.nabble.com/FIFO-Orgination_caller_id-td26487628.html
>>>
>>> There seems to be no easy way to customize the caller ID on  
>>> originated calls from the FIFO to on-hook agents who were  
>>> registered dynamically.  Anthony states a method to do it using  
>>> static entries in the fifo conf file, and the general rationale is  
>>> that SCREEN POPS be the preferred method, with the added nudge  
>>> that good SIP phones can change the caller ID when the bridge is  
>>> complete as well, which is all well and good.
>>>
>>> The problem is for my application, all on-hook agents are using  
>>> cellular phones, and they register dynamically.  Also, none of my  
>>> agents are in front of a computer, so a SIP display update on the  
>>> phone or screen pop on the computer in front of them is not really  
>>> an option, and the only way they can identify calls from my FIFO  
>>> right now is because they are the ones with NO CALLER ID (in other  
>>> words, their mobile phones do not display the name, and the number  
>>> is not recognized because it is set by FreeSWITCH to be "fifo 
>>> +fifoname" instead of being numeric.  This is far from ideal.
>>>
>>> I am wondering if there is anyone on the list who knows how to  
>>> configure the origination_caller_id_number/name variables for  
>>> dynamically registered on-hook agents so that the caller ID from  
>>> the FIFO customer's incoming call is displayed to them instead of  
>>> the above mangled caller ID?
>>>
>>> I'm not disagreeing that it is an old-skewl way of thought, but in  
>>> actuality it is just a way to interface with old-school telephony  
>>> devices (i.e. non-Smartphone mobile phones) and I am not sure how  
>>> to accomplish this.  Any help/input would be appreciated.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From stevesteffler at shaw.ca  Fri Jan 15 21:17:29 2010
From: stevesteffler at shaw.ca (Steve Steffler)
Date: Fri, 15 Jan 2010 22:17:29 -0700
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
Message-ID: <776BA929-D7EC-4366-AF49-348F0E11488B@shaw.ca>


Costa,

I wrote this script to handle fax2email (but not email2fax).  It uses variables you set in the dialplan in advance for the email address for that fax DID.

http://steffler.info/wp-content/uploads/2009/06/process-rxfax.py

Regards,
Steve

On Jan 14, 2010, at 7:54 AM, Costa Zikalala wrote:

> Hi All
> 
> Has anyone worked on a email2fax script for mod_fax? 
> If not how much would it cost for some genius here to quickly whip-up one?
> 
> Ideally both email2fax and fax2email should come standard with mod_fax or is this not possible?
> 
> Thanks 
> Costa
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From yehavi.bourvine at gmail.com  Fri Jan 15 21:57:07 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 16 Jan 2010 07:57:07 +0200
Subject: [Freeswitch-users] Polycom configuration for FreeSWITCH
In-Reply-To: <003701ca964b$3241b100$96c51300$@net>
References: <003701ca964b$3241b100$96c51300$@net>
Message-ID: <d361ca7b1001152157x45b8d8c4l29625c763b24157f@mail.gmail.com>

I am working with various Polycom phones; I'll send you sample configuration
files next week (I am at home now). In the meantime, please send me your
requirenents so I may incorporate some of them into the files.

Have you managed to boot them from your TFTP/FTP./HTTP server? As long as
you did not provision them through a server you can do that through the
phone's WEB interface, but it is very limited and lacks a lot of
configuration options. I do the provisioning via a TFTP server.

                    Regards, __Yehavi:

 2010/1/16 Adam Ford <lists at redbonez.net>

>   Does anyone who has successfully implemented Polycom IP301/501s, with
> FreeSWITCH, have a configuration they can share with me as an example? I
> have read the Polycom Admin Guide several times and understand what the
> settings are/do, I am just not sure which FreeSWITCH supports, which it
> doesn?t, and which need special configuration to work with FreeSWITCH.
>
>
>
> Thanks in advance,
>
> -Adam
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Fri Jan 15 22:16:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 16 Jan 2010 01:16:18 -0500
Subject: [Freeswitch-users] Polycom configuration for FreeSWITCH
In-Reply-To: <d361ca7b1001152157x45b8d8c4l29625c763b24157f@mail.gmail.com>
References: <003701ca964b$3241b100$96c51300$@net>
	<d361ca7b1001152157x45b8d8c4l29625c763b24157f@mail.gmail.com>
Message-ID: <84F040CA-3FA1-4A5D-A417-B4C4F1B21E43@jerris.com>

If people have phone config examples, a good place to share them is on  
the wiki.

Mike

On Jan 16, 2010, at 12:57 AM, Yehavi Bourvine  
<yehavi.bourvine at gmail.com> wrote:

> I am working with various Polycom phones; I'll send you sample  
> configuration files next week (I am at home now). In the meantime,  
> please send me your requirenents so I may incorporate some of them  
> into the files.
>
> Have you managed to boot them from your TFTP/FTP./HTTP server? As  
> long as you did not provision them through a server you can do that  
> through the phone's WEB interface, but it is very limited and lacks  
> a lot of configuration options. I do the provisioning via a TFTP  
> server.
>
>                     Regards, __Yehavi:
>
> 2010/1/16 Adam Ford <lists at redbonez.net>
> Does anyone who has successfully implemented Polycom IP301/501s,  
> with FreeSWITCH, have a configuration they can share with me as an  
> example? I have read the Polycom Admin Guide several times and  
> understand what the settings are/do, I am just not sure which  
> FreeSWITCH supports, which it doesn?t, and which need special config 
> uration to work with FreeSWITCH.
>
>
>
> Thanks in advance,
>
> -Adam
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From kawarod at laposte.net  Fri Jan 15 22:22:53 2010
From: kawarod at laposte.net (rod)
Date: Sat, 16 Jan 2010 10:22:53 +0400
Subject: [Freeswitch-users] Eavesdrop in LUA
In-Reply-To: <007201ca95bb$ba673770$2f35a650$@com>
References: <4B4ED32E.30706@laposte.net> <4B4F33C7.6020403@laposte.net>
	<007201ca95bb$ba673770$2f35a650$@com>
Message-ID: <4B515B3D.9020309@laposte.net>

Hi Pete,

to get the beginning of the communication, I find this:
use the pre-answer/ringback command before bridging the call, this will 
issue a 183 with ringback in RTP.  Doing this, eavesdrop application  
can listen  the  caller  A talking  in the phone (and the ringback tone) 
even if the call is not connected to B :o

in the dialplan I did this:
<action application="set" data="ringback=$${fr-ring}"/>
<action application="pre_answer"/>
<action application="sleep" data="500"/>
<actionapplication="bridge"data="{ecoute=${caller_id_number}}sofia/gateway/${caller_id_number}/${destination_numbe\
r}"/>

A colleague wrote a perl script using mod_event that looks for when a 
call that should be eavesdrop is connecting and originate a call to C 
using eavesdrop on A leg. I will ask my colleague/boss if he's okay to 
share his script on the wiki. I think, he'll be okay, but prefer asking 
before.

Give a try to pre-anwer with lua, and let me know if you could eavesdrop 
to a call before the call is exchanging media.
Are you ok to share your lua script, so that we could document the 
eavesdrop page ?

regards,
rod

Pete Mueller a ?crit :
> I had a similar problem.  I solved it by first making bridging the call
> between A and B.
> Then originate C with a LUA script, the last line of which is:
>
> session:execute("eavesdrop", uuid_of_a_leg)
>
> The down side here is that A and B can talk while C is ringing, but in my
> case that is not a problem.
> -p
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod
> Sent: Thursday, January 14, 2010 8:10 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Eavesdrop in LUA
>
> Hi all,
>
> I have an incomplete solution for those interested.
>
> I did it like this in dialplan:
>         <action application="set" data="api_on_answer=originate 
> sofia/internal/2000 at 10.10.10.10 &eavesdrop(${uuid})"/>-->
>         <action application="bridge" 
> data="{ecoute=${caller_id_number}}sofia/gateway/${caller_id_number}/${destin
> ation_number}"/>
>
> so when a call is setup, FS initiate a new call to 2000 and eavesdrop 
> the call.
> But I have a small problem, the callee receives no sound until the 
> eavesdropper send a SIP reply, so there is a 2-3 seconds delay before 
> caller and callee can talk each other.
>
> rod
>
>
> rod a ?crit :
>   
>> Hi all,
>>
>> I'm trying to do this in LUA:
>>      A call B
>>
>> and I'd like to setup a new call to C with eavesdrop of A conversation 
>> with B.
>>
>> I have no idea how to do this if someone can help.
>> I switched to LUA cause I see no way to achieve this with dialplan 
>> (snippets are welcome).
>>
>> regards,
>> rod
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>   
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   



From woodydickson at gmail.com  Fri Jan 15 22:36:32 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 16 Jan 2010 14:36:32 +0800
Subject: [Freeswitch-users] ODBC Not Available!
Message-ID: <a44688a1001152236y57f8fe8bk5aced53cfd41e425@mail.gmail.com>

Hello,

I am trying to get ODBC within the voicemail module to work, but I am
getting the following error:

2010-01-16 14:32:32.584124 [CRIT] switch_core_sqldb.c:306 Failure! OBDC NOT
AVAILABLE!
2010-01-16 14:32:32.584124 [ERR] mod_voicemail.c:214 Error Opening DB


[root at e-d freeswitch]# isql my_odbc
+---------------------------------------+
| Connected!                            |
|                                       |
| sql-statement                         |
| help [tablename]                      |
| quit                                  |
|                                       |
+---------------------------------------+

In my voicemail.conf.xml, I have:
      <param name="odbc-dsn" value="my_odbc:root:123456"/>


Does anyone know what may be wrong with my config?

Thanks for your help.

woody
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From mike at jerris.com  Fri Jan 15 23:15:49 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 16 Jan 2010 02:15:49 -0500
Subject: [Freeswitch-users] ODBC Not Available!
In-Reply-To: <a44688a1001152236y57f8fe8bk5aced53cfd41e425@mail.gmail.com>
References: <a44688a1001152236y57f8fe8bk5aced53cfd41e425@mail.gmail.com>
Message-ID: <6461720E-23F5-4C6B-9D77-3042A853F14D@jerris.com>

It looks like freeswitch was built without odbc support.  Install odbc  
libs and dev packages, before you run configure to build with odbc  
support.

Mike

On Jan 16, 2010, at 1:36 AM, Woody Dickson <woodydickson at gmail.com>  
wrote:

> Hello,
>
> I am trying to get ODBC within the voicemail module to work, but I  
> am getting the following error:
>
> 2010-01-16 14:32:32.584124 [CRIT] switch_core_sqldb.c:306 Failure!  
> OBDC NOT AVAILABLE!
> 2010-01-16 14:32:32.584124 [ERR] mod_voicemail.c:214 Error Opening DB
>
>
> [root at e-d freeswitch]# isql my_odbc
> +---------------------------------------+
> | Connected!                            |
> |                                       |
> | sql-statement                         |
> | help [tablename]                      |
> | quit                                  |
> |                                       |
> +---------------------------------------+
>
> In my voicemail.conf.xml, I have:
>       <param name="odbc-dsn" value="my_odbc:root:123456"/>
>
>
> Does anyone know what may be wrong with my config?
>
> Thanks for your help.
>
> woody
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From sharad at coraltele.com  Fri Jan 15 23:27:40 2010
From: sharad at coraltele.com (Sharad)
Date: Fri, 15 Jan 2010 23:27:40 -0800 (PST)
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
Message-ID: <1263626860891-4403364.post@n2.nabble.com>


We also would like to work on the same. Plz let me know all the possibilities
-

1.  Email body to be sent as fax body.
2.  What about attached document to email ?
3.  Covering page ?

Plz let us know the possibilities, so that me a7 my team can start the work.

regards
sharad




Aza1 wrote:
> 
> Hi All
> 
> Has anyone worked on a email2fax script for mod_fax?
> If not how much would it cost for some genius here to quickly whip-up one?
> 
> Ideally both email2fax and fax2email should come standard with mod_fax or
> is
> this not possible?
> 
> Thanks
> Costa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/Mod-Fax-tp4393083p4403364.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From sharad at coraltele.com  Fri Jan 15 23:36:37 2010
From: sharad at coraltele.com (Sharad)
Date: Fri, 15 Jan 2010 23:36:37 -0800 (PST)
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <1263626860891-4403364.post@n2.nabble.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
	<1263626860891-4403364.post@n2.nabble.com>
Message-ID: <1263627397511-4403377.post@n2.nabble.com>


Just adding one more thing that first we are getting this done from a web
page...



Sharad wrote:
> 
> We also would like to work on the same. Plz let me know all the
> possibilities -
> 
> 1.  Email body to be sent as fax body.
> 2.  What about attached document to email ?
> 3.  Covering page ?
> 
> Plz let us know the possibilities, so that me a7 my team can start the
> work.
> 
> regards
> sharad
> 
> 
> 
> 
> Aza1 wrote:
>> 
>> Hi All
>> 
>> Has anyone worked on a email2fax script for mod_fax?
>> If not how much would it cost for some genius here to quickly whip-up
>> one?
>> 
>> Ideally both email2fax and fax2email should come standard with mod_fax or
>> is
>> this not possible?
>> 
>> Thanks
>> Costa
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 

-- 
View this message in context: http://n2.nabble.com/Mod-Fax-tp4393083p4403377.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From peder at networkoblivion.com  Sat Jan 16 05:08:35 2010
From: peder at networkoblivion.com (Peder)
Date: Sat, 16 Jan 2010 07:08:35 -0600
Subject: [Freeswitch-users] Polycom configuration for FreeSWITCH
In-Reply-To: <003701ca964b$3241b100$96c51300$@net>
References: <003701ca964b$3241b100$96c51300$@net>
Message-ID: <052c01ca96ad$02d7b4c0$08871e40$@com>

It is usually better if you ask a specific question, rather than just a
general "how do I configure it".  Freeswitch supports standard registration,
shared line, SCA ( thanks to a lot of work by the dev team recently), one
touch voicemail, mwi, conference, transfer, blind transfer, hold, DND, etc.


 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam
Ford
Sent: Friday, January 15, 2010 7:28 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Polycom configuration for FreeSWITCH

 

Does anyone who has successfully implemented Polycom IP301/501s, with
FreeSWITCH, have a configuration they can share with me as an example? I
have read the Polycom Admin Guide several times and understand what the
settings are/do, I am just not sure which FreeSWITCH supports, which it
doesn't, and which need special configuration to work with FreeSWITCH.

 

Thanks in advance,

-Adam

 

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From math.parent at gmail.com  Sat Jan 16 05:40:59 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Sat, 16 Jan 2010 14:40:59 +0100
Subject: [Freeswitch-users] Email2Pdf
In-Reply-To: <59daa2cd1001141603w462cdc84pb8fa721f5924cabc@mail.gmail.com>
References: <59daa2cd1001141603w462cdc84pb8fa721f5924cabc@mail.gmail.com>
Message-ID: <960738411001160540j197dfc5ay86f5e2e5990550be@mail.gmail.com>

On Fri, Jan 15, 2010 at 1:03 AM, Costa Zikalala
<costa.zikalala at gmail.com> wrote:
> Hi Mathieu
>
> Thank you for your very wonderful script.

You are talking about
http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/sathieu/email2pdf/

> I'm just trying understand it a bit and am not too good with Perl.
> Kindly confirm the folowing for me:
> - Only attachments and not the whole email is converted to PDF?

No, the message body should also be converted (in the script, this is
call $header ;).

> - Is the PDF output saved in the variable $file_out?
Yes, or to stdout, if $file_out is "-". more info with:
./email2pdf --man

> - Will you be completing your script as Email2Fax for Freeswitch?
The remaining thing will be added to FS wiki (don't have time now), as
this is site-specific:
- postfix config (see http://hylafax.sourceforge.net/howto/faxing.php#ss5.4)
- script email2fax:
 + guess From: fax number from the email
 + run email2pdf
 + convert to tiff
 + call txfax within FS

Cheers

Mathieu Parent

> Thanks again,
> Costa
>
>



From mailinglist at fribert.dk  Sat Jan 16 05:50:45 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Sat, 16 Jan 2010 14:50:45 +0100
Subject: [Freeswitch-users] How do I carry dial-in number to extension?
Message-ID: <4B51D245020000E1000003C0@mail.fribert.dk>

I have two DID's registered at my SIP.
This works nicely, I've created groups for holding the local extensions, and one of my phones can subscribe to two SIP accounts, so I can distinguish between if it's one or the other DID (it will show which SIP account is called in the display).
 
Problem is, another one of the phones can only subscribe to one SIP, and has no display (linksys sipura 901), any idea how to accomplish that I can distinnguish between the DID's on that one?
 
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From mike at jerris.com  Sat Jan 16 07:53:29 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 16 Jan 2010 10:53:29 -0500
Subject: [Freeswitch-users] How do I carry dial-in number to extension?
In-Reply-To: <4B51D245020000E1000003C0@mail.fribert.dk>
References: <4B51D245020000E1000003C0@mail.fribert.dk>
Message-ID: <54E291D4-6954-49DD-91FC-50F927C89532@jerris.com>



On Jan 16, 2010, at 8:50 AM, "mailinglist" <mailinglist at fribert.dk>  
wrote:

> I have two DID's registered at my SIP.

At your sip what?

> This works nicely, I've created groups for holding the local  
> extensions, and one of my phones can subscribe to two SIP accounts,  
> so I can distinguish between if it's one or the other DID (it will  
> show which SIP account is called in the display).
>

Are you talking about a phone registering or subscribing to freeswitch  
here?

> Problem is, another one of the phones can only subscribe to one SIP,  
> and has no display (linksys sipura 901), any idea how to accomplish  
> that I can distinnguish between the DID's on that one?
>

So if I understand you want to make a single line no display phone and  
send it multiple dids to it and be able to tell what did called?   
Really, the phone has no way to display this, but you could play a  
sound when they answer the phone.

Mike

>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From dome at tel.co.th  Sat Jan 16 09:17:02 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 17 Jan 2010 00:17:02 +0700
Subject: [Freeswitch-users] How to register from FS to Huawei SoftX3000
Message-ID: <8ccbff061001160917m22edfed8n291ea8a07518e4f7@mail.gmail.com>

Dear sir,
          I found some provider use Huawei SoftX3000 and can limit use
call from they softphone only. (use eyeball SDK).
They can limit some account can register and call by sip server like
an FS and Asterisk. but some account can't. (register and call by
softphone). and i don't know how they can do that.
So i try to use wireshark to debug sip headeder when use softphone
with both account type. it's nothing diferent.
          I want to use both account work by FS register to  Huawei
SoftX3000. Can someone help me. i can give you softphone and both
account type for test.


Best Regards.

Dome C.



From jmesquita at freeswitch.org  Sat Jan 16 09:17:09 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 16 Jan 2010 14:17:09 -0300
Subject: [Freeswitch-users] RTCP information
In-Reply-To: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE09AC568494@SBS2008SERVER.consiglia.local>
References: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE09AC568494@SBS2008SERVER.consiglia.local>
Message-ID: <ebbc2d551001160917n7b0278dcv4b4c0bf30dac3a91@mail.gmail.com>

No, there isn't..

Jo?o Mesquita


On Fri, Jan 15, 2010 at 2:02 PM, Jon Bruel <jbr at consiglia.dk> wrote:

>  In a real setup with 5-20 VoIP calls a day, every now and then there are
> some problems with sound quality, and I need some tools to investigate the
> cause.
>
> The phones support RTCP, and I would like to hear if I can get the FS to
> relay those packets to some kind of logger, including the signalling
> information?
>
> /Jon
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sat Jan 16 10:27:28 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 16 Jan 2010 13:27:28 -0500
Subject: [Freeswitch-users] How to register from FS to Huawei SoftX3000
In-Reply-To: <8ccbff061001160917m22edfed8n291ea8a07518e4f7@mail.gmail.com>
References: <8ccbff061001160917m22edfed8n291ea8a07518e4f7@mail.gmail.com>
Message-ID: <77F7CA9E-060E-4645-83D9-034B46C1C843@jerris.com>

Try changing the user agent, thats the only thing I can think they would be using.

Mike

On Jan 16, 2010, at 12:17 PM, Dome Charoenyost wrote:

> Dear sir,
>          I found some provider use Huawei SoftX3000 and can limit use
> call from they softphone only. (use eyeball SDK).
> They can limit some account can register and call by sip server like
> an FS and Asterisk. but some account can't. (register and call by
> softphone). and i don't know how they can do that.
> So i try to use wireshark to debug sip headeder when use softphone
> with both account type. it's nothing diferent.
>          I want to use both account work by FS register to  Huawei
> SoftX3000. Can someone help me. i can give you softphone and both
> account type for test.
> 




From lawwton at gmail.com  Sat Jan 16 11:58:44 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 16 Jan 2010 14:58:44 -0500
Subject: [Freeswitch-users] reloadxml question
Message-ID: <5fe6fa8f1001161158t54300ae5r62845f3baa1f97f5@mail.gmail.com>

All:

I am starting to play a bit with FS to get more familiar with it. I am
trying the following:

1- sofia status  // command for example to see the list of profiles.

2- I then went ahead and renamed the example.com.xml under
conf/directory/default to example.com.xml.orig

3- I ran then reloadxml using the fs_cli

4- sofia status still displays the same info as before for
example.conf as a gateway.

I then went ahead and stopped FW and re-started it. After I did this I
re-ran sofia status and this time the example.conf gateway that was
previously listed went away as it was supposed to.

Am I missing something? Is reloadxml the proper way to reload
configuration changes. If it is why is it not working for this case?

Thanks in advance,

Alfredo



From costa.zikalala at gmail.com  Sat Jan 16 11:58:47 2010
From: costa.zikalala at gmail.com (Costa Zikalala)
Date: Sat, 16 Jan 2010 21:58:47 +0200
Subject: [Freeswitch-users] Email2Pdf
In-Reply-To: <960738411001160540j197dfc5ay86f5e2e5990550be@mail.gmail.com>
References: <59daa2cd1001141603w462cdc84pb8fa721f5924cabc@mail.gmail.com> 
	<960738411001160540j197dfc5ay86f5e2e5990550be@mail.gmail.com>
Message-ID: <59daa2cd1001161158h51629821u80afa1168d7be796@mail.gmail.com>

This will be a very valuable addition to the Mod_Fax Wiki.

Thanks



2010/1/16 Mathieu Parent <math.parent at gmail.com>

> On Fri, Jan 15, 2010 at 1:03 AM, Costa Zikalala
> <costa.zikalala at gmail.com> wrote:
> > Hi Mathieu
> >
> > Thank you for your very wonderful script.
>
> You are talking about
> http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/sathieu/email2pdf/
>
> > I'm just trying understand it a bit and am not too good with Perl.
> > Kindly confirm the folowing for me:
> > - Only attachments and not the whole email is converted to PDF?
>
> No, the message body should also be converted (in the script, this is
> call $header ;).
>
> > - Is the PDF output saved in the variable $file_out?
> Yes, or to stdout, if $file_out is "-". more info with:
> ./email2pdf --man
>
> > - Will you be completing your script as Email2Fax for Freeswitch?
> The remaining thing will be added to FS wiki (don't have time now), as
> this is site-specific:
> - postfix config (see
> http://hylafax.sourceforge.net/howto/faxing.php#ss5.4)
> - script email2fax:
>  + guess From: fax number from the email
>  + run email2pdf
>  + convert to tiff
>  + call txfax within FS
>
> Cheers
>
> Mathieu Parent
>
> > Thanks again,
> > Costa
> >
> >
>
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From costa.zikalala at gmail.com  Sat Jan 16 12:05:22 2010
From: costa.zikalala at gmail.com (Costa Zikalala)
Date: Sat, 16 Jan 2010 22:05:22 +0200
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <776BA929-D7EC-4366-AF49-348F0E11488B@shaw.ca>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com> 
	<776BA929-D7EC-4366-AF49-348F0E11488B@shaw.ca>
Message-ID: <59daa2cd1001161205h3f6feeddla0be8e1fa9311666@mail.gmail.com>

Yes Steve, I'm already using that for fax2email.
I'm now trying to do things in the opposite direction.

*A realy great script by the way*

Thanks
Costa


2010/1/16 Steve Steffler <stevesteffler at shaw.ca>

>
> Costa,
>
> I wrote this script to handle fax2email (but not email2fax).  It uses
> variables you set in the dialplan in advance for the email address for that
> fax DID.
>
> http://steffler.info/wp-content/uploads/2009/06/process-rxfax.py
>
> Regards,
> Steve
>
> On Jan 14, 2010, at 7:54 AM, Costa Zikalala wrote:
>
> Hi All
>
> Has anyone worked on a email2fax script for mod_fax?
> If not how much would it cost for some genius here to quickly whip-up one?
>
> Ideally both email2fax and fax2email should come standard with mod_fax or
> is this not possible?
>
> Thanks
> Costa
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Sat Jan 16 12:16:18 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sat, 16 Jan 2010 15:16:18 -0500
Subject: [Freeswitch-users] reloadxml question
In-Reply-To: <5fe6fa8f1001161158t54300ae5r62845f3baa1f97f5@mail.gmail.com>
References: <5fe6fa8f1001161158t54300ae5r62845f3baa1f97f5@mail.gmail.com>
Message-ID: <3E8FB576-F49D-49A1-B3E2-8C6F5D3A6059@avgs.ca>

reloadxml preprocesses the xml config again, it does not tell modules  
to reload their configuration from that file.

In order to delete a gateway, you do: sofia profile <profilename>  
killgw <gwname>

If you make changes to a gateway and want to reload it, you first  
delete it with that command and then do:
	sofia profile <profilename> rescan

If you don't have any active calls on the box, restarting the profile  
will do the equivalent of killgw+rescan.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 16-Jan-10, at 2:58 PM, Alfredo Quiroga-Villamil wrote:

> All:
>
> I am starting to play a bit with FS to get more familiar with it. I am
> trying the following:
>
> 1- sofia status  // command for example to see the list of profiles.
>
> 2- I then went ahead and renamed the example.com.xml under
> conf/directory/default to example.com.xml.orig
>
> 3- I ran then reloadxml using the fs_cli
>
> 4- sofia status still displays the same info as before for
> example.conf as a gateway.
>
> I then went ahead and stopped FW and re-started it. After I did this I
> re-ran sofia status and this time the example.conf gateway that was
> previously listed went away as it was supposed to.
>
> Am I missing something? Is reloadxml the proper way to reload
> configuration changes. If it is why is it not working for this case?
>
> Thanks in advance,
>
> Alfredo
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mohamedezz.fci at gmail.com  Sat Jan 16 13:06:42 2010
From: mohamedezz.fci at gmail.com (Mohamed Hassan)
Date: Sat, 16 Jan 2010 23:06:42 +0200
Subject: [Freeswitch-users] No media after Originate
Message-ID: <f97004c31001161306o10517d1y9e0cf0229bce3589@mail.gmail.com>

Hi Freeswitch users
thank you in advance

as i am new to freeswitch iam facing a problem while writing a Callback service
i tried to bridge the two legs through one sip provider the call
estaplished successfully but without media passing
and after some searching i found that there is a bug with sofia
loopback so i shoulkd handle the two legs with different sip gateways
and after trying i found the same error


using cli :
originate {ignore_early_media=true}sofia/gateway/prov/00xxxxxxxxxxx
9999 XML callback

and here the 9999 context:
<context name="callback">
<extension name="9999">
<condition field="destination_number" expression="9999">
<action application="sleep" data="5000"/>
<action application="answer"/>
<action application="set" data="ringback=${us-ring}"/>
<action application="bridge"
data="[ignore_early_media=false]sofia/gateway/prov/00xxxxxxxxxx"/>
</condition>
</extension>
</context>



From mailinglist at fribert.dk  Sat Jan 16 13:33:30 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Sat, 16 Jan 2010 22:33:30 +0100
Subject: [Freeswitch-users] Svar: Re: How do I carry dial-in number to	
	extension?
In-Reply-To: <54E291D4-6954-49DD-91FC-50F927C89532@jerris.com>
References: <4B51D245020000E1000003C0@mail.fribert.dk>
	<54E291D4-6954-49DD-91FC-50F927C89532@jerris.com>
Message-ID: <4B523EBA020000E1000003C5@mail.fribert.dk>

>> I have two DID's registered at my SIP.
> At your sip what?
 
provider
 
>> This works nicely, I've created groups for holding the local  
>> extensions, and one of my phones can subscribe to two SIP accounts,  
>> so I can distinguish between if it's one or the other DID (it will  
>> show which SIP account is called in the display).
> Are you talking about a phone registering or subscribing to freeswitch  
> here?
 
FS is subscribing to two accounts at my provider.
I have a couple of phones subscribing to the FS.
Two phones has the ability of several sipaccounts, the last has one SIP subscription.
 
>> Problem is, another one of the phones can only subscribe to one SIP,  
>> and has no display (linksys sipura 901), any idea how to accomplish  
>> that I can distinnguish between the DID's on that one?
> So if I understand you want to make a single line no display phone and  
> send it multiple dids to it and be able to tell what did called?   
> Really, the phone has no way to display this, but you could play a  
> sound when they answer the phone.
 
Ahh, yes that sounds interesting, I thought of something like 'distinctive ringing' the phone seams to support that, but I have no idea how to use that.
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From lawwton at gmail.com  Sat Jan 16 13:39:41 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 16 Jan 2010 16:39:41 -0500
Subject: [Freeswitch-users] reloadxml question
In-Reply-To: <3E8FB576-F49D-49A1-B3E2-8C6F5D3A6059@avgs.ca>
References: <5fe6fa8f1001161158t54300ae5r62845f3baa1f97f5@mail.gmail.com>
	<3E8FB576-F49D-49A1-B3E2-8C6F5D3A6059@avgs.ca>
Message-ID: <5fe6fa8f1001161339m38c4c44ej71d7a30be4e3fb71@mail.gmail.com>

Thanks Mathieu, appreciate the help.

Alfredo

On Sat, Jan 16, 2010 at 3:16 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> reloadxml preprocesses the xml config again, it does not tell modules
> to reload their configuration from that file.
>
> In order to delete a gateway, you do: sofia profile <profilename>
> killgw <gwname>
>
> If you make changes to a gateway and want to reload it, you first
> delete it with that command and then do:
> ? ? ? ?sofia profile <profilename> rescan
>
> If you don't have any active calls on the box, restarting the profile
> will do the equivalent of killgw+rescan.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 16-Jan-10, at 2:58 PM, Alfredo Quiroga-Villamil wrote:
>
>> All:
>>
>> I am starting to play a bit with FS to get more familiar with it. I am
>> trying the following:
>>
>> 1- sofia status ?// command for example to see the list of profiles.
>>
>> 2- I then went ahead and renamed the example.com.xml under
>> conf/directory/default to example.com.xml.orig
>>
>> 3- I ran then reloadxml using the fs_cli
>>
>> 4- sofia status still displays the same info as before for
>> example.conf as a gateway.
>>
>> I then went ahead and stopped FW and re-started it. After I did this I
>> re-ran sofia status and this time the example.conf gateway that was
>> previously listed went away as it was supposed to.
>>
>> Am I missing something? Is reloadxml the proper way to reload
>> configuration changes. If it is why is it not working for this case?
>>
>> Thanks in advance,
>>
>> Alfredo
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Sat Jan 16 13:44:03 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 16 Jan 2010 15:44:03 -0600
Subject: [Freeswitch-users] No media after Originate
In-Reply-To: <f97004c31001161306o10517d1y9e0cf0229bce3589@mail.gmail.com>
References: <f97004c31001161306o10517d1y9e0cf0229bce3589@mail.gmail.com>
Message-ID: <3081C466-F735-47F1-BAEC-2B497F8F08D8@freeswitch.org>

Is nat involved?

/b

On Jan 16, 2010, at 3:06 PM, Mohamed Hassan wrote:

> estaplished successfully but without media passing




From brian at freeswitch.org  Sat Jan 16 13:46:00 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 16 Jan 2010 15:46:00 -0600
Subject: [Freeswitch-users] reloadxml question
In-Reply-To: <3E8FB576-F49D-49A1-B3E2-8C6F5D3A6059@avgs.ca>
References: <5fe6fa8f1001161158t54300ae5r62845f3baa1f97f5@mail.gmail.com>
	<3E8FB576-F49D-49A1-B3E2-8C6F5D3A6059@avgs.ca>
Message-ID: <E9FF7C20-8D77-4DA0-9B2D-DD75091C1457@freeswitch.org>

Also noted killing a gateway and reinstalling it doesn't touch currently active calls that happened to have used that gateway.
/b

On Jan 16, 2010, at 2:16 PM, Mathieu Rene wrote:

> If you don't have any active calls on the box, restarting the profile  
> will do the equivalent of killgw+rescan.




From mike at jerris.com  Sat Jan 16 14:24:49 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 16 Jan 2010 17:24:49 -0500
Subject: [Freeswitch-users] Svar: Re: How do I carry dial-in number to
	extension?
In-Reply-To: <4B523EBA020000E1000003C5@mail.fribert.dk>
References: <4B51D245020000E1000003C0@mail.fribert.dk>
	<54E291D4-6954-49DD-91FC-50F927C89532@jerris.com>
	<4B523EBA020000E1000003C5@mail.fribert.dk>
Message-ID: <F65EBF8D-0194-47BF-897F-FF8C2CE68C98@jerris.com>


On Jan 16, 2010, at 4:33 PM, mailinglist wrote:

> >> I have two DID's registered at my SIP.
> > At your sip what?
>  
> provider
>  
> >> This works nicely, I've created groups for holding the local  
> >> extensions, and one of my phones can subscribe to two SIP accounts,  
> >> so I can distinguish between if it's one or the other DID (it will  
> >> show which SIP account is called in the display).
> > Are you talking about a phone registering or subscribing to freeswitch  
> > here?
>  
> FS is subscribing to two accounts at my provider.
> I have a couple of phones subscribing to the FS.
> Two phones has the ability of several sipaccounts, the last has one SIP subscription.

I think you are trying to say sip registration, not subscription.  subscription is totally different but still a concept in sip.

> >> Problem is, another one of the phones can only subscribe to one SIP,  
> >> and has no display (linksys sipura 901), any idea how to accomplish  
> >> that I can distinnguish between the DID's on that one?
> > So if I understand you want to make a single line no display phone and  
> > send it multiple dids to it and be able to tell what did called?   
> > Really, the phone has no way to display this, but you could play a  
> > sound when they answer the phone.
>  
> Ahh, yes that sounds interesting, I thought of something like 'distinctive ringing' the phone seams to support that, but I have no idea how to use that.

Its quite possible this phone has other features for this, if so, read up on what they are and how they work and provide back some details and we can help with how to do this in freeswitch.

Mike

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From mohamedezz.fci at gmail.com  Sat Jan 16 15:22:54 2010
From: mohamedezz.fci at gmail.com (Mohamed Hassan)
Date: Sun, 17 Jan 2010 01:22:54 +0200
Subject: [Freeswitch-users] No media after Originate
In-Reply-To: <3081C466-F735-47F1-BAEC-2B497F8F08D8@freeswitch.org>
References: <f97004c31001161306o10517d1y9e0cf0229bce3589@mail.gmail.com>
	<3081C466-F735-47F1-BAEC-2B497F8F08D8@freeswitch.org>
Message-ID: <f97004c31001161522g65a0da7agec1eabffa321692d@mail.gmail.com>

There is no nat my server has public ip and not nated
and my sip provider too  as i can make regular calls through the same
provider without originate

On Sat, Jan 16, 2010 at 11:44 PM, Brian West <brian at freeswitch.org> wrote:
> Is nat involved?
>
> /b
>
> On Jan 16, 2010, at 3:06 PM, Mohamed Hassan wrote:
>
>> estaplished successfully but without media passing
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From chris.chen2004 at gmail.com  Sat Jan 16 17:37:46 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Sat, 16 Jan 2010 20:37:46 -0500
Subject: [Freeswitch-users] Originate_timeout not working with latest SVN
	trunk
Message-ID: <507898381001161737x17435f9ds33d9585aa83418bc@mail.gmail.com>

Hi there, if anybody has the similar issue like me regarding the
originate_timeout variable.

I have the following dialplan:

<extension name="Support_Extension">
      <condition field="destination_number" expression="^(support)$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="bind_meta_app" data="1 b s
execute_extension::dx XML features_nln"/>
        <action application="bind_meta_app" data="2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftim$
        <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features_nln"/>
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="call_timeout=10"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="db"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
        <action application="db"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
        <action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}$
        <action application="db"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
        <action application="bridge" data="user/1000 at My_FS_Public_IP
,user/1001 at My_FS_Public_IP"/>
        <action application="set" data="originate_timeout=15"/>
        <action application="bridge"
data="sofia/gateway/unlimitel/6475552222"/>
        <action application="bridge"
data="sofia/gateway/unlimitel/4165552222"/>
         <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="voicemail" data="default_nln ${domain_name}
1001"/>
      </condition>
    </extension>


in the past the call_timeout and originate_timeout all working as expected,
but with latest SVN trunk since 16318, the originate_timeout not working,
once the call forwarded to the cell phone by
  <action application="bridge" data="sofia/gateway/unlimitel/6475552222"/>
        <action application="bridge"
data="sofia/gateway/unlimitel/4165552222"/>
it never timed out
so the voicemail never got executed.
 Any helps are greatly appreciated
Thanks,

Chris
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From lawwton at gmail.com  Sat Jan 16 17:38:22 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 16 Jan 2010 20:38:22 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
Message-ID: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>

All:

Any ideas why there is an alias here? What does that exactly mean? Do
I need to have that? How do I remove that?

freeswitch at internal> sofia status
                     Name          Type
Data      State
=================================================================================================
                 external       profile
sip:mod_sofia at 192.168.1.148:5080      RUNNING (0)
                 internal       profile
sip:mod_sofia at 192.168.1.148:5060      RUNNING (0)
            192.168.1.148         alias
internal      ALIASED
=================================================================================================
2 profiles 1 alias

Lots of questions there all trying to figure out why it's showing up there.

Thanks in advance,

Alfredo Q-V



From lawwton at gmail.com  Sat Jan 16 17:49:26 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 16 Jan 2010 20:49:26 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
Message-ID: <5fe6fa8f1001161749m1c60fb77i579c7e7ecfe98e98@mail.gmail.com>

All:

I was able to eliminate the alias by setting the parameter in the
internal.xml file under sip_profiles to false. However to actually
load the change I had to restart FS. Is there a way to reload this
kind of change without restarting FS (not dropping calls).

Thanks in advance,

Alfredo

On Sat, Jan 16, 2010 at 8:38 PM, Alfredo Quiroga-Villamil
<lawwton at gmail.com> wrote:
> All:
>
> Any ideas why there is an alias here? What does that exactly mean? Do
> I need to have that? How do I remove that?
>
> freeswitch at internal> sofia status
> ? ? ? ? ? ? ? ? ? ? Name ? ? ? ? ?Type
> Data ? ? ?State
> =================================================================================================
> ? ? ? ? ? ? ? ? external ? ? ? profile
> sip:mod_sofia at 192.168.1.148:5080 ? ? ?RUNNING (0)
> ? ? ? ? ? ? ? ? internal ? ? ? profile
> sip:mod_sofia at 192.168.1.148:5060 ? ? ?RUNNING (0)
> ? ? ? ? ? ?192.168.1.148 ? ? ? ? alias
> internal ? ? ?ALIASED
> =================================================================================================
> 2 profiles 1 alias
>
> Lots of questions there all trying to figure out why it's showing up there.
>
> Thanks in advance,
>
> Alfredo Q-V
>

From mike at jerris.com  Sat Jan 16 17:58:40 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 16 Jan 2010 20:58:40 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
Message-ID: <9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>

it lets you use your ip as the profile name as well.  There are some things in the default configs that take advantage of and assume that the profile name is the domain name.  In the case of the default configs, we use the detected ip address for this.  If you remove it, things will probably break unless you have devices that all work right, dns setup right, and all your devices dns.  

Mike

On Jan 16, 2010, at 8:38 PM, Alfredo Quiroga-Villamil wrote:

> All:
> 
> Any ideas why there is an alias here? What does that exactly mean? Do
> I need to have that? How do I remove that?
> 
> freeswitch at internal> sofia status
>                     Name          Type
> Data      State
> =================================================================================================
>                 external       profile
> sip:mod_sofia at 192.168.1.148:5080      RUNNING (0)
>                 internal       profile
> sip:mod_sofia at 192.168.1.148:5060      RUNNING (0)
>            192.168.1.148         alias
> internal      ALIASED
> =================================================================================================
> 2 profiles 1 alias
> 
> Lots of questions there all trying to figure out why it's showing up there.
> 
> Thanks in advance,
> 
> Alfredo Q-V




From lawwton at gmail.com  Sat Jan 16 18:09:55 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 16 Jan 2010 21:09:55 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>
Message-ID: <5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>

Appreciate the response Mike.

That's a little different than everything I've seen in other systems.
I don't want to be forced to display 3 different profiles when in
reality one is just an alias name. It says alias and all; but I find
that a bit repetitive, specially if I remove it and it breaks things.

When a change is made on one of the sip_profiles, take internal.xml
for example where I changed a parameter from true to false. What
command needs to be ran to reload the changes (non-disruptive).

Thanks in advance,

Alfredo

On Sat, Jan 16, 2010 at 8:58 PM, Michael Jerris <mike at jerris.com> wrote:
> it lets you use your ip as the profile name as well. ?There are some things in the default configs that take advantage of and assume that the profile name is the domain name. ?In the case of the default configs, we use the detected ip address for this. ?If you remove it, things will probably break unless you have devices that all work right, dns setup right, and all your devices dns.
>
> Mike
>
> On Jan 16, 2010, at 8:38 PM, Alfredo Quiroga-Villamil wrote:
>
>> All:
>>
>> Any ideas why there is an alias here? What does that exactly mean? Do
>> I need to have that? How do I remove that?
>>
>> freeswitch at internal> sofia status
>> ? ? ? ? ? ? ? ? ? ? Name ? ? ? ? ?Type
>> Data ? ? ?State
>> =================================================================================================
>> ? ? ? ? ? ? ? ? external ? ? ? profile
>> sip:mod_sofia at 192.168.1.148:5080 ? ? ?RUNNING (0)
>> ? ? ? ? ? ? ? ? internal ? ? ? profile
>> sip:mod_sofia at 192.168.1.148:5060 ? ? ?RUNNING (0)
>> ? ? ? ? ? ?192.168.1.148 ? ? ? ? alias
>> internal ? ? ?ALIASED
>> =================================================================================================
>> 2 profiles 1 alias
>>
>> Lots of questions there all trying to figure out why it's showing up there.
>>
>> Thanks in advance,
>>
>> Alfredo Q-V
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mastermind202 at gmail.com  Sat Jan 16 18:39:26 2010
From: mastermind202 at gmail.com (mm_202)
Date: Sat, 16 Jan 2010 21:39:26 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>
	<5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>
Message-ID: <63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>

On Sat, Jan 16, 2010 at 9:09 PM, Alfredo Quiroga-Villamil <lawwton at gmail.com
> wrote:

> Appreciate the response Mike.
>
> That's a little different than everything I've seen in other systems.
> I don't want to be forced to display 3 different profiles when in
> reality one is just an alias name. It says alias and all; but I find
> that a bit repetitive, specially if I remove it and it breaks things.
>
> When a change is made on one of the sip_profiles, take internal.xml
> for example where I changed a parameter from true to false. What
> command needs to be ran to reload the changes (non-disruptive).
>
> Thanks in advance,
>
> Alfredo
>
> On Sat, Jan 16, 2010 at 8:58 PM, Michael Jerris <mike at jerris.com> wrote:
> > it lets you use your ip as the profile name as well.  There are some
> things in the default configs that take advantage of and assume that the
> profile name is the domain name.  In the case of the default configs, we use
> the detected ip address for this.  If you remove it, things will probably
> break unless you have devices that all work right, dns setup right, and all
> your devices dns.
> >
> > Mike
> >
> > On Jan 16, 2010, at 8:38 PM, Alfredo Quiroga-Villamil wrote:
> >
> >> All:
> >>
> >> Any ideas why there is an alias here? What does that exactly mean? Do
> >> I need to have that? How do I remove that?
> >>
> >> freeswitch at internal> sofia status
> >>                     Name          Type
> >> Data      State
> >>
> =================================================================================================
> >>                 external       profile
> >> sip:mod_sofia at 192.168.1.148:5080      RUNNING (0)
> >>                 internal       profile
> >> sip:mod_sofia at 192.168.1.148:5060      RUNNING (0)
> >>            192.168.1.148         alias
> >> internal      ALIASED
> >>
> =================================================================================================
> >> 2 profiles 1 alias
> >>
> >> Lots of questions there all trying to figure out why it's showing up
> there.
> >>
> >> Thanks in advance,
> >>
> >> Alfredo Q-V
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


Alfredo,
Use 'reloadxml' and then for mod_sofia to actually use the changes, you'll
have to run 'sofia profile internal restart', but that will drop any calls
that are on that profile.

-- mm_202.
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From lawwton at gmail.com  Sat Jan 16 19:00:08 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sat, 16 Jan 2010 22:00:08 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>
	<5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>
	<63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>
Message-ID: <5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>

mm_202:

Appreciate the response. So, essentially what that means is that there
is no way to reload changes unless we do a restart which will drop
calls? I apologize before hand for saying this; but that's really
really bad when you have a production server to which changes are made
sometimes through out the day, new turnups/disconnects, etc...

I actually just did the following test:

a) Made sure I removed the alias from my internal profile. Verified it
was gone with "sofia status"

b) Edit the internal.xml file again and reset the flag to true for the alias.

c) Ran: sofia profile internal rescan  reloadxml

I received the expected message about the alias being added:

freeswitch at internal> 2010-01-16 22:17:49.778272 [INFO]
switch_time.c:812 Timezone reloaded 530 definitions
2010-01-16 22:17:49.778272 [DEBUG] sofia.c:2252 Duration out of
bounds, using default of 2000!
2010-01-16 22:17:49.778272 [NOTICE] sofia.c:1804 Adding Alias
[192.168.1.148] for profile [internal]

d) I then edited the file again and set the flag again to false

e) I then re-ran: sofia profile internal rescan  reloadxml again to
find out that the alias is not removed this time around. In other
words, the alias is only added with a re-scan after the flag is set to
true. It's not removed with rescan if the flag is set to false.

So, two big concerns now:

1- If there is no way to reload things dynamically without disrupting
service, is this by design? Specially the sip_profiles part, that's
really important.

2- Why are the a-e) steps above partially working, it might seem at
first glance that sofia profile internal rescan  reloadxml should do
what it says, to re-scan for new changes and load them for that
profile. Is this a bug or this is by design?

All:

Feel free to keep me honest here and let me know if I am doing
something wrong since I just started playing with FS today.

Appreciate the help,

Alfredo

That command

On Sat, Jan 16, 2010 at 9:39 PM, mm_202 <mastermind202 at gmail.com> wrote:
>
> On Sat, Jan 16, 2010 at 9:09 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> Appreciate the response Mike.
>>
>> That's a little different than everything I've seen in other systems.
>> I don't want to be forced to display 3 different profiles when in
>> reality one is just an alias name. It says alias and all; but I find
>> that a bit repetitive, specially if I remove it and it breaks things.
>>
>> When a change is made on one of the sip_profiles, take internal.xml
>> for example where I changed a parameter from true to false. What
>> command needs to be ran to reload the changes (non-disruptive).
>>
>> Thanks in advance,
>>
>> Alfredo
>>
>> On Sat, Jan 16, 2010 at 8:58 PM, Michael Jerris <mike at jerris.com> wrote:
>> > it lets you use your ip as the profile name as well. ?There are some
>> > things in the default configs that take advantage of and assume that the
>> > profile name is the domain name. ?In the case of the default configs, we use
>> > the detected ip address for this. ?If you remove it, things will probably
>> > break unless you have devices that all work right, dns setup right, and all
>> > your devices dns.
>> >
>> > Mike
>> >
>> > On Jan 16, 2010, at 8:38 PM, Alfredo Quiroga-Villamil wrote:
>> >
>> >> All:
>> >>
>> >> Any ideas why there is an alias here? What does that exactly mean? Do
>> >> I need to have that? How do I remove that?
>> >>
>> >> freeswitch at internal> sofia status
>> >> ? ? ? ? ? ? ? ? ? ? Name ? ? ? ? ?Type
>> >> Data ? ? ?State
>> >>
>> >> =================================================================================================
>> >> ? ? ? ? ? ? ? ? external ? ? ? profile
>> >> sip:mod_sofia at 192.168.1.148:5080 ? ? ?RUNNING (0)
>> >> ? ? ? ? ? ? ? ? internal ? ? ? profile
>> >> sip:mod_sofia at 192.168.1.148:5060 ? ? ?RUNNING (0)
>> >> ? ? ? ? ? ?192.168.1.148 ? ? ? ? alias
>> >> internal ? ? ?ALIASED
>> >>
>> >> =================================================================================================
>> >> 2 profiles 1 alias
>> >>
>> >> Lots of questions there all trying to figure out why it's showing up
>> >> there.
>> >>
>> >> Thanks in advance,
>> >>
>> >> Alfredo Q-V
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> Alfredo,
> Use 'reloadxml' and then for mod_sofia to actually use the changes, you'll
> have to run 'sofia profile internal restart', but that will drop any calls
> that are on that profile.
>
> -- mm_202.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From mastermind202 at gmail.com  Sat Jan 16 19:35:28 2010
From: mastermind202 at gmail.com (mm_202)
Date: Sat, 16 Jan 2010 22:35:28 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>
	<5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>
	<63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>
	<5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>
Message-ID: <63de75711001161935w2d1e204ek81ebe1143afb1497@mail.gmail.com>

On Sat, Jan 16, 2010 at 10:00 PM, Alfredo Quiroga-Villamil <
lawwton at gmail.com> wrote:

> mm_202:
>
> Appreciate the response. So, essentially what that means is that there
> is no way to reload changes unless we do a restart which will drop
> calls? I apologize before hand for saying this; but that's really
> really bad when you have a production server to which changes are made
> sometimes through out the day, new turnups/disconnects, etc...
>
> I actually just did the following test:
>
> a) Made sure I removed the alias from my internal profile. Verified it
> was gone with "sofia status"
>
> b) Edit the internal.xml file again and reset the flag to true for the
> alias.
>
> c) Ran: sofia profile internal rescan  reloadxml
>
> I received the expected message about the alias being added:
>
> freeswitch at internal> 2010-01-16 22:17:49.778272 [INFO]
> switch_time.c:812 Timezone reloaded 530 definitions
> 2010-01-16 22:17:49.778272 [DEBUG] sofia.c:2252 Duration out of
> bounds, using default of 2000!
> 2010-01-16 22:17:49.778272 [NOTICE] sofia.c:1804 Adding Alias
> [192.168.1.148] for profile [internal]
>
> d) I then edited the file again and set the flag again to false
>
> e) I then re-ran: sofia profile internal rescan  reloadxml again to
> find out that the alias is not removed this time around. In other
> words, the alias is only added with a re-scan after the flag is set to
> true. It's not removed with rescan if the flag is set to false.
>
> So, two big concerns now:
>
> 1- If there is no way to reload things dynamically without disrupting
> service, is this by design? Specially the sip_profiles part, that's
> really important.
>
> 2- Why are the a-e) steps above partially working, it might seem at
> first glance that sofia profile internal rescan  reloadxml should do
> what it says, to re-scan for new changes and load them for that
> profile. Is this a bug or this is by design?
>
> All:
>
> Feel free to keep me honest here and let me know if I am doing
> something wrong since I just started playing with FS today.
>
> Appreciate the help,
>
> Alfredo
>
> That command
>
> On Sat, Jan 16, 2010 at 9:39 PM, mm_202 <mastermind202 at gmail.com> wrote:
> >
> > On Sat, Jan 16, 2010 at 9:09 PM, Alfredo Quiroga-Villamil
> > <lawwton at gmail.com> wrote:
> >>
> >> Appreciate the response Mike.
> >>
> >> That's a little different than everything I've seen in other systems.
> >> I don't want to be forced to display 3 different profiles when in
> >> reality one is just an alias name. It says alias and all; but I find
> >> that a bit repetitive, specially if I remove it and it breaks things.
> >>
> >> When a change is made on one of the sip_profiles, take internal.xml
> >> for example where I changed a parameter from true to false. What
> >> command needs to be ran to reload the changes (non-disruptive).
> >>
> >> Thanks in advance,
> >>
> >> Alfredo
> >>
> >> On Sat, Jan 16, 2010 at 8:58 PM, Michael Jerris <mike at jerris.com>
> wrote:
> >> > it lets you use your ip as the profile name as well.  There are some
> >> > things in the default configs that take advantage of and assume that
> the
> >> > profile name is the domain name.  In the case of the default configs,
> we use
> >> > the detected ip address for this.  If you remove it, things will
> probably
> >> > break unless you have devices that all work right, dns setup right,
> and all
> >> > your devices dns.
> >> >
> >> > Mike
> >> >
> >> > On Jan 16, 2010, at 8:38 PM, Alfredo Quiroga-Villamil wrote:
> >> >
> >> >> All:
> >> >>
> >> >> Any ideas why there is an alias here? What does that exactly mean? Do
> >> >> I need to have that? How do I remove that?
> >> >>
> >> >> freeswitch at internal> sofia status
> >> >>                     Name          Type
> >> >> Data      State
> >> >>
> >> >>
> =================================================================================================
> >> >>                 external       profile
> >> >> sip:mod_sofia at 192.168.1.148:5080      RUNNING (0)
> >> >>                 internal       profile
> >> >> sip:mod_sofia at 192.168.1.148:5060      RUNNING (0)
> >> >>            192.168.1.148         alias
> >> >> internal      ALIASED
> >> >>
> >> >>
> =================================================================================================
> >> >> 2 profiles 1 alias
> >> >>
> >> >> Lots of questions there all trying to figure out why it's showing up
> >> >> there.
> >> >>
> >> >> Thanks in advance,
> >> >>
> >> >> Alfredo Q-V
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > Alfredo,
> > Use 'reloadxml' and then for mod_sofia to actually use the changes,
> you'll
> > have to run 'sofia profile internal restart', but that will drop any
> calls
> > that are on that profile.
> >
> > -- mm_202.
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

Alfredo,

In a production system, you *shouldnt* have to change the main profiles at
all.
When adding / removing gateways, you can use 'sofia profile [profilename]
rescan',
that will not drop calls, only if you run 'reload'.

As far as why the rescan adds the alias but doesnt remove it, I would say
that is by design.
That way you cant 'break' anything if your dialplan is using that alias.
But someone more
experienced than me may have a better / more accurate answer.

-- mm_202.
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From dujinfang at gmail.com  Sat Jan 16 19:37:31 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 17 Jan 2010 11:37:31 +0800
Subject: [Freeswitch-users] Questions about mod_fifo
In-Reply-To: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
References: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
Message-ID: <23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>

2010/1/15, Jingwei Yang <jingwei.yang at gmail.com>:
> Hi Guys,
>
> I'm implementing an ACD system using ESL and mod_fifo. Based on what Anthony
> suggested in this post:
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html
>
> *You can make an event socket application that listens for FIFO events and
> keeps track of what FIFOs are currently busy and when there are people
> waiting you can have that script generate a call to a group of SIP phones so
> when the first one answers, it sends them in as an agent where they can
> field the calls.
> *
>
> 1. How should I handle the concurrent issues? If I bridge a user to two
> agents and both of them answers, how does FS take care of this situation?
> Will a slower agent get a busy tone automatically?
>

I think it just follow the standard originate dialstring rules.

> 2. If the socket application is brought up after some users have called in,
> what command should I use to check the busy queues? fifo list?
>
Yes. Perhaps you can also check the fifo db, either sqlite or ODBC.

> 3. Am I using fifo list and fifo count correctly?
>
> here's the testing dialplan:
>
>     <extension name="in_fifo">
>       <condition field="destination_number" expression="^779$">
>         <action application="set" data="fifo_music=$${hold_music}"/>
>         <action application="answer"/>
>         <action application="fifo" data="*myq *in"/>
>       </condition>
>     </extension>
>
> when a call comes in and gets queued, these are the results of some commands
> I tried.
>
> freeswitch at localhost.localdomain> fifo list
> API CALL [fifo(list)] output:
> <fifo_report>
>   <fifo name="cool_fifo at 192.168.1.104" consumer_count="0" caller_count="0"
> waiting_count="0" importance="0">
>     <callers></callers>
>     <consumers></consumers>
>   </fifo>
> </fifo_report>
>
>
> freeswitch at localhost.localdomain> fifo list myq
> API CALL [fifo(list myq)] output:
> <fifo_report></fifo_report>
>
>
> freeswitch at localhost.localdomain> fifo count myq
> API CALL [fifo(count myq)] output:
> none
>
> It seems *myq* doesn't get created at all? Please enlighten.
>
> Thanks and best regards,
> -Jingwei
>
AFAIK, thant means the channel didn't queued in. Did you see any error
logs? I think you need to remove the stars in   <action
application="fifo" data="*myq *in"/>.



From lawwton at gmail.com  Sat Jan 16 21:00:21 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sun, 17 Jan 2010 00:00:21 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <63de75711001161935w2d1e204ek81ebe1143afb1497@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>
	<5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>
	<63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>
	<5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>
	<63de75711001161935w2d1e204ek81ebe1143afb1497@mail.gmail.com>
Message-ID: <5fe6fa8f1001162100n699077faw8ac1e8cef4e4ea5@mail.gmail.com>

mm_202:

Thanks for the reply.

I don't agree with your statement: "In a production system, you
*shouldnt* have to change the main profiles at all."

I think it's always good to be able to reload things dynamically in a
non-disruptive way.

The alias is a bit unusual. To be honest I had never seen that before
in any of the Telecommunication Systems I've worked on. Not that is
not a good idea, just one that at first glance doesn't seem too clear
for some reason. Specially when you see two profiles both representing
the same object listed under "sofia status". Being forced to have the
alias doesn't seem like an appealing option, should be optional I
think. I would like if possible a more detailed explanation on what
would break if not present.

In any case, I am sure there is a reason for these things. I am trying
to understand how they all work together. Things are not as apparent
in FS at first glance when compared to other systems; but the platform
seems to be built with the purpose of offering a lot of flexibility
which is really good.

Appreciate the feedback.

Alfredo Q-V

Appreciate the help,

Alfredo



On Sat, Jan 16, 2010 at 10:35 PM, mm_202 <mastermind202 at gmail.com> wrote:
>
> On Sat, Jan 16, 2010 at 10:00 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> mm_202:
>>
>> Appreciate the response. So, essentially what that means is that there
>> is no way to reload changes unless we do a restart which will drop
>> calls? I apologize before hand for saying this; but that's really
>> really bad when you have a production server to which changes are made
>> sometimes through out the day, new turnups/disconnects, etc...
>>
>> I actually just did the following test:
>>
>> a) Made sure I removed the alias from my internal profile. Verified it
>> was gone with "sofia status"
>>
>> b) Edit the internal.xml file again and reset the flag to true for the
>> alias.
>>
>> c) Ran: sofia profile internal rescan ?reloadxml
>>
>> I received the expected message about the alias being added:
>>
>> freeswitch at internal> 2010-01-16 22:17:49.778272 [INFO]
>> switch_time.c:812 Timezone reloaded 530 definitions
>> 2010-01-16 22:17:49.778272 [DEBUG] sofia.c:2252 Duration out of
>> bounds, using default of 2000!
>> 2010-01-16 22:17:49.778272 [NOTICE] sofia.c:1804 Adding Alias
>> [192.168.1.148] for profile [internal]
>>
>> d) I then edited the file again and set the flag again to false
>>
>> e) I then re-ran: sofia profile internal rescan ?reloadxml again to
>> find out that the alias is not removed this time around. In other
>> words, the alias is only added with a re-scan after the flag is set to
>> true. It's not removed with rescan if the flag is set to false.
>>
>> So, two big concerns now:
>>
>> 1- If there is no way to reload things dynamically without disrupting
>> service, is this by design? Specially the sip_profiles part, that's
>> really important.
>>
>> 2- Why are the a-e) steps above partially working, it might seem at
>> first glance that sofia profile internal rescan ?reloadxml should do
>> what it says, to re-scan for new changes and load them for that
>> profile. Is this a bug or this is by design?
>>
>> All:
>>
>> Feel free to keep me honest here and let me know if I am doing
>> something wrong since I just started playing with FS today.
>>
>> Appreciate the help,
>>
>> Alfredo
>>
>> That command
>>
>> On Sat, Jan 16, 2010 at 9:39 PM, mm_202 <mastermind202 at gmail.com> wrote:
>> >
>> > On Sat, Jan 16, 2010 at 9:09 PM, Alfredo Quiroga-Villamil
>> > <lawwton at gmail.com> wrote:
>> >>
>> >> Appreciate the response Mike.
>> >>
>> >> That's a little different than everything I've seen in other systems.
>> >> I don't want to be forced to display 3 different profiles when in
>> >> reality one is just an alias name. It says alias and all; but I find
>> >> that a bit repetitive, specially if I remove it and it breaks things.
>> >>
>> >> When a change is made on one of the sip_profiles, take internal.xml
>> >> for example where I changed a parameter from true to false. What
>> >> command needs to be ran to reload the changes (non-disruptive).
>> >>
>> >> Thanks in advance,
>> >>
>> >> Alfredo
>> >>
>> >> On Sat, Jan 16, 2010 at 8:58 PM, Michael Jerris <mike at jerris.com>
>> >> wrote:
>> >> > it lets you use your ip as the profile name as well. ?There are some
>> >> > things in the default configs that take advantage of and assume that
>> >> > the
>> >> > profile name is the domain name. ?In the case of the default configs,
>> >> > we use
>> >> > the detected ip address for this. ?If you remove it, things will
>> >> > probably
>> >> > break unless you have devices that all work right, dns setup right,
>> >> > and all
>> >> > your devices dns.
>> >> >
>> >> > Mike
>> >> >
>> >> > On Jan 16, 2010, at 8:38 PM, Alfredo Quiroga-Villamil wrote:
>> >> >
>> >> >> All:
>> >> >>
>> >> >> Any ideas why there is an alias here? What does that exactly mean?
>> >> >> Do
>> >> >> I need to have that? How do I remove that?
>> >> >>
>> >> >> freeswitch at internal> sofia status
>> >> >> ? ? ? ? ? ? ? ? ? ? Name ? ? ? ? ?Type
>> >> >> Data ? ? ?State
>> >> >>
>> >> >>
>> >> >> =================================================================================================
>> >> >> ? ? ? ? ? ? ? ? external ? ? ? profile
>> >> >> sip:mod_sofia at 192.168.1.148:5080 ? ? ?RUNNING (0)
>> >> >> ? ? ? ? ? ? ? ? internal ? ? ? profile
>> >> >> sip:mod_sofia at 192.168.1.148:5060 ? ? ?RUNNING (0)
>> >> >> ? ? ? ? ? ?192.168.1.148 ? ? ? ? alias
>> >> >> internal ? ? ?ALIASED
>> >> >>
>> >> >>
>> >> >> =================================================================================================
>> >> >> 2 profiles 1 alias
>> >> >>
>> >> >> Lots of questions there all trying to figure out why it's showing up
>> >> >> there.
>> >> >>
>> >> >> Thanks in advance,
>> >> >>
>> >> >> Alfredo Q-V
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > Alfredo,
>> > Use 'reloadxml' and then for mod_sofia to actually use the changes,
>> > you'll
>> > have to run 'sofia profile internal restart', but that will drop any
>> > calls
>> > that are on that profile.
>> >
>> > -- mm_202.
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> Alfredo,
>
> In a production system, you *shouldnt* have to change the main profiles at
> all.
> When adding / removing gateways, you can use 'sofia profile [profilename]
> rescan',
> that will not drop calls, only if you run 'reload'.
>
> As far as why the rescan adds the alias but doesnt remove it, I would say
> that is by design.
> That way you cant 'break' anything if your dialplan is using that alias.
> But someone more
> experienced than me may have a better / more accurate answer.
>
> -- mm_202.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From anthony.minessale at gmail.com  Sat Jan 16 21:20:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 16 Jan 2010 23:20:56 -0600
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <191c3a031001162120k50fcca08l667fd3b842d5ebbd@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>
	<5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>
	<63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>
	<5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>
	<63de75711001161935w2d1e204ek81ebe1143afb1497@mail.gmail.com>
	<5fe6fa8f1001162100n699077faw8ac1e8cef4e4ea5@mail.gmail.com>
	<191c3a031001162117u413f0d5bi1903c69e519c3632@mail.gmail.com>
	<191c3a031001162120k50fcca08l667fd3b842d5ebbd@mail.gmail.com>
Message-ID: <191c3a031001162120j499f0e9xe359bb9a97607415@mail.gmail.com>

We do offer a triple your money back guarentee!

Many params can change without restarting and many can't ip/port fo
example.  FYI, You don't sound sincere when you use the term appriciate when
you are arguing with these guys voulenteering to help explain it to you.
You should make sure you make the most of their assistance as they could
have just said rtfm

On Jan 16, 2010 11:07 PM, "Alfredo Quiroga-Villamil" <lawwton at gmail.com>
wrote:

mm_202:

Thanks for the reply.

I don't agree with your statement: "In a production system, you

*shouldnt* have to change the main profiles at all."
I think it's always good to be able to reload things dynamically in a
non-disruptive way.

The alias is a bit unusual. To be honest I had never seen that before
in any of the Telecommunication Systems I've worked on. Not that is
not a good idea, just one that at first glance doesn't seem too clear
for some reason. Specially when you see two profiles both representing
the same object listed under "sofia status". Being forced to have the
alias doesn't seem like an appealing option, should be optional I
think. I would like if possible a more detailed explanation on what
would break if not present.

In any case, I am sure there is a reason for these things. I am trying
to understand how they all work together. Things are not as apparent
in FS at first glance when compared to other systems; but the platform
seems to be built with the purpose of offering a lot of flexibility
which is really good.

Appreciate the feedback.

Alfredo Q-V

Appreciate the help,

Alfredo

On Sat, Jan 16, 2010 at 10:35 PM, mm_202 <mastermind202 at gmail.com> wrote: >
> On Sat, Jan 16, 201...
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From lawwton at gmail.com  Sun Jan 17 06:07:25 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sun, 17 Jan 2010 09:07:25 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <191c3a031001162120j499f0e9xe359bb9a97607415@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<9DCCE726-AD2A-4980-8FE0-57F92018F47A@jerris.com>
	<5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>
	<63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>
	<5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>
	<63de75711001161935w2d1e204ek81ebe1143afb1497@mail.gmail.com>
	<5fe6fa8f1001162100n699077faw8ac1e8cef4e4ea5@mail.gmail.com>
	<191c3a031001162117u413f0d5bi1903c69e519c3632@mail.gmail.com>
	<191c3a031001162120k50fcca08l667fd3b842d5ebbd@mail.gmail.com>
	<191c3a031001162120j499f0e9xe359bb9a97607415@mail.gmail.com>
Message-ID: <5fe6fa8f1001170607v52c0516egc6f69a60a934722f@mail.gmail.com>

I am pretty sincere when I say "I appreciate the help", otherwise I
wouldn't say it. I don't think I've argued in any of my previous
statements and I've tried to be very polite when raising questions or
concerns. Disagreeing in a polite way doesn't necessarily mean that I
am 100% rejecting the idea and by the way it's not a bad thing to
disagree. No one said "You are wrong"; but "I disagree". There is a
difference there.

If you expect to put a system out there and new users that have used
other ones not to ask why this or that I think you'll keep running
into this with other people. No one is criticizing or putting down the
system here, on the contrary I am trying to simply find out how it
works so I can use it. But things like reloading configurations and
specific system details that are quite different when compared to
others (alias for instance which I am still not sure what would break
if removed) do require some asking if after reading the online
documentation and previous archived messages the reason is still not
clear.

So I am not sure how you expect people to ask questions or if this is
more a "this is what it is" take it or leave it dictatorship approach.
Not the paradigm followed in open source projects where collaboration,
asking and questioning is usually the  way to go.

Interestingly enough in this one case I DON'T APPRECIATE YOUR
RESPONSE. Now tell me if that doesn't sound sincere to you. In any
case I will continue to play with the system and possibly ask
questions as well.

Alfredo

On Sun, Jan 17, 2010 at 12:20 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> We do offer a triple your money back guarentee!
>
> Many params can change without restarting and many can't ip/port fo
> example.? FYI, You don't sound sincere when you use the term appriciate when
> you are arguing with these guys voulenteering to help explain it to you.
> You should make sure you make the most of their assistance as they could
> have just said rtfm
>
> On Jan 16, 2010 11:07 PM, "Alfredo Quiroga-Villamil" <lawwton at gmail.com>
> wrote:
>
> mm_202:
>
> Thanks for the reply.
>
> I don't agree with your statement: "In a production system, you
>
> *shouldnt* have to change the main profiles at all."
>
> I think it's always good to be able to reload things dynamically in a
> non-disruptive way.
>
> The alias is a bit unusual. To be honest I had never seen that before
> in any of the Telecommunication Systems I've worked on. Not that is
> not a good idea, just one that at first glance doesn't seem too clear
> for some reason. Specially when you see two profiles both representing
> the same object listed under "sofia status". Being forced to have the
> alias doesn't seem like an appealing option, should be optional I
> think. I would like if possible a more detailed explanation on what
> would break if not present.
>
> In any case, I am sure there is a reason for these things. I am trying
> to understand how they all work together. Things are not as apparent
> in FS at first glance when compared to other systems; but the platform
> seems to be built with the purpose of offering a lot of flexibility
> which is really good.
>
> Appreciate the feedback.
>
> Alfredo Q-V
>
> Appreciate the help,
>
> Alfredo
>
> On Sat, Jan 16, 2010 at 10:35 PM, mm_202 <mastermind202 at gmail.com> wrote: >
>> On Sat, Jan 16, 201...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From max.bridgewater at gmail.com  Sun Jan 17 10:55:34 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 17 Jan 2010 13:55:34 -0500
Subject: [Freeswitch-users] Trouble with Originate/Bridge from ESL
Message-ID: <c55d46951001171055p41f8e01dje6a5427fa2f3aeed@mail.gmail.com>

Hi,

The following command works great from the command line:
originate
{ringback=\'%(2000,4000,440.0,480.0)\'}sofia/internal/1004%74.63.243.52
&bridge(sofia/internal/1005%74.63.243.52)

But this one isn't working from the ESL:
api originate
{ringback=\'%(2000,4000,440.0,480.0)\'}sofia/internal/1004%74.63.243.52
&bridge(sofia/internal/1005%74.63.243.52)

The observed behavior is following: The A leg is dialed, then B leg is also
dialed but immediately followed by a hangup. That is, the B user doesn't
even has time to answer.

Any idea what I'm doing wrong?

Thanks,
Max.
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From devel at thom.fr.eu.org  Sun Jan 17 11:45:38 2010
From: devel at thom.fr.eu.org (devel at thom.fr.eu.org)
Date: Sun, 17 Jan 2010 20:45:38 +0100
Subject: [Freeswitch-users] No hangup tone after zap channel closed,
	tones 	in general
In-Reply-To: <87f2f3b91001151124r596418abta126ebdaae10465@mail.gmail.com>
References: <8b18ffe229516c90285de8a54a00e845@thom.fr.eu.org>	<87f2f3b91001141723u4b0b12e1rd0f922db493492cd@mail.gmail.com>	<001201ca95d1$045eb6e0$0d1c24a0$@fr.eu.org>
	<87f2f3b91001151124r596418abta126ebdaae10465@mail.gmail.com>
Message-ID: <002701ca97ad$a523e590$ef6bb0b0$@fr.eu.org>

It?s not exactly that. I expected an FXS port would by itself generate a
busy tone after a call (initiated or not by this port) is terminated by the
other end. Doing this permit the phone connected to this port to detect the
end of communication and hang up automatically.

 

That said, I guess the dialplan example would work when the FXS port is the
A leg, but what should I do when the FXS port is the B leg ?

 

Thanks

 

Fran?ois

 

De : freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael
Collins
Envoy? : vendredi 15 janvier 2010 20:24
? : freeswitch-users at lists.freeswitch.org
Objet : Re: [Freeswitch-users] No hangup tone after zap channel closed,
tones in general

 

 

On Fri, Jan 15, 2010 at 2:53 AM, <devel at thom.fr.eu.org> wrote:

Thank you for the link. I googled through but could not find anything
relevant.

 

So then with my FXS port, do I have to, when a call is over, bridge the
channel (which is either A or B leg depending on the cases) to an extension
with for instance

<action application="playback" data="tone_stream://%(500,500,440)"/>

if you're just trying to manually send out that tone then yes, you can just
add the line in your dialplan. You can then hangup after playing the tone.
The other end will have to decide what to do on its own.
-MC

 

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From max.bridgewater at gmail.com  Sun Jan 17 12:51:12 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 17 Jan 2010 15:51:12 -0500
Subject: [Freeswitch-users] Trouble with Originate/Bridge from ESL
In-Reply-To: <c55d46951001171055p41f8e01dje6a5427fa2f3aeed@mail.gmail.com>
References: <c55d46951001171055p41f8e01dje6a5427fa2f3aeed@mail.gmail.com>
Message-ID: <c55d46951001171251l42dee174kee35eeadaf1ef40@mail.gmail.com>

One more piece of information: the call is being terminated by Freeswitch
with the event:

Event-Name: CHANNEL_HANGUP
Hangup-Cause: NO_ANSWER

Which is strange because B leg doesn't even have the time to answer.


On Sun, Jan 17, 2010 at 1:55 PM, Max Bridgewater
<max.bridgewater at gmail.com>wrote:

> Hi,
>
> The following command works great from the command line:
> originate
> {ringback=\'%(2000,4000,440.0,480.0)\'}sofia/internal/1004%74.63.243.52
> &bridge(sofia/internal/1005%74.63.243.52)
>
> But this one isn't working from the ESL:
> api originate
> {ringback=\'%(2000,4000,440.0,480.0)\'}sofia/internal/1004%74.63.243.52
> &bridge(sofia/internal/1005%74.63.243.52)
>
> The observed behavior is following: The A leg is dialed, then B leg is also
> dialed but immediately followed by a hangup. That is, the B user doesn't
> even has time to answer.
>
> Any idea what I'm doing wrong?
>
> Thanks,
> Max.
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From mastermind202 at gmail.com  Sun Jan 17 13:53:52 2010
From: mastermind202 at gmail.com (mm_202)
Date: Sun, 17 Jan 2010 16:53:52 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <5fe6fa8f1001170607v52c0516egc6f69a60a934722f@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<5fe6fa8f1001161809m490a524cn66189a7bf8952e6b@mail.gmail.com>
	<63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>
	<5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>
	<63de75711001161935w2d1e204ek81ebe1143afb1497@mail.gmail.com>
	<5fe6fa8f1001162100n699077faw8ac1e8cef4e4ea5@mail.gmail.com>
	<191c3a031001162117u413f0d5bi1903c69e519c3632@mail.gmail.com>
	<191c3a031001162120k50fcca08l667fd3b842d5ebbd@mail.gmail.com>
	<191c3a031001162120j499f0e9xe359bb9a97607415@mail.gmail.com>
	<5fe6fa8f1001170607v52c0516egc6f69a60a934722f@mail.gmail.com>
Message-ID: <63de75711001171353hca2a330he29cf2760f355a3d@mail.gmail.com>

On Sun, Jan 17, 2010 at 9:07 AM, Alfredo Quiroga-Villamil <lawwton at gmail.com
> wrote:

> I am pretty sincere when I say "I appreciate the help", otherwise I
> wouldn't say it. I don't think I've argued in any of my previous
> statements and I've tried to be very polite when raising questions or
> concerns. Disagreeing in a polite way doesn't necessarily mean that I
> am 100% rejecting the idea and by the way it's not a bad thing to
> disagree. No one said "You are wrong"; but "I disagree". There is a
> difference there.
>
> If you expect to put a system out there and new users that have used
> other ones not to ask why this or that I think you'll keep running
> into this with other people. No one is criticizing or putting down the
> system here, on the contrary I am trying to simply find out how it
> works so I can use it. But things like reloading configurations and
> specific system details that are quite different when compared to
> others (alias for instance which I am still not sure what would break
> if removed) do require some asking if after reading the online
> documentation and previous archived messages the reason is still not
> clear.
>
> So I am not sure how you expect people to ask questions or if this is
> more a "this is what it is" take it or leave it dictatorship approach.
> Not the paradigm followed in open source projects where collaboration,
> asking and questioning is usually the  way to go.
>
> Interestingly enough in this one case I DON'T APPRECIATE YOUR
> RESPONSE. Now tell me if that doesn't sound sincere to you. In any
> case I will continue to play with the system and possibly ask
> questions as well.
>
> Alfredo
>
> On Sun, Jan 17, 2010 at 12:20 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > We do offer a triple your money back guarentee!
> >
> > Many params can change without restarting and many can't ip/port fo
> > example.  FYI, You don't sound sincere when you use the term appriciate
> when
> > you are arguing with these guys voulenteering to help explain it to you.
> > You should make sure you make the most of their assistance as they could
> > have just said rtfm
> >
> > On Jan 16, 2010 11:07 PM, "Alfredo Quiroga-Villamil" <lawwton at gmail.com>
> > wrote:
> >
> > mm_202:
> >
> > Thanks for the reply.
> >
> > I don't agree with your statement: "In a production system, you
> >
> > *shouldnt* have to change the main profiles at all."
> >
> > I think it's always good to be able to reload things dynamically in a
> > non-disruptive way.
> >
> > The alias is a bit unusual. To be honest I had never seen that before
> > in any of the Telecommunication Systems I've worked on. Not that is
> > not a good idea, just one that at first glance doesn't seem too clear
> > for some reason. Specially when you see two profiles both representing
> > the same object listed under "sofia status". Being forced to have the
> > alias doesn't seem like an appealing option, should be optional I
> > think. I would like if possible a more detailed explanation on what
> > would break if not present.
> >
> > In any case, I am sure there is a reason for these things. I am trying
> > to understand how they all work together. Things are not as apparent
> > in FS at first glance when compared to other systems; but the platform
> > seems to be built with the purpose of offering a lot of flexibility
> > which is really good.
> >
> > Appreciate the feedback.
> >
> > Alfredo Q-V
> >
> > Appreciate the help,
> >
> > Alfredo
> >
> > On Sat, Jan 16, 2010 at 10:35 PM, mm_202 <mastermind202 at gmail.com>
> wrote: >
> >> On Sat, Jan 16, 201...
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

Alfredo,

What I meant by "In a production system, you *shouldnt* have to change the
main profiles at all", I meant core things like IP address / port; once they
are setup, they shouldnt change. FreeSWITCH is extremely dynamic and almost
everything can be changed on the fly without impacting production.

When I first started using FS, I was also a bit curious why there were
profile aliases, but I assure you that they are NOT a bad thing.  Just play
with FS some more and get a feel for it, you may even run into a situation
when you actually need the aliases :)

As for Anthony Minessale (anthm), he is the main creator of FreeSWITCH.  He
constantly deals with new users critizing FS before even using it and/or
reading the docs, hence he can come off quite direct sometimes. I would
guess that there was just a misunderstanding between you two.

Regardless, hopefully you'll continue to play with FS and see what myself
and so many other people did, -- a kickass system!

-- mm_202.
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From max.bridgewater at gmail.com  Sun Jan 17 14:02:21 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 17 Jan 2010 17:02:21 -0500
Subject: [Freeswitch-users] Trouble with Originate/Bridge from ESL
	[SOLVED]
Message-ID: <c55d46951001171402i6f906972p896c5b8c584bf919@mail.gmail.com>

Given that I'm using Java, I had to escape the ringback value's quote twice:

originate
{ringback=\\'%(2000,4000,440.0,480.0)\\'}sofia/internal/1004%74.63.243.52
&bridge(sofia/internal/1005%74.63.243.52)


On Sun, Jan 17, 2010 at 3:51 PM, Max Bridgewater
<max.bridgewater at gmail.com>wrote:

> One more piece of information: the call is being terminated by Freeswitch
> with the event:
>
> Event-Name: CHANNEL_HANGUP
> Hangup-Cause: NO_ANSWER
>
> Which is strange because B leg doesn't even have the time to answer.
>
>
>
> On Sun, Jan 17, 2010 at 1:55 PM, Max Bridgewater <
> max.bridgewater at gmail.com> wrote:
>
>> Hi,
>>
>> The following command works great from the command line:
>> originate
>> {ringback=\'%(2000,4000,440.0,480.0)\'}sofia/internal/1004%74.63.243.52
>> &bridge(sofia/internal/1005%74.63.243.52)
>>
>> But this one isn't working from the ESL:
>> api originate
>> {ringback=\'%(2000,4000,440.0,480.0)\'}sofia/internal/1004%74.63.243.52
>> &bridge(sofia/internal/1005%74.63.243.52)
>>
>> The observed behavior is following: The A leg is dialed, then B leg is
>> also dialed but immediately followed by a hangup. That is, the B user
>> doesn't even has time to answer.
>>
>> Any idea what I'm doing wrong?
>>
>> Thanks,
>> Max.
>
>
>
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From lists at redbonez.net  Sun Jan 17 14:20:58 2010
From: lists at redbonez.net (Adam Ford)
Date: Sun, 17 Jan 2010 15:20:58 -0700
Subject: [Freeswitch-users] Polycom configuration for FreeSWITCH
In-Reply-To: <d361ca7b1001152157x45b8d8c4l29625c763b24157f@mail.gmail.com>
References: <003701ca964b$3241b100$96c51300$@net>
	<d361ca7b1001152157x45b8d8c4l29625c763b24157f@mail.gmail.com>
Message-ID: <26AF402E733A4882B088F846AC2AE3CD@redbonez>

Thank you very much.  Mike is right, if you would be willing to post a
breakdown of the configurations that relate to FreeSWITCH on the wiki that
would be great.  I understand it would be easier to answer a specific
question, but I am not really looking for anything specific just a
comparison.  Yes, I have all the phones booting from an FTP server, they are
running the latest bootrom and SIP software that they support (4.1.4 &
3.1.4), and they register with FreeSWITCH ok.  I am mostly just looking for
a working config as an example to make sure I am not missing something, or
maybe not using the best settings for FreeSWITCH. 

 

I greatly appreciate the help,

 

-Adam

 

  

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yehavi
Bourvine
Sent: Friday, January 15, 2010 10:57 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom configuration for FreeSWITCH

 

I am working with various Polycom phones; I'll send you sample configuration
files next week (I am at home now). In the meantime, please send me your
requirenents so I may incorporate some of them into the files.

 

Have you managed to boot them from your TFTP/FTP./HTTP server? As long as
you did not provision them through a server you can do that through the
phone's WEB interface, but it is very limited and lacks a lot of
configuration options. I do the provisioning via a TFTP server.

 

                    Regards, __Yehavi:

2010/1/16 Adam Ford <lists at redbonez.net>

Does anyone who has successfully implemented Polycom IP301/501s, with
FreeSWITCH, have a configuration they can share with me as an example? I
have read the Polycom Admin Guide several times and understand what the
settings are/do, I am just not sure which FreeSWITCH supports, which it
doesn't, and which need special configuration to work with FreeSWITCH.

 

Thanks in advance,

-Adam

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org <http://www.freeswitch.org/> 

 

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From lawwton at gmail.com  Sun Jan 17 14:54:41 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sun, 17 Jan 2010 17:54:41 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <63de75711001171353hca2a330he29cf2760f355a3d@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<63de75711001161839r48f900e5o3ca04165219d67ff@mail.gmail.com>
	<5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>
	<63de75711001161935w2d1e204ek81ebe1143afb1497@mail.gmail.com>
	<5fe6fa8f1001162100n699077faw8ac1e8cef4e4ea5@mail.gmail.com>
	<191c3a031001162117u413f0d5bi1903c69e519c3632@mail.gmail.com>
	<191c3a031001162120k50fcca08l667fd3b842d5ebbd@mail.gmail.com>
	<191c3a031001162120j499f0e9xe359bb9a97607415@mail.gmail.com>
	<5fe6fa8f1001170607v52c0516egc6f69a60a934722f@mail.gmail.com>
	<63de75711001171353hca2a330he29cf2760f355a3d@mail.gmail.com>
Message-ID: <5fe6fa8f1001171454g6bfad0dy7a9d87238be5d3a0@mail.gmail.com>

Thanks mm_202.

Yeah, I figured he was having a bad day or something, don't think my
reply to you was offensive or contained criticism in any way. So I
said that I really appreciated your help which I of course meant. To
see his response was a bit of a shocker since I made sure I stated I
understood most of the time things are done in a way for a reason. I
had also spent a bit of time reading up documentation and trying
things before I posted the question.

So far I've been reading and trying to come up to speed and I really
like what I see. I will definitely continue to explore it and
hopefully use it in a production system one day.

Regards,

Alfredo

On Sun, Jan 17, 2010 at 4:53 PM, mm_202 <mastermind202 at gmail.com> wrote:
>
>
> On Sun, Jan 17, 2010 at 9:07 AM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> I am pretty sincere when I say "I appreciate the help", otherwise I
>> wouldn't say it. I don't think I've argued in any of my previous
>> statements and I've tried to be very polite when raising questions or
>> concerns. Disagreeing in a polite way doesn't necessarily mean that I
>> am 100% rejecting the idea and by the way it's not a bad thing to
>> disagree. No one said "You are wrong"; but "I disagree". There is a
>> difference there.
>>
>> If you expect to put a system out there and new users that have used
>> other ones not to ask why this or that I think you'll keep running
>> into this with other people. No one is criticizing or putting down the
>> system here, on the contrary I am trying to simply find out how it
>> works so I can use it. But things like reloading configurations and
>> specific system details that are quite different when compared to
>> others (alias for instance which I am still not sure what would break
>> if removed) do require some asking if after reading the online
>> documentation and previous archived messages the reason is still not
>> clear.
>>
>> So I am not sure how you expect people to ask questions or if this is
>> more a "this is what it is" take it or leave it dictatorship approach.
>> Not the paradigm followed in open source projects where collaboration,
>> asking and questioning is usually the ?way to go.
>>
>> Interestingly enough in this one case I DON'T APPRECIATE YOUR
>> RESPONSE. Now tell me if that doesn't sound sincere to you. In any
>> case I will continue to play with the system and possibly ask
>> questions as well.
>>
>> Alfredo
>>
>> On Sun, Jan 17, 2010 at 12:20 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > We do offer a triple your money back guarentee!
>> >
>> > Many params can change without restarting and many can't ip/port fo
>> > example.? FYI, You don't sound sincere when you use the term appriciate
>> > when
>> > you are arguing with these guys voulenteering to help explain it to you.
>> > You should make sure you make the most of their assistance as they could
>> > have just said rtfm
>> >
>> > On Jan 16, 2010 11:07 PM, "Alfredo Quiroga-Villamil" <lawwton at gmail.com>
>> > wrote:
>> >
>> > mm_202:
>> >
>> > Thanks for the reply.
>> >
>> > I don't agree with your statement: "In a production system, you
>> >
>> > *shouldnt* have to change the main profiles at all."
>> >
>> > I think it's always good to be able to reload things dynamically in a
>> > non-disruptive way.
>> >
>> > The alias is a bit unusual. To be honest I had never seen that before
>> > in any of the Telecommunication Systems I've worked on. Not that is
>> > not a good idea, just one that at first glance doesn't seem too clear
>> > for some reason. Specially when you see two profiles both representing
>> > the same object listed under "sofia status". Being forced to have the
>> > alias doesn't seem like an appealing option, should be optional I
>> > think. I would like if possible a more detailed explanation on what
>> > would break if not present.
>> >
>> > In any case, I am sure there is a reason for these things. I am trying
>> > to understand how they all work together. Things are not as apparent
>> > in FS at first glance when compared to other systems; but the platform
>> > seems to be built with the purpose of offering a lot of flexibility
>> > which is really good.
>> >
>> > Appreciate the feedback.
>> >
>> > Alfredo Q-V
>> >
>> > Appreciate the help,
>> >
>> > Alfredo
>> >
>> > On Sat, Jan 16, 2010 at 10:35 PM, mm_202 <mastermind202 at gmail.com>
>> > wrote: >
>> >> On Sat, Jan 16, 201...
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> Alfredo,
>
> What I meant by "In a production system, you *shouldnt* have to change the
> main profiles at all", I meant core things like IP address / port; once they
> are setup, they shouldnt change. FreeSWITCH is extremely dynamic and almost
> everything can be changed on the fly without impacting production.
>
> When I first started using FS, I was also a bit curious why there were
> profile aliases, but I assure you that they are NOT a bad thing.? Just play
> with FS some more and get a feel for it, you may even run into a situation
> when you actually need the aliases :)
>
> As for Anthony Minessale (anthm), he is the main creator of FreeSWITCH.? He
> constantly deals with new users critizing FS before even using it and/or
> reading the docs, hence he can come off quite direct sometimes. I would
> guess that there was just a misunderstanding between you two.
>
> Regardless, hopefully you'll continue to play with FS and see what myself
> and so many other people did, -- a kickass system!
>
> -- mm_202.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From mastermind202 at gmail.com  Sun Jan 17 15:09:14 2010
From: mastermind202 at gmail.com (mm_202)
Date: Sun, 17 Jan 2010 18:09:14 -0500
Subject: [Freeswitch-users] sip_profiles - Aliases Question
In-Reply-To: <5fe6fa8f1001171454g6bfad0dy7a9d87238be5d3a0@mail.gmail.com>
References: <5fe6fa8f1001161738y4baf95c1p4daa770790cf35f9@mail.gmail.com>
	<5fe6fa8f1001161900u726d6521gfa602d0cef8d000e@mail.gmail.com>
	<63de75711001161935w2d1e204ek81ebe1143afb1497@mail.gmail.com>
	<5fe6fa8f1001162100n699077faw8ac1e8cef4e4ea5@mail.gmail.com>
	<191c3a031001162117u413f0d5bi1903c69e519c3632@mail.gmail.com>
	<191c3a031001162120k50fcca08l667fd3b842d5ebbd@mail.gmail.com>
	<191c3a031001162120j499f0e9xe359bb9a97607415@mail.gmail.com>
	<5fe6fa8f1001170607v52c0516egc6f69a60a934722f@mail.gmail.com>
	<63de75711001171353hca2a330he29cf2760f355a3d@mail.gmail.com>
	<5fe6fa8f1001171454g6bfad0dy7a9d87238be5d3a0@mail.gmail.com>
Message-ID: <63de75711001171509q67ca6b0ap747fa992f3cd97e2@mail.gmail.com>

On Sun, Jan 17, 2010 at 5:54 PM, Alfredo Quiroga-Villamil <lawwton at gmail.com
> wrote:

> Thanks mm_202.
>
> Yeah, I figured he was having a bad day or something, don't think my
> reply to you was offensive or contained criticism in any way. So I
> said that I really appreciated your help which I of course meant. To
> see his response was a bit of a shocker since I made sure I stated I
> understood most of the time things are done in a way for a reason. I
> had also spent a bit of time reading up documentation and trying
> things before I posted the question.
>
> So far I've been reading and trying to come up to speed and I really
> like what I see. I will definitely continue to explore it and
> hopefully use it in a production system one day.
>
> Regards,
>
> Alfredo
>
> On Sun, Jan 17, 2010 at 4:53 PM, mm_202 <mastermind202 at gmail.com> wrote:
> >
> >
> > On Sun, Jan 17, 2010 at 9:07 AM, Alfredo Quiroga-Villamil
> > <lawwton at gmail.com> wrote:
> >>
> >> I am pretty sincere when I say "I appreciate the help", otherwise I
> >> wouldn't say it. I don't think I've argued in any of my previous
> >> statements and I've tried to be very polite when raising questions or
> >> concerns. Disagreeing in a polite way doesn't necessarily mean that I
> >> am 100% rejecting the idea and by the way it's not a bad thing to
> >> disagree. No one said "You are wrong"; but "I disagree". There is a
> >> difference there.
> >>
> >> If you expect to put a system out there and new users that have used
> >> other ones not to ask why this or that I think you'll keep running
> >> into this with other people. No one is criticizing or putting down the
> >> system here, on the contrary I am trying to simply find out how it
> >> works so I can use it. But things like reloading configurations and
> >> specific system details that are quite different when compared to
> >> others (alias for instance which I am still not sure what would break
> >> if removed) do require some asking if after reading the online
> >> documentation and previous archived messages the reason is still not
> >> clear.
> >>
> >> So I am not sure how you expect people to ask questions or if this is
> >> more a "this is what it is" take it or leave it dictatorship approach.
> >> Not the paradigm followed in open source projects where collaboration,
> >> asking and questioning is usually the  way to go.
> >>
> >> Interestingly enough in this one case I DON'T APPRECIATE YOUR
> >> RESPONSE. Now tell me if that doesn't sound sincere to you. In any
> >> case I will continue to play with the system and possibly ask
> >> questions as well.
> >>
> >> Alfredo
> >>
> >> On Sun, Jan 17, 2010 at 12:20 AM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >> > We do offer a triple your money back guarentee!
> >> >
> >> > Many params can change without restarting and many can't ip/port fo
> >> > example.  FYI, You don't sound sincere when you use the term
> appriciate
> >> > when
> >> > you are arguing with these guys voulenteering to help explain it to
> you.
> >> > You should make sure you make the most of their assistance as they
> could
> >> > have just said rtfm
> >> >
> >> > On Jan 16, 2010 11:07 PM, "Alfredo Quiroga-Villamil" <
> lawwton at gmail.com>
> >> > wrote:
> >> >
> >> > mm_202:
> >> >
> >> > Thanks for the reply.
> >> >
> >> > I don't agree with your statement: "In a production system, you
> >> >
> >> > *shouldnt* have to change the main profiles at all."
> >> >
> >> > I think it's always good to be able to reload things dynamically in a
> >> > non-disruptive way.
> >> >
> >> > The alias is a bit unusual. To be honest I had never seen that before
> >> > in any of the Telecommunication Systems I've worked on. Not that is
> >> > not a good idea, just one that at first glance doesn't seem too clear
> >> > for some reason. Specially when you see two profiles both representing
> >> > the same object listed under "sofia status". Being forced to have the
> >> > alias doesn't seem like an appealing option, should be optional I
> >> > think. I would like if possible a more detailed explanation on what
> >> > would break if not present.
> >> >
> >> > In any case, I am sure there is a reason for these things. I am trying
> >> > to understand how they all work together. Things are not as apparent
> >> > in FS at first glance when compared to other systems; but the platform
> >> > seems to be built with the purpose of offering a lot of flexibility
> >> > which is really good.
> >> >
> >> > Appreciate the feedback.
> >> >
> >> > Alfredo Q-V
> >> >
> >> > Appreciate the help,
> >> >
> >> > Alfredo
> >> >
> >> > On Sat, Jan 16, 2010 at 10:35 PM, mm_202 <mastermind202 at gmail.com>
> >> > wrote: >
> >> >> On Sat, Jan 16, 201...
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > Alfredo,
> >
> > What I meant by "In a production system, you *shouldnt* have to change
> the
> > main profiles at all", I meant core things like IP address / port; once
> they
> > are setup, they shouldnt change. FreeSWITCH is extremely dynamic and
> almost
> > everything can be changed on the fly without impacting production.
> >
> > When I first started using FS, I was also a bit curious why there were
> > profile aliases, but I assure you that they are NOT a bad thing.  Just
> play
> > with FS some more and get a feel for it, you may even run into a
> situation
> > when you actually need the aliases :)
> >
> > As for Anthony Minessale (anthm), he is the main creator of FreeSWITCH.
> He
> > constantly deals with new users critizing FS before even using it and/or
> > reading the docs, hence he can come off quite direct sometimes. I would
> > guess that there was just a misunderstanding between you two.
> >
> > Regardless, hopefully you'll continue to play with FS and see what myself
> > and so many other people did, -- a kickass system!
> >
> > -- mm_202.
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


Excellent to hear!  If you have any more questions or even curiosities,
don't hesitate to ask.
The FS community is great in answering even the most perplexing questions.

You should also (if you havent yet) join us on IRC in #freeswitch (
irc.freenode.net), you'll get answers much
quicker and see more of the 'community'.

-- mm_202.
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From jingwei.yang at gmail.com  Sun Jan 17 17:55:09 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 18 Jan 2010 09:55:09 +0800
Subject: [Freeswitch-users] Questions about mod_fifo
In-Reply-To: <23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>
References: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
	<23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>
Message-ID: <13529f9d1001171755v229c2137u4fa5cfe2f30d518@mail.gmail.com>

Thanks for replying. This is my dialplan

    <extension name="in_fifo">
      <condition field="destination_number" expression="^779$">
        <action application="set" data="fifo_music=$${hold_music}"/>
        <action application="answer"/>
        <action application="fifo" data="myq in undef undef"/>
      </condition>
    </extension>

And I created a queue in fifo.conf.xml like this

    <fifo name="myq" importance="0">
      <!--<member timeout="60" simo="1"
lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
    </fifo>

However, I'm still not able to see the incoming call get queued.

freeswitch at localhost.localdomain> fifo list myq
API CALL [fifo(list myq)] output:
<fifo_report>
  <fifo name="myq" consumer_count="0" caller_count="0" waiting_count="0"
importance="0">
    <callers></callers>
    <consumers></consumers>
  </fifo>
</fifo_report>

I tried both mod_skypiax and mod_dingaling, but with the same result.

Regards,
-Jingwei

On Sun, Jan 17, 2010 at 11:37 AM, Seven Du <dujinfang at gmail.com> wrote:

> 2010/1/15, Jingwei Yang <jingwei.yang at gmail.com>:
> > Hi Guys,
> >
> > I'm implementing an ACD system using ESL and mod_fifo. Based on what
> Anthony
> > suggested in this post:
> >
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html
> >
> > *You can make an event socket application that listens for FIFO events
> and
> > keeps track of what FIFOs are currently busy and when there are people
> > waiting you can have that script generate a call to a group of SIP phones
> so
> > when the first one answers, it sends them in as an agent where they can
> > field the calls.
> > *
> >
> > 1. How should I handle the concurrent issues? If I bridge a user to two
> > agents and both of them answers, how does FS take care of this situation?
> > Will a slower agent get a busy tone automatically?
> >
>
> I think it just follow the standard originate dialstring rules.
>
> > 2. If the socket application is brought up after some users have called
> in,
> > what command should I use to check the busy queues? fifo list?
> >
> Yes. Perhaps you can also check the fifo db, either sqlite or ODBC.
>
> > 3. Am I using fifo list and fifo count correctly?
> >
> > here's the testing dialplan:
> >
> >     <extension name="in_fifo">
> >       <condition field="destination_number" expression="^779$">
> >         <action application="set" data="fifo_music=$${hold_music}"/>
> >         <action application="answer"/>
> >         <action application="fifo" data="*myq *in"/>
> >       </condition>
> >     </extension>
> >
> > when a call comes in and gets queued, these are the results of some
> commands
> > I tried.
> >
> > freeswitch at localhost.localdomain> fifo list
> > API CALL [fifo(list)] output:
> > <fifo_report>
> >   <fifo name="cool_fifo at 192.168.1.104" consumer_count="0"
> caller_count="0"
> > waiting_count="0" importance="0">
> >     <callers></callers>
> >     <consumers></consumers>
> >   </fifo>
> > </fifo_report>
> >
> >
> > freeswitch at localhost.localdomain> fifo list myq
> > API CALL [fifo(list myq)] output:
> > <fifo_report></fifo_report>
> >
> >
> > freeswitch at localhost.localdomain> fifo count myq
> > API CALL [fifo(count myq)] output:
> > none
> >
> > It seems *myq* doesn't get created at all? Please enlighten.
> >
> > Thanks and best regards,
> > -Jingwei
> >
> AFAIK, thant means the channel didn't queued in. Did you see any error
> logs? I think you need to remove the stars in   <action
> application="fifo" data="*myq *in"/>.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From darklion11 at yahoo.com  Sun Jan 17 18:13:32 2010
From: darklion11 at yahoo.com (Edmar Cruz)
Date: Sun, 17 Jan 2010 18:13:32 -0800 (PST)
Subject: [Freeswitch-users]  Change Ip to Domain Name
Message-ID: <27104680.post@talk.nabble.com>


Dear All,

   How can i change the domain of my freeswitch 52.236.125.12 to
sip.grandminister.com to be able to detect the presence of the user whos
online or not...

Thanks
Edmar
-- 
View this message in context: http://old.nabble.com/Change-Ip-to-Domain-Name-tp27104680p27104680.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From dujinfang at gmail.com  Sun Jan 17 19:42:28 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 18 Jan 2010 11:42:28 +0800
Subject: [Freeswitch-users] Questions about mod_fifo
In-Reply-To: <13529f9d1001171755v229c2137u4fa5cfe2f30d518@mail.gmail.com>
References: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
	<23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>
	<13529f9d1001171755v229c2137u4fa5cfe2f30d518@mail.gmail.com>
Message-ID: <23f91031001171942h42e655c9vbc5ecb4165b7aca9@mail.gmail.com>

better to pastebin your log.

2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> Thanks for replying. This is my dialplan
>
> ??? <extension name="in_fifo">
> ????? <condition field="destination_number" expression="^779$">
> ??????? <action application="set" data="fifo_music=$${hold_music}"/>
> ??????? <action application="answer"/>
> ??????? <action application="fifo" data="myq in undef undef"/>
> ????? </condition>
> ??? </extension>
>
> And I created a queue in fifo.conf.xml like this
>
> ??? <fifo name="myq" importance="0">
> ????? <!--<member timeout="60" simo="1"
> lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
> ??? </fifo>
>
> However, I'm still not able to see the incoming call get queued.
>
> freeswitch at localhost.localdomain> fifo list myq
> API CALL [fifo(list myq)] output:
> <fifo_report>
> ? <fifo name="myq" consumer_count="0" caller_count="0" waiting_count="0"
> importance="0">
> ??? <callers></callers>
> ??? <consumers></consumers>
> ? </fifo>
> </fifo_report>
>
> I tried both mod_skypiax and mod_dingaling, but with the same result.
>
> Regards,
> -Jingwei
>
> On Sun, Jan 17, 2010 at 11:37 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> 2010/1/15, Jingwei Yang <jingwei.yang at gmail.com>:
>> > Hi Guys,
>> >
>> > I'm implementing an ACD system using ESL and mod_fifo. Based on what
>> > Anthony
>> > suggested in this post:
>> >
>> > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html
>> >
>> > *You can make an event socket application that listens for FIFO events
>> > and
>> > keeps track of what FIFOs are currently busy and when there are people
>> > waiting you can have that script generate a call to a group of SIP
>> > phones so
>> > when the first one answers, it sends them in as an agent where they can
>> > field the calls.
>> > *
>> >
>> > 1. How should I handle the concurrent issues? If I bridge a user to two
>> > agents and both of them answers, how does FS take care of this
>> > situation?
>> > Will a slower agent get a busy tone automatically?
>> >
>>
>> I think it just follow the standard originate dialstring rules.
>>
>> > 2. If the socket application is brought up after some users have called
>> > in,
>> > what command should I use to check the busy queues? fifo list?
>> >
>> Yes. Perhaps you can also check the fifo db, either sqlite or ODBC.
>>
>> > 3. Am I using fifo list and fifo count correctly?
>> >
>> > here's the testing dialplan:
>> >
>> > ? ? <extension name="in_fifo">
>> > ? ? ? <condition field="destination_number" expression="^779$">
>> > ? ? ? ? <action application="set" data="fifo_music=$${hold_music}"/>
>> > ? ? ? ? <action application="answer"/>
>> > ? ? ? ? <action application="fifo" data="*myq *in"/>
>> > ? ? ? </condition>
>> > ? ? </extension>
>> >
>> > when a call comes in and gets queued, these are the results of some
>> > commands
>> > I tried.
>> >
>> > freeswitch at localhost.localdomain> fifo list
>> > API CALL [fifo(list)] output:
>> > <fifo_report>
>> > ? <fifo name="cool_fifo at 192.168.1.104" consumer_count="0"
>> > caller_count="0"
>> > waiting_count="0" importance="0">
>> > ? ? <callers></callers>
>> > ? ? <consumers></consumers>
>> > ? </fifo>
>> > </fifo_report>
>> >
>> >
>> > freeswitch at localhost.localdomain> fifo list myq
>> > API CALL [fifo(list myq)] output:
>> > <fifo_report></fifo_report>
>> >
>> >
>> > freeswitch at localhost.localdomain> fifo count myq
>> > API CALL [fifo(count myq)] output:
>> > none
>> >
>> > It seems *myq* doesn't get created at all? Please enlighten.
>> >
>> > Thanks and best regards,
>> > -Jingwei
>> >
>> AFAIK, thant means the channel didn't queued in. Did you see any error
>> logs? I think you need to remove the stars in ? <action
>> application="fifo" data="*myq *in"/>.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Sun Jan 17 20:13:20 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 17 Jan 2010 22:13:20 -0600
Subject: [Freeswitch-users] Change Ip to Domain Name
In-Reply-To: <27104680.post@talk.nabble.com>
References: <27104680.post@talk.nabble.com>
Message-ID: <FF1ABD07-18BA-412B-9F3B-D0178D878883@freeswitch.org>

Change it in the config and setup proper DNS.

/b

On Jan 17, 2010, at 8:13 PM, Edmar Cruz wrote:

> 
> Dear All,
> 
>   How can i change the domain of my freeswitch 52.236.125.12 to
> sip.grandminister.com to be able to detect the presence of the user whos
> online or not...
> 
> Thanks
> Edmar

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From brian at freeswitch.org  Sun Jan 17 20:15:33 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 17 Jan 2010 22:15:33 -0600
Subject: [Freeswitch-users] Trouble with Originate/Bridge from ESL
	[SOLVED]
In-Reply-To: <c55d46951001171402i6f906972p896c5b8c584bf919@mail.gmail.com>
References: <c55d46951001171402i6f906972p896c5b8c584bf919@mail.gmail.com>
Message-ID: <062F43CC-5C35-4415-B202-81551BD71E1C@freeswitch.org>

Its not just java!  :P 

/b

On Jan 17, 2010, at 4:02 PM, Max Bridgewater wrote:

> Given that I'm using Java, I had to escape the ringback value's quote twice:
> 
> originate {ringback=\\'%(2000,4000,440.0,480.0)\\'}sofia/internal/1004%74.63.243.52 &bridge(sofia/internal/1005%74.63.243.52)




From brian at freeswitch.org  Sun Jan 17 20:15:53 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 17 Jan 2010 22:15:53 -0600
Subject: [Freeswitch-users] Trouble with Originate/Bridge from ESL
	[SOLVED]
In-Reply-To: <c55d46951001171402i6f906972p896c5b8c584bf919@mail.gmail.com>
References: <c55d46951001171402i6f906972p896c5b8c584bf919@mail.gmail.com>
Message-ID: <5696EFC4-8DC0-42B0-A909-02CA03000E84@freeswitch.org>

Oh btw happen to wanna write me some sample ESL java samples to put in tree.. see the examples in the perl folder.

/b

On Jan 17, 2010, at 4:02 PM, Max Bridgewater wrote:

> Given that I'm using Java, I had to escape the ringback value's quote twice:
> 
> originate {ringback=\\'%(2000,4000,440.0,480.0)\\'}sofia/internal/1004%74.63.243.52 &bridge(sofia/internal/1005%74.63.243.52)




From brian at freeswitch.org  Sun Jan 17 20:16:36 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 17 Jan 2010 22:16:36 -0600
Subject: [Freeswitch-users] No media after Originate
In-Reply-To: <f97004c31001161522g65a0da7agec1eabffa321692d@mail.gmail.com>
References: <f97004c31001161306o10517d1y9e0cf0229bce3589@mail.gmail.com>
	<3081C466-F735-47F1-BAEC-2B497F8F08D8@freeswitch.org>
	<f97004c31001161522g65a0da7agec1eabffa321692d@mail.gmail.com>
Message-ID: <D44E6C4F-58D0-4925-B025-2C34FB6E9D53@freeswitch.org>

Are you trying to do bypass media?

/b

On Jan 16, 2010, at 5:22 PM, Mohamed Hassan wrote:

> There is no nat my server has public ip and not nated
> and my sip provider too  as i can make regular calls through the same
> provider without originate




From max.bridgewater at gmail.com  Sun Jan 17 20:23:27 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 17 Jan 2010 23:23:27 -0500
Subject: [Freeswitch-users] Trouble with Originate/Bridge from ESL
	[SOLVED]
In-Reply-To: <5696EFC4-8DC0-42B0-A909-02CA03000E84@freeswitch.org>
References: <c55d46951001171402i6f906972p896c5b8c584bf919@mail.gmail.com>
	<5696EFC4-8DC0-42B0-A909-02CA03000E84@freeswitch.org>
Message-ID: <c55d46951001172023w1b2e5d4fm22f23d7d58d079a4@mail.gmail.com>

I might. Stay tuned. Self imposed deadline: Feb 14th ;).

On Sun, Jan 17, 2010 at 11:15 PM, Brian West <brian at freeswitch.org> wrote:

> Oh btw happen to wanna write me some sample ESL java samples to put in
> tree.. see the examples in the perl folder.
>
> /b
>
> On Jan 17, 2010, at 4:02 PM, Max Bridgewater wrote:
>
> > Given that I'm using Java, I had to escape the ringback value's quote
> twice:
> >
> > originate
> {ringback=\\'%(2000,4000,440.0,480.0)\\'}sofia/internal/1004%74.63.243.52
> &bridge(sofia/internal/1005%74.63.243.52)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jingwei.yang at gmail.com  Sun Jan 17 22:48:51 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 18 Jan 2010 14:48:51 +0800
Subject: [Freeswitch-users] Questions about mod_fifo
In-Reply-To: <23f91031001171942h42e655c9vbc5ecb4165b7aca9@mail.gmail.com>
References: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
	<23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>
	<13529f9d1001171755v229c2137u4fa5cfe2f30d518@mail.gmail.com>
	<23f91031001171942h42e655c9vbc5ecb4165b7aca9@mail.gmail.com>
Message-ID: <13529f9d1001172248x1a1ddc8arb7f4aba2e71ffb0f@mail.gmail.com>

Logs submitted: http://pastebin.freeswitch.org/11836

I was trying to check whether the call had been added into the queue via
telnet, but failed to find the fifo events. Here's my simplified dialplan:

<extension name="in_fifo">
   <condition field="destination_number" expression="^779$">
     <action application="answer"/>
     <action application="fifo" data="myq in undef undef"/>
  </condition>
</extension>

Please advise where went wrong.

Thanks and best regards,
-Jingwei


On Mon, Jan 18, 2010 at 11:42 AM, Seven Du <dujinfang at gmail.com> wrote:

> better to pastebin your log.
>
> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> > Thanks for replying. This is my dialplan
> >
> >     <extension name="in_fifo">
> >       <condition field="destination_number" expression="^779$">
> >         <action application="set" data="fifo_music=$${hold_music}"/>
> >         <action application="answer"/>
> >         <action application="fifo" data="myq in undef undef"/>
> >       </condition>
> >     </extension>
> >
> > And I created a queue in fifo.conf.xml like this
> >
> >     <fifo name="myq" importance="0">
> >       <!--<member timeout="60" simo="1"
> > lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
> >     </fifo>
> >
> > However, I'm still not able to see the incoming call get queued.
> >
> > freeswitch at localhost.localdomain> fifo list myq
> > API CALL [fifo(list myq)] output:
> > <fifo_report>
> >   <fifo name="myq" consumer_count="0" caller_count="0" waiting_count="0"
> > importance="0">
> >     <callers></callers>
> >     <consumers></consumers>
> >   </fifo>
> > </fifo_report>
> >
> > I tried both mod_skypiax and mod_dingaling, but with the same result.
> >
> > Regards,
> > -Jingwei
> >
> > On Sun, Jan 17, 2010 at 11:37 AM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> 2010/1/15, Jingwei Yang <jingwei.yang at gmail.com>:
> >> > Hi Guys,
> >> >
> >> > I'm implementing an ACD system using ESL and mod_fifo. Based on what
> >> > Anthony
> >> > suggested in this post:
> >> >
> >> >
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html
> >> >
> >> > *You can make an event socket application that listens for FIFO events
> >> > and
> >> > keeps track of what FIFOs are currently busy and when there are people
> >> > waiting you can have that script generate a call to a group of SIP
> >> > phones so
> >> > when the first one answers, it sends them in as an agent where they
> can
> >> > field the calls.
> >> > *
> >> >
> >> > 1. How should I handle the concurrent issues? If I bridge a user to
> two
> >> > agents and both of them answers, how does FS take care of this
> >> > situation?
> >> > Will a slower agent get a busy tone automatically?
> >> >
> >>
> >> I think it just follow the standard originate dialstring rules.
> >>
> >> > 2. If the socket application is brought up after some users have
> called
> >> > in,
> >> > what command should I use to check the busy queues? fifo list?
> >> >
> >> Yes. Perhaps you can also check the fifo db, either sqlite or ODBC.
> >>
> >> > 3. Am I using fifo list and fifo count correctly?
> >> >
> >> > here's the testing dialplan:
> >> >
> >> >     <extension name="in_fifo">
> >> >       <condition field="destination_number" expression="^779$">
> >> >         <action application="set" data="fifo_music=$${hold_music}"/>
> >> >         <action application="answer"/>
> >> >         <action application="fifo" data="*myq *in"/>
> >> >       </condition>
> >> >     </extension>
> >> >
> >> > when a call comes in and gets queued, these are the results of some
> >> > commands
> >> > I tried.
> >> >
> >> > freeswitch at localhost.localdomain> fifo list
> >> > API CALL [fifo(list)] output:
> >> > <fifo_report>
> >> >   <fifo name="cool_fifo at 192.168.1.104" consumer_count="0"
> >> > caller_count="0"
> >> > waiting_count="0" importance="0">
> >> >     <callers></callers>
> >> >     <consumers></consumers>
> >> >   </fifo>
> >> > </fifo_report>
> >> >
> >> >
> >> > freeswitch at localhost.localdomain> fifo list myq
> >> > API CALL [fifo(list myq)] output:
> >> > <fifo_report></fifo_report>
> >> >
> >> >
> >> > freeswitch at localhost.localdomain> fifo count myq
> >> > API CALL [fifo(count myq)] output:
> >> > none
> >> >
> >> > It seems *myq* doesn't get created at all? Please enlighten.
> >> >
> >> > Thanks and best regards,
> >> > -Jingwei
> >> >
> >> AFAIK, thant means the channel didn't queued in. Did you see any error
> >> logs? I think you need to remove the stars in   <action
> >> application="fifo" data="*myq *in"/>.
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mailinglist at fribert.dk  Sun Jan 17 22:50:30 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Mon, 18 Jan 2010 07:50:30 +0100
Subject: [Freeswitch-users] How do I invite group to join existing call?
Message-ID: <4B5412C6020000E1000003D6@mail.fribert.dk>

Hi All

I would like to be able to invite a group / global to join an existing call, but how do I accomplish this, can it be done?


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From dujinfang at gmail.com  Sun Jan 17 23:25:23 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 18 Jan 2010 15:25:23 +0800
Subject: [Freeswitch-users] Questions about mod_fifo
In-Reply-To: <13529f9d1001172248x1a1ddc8arb7f4aba2e71ffb0f@mail.gmail.com>
References: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
	<23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>
	<13529f9d1001171755v229c2137u4fa5cfe2f30d518@mail.gmail.com>
	<23f91031001171942h42e655c9vbc5ecb4165b7aca9@mail.gmail.com>
	<13529f9d1001172248x1a1ddc8arb7f4aba2e71ffb0f@mail.gmail.com>
Message-ID: <23f91031001172325n2dd0f462ga202abec3a79d019@mail.gmail.com>

did you happened to run "show channels" ?

clearly it's a dialplan problem other than a fifo one.


Dialplan: skypiax/interface8 Regex (PASS) [eavesdrop]
destination_number(779) =~ /^779$/ break=on-false
Dialplan: skypiax/interface8 Action answer()
Dialplan: skypiax/interface8 Action
set(eavesdrop_indicate_failed=tone_stream://%(500, 0, 320))
Dialplan: skypiax/interface8 Action
set(eavesdrop_indicate_new=tone_stream://%(500, 0, 620))
Dialplan: skypiax/interface8 Action
set(eavesdrop_indicate_idle=tone_stream://%(250, 0, 920))
Dialplan: skypiax/interface8 Action eavesdrop(all)



2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> Logs submitted: http://pastebin.freeswitch.org/11836
>
> I was trying to check whether the call had been added into the queue via
> telnet, but failed to find the fifo events. Here's my simplified dialplan:
>
> <extension name="in_fifo">
> ?? <condition field="destination_number" expression="^779$">
> ???? <action application="answer"/>
> ???? <action application="fifo" data="myq in undef undef"/>
> ? </condition>
> </extension>
>
> Please advise where went wrong.
>
> Thanks and best regards,
> -Jingwei
>
>
> On Mon, Jan 18, 2010 at 11:42 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> better to pastebin your log.
>>
>> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
>> > Thanks for replying. This is my dialplan
>> >
>> > ??? <extension name="in_fifo">
>> > ????? <condition field="destination_number" expression="^779$">
>> > ??????? <action application="set" data="fifo_music=$${hold_music}"/>
>> > ??????? <action application="answer"/>
>> > ??????? <action application="fifo" data="myq in undef undef"/>
>> > ????? </condition>
>> > ??? </extension>
>> >
>> > And I created a queue in fifo.conf.xml like this
>> >
>> > ??? <fifo name="myq" importance="0">
>> > ????? <!--<member timeout="60" simo="1"
>> > lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
>> > ??? </fifo>
>> >
>> > However, I'm still not able to see the incoming call get queued.
>> >
>> > freeswitch at localhost.localdomain> fifo list myq
>> > API CALL [fifo(list myq)] output:
>> > <fifo_report>
>> > ? <fifo name="myq" consumer_count="0" caller_count="0" waiting_count="0"
>> > importance="0">
>> > ??? <callers></callers>
>> > ??? <consumers></consumers>
>> > ? </fifo>
>> > </fifo_report>
>> >
>> > I tried both mod_skypiax and mod_dingaling, but with the same result.
>> >
>> > Regards,
>> > -Jingwei
>> >
>> > On Sun, Jan 17, 2010 at 11:37 AM, Seven Du <dujinfang at gmail.com> wrote:
>> >>
>> >> 2010/1/15, Jingwei Yang <jingwei.yang at gmail.com>:
>> >> > Hi Guys,
>> >> >
>> >> > I'm implementing an ACD system using ESL and mod_fifo. Based on what
>> >> > Anthony
>> >> > suggested in this post:
>> >> >
>> >> >
>> >> > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html
>> >> >
>> >> > *You can make an event socket application that listens for FIFO
>> >> > events
>> >> > and
>> >> > keeps track of what FIFOs are currently busy and when there are
>> >> > people
>> >> > waiting you can have that script generate a call to a group of SIP
>> >> > phones so
>> >> > when the first one answers, it sends them in as an agent where they
>> >> > can
>> >> > field the calls.
>> >> > *
>> >> >
>> >> > 1. How should I handle the concurrent issues? If I bridge a user to
>> >> > two
>> >> > agents and both of them answers, how does FS take care of this
>> >> > situation?
>> >> > Will a slower agent get a busy tone automatically?
>> >> >
>> >>
>> >> I think it just follow the standard originate dialstring rules.
>> >>
>> >> > 2. If the socket application is brought up after some users have
>> >> > called
>> >> > in,
>> >> > what command should I use to check the busy queues? fifo list?
>> >> >
>> >> Yes. Perhaps you can also check the fifo db, either sqlite or ODBC.
>> >>
>> >> > 3. Am I using fifo list and fifo count correctly?
>> >> >
>> >> > here's the testing dialplan:
>> >> >
>> >> > ? ? <extension name="in_fifo">
>> >> > ? ? ? <condition field="destination_number" expression="^779$">
>> >> > ? ? ? ? <action application="set" data="fifo_music=$${hold_music}"/>
>> >> > ? ? ? ? <action application="answer"/>
>> >> > ? ? ? ? <action application="fifo" data="*myq *in"/>
>> >> > ? ? ? </condition>
>> >> > ? ? </extension>
>> >> >
>> >> > when a call comes in and gets queued, these are the results of some
>> >> > commands
>> >> > I tried.
>> >> >
>> >> > freeswitch at localhost.localdomain> fifo list
>> >> > API CALL [fifo(list)] output:
>> >> > <fifo_report>
>> >> > ? <fifo name="cool_fifo at 192.168.1.104" consumer_count="0"
>> >> > caller_count="0"
>> >> > waiting_count="0" importance="0">
>> >> > ? ? <callers></callers>
>> >> > ? ? <consumers></consumers>
>> >> > ? </fifo>
>> >> > </fifo_report>
>> >> >
>> >> >
>> >> > freeswitch at localhost.localdomain> fifo list myq
>> >> > API CALL [fifo(list myq)] output:
>> >> > <fifo_report></fifo_report>
>> >> >
>> >> >
>> >> > freeswitch at localhost.localdomain> fifo count myq
>> >> > API CALL [fifo(count myq)] output:
>> >> > none
>> >> >
>> >> > It seems *myq* doesn't get created at all? Please enlighten.
>> >> >
>> >> > Thanks and best regards,
>> >> > -Jingwei
>> >> >
>> >> AFAIK, thant means the channel didn't queued in. Did you see any error
>> >> logs? I think you need to remove the stars in ? <action
>> >> application="fifo" data="*myq *in"/>.
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From kond at nstel.ru  Sun Jan 17 23:26:56 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Mon, 18 Jan 2010 10:26:56 +0300
Subject: [Freeswitch-users] eavesdrop problem?
In-Reply-To: <191c3a031001151445n51ba1514rb387179bb837c558@mail.gmail.com>
Message-ID: <20100118072655.F29E011F68@mail.nstel.ru>

Anthony,

Inserting 

<action application="hash"
data="insert/${domain_name}-spymap/${destination_number}/${uuid}"/>

into <extension name="global" continue="true"> in the dialplan

looks to work ok. I can now eavesdrop a colee.

By the way, should I do something to remove a uuid from the database when
the call is ended? Or will it be removed automatically?

Thanks and regards,

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Saturday, January 16, 2010 1:46 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] eavesdrop problem?

 

yes, exactly.
That is a demo of how you could possibly store a uuid by inserting them into
the db keyed from your user extension in the caller id.

if you do show channel and you see the uuid for each leg that is the
argument eavesdrop takes.
you can also do "all" in place of a uuid so you can cycle all the calls with
DTMF




On Fri, Jan 15, 2010 at 11:41 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Anthony,

Thanks for the reply.

Can you please point me to the document where I could read about it?

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop does not say
anything about it.

But let me guess: I should add 

<action application="hash"
data="insert/${domain_name}-spymap/${destination_number}/${uuid}"/>

Into <extension name="global" continue="true"> in the dialplan.

Am I close?

Thanks and regards,

Nikolay.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Friday, January 15, 2010 7:05 PM


To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] eavesdrop problem?

 

don't bother,



only inbound legs are added to the db that is used to lookup for eavesdrop
because the action is in the dialplan.
The extensions to eavesdrop you are using are just a demo to show you how to
work it.
you need to know the uuid of the channel you are trying to eavesdrop on
before you can do what you want.




On Fri, Jan 15, 2010 at 8:33 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

Hi all,

 

I want to use eavesdrop application.

Playing with it I found that when one tries to eavesdrop caller the feature
works ok.

But when trying to eavesdrop callee eavesdrop attempt failes.

I just updated to the latest version from http://latest.freeswitch.org
<http://latest.freeswitch.org/>  

[freeswitch at freeswitch log]$ fs_cli -x version

FreeSWITCH Version 1.0.5-20100115-0400 (16318M)

 

My setup is as following:

I have 3 internal extensions: 2853, 2840, 2879 (all are xlite).

1. 2879 calls 2840. Then 2853 dials 882879 - eavesdrop worked as expected.

2. 2840 calls 2879. Then 2853 dials 882879 - eavesdrop failed. 

 

I attached logs for both cases.

 

I don't believe it's intended behavior. 

 

Can anybody please advise if it is a configuration or a software problem?

 

Thanks and regards,

Nikolay.


_______________________________________________
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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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<mailto:MSN%3Aanthony_minessale at hotmail.com> 
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IRC: irc.freenode.net #freeswitch

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<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


_______________________________________________
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http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From jingwei.yang at gmail.com  Mon Jan 18 01:01:31 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 18 Jan 2010 17:01:31 +0800
Subject: [Freeswitch-users] Questions about mod_fifo
In-Reply-To: <23f91031001172325n2dd0f462ga202abec3a79d019@mail.gmail.com>
References: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
	<23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>
	<13529f9d1001171755v229c2137u4fa5cfe2f30d518@mail.gmail.com>
	<23f91031001171942h42e655c9vbc5ecb4165b7aca9@mail.gmail.com>
	<13529f9d1001172248x1a1ddc8arb7f4aba2e71ffb0f@mail.gmail.com>
	<23f91031001172325n2dd0f462ga202abec3a79d019@mail.gmail.com>
Message-ID: <13529f9d1001180101y6d345297y4d152628d6d6113e@mail.gmail.com>

Yes, I'm able to see the inbound channel created:

freeswitch at localhost.localdomain> show channels
API CALL [show(channels)] output:
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure,hostname,presence_id,presence_data
6d347eaa-aea8-47e3-b915-8cfdf0f8fd9c,inbound,2010-01-19
17:04:22,1263891862,skypiax/interface5,CS_EXECUTE,Jingwei,jingwei.yang,,779,eavesdrop,all,XML,default,L16,16000,L16,16000,,localhost.localdomain,,

Hmmm, may I know how you could tell it's a dialplan problem?

Regards,
-Jingwei

On Mon, Jan 18, 2010 at 3:25 PM, Seven Du <dujinfang at gmail.com> wrote:

> did you happened to run "show channels" ?
>
> clearly it's a dialplan problem other than a fifo one.
>
>
> Dialplan: skypiax/interface8 Regex (PASS) [eavesdrop]
> destination_number(779) =~ /^779$/ break=on-false
> Dialplan: skypiax/interface8 Action answer()
> Dialplan: skypiax/interface8 Action
> set(eavesdrop_indicate_failed=tone_stream://%(500, 0, 320))
> Dialplan: skypiax/interface8 Action
> set(eavesdrop_indicate_new=tone_stream://%(500, 0, 620))
> Dialplan: skypiax/interface8 Action
> set(eavesdrop_indicate_idle=tone_stream://%(250, 0, 920))
> Dialplan: skypiax/interface8 Action eavesdrop(all)
>
>
>
> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> > Logs submitted: http://pastebin.freeswitch.org/11836
> >
> > I was trying to check whether the call had been added into the queue via
> > telnet, but failed to find the fifo events. Here's my simplified
> dialplan:
> >
> > <extension name="in_fifo">
> >    <condition field="destination_number" expression="^779$">
> >      <action application="answer"/>
> >      <action application="fifo" data="myq in undef undef"/>
> >   </condition>
> > </extension>
> >
> > Please advise where went wrong.
> >
> > Thanks and best regards,
> > -Jingwei
> >
> >
> > On Mon, Jan 18, 2010 at 11:42 AM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> better to pastebin your log.
> >>
> >> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> >> > Thanks for replying. This is my dialplan
> >> >
> >> >     <extension name="in_fifo">
> >> >       <condition field="destination_number" expression="^779$">
> >> >         <action application="set" data="fifo_music=$${hold_music}"/>
> >> >         <action application="answer"/>
> >> >         <action application="fifo" data="myq in undef undef"/>
> >> >       </condition>
> >> >     </extension>
> >> >
> >> > And I created a queue in fifo.conf.xml like this
> >> >
> >> >     <fifo name="myq" importance="0">
> >> >       <!--<member timeout="60" simo="1"
> >> > lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
> >> >     </fifo>
> >> >
> >> > However, I'm still not able to see the incoming call get queued.
> >> >
> >> > freeswitch at localhost.localdomain> fifo list myq
> >> > API CALL [fifo(list myq)] output:
> >> > <fifo_report>
> >> >   <fifo name="myq" consumer_count="0" caller_count="0"
> waiting_count="0"
> >> > importance="0">
> >> >     <callers></callers>
> >> >     <consumers></consumers>
> >> >   </fifo>
> >> > </fifo_report>
> >> >
> >> > I tried both mod_skypiax and mod_dingaling, but with the same result.
> >> >
> >> > Regards,
> >> > -Jingwei
> >> >
> >> > On Sun, Jan 17, 2010 at 11:37 AM, Seven Du <dujinfang at gmail.com>
> wrote:
> >> >>
> >> >> 2010/1/15, Jingwei Yang <jingwei.yang at gmail.com>:
> >> >> > Hi Guys,
> >> >> >
> >> >> > I'm implementing an ACD system using ESL and mod_fifo. Based on
> what
> >> >> > Anthony
> >> >> > suggested in this post:
> >> >> >
> >> >> >
> >> >> >
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html
> >> >> >
> >> >> > *You can make an event socket application that listens for FIFO
> >> >> > events
> >> >> > and
> >> >> > keeps track of what FIFOs are currently busy and when there are
> >> >> > people
> >> >> > waiting you can have that script generate a call to a group of SIP
> >> >> > phones so
> >> >> > when the first one answers, it sends them in as an agent where they
> >> >> > can
> >> >> > field the calls.
> >> >> > *
> >> >> >
> >> >> > 1. How should I handle the concurrent issues? If I bridge a user to
> >> >> > two
> >> >> > agents and both of them answers, how does FS take care of this
> >> >> > situation?
> >> >> > Will a slower agent get a busy tone automatically?
> >> >> >
> >> >>
> >> >> I think it just follow the standard originate dialstring rules.
> >> >>
> >> >> > 2. If the socket application is brought up after some users have
> >> >> > called
> >> >> > in,
> >> >> > what command should I use to check the busy queues? fifo list?
> >> >> >
> >> >> Yes. Perhaps you can also check the fifo db, either sqlite or ODBC.
> >> >>
> >> >> > 3. Am I using fifo list and fifo count correctly?
> >> >> >
> >> >> > here's the testing dialplan:
> >> >> >
> >> >> >     <extension name="in_fifo">
> >> >> >       <condition field="destination_number" expression="^779$">
> >> >> >         <action application="set"
> data="fifo_music=$${hold_music}"/>
> >> >> >         <action application="answer"/>
> >> >> >         <action application="fifo" data="*myq *in"/>
> >> >> >       </condition>
> >> >> >     </extension>
> >> >> >
> >> >> > when a call comes in and gets queued, these are the results of some
> >> >> > commands
> >> >> > I tried.
> >> >> >
> >> >> > freeswitch at localhost.localdomain> fifo list
> >> >> > API CALL [fifo(list)] output:
> >> >> > <fifo_report>
> >> >> >   <fifo name="cool_fifo at 192.168.1.104" consumer_count="0"
> >> >> > caller_count="0"
> >> >> > waiting_count="0" importance="0">
> >> >> >     <callers></callers>
> >> >> >     <consumers></consumers>
> >> >> >   </fifo>
> >> >> > </fifo_report>
> >> >> >
> >> >> >
> >> >> > freeswitch at localhost.localdomain> fifo list myq
> >> >> > API CALL [fifo(list myq)] output:
> >> >> > <fifo_report></fifo_report>
> >> >> >
> >> >> >
> >> >> > freeswitch at localhost.localdomain> fifo count myq
> >> >> > API CALL [fifo(count myq)] output:
> >> >> > none
> >> >> >
> >> >> > It seems *myq* doesn't get created at all? Please enlighten.
> >> >> >
> >> >> > Thanks and best regards,
> >> >> > -Jingwei
> >> >> >
> >> >> AFAIK, thant means the channel didn't queued in. Did you see any
> error
> >> >> logs? I think you need to remove the stars in   <action
> >> >> application="fifo" data="*myq *in"/>.
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From mike at jerris.com  Mon Jan 18 01:15:52 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 Jan 2010 04:15:52 -0500
Subject: [Freeswitch-users] IMPORTANT -- Call for bug updates
Message-ID: <7E869910-75D2-40C3-8FA3-849FE1854F4D@jerris.com>

To all FreeSWITCH users-

	If you have bugs open on http://jira.freeswitch.org please login today and post a status update on these bugs (even if they appear to be awaiting comment by the development team).  If you have a patch, please update these patches to svn trunk so that they may be reviewed. I know many patches have been sitting for quite some time but I will make a strong effort to review and merge patches that are ready to go in.  If you have a bug, please update to svn trunk and comment if the issue still exists or is now fixed in trunk.  If you bug has a comment requesting more information, please provide it.  If you don't have any open bugs, and are not currently using recent svn trunk, I would appreciate it if you could carve a little bit of time out of your days and test out trunk.  Feel free to look through jira and find a bug you are interested in and test it as well.  We are working towards having the most stable and feature rich release of FreeSWITCH yet and we need your support and assistance to do so.  As always, if you have a new bug to file, please do so as soon as possible and try to get as much information as possible on the bug.  There are some good guidelines for reporting at http://wiki.freeswitch.org/wiki/Reporting_Bugs  As always, many thanks to the community for all your hard work and support.

Mike


From dujinfang at gmail.com  Mon Jan 18 02:19:07 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 18 Jan 2010 18:19:07 +0800
Subject: [Freeswitch-users] Questions about mod_fifo
In-Reply-To: <13529f9d1001180101y6d345297y4d152628d6d6113e@mail.gmail.com>
References: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
	<23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>
	<13529f9d1001171755v229c2137u4fa5cfe2f30d518@mail.gmail.com>
	<23f91031001171942h42e655c9vbc5ecb4165b7aca9@mail.gmail.com>
	<13529f9d1001172248x1a1ddc8arb7f4aba2e71ffb0f@mail.gmail.com>
	<23f91031001172325n2dd0f462ga202abec3a79d019@mail.gmail.com>
	<13529f9d1001180101y6d345297y4d152628d6d6113e@mail.gmail.com>
Message-ID: <23f91031001180219x3a46111bo8f081c59a78a3e0@mail.gmail.com>

from the log you can see that it routed to eavesdrop 779 but not
fifo_in 779 as you expected. In other words, either there are two 779
entries in your dialplan or your fifo_in 779 dialplan not been loaded
into freeswitch. Did you run reloadxml after you edited the dialplan
file?

2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> Yes, I'm able to see the inbound channel created:
>
> freeswitch at localhost.localdomain> show channels
> API CALL [show(channels)] output:
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure,hostname,presence_id,presence_data
> 6d347eaa-aea8-47e3-b915-8cfdf0f8fd9c,inbound,2010-01-19
> 17:04:22,1263891862,skypiax/interface5,CS_EXECUTE,Jingwei,jingwei.yang,,779,eavesdrop,all,XML,default,L16,16000,L16,16000,,localhost.localdomain,,
>
> Hmmm, may I know how you could tell it's a dialplan problem?
>
> Regards,
> -Jingwei
>
> On Mon, Jan 18, 2010 at 3:25 PM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> did you happened to run "show channels" ?
>>
>> clearly it's a dialplan problem other than a fifo one.
>>
>>
>> Dialplan: skypiax/interface8 Regex (PASS) [eavesdrop]
>> destination_number(779) =~ /^779$/ break=on-false
>> Dialplan: skypiax/interface8 Action answer()
>> Dialplan: skypiax/interface8 Action
>> set(eavesdrop_indicate_failed=tone_stream://%(500, 0, 320))
>> Dialplan: skypiax/interface8 Action
>> set(eavesdrop_indicate_new=tone_stream://%(500, 0, 620))
>> Dialplan: skypiax/interface8 Action
>> set(eavesdrop_indicate_idle=tone_stream://%(250, 0, 920))
>> Dialplan: skypiax/interface8 Action eavesdrop(all)
>>
>>
>>
>> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
>> > Logs submitted: http://pastebin.freeswitch.org/11836
>> >
>> > I was trying to check whether the call had been added into the queue via
>> > telnet, but failed to find the fifo events. Here's my simplified
>> > dialplan:
>> >
>> > <extension name="in_fifo">
>> > ?? <condition field="destination_number" expression="^779$">
>> > ???? <action application="answer"/>
>> > ???? <action application="fifo" data="myq in undef undef"/>
>> > ? </condition>
>> > </extension>
>> >
>> > Please advise where went wrong.
>> >
>> > Thanks and best regards,
>> > -Jingwei
>> >
>> >
>> > On Mon, Jan 18, 2010 at 11:42 AM, Seven Du <dujinfang at gmail.com> wrote:
>> >>
>> >> better to pastebin your log.
>> >>
>> >> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
>> >> > Thanks for replying. This is my dialplan
>> >> >
>> >> > ??? <extension name="in_fifo">
>> >> > ????? <condition field="destination_number" expression="^779$">
>> >> > ??????? <action application="set" data="fifo_music=$${hold_music}"/>
>> >> > ??????? <action application="answer"/>
>> >> > ??????? <action application="fifo" data="myq in undef undef"/>
>> >> > ????? </condition>
>> >> > ??? </extension>
>> >> >
>> >> > And I created a queue in fifo.conf.xml like this
>> >> >
>> >> > ??? <fifo name="myq" importance="0">
>> >> > ????? <!--<member timeout="60" simo="1"
>> >> > lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
>> >> > ??? </fifo>
>> >> >
>> >> > However, I'm still not able to see the incoming call get queued.
>> >> >
>> >> > freeswitch at localhost.localdomain> fifo list myq
>> >> > API CALL [fifo(list myq)] output:
>> >> > <fifo_report>
>> >> > ? <fifo name="myq" consumer_count="0" caller_count="0"
>> >> > waiting_count="0"
>> >> > importance="0">
>> >> > ??? <callers></callers>
>> >> > ??? <consumers></consumers>
>> >> > ? </fifo>
>> >> > </fifo_report>
>> >> >
>> >> > I tried both mod_skypiax and mod_dingaling, but with the same result.
>> >> >
>> >> > Regards,
>> >> > -Jingwei
>> >> >
>> >> > On Sun, Jan 17, 2010 at 11:37 AM, Seven Du <dujinfang at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> 2010/1/15, Jingwei Yang <jingwei.yang at gmail.com>:
>> >> >> > Hi Guys,
>> >> >> >
>> >> >> > I'm implementing an ACD system using ESL and mod_fifo. Based on
>> >> >> > what
>> >> >> > Anthony
>> >> >> > suggested in this post:
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html
>> >> >> >
>> >> >> > *You can make an event socket application that listens for FIFO
>> >> >> > events
>> >> >> > and
>> >> >> > keeps track of what FIFOs are currently busy and when there are
>> >> >> > people
>> >> >> > waiting you can have that script generate a call to a group of SIP
>> >> >> > phones so
>> >> >> > when the first one answers, it sends them in as an agent where
>> >> >> > they
>> >> >> > can
>> >> >> > field the calls.
>> >> >> > *
>> >> >> >
>> >> >> > 1. How should I handle the concurrent issues? If I bridge a user
>> >> >> > to
>> >> >> > two
>> >> >> > agents and both of them answers, how does FS take care of this
>> >> >> > situation?
>> >> >> > Will a slower agent get a busy tone automatically?
>> >> >> >
>> >> >>
>> >> >> I think it just follow the standard originate dialstring rules.
>> >> >>
>> >> >> > 2. If the socket application is brought up after some users have
>> >> >> > called
>> >> >> > in,
>> >> >> > what command should I use to check the busy queues? fifo list?
>> >> >> >
>> >> >> Yes. Perhaps you can also check the fifo db, either sqlite or ODBC.
>> >> >>
>> >> >> > 3. Am I using fifo list and fifo count correctly?
>> >> >> >
>> >> >> > here's the testing dialplan:
>> >> >> >
>> >> >> > ? ? <extension name="in_fifo">
>> >> >> > ? ? ? <condition field="destination_number" expression="^779$">
>> >> >> > ? ? ? ? <action application="set"
>> >> >> > data="fifo_music=$${hold_music}"/>
>> >> >> > ? ? ? ? <action application="answer"/>
>> >> >> > ? ? ? ? <action application="fifo" data="*myq *in"/>
>> >> >> > ? ? ? </condition>
>> >> >> > ? ? </extension>
>> >> >> >
>> >> >> > when a call comes in and gets queued, these are the results of
>> >> >> > some
>> >> >> > commands
>> >> >> > I tried.
>> >> >> >
>> >> >> > freeswitch at localhost.localdomain> fifo list
>> >> >> > API CALL [fifo(list)] output:
>> >> >> > <fifo_report>
>> >> >> > ? <fifo name="cool_fifo at 192.168.1.104" consumer_count="0"
>> >> >> > caller_count="0"
>> >> >> > waiting_count="0" importance="0">
>> >> >> > ? ? <callers></callers>
>> >> >> > ? ? <consumers></consumers>
>> >> >> > ? </fifo>
>> >> >> > </fifo_report>
>> >> >> >
>> >> >> >
>> >> >> > freeswitch at localhost.localdomain> fifo list myq
>> >> >> > API CALL [fifo(list myq)] output:
>> >> >> > <fifo_report></fifo_report>
>> >> >> >
>> >> >> >
>> >> >> > freeswitch at localhost.localdomain> fifo count myq
>> >> >> > API CALL [fifo(count myq)] output:
>> >> >> > none
>> >> >> >
>> >> >> > It seems *myq* doesn't get created at all? Please enlighten.
>> >> >> >
>> >> >> > Thanks and best regards,
>> >> >> > -Jingwei
>> >> >> >
>> >> >> AFAIK, thant means the channel didn't queued in. Did you see any
>> >> >> error
>> >> >> logs? I think you need to remove the stars in ? <action
>> >> >> application="fifo" data="*myq *in"/>.
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From jingwei.yang at gmail.com  Mon Jan 18 02:39:54 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 18 Jan 2010 18:39:54 +0800
Subject: [Freeswitch-users] Questions about mod_fifo
In-Reply-To: <23f91031001180219x3a46111bo8f081c59a78a3e0@mail.gmail.com>
References: <13529f9d1001150251q51304522x7d449394f53782df@mail.gmail.com>
	<23f91031001161937v20b16873w1be9e55d29973c94@mail.gmail.com>
	<13529f9d1001171755v229c2137u4fa5cfe2f30d518@mail.gmail.com>
	<23f91031001171942h42e655c9vbc5ecb4165b7aca9@mail.gmail.com>
	<13529f9d1001172248x1a1ddc8arb7f4aba2e71ffb0f@mail.gmail.com>
	<23f91031001172325n2dd0f462ga202abec3a79d019@mail.gmail.com>
	<13529f9d1001180101y6d345297y4d152628d6d6113e@mail.gmail.com>
	<23f91031001180219x3a46111bo8f081c59a78a3e0@mail.gmail.com>
Message-ID: <13529f9d1001180239j70a00cefo334539af9304725c@mail.gmail.com>

Yes, you're right!! There's a eavesdrop 779 matched first. Thank you so
much!

On Mon, Jan 18, 2010 at 6:19 PM, Seven Du <dujinfang at gmail.com> wrote:

> from the log you can see that it routed to eavesdrop 779 but not
> fifo_in 779 as you expected. In other words, either there are two 779
> entries in your dialplan or your fifo_in 779 dialplan not been loaded
> into freeswitch. Did you run reloadxml after you edited the dialplan
> file?
>
> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> > Yes, I'm able to see the inbound channel created:
> >
> > freeswitch at localhost.localdomain> show channels
> > API CALL [show(channels)] output:
> >
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure,hostname,presence_id,presence_data
> > 6d347eaa-aea8-47e3-b915-8cfdf0f8fd9c,inbound,2010-01-19
> >
> 17:04:22,1263891862,skypiax/interface5,CS_EXECUTE,Jingwei,jingwei.yang,,779,eavesdrop,all,XML,default,L16,16000,L16,16000,,localhost.localdomain,,
> >
> > Hmmm, may I know how you could tell it's a dialplan problem?
> >
> > Regards,
> > -Jingwei
> >
> > On Mon, Jan 18, 2010 at 3:25 PM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> did you happened to run "show channels" ?
> >>
> >> clearly it's a dialplan problem other than a fifo one.
> >>
> >>
> >> Dialplan: skypiax/interface8 Regex (PASS) [eavesdrop]
> >> destination_number(779) =~ /^779$/ break=on-false
> >> Dialplan: skypiax/interface8 Action answer()
> >> Dialplan: skypiax/interface8 Action
> >> set(eavesdrop_indicate_failed=tone_stream://%(500, 0, 320))
> >> Dialplan: skypiax/interface8 Action
> >> set(eavesdrop_indicate_new=tone_stream://%(500, 0, 620))
> >> Dialplan: skypiax/interface8 Action
> >> set(eavesdrop_indicate_idle=tone_stream://%(250, 0, 920))
> >> Dialplan: skypiax/interface8 Action eavesdrop(all)
> >>
> >>
> >>
> >> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> >> > Logs submitted: http://pastebin.freeswitch.org/11836
> >> >
> >> > I was trying to check whether the call had been added into the queue
> via
> >> > telnet, but failed to find the fifo events. Here's my simplified
> >> > dialplan:
> >> >
> >> > <extension name="in_fifo">
> >> >    <condition field="destination_number" expression="^779$">
> >> >      <action application="answer"/>
> >> >      <action application="fifo" data="myq in undef undef"/>
> >> >   </condition>
> >> > </extension>
> >> >
> >> > Please advise where went wrong.
> >> >
> >> > Thanks and best regards,
> >> > -Jingwei
> >> >
> >> >
> >> > On Mon, Jan 18, 2010 at 11:42 AM, Seven Du <dujinfang at gmail.com>
> wrote:
> >> >>
> >> >> better to pastebin your log.
> >> >>
> >> >> 2010/1/18 Jingwei Yang <jingwei.yang at gmail.com>:
> >> >> > Thanks for replying. This is my dialplan
> >> >> >
> >> >> >     <extension name="in_fifo">
> >> >> >       <condition field="destination_number" expression="^779$">
> >> >> >         <action application="set"
> data="fifo_music=$${hold_music}"/>
> >> >> >         <action application="answer"/>
> >> >> >         <action application="fifo" data="myq in undef undef"/>
> >> >> >       </condition>
> >> >> >     </extension>
> >> >> >
> >> >> > And I created a queue in fifo.conf.xml like this
> >> >> >
> >> >> >     <fifo name="myq" importance="0">
> >> >> >       <!--<member timeout="60" simo="1"
> >> >> > lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
> >> >> >     </fifo>
> >> >> >
> >> >> > However, I'm still not able to see the incoming call get queued.
> >> >> >
> >> >> > freeswitch at localhost.localdomain> fifo list myq
> >> >> > API CALL [fifo(list myq)] output:
> >> >> > <fifo_report>
> >> >> >   <fifo name="myq" consumer_count="0" caller_count="0"
> >> >> > waiting_count="0"
> >> >> > importance="0">
> >> >> >     <callers></callers>
> >> >> >     <consumers></consumers>
> >> >> >   </fifo>
> >> >> > </fifo_report>
> >> >> >
> >> >> > I tried both mod_skypiax and mod_dingaling, but with the same
> result.
> >> >> >
> >> >> > Regards,
> >> >> > -Jingwei
> >> >> >
> >> >> > On Sun, Jan 17, 2010 at 11:37 AM, Seven Du <dujinfang at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >> 2010/1/15, Jingwei Yang <jingwei.yang at gmail.com>:
> >> >> >> > Hi Guys,
> >> >> >> >
> >> >> >> > I'm implementing an ACD system using ESL and mod_fifo. Based on
> >> >> >> > what
> >> >> >> > Anthony
> >> >> >> > suggested in this post:
> >> >> >> >
> >> >> >> >
> >> >> >> >
> >> >> >> >
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg01652.html
> >> >> >> >
> >> >> >> > *You can make an event socket application that listens for FIFO
> >> >> >> > events
> >> >> >> > and
> >> >> >> > keeps track of what FIFOs are currently busy and when there are
> >> >> >> > people
> >> >> >> > waiting you can have that script generate a call to a group of
> SIP
> >> >> >> > phones so
> >> >> >> > when the first one answers, it sends them in as an agent where
> >> >> >> > they
> >> >> >> > can
> >> >> >> > field the calls.
> >> >> >> > *
> >> >> >> >
> >> >> >> > 1. How should I handle the concurrent issues? If I bridge a user
> >> >> >> > to
> >> >> >> > two
> >> >> >> > agents and both of them answers, how does FS take care of this
> >> >> >> > situation?
> >> >> >> > Will a slower agent get a busy tone automatically?
> >> >> >> >
> >> >> >>
> >> >> >> I think it just follow the standard originate dialstring rules.
> >> >> >>
> >> >> >> > 2. If the socket application is brought up after some users have
> >> >> >> > called
> >> >> >> > in,
> >> >> >> > what command should I use to check the busy queues? fifo list?
> >> >> >> >
> >> >> >> Yes. Perhaps you can also check the fifo db, either sqlite or
> ODBC.
> >> >> >>
> >> >> >> > 3. Am I using fifo list and fifo count correctly?
> >> >> >> >
> >> >> >> > here's the testing dialplan:
> >> >> >> >
> >> >> >> >     <extension name="in_fifo">
> >> >> >> >       <condition field="destination_number" expression="^779$">
> >> >> >> >         <action application="set"
> >> >> >> > data="fifo_music=$${hold_music}"/>
> >> >> >> >         <action application="answer"/>
> >> >> >> >         <action application="fifo" data="*myq *in"/>
> >> >> >> >       </condition>
> >> >> >> >     </extension>
> >> >> >> >
> >> >> >> > when a call comes in and gets queued, these are the results of
> >> >> >> > some
> >> >> >> > commands
> >> >> >> > I tried.
> >> >> >> >
> >> >> >> > freeswitch at localhost.localdomain> fifo list
> >> >> >> > API CALL [fifo(list)] output:
> >> >> >> > <fifo_report>
> >> >> >> >   <fifo name="cool_fifo at 192.168.1.104" consumer_count="0"
> >> >> >> > caller_count="0"
> >> >> >> > waiting_count="0" importance="0">
> >> >> >> >     <callers></callers>
> >> >> >> >     <consumers></consumers>
> >> >> >> >   </fifo>
> >> >> >> > </fifo_report>
> >> >> >> >
> >> >> >> >
> >> >> >> > freeswitch at localhost.localdomain> fifo list myq
> >> >> >> > API CALL [fifo(list myq)] output:
> >> >> >> > <fifo_report></fifo_report>
> >> >> >> >
> >> >> >> >
> >> >> >> > freeswitch at localhost.localdomain> fifo count myq
> >> >> >> > API CALL [fifo(count myq)] output:
> >> >> >> > none
> >> >> >> >
> >> >> >> > It seems *myq* doesn't get created at all? Please enlighten.
> >> >> >> >
> >> >> >> > Thanks and best regards,
> >> >> >> > -Jingwei
> >> >> >> >
> >> >> >> AFAIK, thant means the channel didn't queued in. Did you see any
> >> >> >> error
> >> >> >> logs? I think you need to remove the stars in   <action
> >> >> >> application="fifo" data="*myq *in"/>.
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From n.geordzhev at gmail.com  Mon Jan 18 03:00:03 2010
From: n.geordzhev at gmail.com (Nikolai Geordzhev)
Date: Mon, 18 Jan 2010 13:00:03 +0200
Subject: [Freeswitch-users] database disk image is malformed ISSUE
Message-ID: <c8df2a131001180300m25419dd8y65ff80ce64824cef@mail.gmail.com>

Hi Guys,

I have an issue playing with FS as a registrar server. I have made some
tests with 3000 subscribers registering every 600 seconds running for days
and everything went fine.
Then I have tried with 6000 subscribers registering every 3600 seconds and
after some time ( between 1 and 2 days) I received  [ERR]
switch_core_sqldb.c:662 SQL ERR [database disk image is malformed] message
in the freeswitch.log file. The only solution I have found is deleting the
db folder ( mounted in tmpfs) and restarting the application.
When I measure the packets/sec rates of both setups i see 13 pack/s for the
first setup and 7 pack/s for the second one. Can somebody advise what can
cause this Error and if there is some kind of a solution.

Thanks and best regards
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From lakindia89 at gmail.com  Mon Jan 18 03:22:20 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 18 Jan 2010 16:52:20 +0530
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <7d79b3931001132357i36bbb482jdf2bbdd3aea2a583@mail.gmail.com>
References: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
	<191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>
	<7d79b3931001132357i36bbb482jdf2bbdd3aea2a583@mail.gmail.com>
Message-ID: <7d79b3931001180322p74a1227qe0c2199a77cbfe2@mail.gmail.com>

Here is the result

Program:
require ESL;
use IO::Socket::INET;
use Data::Dumper;

my $ip = "192.168.1.222";
my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
Proto => 'tcp',  Listen => 1,  Reuse => 1 );
die "Could not create socket: $!\n" unless $sock;


for(;;) {
    my $new_sock = $sock->accept();
    next if (not defined ($new_sock));
    my $pid = fork();
    if ($pid) {
        close($new_sock);
        next;
    }
    print "CHILD PID: $$\n";
    &register_Signals_Child();
    sub register_Signals_Child() {
        foreach ( keys %SIG ) {
            $SIG{$_} = 'Handler';
        }
    }

    sub Handler() {
        my $handle=$_[0];
        if($handle eq "INT") {
            print "CHILD $$: SIGNAL SIG$handle is generated\n";
        }
        else
        {
            print "CHILD $$: Received $handle\n";
        }
    }
    my $host = $new_sock->sockhost();
    my $fd = fileno($new_sock);

    my $con = new ESL::ESLconnection($fd);
    my $info = $con->getInfo();

    my $uuid = $info->getHeader("unique-id");

    printf "Connected call %s, from %s\n", $uuid,
$info->getHeader("caller-caller-id-number");
    my $r=$con->execute("answer");
    print Dumper $r;
    $con->events("plain","all");
    my
$re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
    while($con->connected()) {
        my $e = $con->recvEvent();

        if ($e) {
            my $name = $e->getHeader("event-name");
            print "EVENT [$name]\n";
            if ($name eq "DTMF") {
                my $digit = $e->getHeader("dtmf-digit");
                print "$digit\n";
            }
        }
    }
    close($new_sock);
}

I executed the program and the following things were printed

CHILD PID: 6778
Connected call e0d1001a-03f4-11df-b002-db488337e0ea, from 1001
$VAR1 = 0;
EVENT [CHANNEL_EXECUTE]
EVENT [CHANNEL_ANSWER]
EVENT [CHANNEL_EXECUTE_COMPLETE]
EVENT [COMMAND]
EVENT [CHANNEL_EXECUTE]
EVENT [HEARTBEAT]
EVENT [RE_SCHEDULE]
EVENT [CHANNEL_EXECUTE_COMPLETE]

Then from another shell I executed kill -2 6778, the result is follows
CHILD 6778: SIGNAL SIGINT is generated
EVENT [SERVER_DISCONNECTED]

But the child process is still running as expected.
But  I don't know why I received SERVER_DISCONNECTED from freeswitch???





On Thu, Jan 14, 2010 at 1:27 PM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> I taught the signal handler will be inherited by the child process. It also
> does like that.
> After making a call, If I press ctrl + c, the above program printed
> PARENT PID: Signal SIGINT is generated
> CHILD PID: Signal SIGINT is generated.
>
> So I think the sigal handlers will be inherited to the child.
> Anyway I'll also try registering signal handlers in child also, and then
> I'll come back with that result.
>
> Thanks....
> On Wed, Jan 13, 2010 at 9:48 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> you would have to register signals in your child process too
>>
>>   On Wed, Jan 13, 2010 at 3:13 AM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>>  Hi all,
>>>
>>> I've done a sample program (In perl ESL) , which play a file to the
>>> caller and then it will call recvEvent() and print the event name. I've
>>> handled signals also.
>>>
>>> When I send SIGINT to my program (Perl), the signal handler is called and
>>> I can see the print output. But in the same time,  I received
>>> SERVER_DISCONNECTED from freeswitch as event.
>>>
>>> I don't know why I received SERVER_DISCONNECTED from freeswitch. Is it
>>> because, the recvEvent() from perl internally  calls the recvevent function
>>> in the Esl.c and when it waits to receive the information from socket, the
>>> signal occurred???
>>>
>>> Please clarify me!!
>>>
>>> Here is my program
>>> require ESL;
>>> use IO::Socket::INET;
>>> use Data::Dumper;
>>>
>>> my $ip = "192.168.1.222";
>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>> die "Could not create socket: $!\n" unless $sock;
>>> &register_Signals();
>>>
>>> for(;;) {
>>>     my $new_sock = $sock->accept();
>>>     next if (not defined ($new_sock));
>>>     my $pid = fork();
>>>     if ($pid) {
>>>         close($new_sock);
>>>         next;
>>>     }
>>>     print "CHILD PID: $$\n";
>>>     my $host = $new_sock->sockhost();
>>>     my $fd = fileno($new_sock);
>>>
>>>     my $con = new ESL::ESLconnection($fd);
>>>     my $info = $con->getInfo();
>>>
>>>     my $uuid = $info->getHeader("unique-id");
>>>
>>>     printf "Connected call %s, from %s\n", $uuid,
>>> $info->getHeader("caller-caller-id-number");
>>>     my $r=$con->execute("answer");
>>>     print Dumper $r;
>>>     $con->events("plain","all");
>>>     my
>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>
>>>     while($con->connected()) {
>>>         my $e = $con->recvEvent();
>>>
>>>         if ($e) {
>>>             my $name = $e->getHeader("event-name");
>>>             print "EVENT [$name]\n";
>>>             if ($name eq "DTMF") {
>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>                 print "$digit\n";
>>>             }
>>>         }
>>>     }
>>>     close($new_sock);
>>> }
>>> sub register_Signals() {
>>>     foreach ( keys %SIG ) {
>>>         $SIG{$_} = 'sig_Handler';
>>>     }
>>> }
>>>
>>> sub sig_Handler() {
>>>     my $handle=$_[0];
>>>     if($handle eq "INT") {
>>>         print "$$: SIGNAL SIG$handle is generated\n";
>>>     }
>>> }
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From mike at jerris.com  Mon Jan 18 03:28:04 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 Jan 2010 06:28:04 -0500
Subject: [Freeswitch-users] database disk image is malformed ISSUE
In-Reply-To: <c8df2a131001180300m25419dd8y65ff80ce64824cef@mail.gmail.com>
References: <c8df2a131001180300m25419dd8y65ff80ce64824cef@mail.gmail.com>
Message-ID: <206AA031-94DA-4AAB-8761-90A06A80E9F5@jerris.com>

sqlite doesn't scale.  If you want to do anything serious, use odbc and a real database instead.

Mike

On Jan 18, 2010, at 6:00 AM, Nikolai Geordzhev wrote:

> Hi Guys,
> 
> I have an issue playing with FS as a registrar server. I have made some tests with 3000 subscribers registering every 600 seconds running for days and everything went fine. 
> Then I have tried with 6000 subscribers registering every 3600 seconds and after some time ( between 1 and 2 days) I received  [ERR] switch_core_sqldb.c:662 SQL ERR [database disk image is malformed] message in the freeswitch.log file. The only solution I have found is deleting the db folder ( mounted in tmpfs) and restarting the application.
> When I measure the packets/sec rates of both setups i see 13 pack/s for the first setup and 7 pack/s for the second one. Can somebody advise what can cause this Error and if there is some kind of a solution. 




From Prometheus001 at gmx.net  Mon Jan 18 03:56:36 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 18 Jan 2010 12:56:36 +0100
Subject: [Freeswitch-users] Multiple registrations: Only calling
 the	first element in the list in this mode
In-Reply-To: <4B50C757.3050901@gmx.net>
References: <4B4DADD5.3010507@gmx.net>
	<b39273531001130450k16af27d2v209a300f441d39d8@mail.gmail.com>
	<4B50C757.3050901@gmx.net>
Message-ID: <4B544C74.1000301@gmx.net>

I just found out that this method unfortunately had a side effect,

when I use
    ":_:" the caller does not receive a dialtone
If I change to
    "," the dialtone is there

Any clue how I can work around this?

Best regards
Peter

Peter P GMX schrieb:
> Thanks Rupa,
>
> this worked. I have documented this in the wiki:
>     http://wiki.freeswitch.org/wiki/Ring_group
>
> Best regards
> Peter
>
> Rupa Schomaker schrieb:
>   
>> Try:
>>
>> bridge user/100 at domain:_:user/101 at domain:_:user/102 at domain
>>
>> Then document it up if it works.
>>
>> On Wed, Jan 13, 2010 at 5:26 AM, Peter P GMX <Prometheus001 at gmx.net
>> <mailto:Prometheus001 at gmx.net>> wrote:
>>
>>     Hello,
>>
>>     I habe the following behaviour
>>
>>     when I call a user which is registered twice with 2 phones via
>>        bridge user/100 at domain
>>     both phones are ringing. This is correct as I allow multiple
>>     registrations in a profile
>>
>>     However when I call multiple endpoints via
>>        bridge user/100 at domain,user/101 at domain,user/102 at domain
>>     only one phone with number100 is ringing.
>>
>>     Console log shows "Only calling the first element in the list in this
>>     mode.":
>>     2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734
>>     variable
>>     string 0 = [presence_id=100 at domain]
>>     2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734
>>     variable
>>     string 1 = [transfer_fallback_extension=100]
>>     2010-01-12 19:52:18.236361 [WARNING] switch_ivr_originate.c:2048 Only
>>     calling the first element in the list in this mode.
>>     2010-01-12 19:52:18.236361 [NOTICE] switch_channel.c:613 New Channel
>>     sofia/internal/sip:100 at 10.11.12.203:2048
>>     <http://sip:100 at 10.11.12.203:2048>
>>     [9b95fcdc-ffab-11de-9ba2-13b2daa7ce61]
>>
>>     Is there any way to work around this? I need all phones to be
>>     ringing in
>>     this scenario.
>>
>>     Best regards
>>     Peter
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> -Rupa
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>   
>>     
>
>   



From rupa at rupa.com  Mon Jan 18 04:33:34 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 18 Jan 2010 06:33:34 -0600
Subject: [Freeswitch-users] Multiple registrations: Only calling the
	first element in the list in this mode
In-Reply-To: <4B544C74.1000301@gmx.net>
References: <4B4DADD5.3010507@gmx.net>
	<b39273531001130450k16af27d2v209a300f441d39d8@mail.gmail.com>
	<4B50C757.3050901@gmx.net> <4B544C74.1000301@gmx.net>
Message-ID: <b39273531001180433m79bced04ja60225615588577c@mail.gmail.com>

Can you open a ticket on jira for that?

By dialtone, do you mean ringback (call progress)?

On Mon, Jan 18, 2010 at 5:56 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> I just found out that this method unfortunately had a side effect,
>
> when I use
>    ":_:" the caller does not receive a dialtone
> If I change to
>    "," the dialtone is there
>
> Any clue how I can work around this?
>
> Best regards
> Peter
>
> Peter P GMX schrieb:
> > Thanks Rupa,
> >
> > this worked. I have documented this in the wiki:
> >     http://wiki.freeswitch.org/wiki/Ring_group
> >
> > Best regards
> > Peter
> >
> > Rupa Schomaker schrieb:
> >
> >> Try:
> >>
> >> bridge user/100 at domain:_:user/101 at domain:_:user/102 at domain
> >>
> >> Then document it up if it works.
> >>
> >> On Wed, Jan 13, 2010 at 5:26 AM, Peter P GMX <Prometheus001 at gmx.net
> >> <mailto:Prometheus001 at gmx.net>> wrote:
> >>
> >>     Hello,
> >>
> >>     I habe the following behaviour
> >>
> >>     when I call a user which is registered twice with 2 phones via
> >>        bridge user/100 at domain
> >>     both phones are ringing. This is correct as I allow multiple
> >>     registrations in a profile
> >>
> >>     However when I call multiple endpoints via
> >>        bridge user/100 at domain,user/101 at domain,user/102 at domain
> >>     only one phone with number100 is ringing.
> >>
> >>     Console log shows "Only calling the first element in the list in
> this
> >>     mode.":
> >>     2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734
> >>     variable
> >>     string 0 = [presence_id=100 at domain]
> >>     2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734
> >>     variable
> >>     string 1 = [transfer_fallback_extension=100]
> >>     2010-01-12 19:52:18.236361 [WARNING] switch_ivr_originate.c:2048
> Only
> >>     calling the first element in the list in this mode.
> >>     2010-01-12 19:52:18.236361 [NOTICE] switch_channel.c:613 New Channel
> >>     sofia/internal/sip:100 at 10.11.12.203:2048
> >>     <http://sip:100 at 10.11.12.203:2048>
> >>     [9b95fcdc-ffab-11de-9ba2-13b2daa7ce61]
> >>
> >>     Is there any way to work around this? I need all phones to be
> >>     ringing in
> >>     this scenario.
> >>
> >>     Best regards
> >>     Peter
> >>
> >>     _______________________________________________
> >>     FreeSWITCH-users mailing list
> >>     FreeSWITCH-users at lists.freeswitch.org
> >>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>     http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >> --
> >> -Rupa
> >> ------------------------------------------------------------------------
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From Prometheus001 at gmx.net  Mon Jan 18 05:48:31 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 18 Jan 2010 14:48:31 +0100
Subject: [Freeswitch-users] Multiple registrations: Only calling
 the	first element in the list in this mode
In-Reply-To: <b39273531001180433m79bced04ja60225615588577c@mail.gmail.com>
References: <4B4DADD5.3010507@gmx.net>	<b39273531001130450k16af27d2v209a300f441d39d8@mail.gmail.com>	<4B50C757.3050901@gmx.net>
	<4B544C74.1000301@gmx.net>
	<b39273531001180433m79bced04ja60225615588577c@mail.gmail.com>
Message-ID: <4B5466AF.2060102@gmx.net>

Hello Rupa,

I've opened a JIRA for this. And yes, I mean ringback.

Best regards
Peter

Rupa Schomaker schrieb:
> Can you open a ticket on jira for that?  
>
> By dialtone, do you mean ringback (call progress)?
>
> On Mon, Jan 18, 2010 at 5:56 AM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     I just found out that this method unfortunately had a side effect,
>
>     when I use
>        ":_:" the caller does not receive a dialtone
>     If I change to
>        "," the dialtone is there
>
>     Any clue how I can work around this?
>
>     Best regards
>     Peter
>
>     Peter P GMX schrieb:
>     > Thanks Rupa,
>     >
>     > this worked. I have documented this in the wiki:
>     >     http://wiki.freeswitch.org/wiki/Ring_group
>     >
>     > Best regards
>     > Peter
>     >
>     > Rupa Schomaker schrieb:
>     >
>     >> Try:
>     >>
>     >> bridge user/100 at domain:_:user/101 at domain:_:user/102 at domain
>     >>
>     >> Then document it up if it works.
>     >>
>     >> On Wed, Jan 13, 2010 at 5:26 AM, Peter P GMX
>     <Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>
>     >> <mailto:Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>>>
>     wrote:
>     >>
>     >>     Hello,
>     >>
>     >>     I habe the following behaviour
>     >>
>     >>     when I call a user which is registered twice with 2 phones via
>     >>        bridge user/100 at domain
>     >>     both phones are ringing. This is correct as I allow multiple
>     >>     registrations in a profile
>     >>
>     >>     However when I call multiple endpoints via
>     >>        bridge user/100 at domain,user/101 at domain,user/102 at domain
>     >>     only one phone with number100 is ringing.
>     >>
>     >>     Console log shows "Only calling the first element in the
>     list in this
>     >>     mode.":
>     >>     2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734
>     >>     variable
>     >>     string 0 = [presence_id=100 at domain]
>     >>     2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734
>     >>     variable
>     >>     string 1 = [transfer_fallback_extension=100]
>     >>     2010-01-12 19:52:18.236361 [WARNING]
>     switch_ivr_originate.c:2048 Only
>     >>     calling the first element in the list in this mode.
>     >>     2010-01-12 19:52:18.236361 [NOTICE] switch_channel.c:613
>     New Channel
>     >>     sofia/internal/sip:100 at 10.11.12.203:2048
>     <http://sip:100 at 10.11.12.203:2048>
>     >>     <http://sip:100 at 10.11.12.203:2048>
>     >>     [9b95fcdc-ffab-11de-9ba2-13b2daa7ce61]
>     >>
>     >>     Is there any way to work around this? I need all phones to be
>     >>     ringing in
>     >>     this scenario.
>     >>
>     >>     Best regards
>     >>     Peter
>     >>
>     >>     _______________________________________________
>     >>     FreeSWITCH-users mailing list
>     >>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >>     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>    
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >>     http://www.freeswitch.org
>     >>
>     >>
>     >>
>     >>
>     >> --
>     >> -Rupa
>     >>
>     ------------------------------------------------------------------------
>     >>
>     >> _______________________________________________
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >>
>     >>
>     >
>     >
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> -Rupa
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From brian at freeswitch.org  Mon Jan 18 06:22:18 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 Jan 2010 08:22:18 -0600
Subject: [Freeswitch-users] eavesdrop problem?
In-Reply-To: <20100118072655.F29E011F68@mail.nstel.ru>
References: <20100118072655.F29E011F68@mail.nstel.ru>
Message-ID: <9C65429E-482D-4438-89CE-2CE2E5D73355@freeswitch.org>

It'll overwrite on the next call... or reboot.

You can use the api_hangup_hook to remove it if you wish... see variables page on wiki.

/b

On Jan 18, 2010, at 1:26 AM, Nikolay Kondratyev wrote:

> By the way, should I do something to remove a uuid from the database when the call is ended? Or will it be removed automatically?
> Thanks and regards,
> Nikolay.

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From brian at freeswitch.org  Mon Jan 18 06:27:17 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 Jan 2010 08:27:17 -0600
Subject: [Freeswitch-users] Multiple registrations: Only calling
	the	first element in the list in this mode
In-Reply-To: <4B5466AF.2060102@gmx.net>
References: <4B4DADD5.3010507@gmx.net>	<b39273531001130450k16af27d2v209a300f441d39d8@mail.gmail.com>	<4B50C757.3050901@gmx.net>
	<4B544C74.1000301@gmx.net>
	<b39273531001180433m79bced04ja60225615588577c@mail.gmail.com>
	<4B5466AF.2060102@gmx.net>
Message-ID: <B7AD5B44-7FC6-4703-ADD3-4F13851C9CBF@freeswitch.org>

In this mode you'll have to set the ringback variable.. did you do that?

/b

On Jan 18, 2010, at 7:48 AM, Peter P GMX wrote:

> Hello Rupa,
> 
> I've opened a JIRA for this. And yes, I mean ringback.
> 
> Best regards
> Peter




From lart2150 at gmail.com  Mon Jan 18 08:00:52 2010
From: lart2150 at gmail.com (Brian Engert)
Date: Mon, 18 Jan 2010 10:00:52 -0600
Subject: [Freeswitch-users] Mod_Fax
In-Reply-To: <59daa2cd1001161205h3f6feeddla0be8e1fa9311666@mail.gmail.com>
References: <59daa2cd1001140654r6712bb00wd2399c0e1d34a308@mail.gmail.com>
	<776BA929-D7EC-4366-AF49-348F0E11488B@shaw.ca>
	<59daa2cd1001161205h3f6feeddla0be8e1fa9311666@mail.gmail.com>
Message-ID: <a88e84c21001180800p24e7a182m61ff689827278882@mail.gmail.com>

on a related subject I'm working on a web->fax solution and would be
willing to share my code when I'm done but I'm stuck on getting the
fax status for api_hangup_hook.  I wish I could do something like this

originate {fax_ident=312-123-4567,fax_header='Soliant Consulting -
Brian',api_hangup_hook='system /usr/bin/php
/usr/local/freeswitch/scripts/sentFax.php bob at smith.com
${fax_result_code} ${fax_result_text}
${fax_document_total_pages}'}sofia/gateway/outbound.fax/1004
&txfax(/tmp/fax.tiff)

However freeswitch does not seem to like channel variables inside
api_hangup_hook.  I've thought about using python or lua to send the
fax instead of originate but I don't know how to do the txfax call.

On Sat, Jan 16, 2010 at 2:05 PM, Costa Zikalala
<costa.zikalala at gmail.com> wrote:
> Yes Steve, I'm already using that for fax2email.
> I'm now trying to do things in the opposite direction.
>
> *A realy great script by the way*
>
> Thanks
> Costa
>
>
> 2010/1/16 Steve Steffler <stevesteffler at shaw.ca>
>>
>> Costa,
>> I wrote this script to handle fax2email (but not email2fax). ?It uses
>> variables you set in the dialplan in advance for the email address for that
>> fax DID.
>> http://steffler.info/wp-content/uploads/2009/06/process-rxfax.py
>> Regards,
>> Steve
>> On Jan 14, 2010, at 7:54 AM, Costa Zikalala wrote:
>>
>> Hi All
>>
>> Has anyone worked on a email2fax script for mod_fax?
>> If not how much would it cost for some genius here to quickly whip-up one?
>>
>> Ideally both email2fax and fax2email should come standard with mod_fax or
>> is this not possible?
>>
>> Thanks
>> Costa
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From devel at thom.fr.eu.org  Mon Jan 18 08:03:47 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Mon, 18 Jan 2010 17:03:47 +0100
Subject: [Freeswitch-users] voicemail->email
Message-ID: <79fa7b97c59bbfd35a68a88ce667f82c@thom.fr.eu.org>



Sorry to come back on this topic, but I could not manage to fix this,
and I don't know what to do. 

voicemail to email was working on FS 1.0.3,
but not anymore since upgrade to 1.0.4 (now running1.0.5-20100112-0400
(hacked)) 

When FS sneds the message, sendmail segfaults. Running FS as
root or standard user does not change this. Modifying FS config to run a
script like 

exec tee -a /tmp/fsmail.log | /usr/sbin/sendmail -O
LogLevel=7 -t >> /tmp/fsmail.log 2>&1 

make sendmail segfaults too. 

then
doing su -l freeeswitch then cat /tmp/fsmail.log | /usr/sbin/sendmail -f
freeswitch at mydomain.com -t calle at mydomain.com succeeds. 

Don't know what
to look for now. 

The system was not (except freeswitch) updated between
running with FS 1.0.3 and 1.0.4. 

Anybody can help ? 

Fran?ois 

 
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From anthony.minessale at gmail.com  Mon Jan 18 08:24:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 18 Jan 2010 10:24:00 -0600
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <7d79b3931001180322p74a1227qe0c2199a77cbfe2@mail.gmail.com>
References: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
	<191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>
	<7d79b3931001132357i36bbb482jdf2bbdd3aea2a583@mail.gmail.com>
	<7d79b3931001180322p74a1227qe0c2199a77cbfe2@mail.gmail.com>
Message-ID: <191c3a031001180824q6e364c72g3f789892597e9469@mail.gmail.com>

try a less famous signal like SIGUSR1 it's possible something in perl still
reacts to SIGINT


On Mon, Jan 18, 2010 at 5:22 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Here is the result
>
> Program:
>
> require ESL;
> use IO::Socket::INET;
> use Data::Dumper;
>
> my $ip = "192.168.1.222";
> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
> Proto => 'tcp',  Listen => 1,  Reuse => 1 );
> die "Could not create socket: $!\n" unless $sock;
>
>
> for(;;) {
>     my $new_sock = $sock->accept();
>     next if (not defined ($new_sock));
>     my $pid = fork();
>     if ($pid) {
>         close($new_sock);
>         next;
>     }
>     print "CHILD PID: $$\n";
>     &register_Signals_Child();
>     sub register_Signals_Child() {
>         foreach ( keys %SIG ) {
>             $SIG{$_} = 'Handler';
>         }
>     }
>
>     sub Handler() {
>
>         my $handle=$_[0];
>         if($handle eq "INT") {
>             print "CHILD $$: SIGNAL SIG$handle is generated\n";
>         }
>         else
>         {
>             print "CHILD $$: Received $handle\n";
>
>         }
>     }
>     my $host = $new_sock->sockhost();
>     my $fd = fileno($new_sock);
>
>     my $con = new ESL::ESLconnection($fd);
>     my $info = $con->getInfo();
>
>     my $uuid = $info->getHeader("unique-id");
>
>     printf "Connected call %s, from %s\n", $uuid,
> $info->getHeader("caller-caller-id-number");
>     my $r=$con->execute("answer");
>     print Dumper $r;
>     $con->events("plain","all");
>     my
> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>     while($con->connected()) {
>         my $e = $con->recvEvent();
>
>         if ($e) {
>             my $name = $e->getHeader("event-name");
>             print "EVENT [$name]\n";
>             if ($name eq "DTMF") {
>                 my $digit = $e->getHeader("dtmf-digit");
>                 print "$digit\n";
>             }
>         }
>     }
>     close($new_sock);
> }
>
> I executed the program and the following things were printed
>
> CHILD PID: 6778
> Connected call e0d1001a-03f4-11df-b002-db488337e0ea, from 1001
> $VAR1 = 0;
> EVENT [CHANNEL_EXECUTE]
> EVENT [CHANNEL_ANSWER]
> EVENT [CHANNEL_EXECUTE_COMPLETE]
> EVENT [COMMAND]
> EVENT [CHANNEL_EXECUTE]
> EVENT [HEARTBEAT]
> EVENT [RE_SCHEDULE]
> EVENT [CHANNEL_EXECUTE_COMPLETE]
>
> Then from another shell I executed kill -2 6778, the result is follows
> CHILD 6778: SIGNAL SIGINT is generated
> EVENT [SERVER_DISCONNECTED]
>
> But the child process is still running as expected.
> But  I don't know why I received SERVER_DISCONNECTED from freeswitch???
>
>
>
>
>
>
> On Thu, Jan 14, 2010 at 1:27 PM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> I taught the signal handler will be inherited by the child process. It
>> also does like that.
>> After making a call, If I press ctrl + c, the above program printed
>> PARENT PID: Signal SIGINT is generated
>> CHILD PID: Signal SIGINT is generated.
>>
>> So I think the sigal handlers will be inherited to the child.
>> Anyway I'll also try registering signal handlers in child also, and then
>> I'll come back with that result.
>>
>> Thanks....
>> On Wed, Jan 13, 2010 at 9:48 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> you would have to register signals in your child process too
>>>
>>>   On Wed, Jan 13, 2010 at 3:13 AM, lakshmanan ganapathy <
>>> lakindia89 at gmail.com> wrote:
>>>
>>>>  Hi all,
>>>>
>>>> I've done a sample program (In perl ESL) , which play a file to the
>>>> caller and then it will call recvEvent() and print the event name. I've
>>>> handled signals also.
>>>>
>>>> When I send SIGINT to my program (Perl), the signal handler is called
>>>> and I can see the print output. But in the same time,  I received
>>>> SERVER_DISCONNECTED from freeswitch as event.
>>>>
>>>> I don't know why I received SERVER_DISCONNECTED from freeswitch. Is it
>>>> because, the recvEvent() from perl internally  calls the recvevent function
>>>> in the Esl.c and when it waits to receive the information from socket, the
>>>> signal occurred???
>>>>
>>>> Please clarify me!!
>>>>
>>>> Here is my program
>>>> require ESL;
>>>> use IO::Socket::INET;
>>>> use Data::Dumper;
>>>>
>>>> my $ip = "192.168.1.222";
>>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>>> die "Could not create socket: $!\n" unless $sock;
>>>> &register_Signals();
>>>>
>>>> for(;;) {
>>>>     my $new_sock = $sock->accept();
>>>>     next if (not defined ($new_sock));
>>>>     my $pid = fork();
>>>>     if ($pid) {
>>>>         close($new_sock);
>>>>         next;
>>>>     }
>>>>     print "CHILD PID: $$\n";
>>>>     my $host = $new_sock->sockhost();
>>>>     my $fd = fileno($new_sock);
>>>>
>>>>     my $con = new ESL::ESLconnection($fd);
>>>>     my $info = $con->getInfo();
>>>>
>>>>     my $uuid = $info->getHeader("unique-id");
>>>>
>>>>     printf "Connected call %s, from %s\n", $uuid,
>>>> $info->getHeader("caller-caller-id-number");
>>>>     my $r=$con->execute("answer");
>>>>     print Dumper $r;
>>>>     $con->events("plain","all");
>>>>     my
>>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>>
>>>>     while($con->connected()) {
>>>>         my $e = $con->recvEvent();
>>>>
>>>>         if ($e) {
>>>>             my $name = $e->getHeader("event-name");
>>>>             print "EVENT [$name]\n";
>>>>             if ($name eq "DTMF") {
>>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>>                 print "$digit\n";
>>>>             }
>>>>         }
>>>>     }
>>>>     close($new_sock);
>>>> }
>>>> sub register_Signals() {
>>>>     foreach ( keys %SIG ) {
>>>>         $SIG{$_} = 'sig_Handler';
>>>>     }
>>>> }
>>>>
>>>> sub sig_Handler() {
>>>>     my $handle=$_[0];
>>>>     if($handle eq "INT") {
>>>>         print "$$: SIGNAL SIG$handle is generated\n";
>>>>     }
>>>> }
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From Russell.Mosemann at cune.org  Mon Jan 18 08:47:38 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 18 Jan 2010 16:47:38 -0000
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <191c3a031001180824q6e364c72g3f789892597e9469@mail.gmail.com>
Message-ID: <20100118164738.9814D21DD50@cuneorg-email.cune.pri>

> try a less famous signal like SIGUSR1 it's possible something in perl still
> reacts to SIGINT

Such as an interrupted system call, perhaps? Depending on the operating
system, some system calls restart and some do not. There is not enough
debugging information in the original message to know what is happening
in the program when it is interrupted. If the interrupted system call is
not restarted, it probably returns an "interrupted" error code and needs
to be called, again.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From help at pdscc.com  Mon Jan 18 09:52:49 2010
From: help at pdscc.com (Harondel J. Sibble)
Date: Mon, 18 Jan 2010 09:52:49 -0800
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4 (fixed)
In-Reply-To: <8C20E0B1-0E56-4892-9D36-4D6DF3331244@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20100105065356.AEE0612F5@sinclaire.sibble.net>,
	<8C20E0B1-0E56-4892-9D36-4D6DF3331244@freeswitch.org>
Message-ID: <20100118175251.531A21DB501@sinclaire.sibble.net>

Thanks, I'll do that this week and report back.

When you say the latest lib and client, are you refering to developer only 
versions?  All I have access to are the official releases on the zfone site.

On 7 Jan 2010 at 17:30, Brian West wrote:

> Harondel,
> 	Please update your FreeSWITCH source rev 16204 fixes the SAS passing in all
> cases from my testing.  You'll need the latest ZRTP Lib and zfone
> application to make this work... I'm not too sure Tiviphone does this yet as
> I don't have one to test with.  This also fixes the issue when both sides
> are enrolled.  Next we will fix the video portion so both video and audio
> will go thru zrtp.
> 
> Please try it and let me know.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From freeswitch at aastral.net  Mon Jan 18 10:04:20 2010
From: freeswitch at aastral.net (Bill W)
Date: Mon, 18 Jan 2010 13:04:20 -0500
Subject: [Freeswitch-users] database disk image is malformed ISSUE
In-Reply-To: <206AA031-94DA-4AAB-8761-90A06A80E9F5@jerris.com>
References: <c8df2a131001180300m25419dd8y65ff80ce64824cef@mail.gmail.com>
	<206AA031-94DA-4AAB-8761-90A06A80E9F5@jerris.com>
Message-ID: <4B54A2A4.2040601@aastral.net>

Hey Michael,

Is there a way to get the core.db and fifo.db into ODBC?  I didn't see 
anything on the wiki about that.


Thanks,
Bill W


Michael Jerris wrote:
> sqlite doesn't scale.  If you want to do anything serious, use odbc and a real database instead.
> 



From troy at tlainvestments.com  Mon Jan 18 11:11:31 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Mon, 18 Jan 2010 12:11:31 -0700
Subject: [Freeswitch-users] More playing with sessions in lua
Message-ID: <095F1264-AB44-45A8-A84B-2C528E61E771@tlainvestments.com>

I think there may be other ways to do this, but if I could understand how to do it from lua, I think it would help me understand more about what's going on and make the mod_lua even more valuable to me and anyone else struggling with this kind of issue.

I am trying to answer a call in lua, exchange a bit of information with the caller, then I want to originate a call to another endpoint, and depending on what their response to some audio questions is, connect the original caller to them, or send the original caller away.  The problem is that I want the original caller to her ring tones or music on hold while they're waiting.  In addition to the way presented here, I've tried parking the caller using uuid_park, but still can't figure out how to play music/ringback for them.

session:answer();
session:sleep(1000);
session:execute("playback","pleasehold.wav")

local targetEndpoint = "1100 at default" -- or wherever

-- ringback works only AFTER the bridge line, below.  How do I get it to start immediately?
local destSession = freeswitch.Session("{ringback='myringback.wav'}".. targetEndpoint)	-- this "originates" a call to targetEndpoint

local digit destSession:playAndGetDigits(1,1,3,3000,"#","instructions.wav","[1-3]")

if (digit == "1") then
	freeswitch.bridge(session,destSession)
else
	session:execute("playback","goodbye.wav")
	session:hangup()
	destSession:hangup()
end


I appreciate any help on this.

-Troy



From john at acsol.net  Mon Jan 18 11:24:12 2010
From: john at acsol.net (John)
Date: Mon, 18 Jan 2010 12:24:12 -0700
Subject: [Freeswitch-users] Call Manager
Message-ID: <4B54B55C.3090402@acsol.net>

Is there an active project for a client call control software for 
Freeswitch, such as ShoreTel's or Cisco's Call Manager? Looking for the 
ability for a Windows or Mac user to be able to transfer calls, See 
presence, conference calls etc.

Thanks




From nicolas at medularis.com  Mon Jan 18 11:38:39 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 18 Jan 2010 16:38:39 -0300
Subject: [Freeswitch-users] Takes 5 minutes for FS to start (weird
	Test-Average-Step messages)
Message-ID: <1b46b4e81001181138g37157922sfb4f5ddf5f91dae8@mail.gmail.com>

I got a server on Rackspace Cloud (www.rackspacecloud.com). I compiled
FreeSWITCH from svn trunk and once it starts it works fine, no problem at
all. But, it takes FreeSWITCH about 5 minutes to start. It prints,
literally, hundreds of messages like the ones below, until the value of
"Test:" reaches 0. I looked for info about this on the wiki and the mailing
list but couldn't find anything. Is there a way to suppress this? Why does
it happen? Is it because the server is a virtual machine? Thanks!


2010-01-18 19:12:00.373937 [CONSOLE] switch_time.c:188 Test: 611 Average:
10000 Step: 13
2010-01-18 19:12:01.373937 [CONSOLE] switch_time.c:188 Test: 598 Average:
10000 Step: 12
2010-01-18 19:12:02.373937 [CONSOLE] switch_time.c:188 Test: 586 Average:
10000 Step: 11
2010-01-18 19:12:03.373937 [CONSOLE] switch_time.c:188 Test: 575 Average:
10000 Step: 10
2010-01-18 19:12:04.373937 [CONSOLE] switch_time.c:188 Test: 565 Average:
10000 Step: 9
2010-01-18 19:12:05.373937 [CONSOLE] switch_time.c:188 Test: 556 Average:
10000 Step: 8
2010-01-18 19:12:06.373937 [CONSOLE] switch_time.c:188 Test: 548 Average:
10000 Step: 7
2010-01-18 19:12:07.373937 [CONSOLE] switch_time.c:188 Test: 541 Average:
10000 Step: 6
2010-01-18 19:12:08.373937 [CONSOLE] switch_time.c:188 Test: 535 Average:
10000 Step: 5
2010-01-18 19:12:09.373937 [CONSOLE] switch_time.c:188 Test: 530 Average:
10000 Step: 4
2010-01-18 19:12:10.373937 [CONSOLE] switch_time.c:188 Test: 526 Average:
10000 Step: 3
2010-01-18 19:12:11.373937 [CONSOLE] switch_time.c:188 Test: 523 Average:
10000 Step: 2
2010-01-18 19:12:12.373937 [CONSOLE] switch_time.c:188 Test: 521 Average:
10000 Step: 1
2010-01-18 19:12:13.373937 [CONSOLE] switch_time.c:188 Test: 520 Average:
10000 Step: 1
2010-01-18 19:12:14.373937 [CONSOLE] switch_time.c:188 Test: 519 Average:
10000 Step: 1
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From anthony.minessale at gmail.com  Mon Jan 18 11:44:44 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 18 Jan 2010 13:44:44 -0600
Subject: [Freeswitch-users] More playing with sessions in lua
In-Reply-To: <095F1264-AB44-45A8-A84B-2C528E61E771@tlainvestments.com>
References: <095F1264-AB44-45A8-A84B-2C528E61E771@tlainvestments.com>
Message-ID: <191c3a031001181144g200e0a41i23dfe6813865965b@mail.gmail.com>

as soon as you play please hold there is no longer a chance for signaling
based ringback
you could have used the application ring_ready to send your phone an 180
ringing if you did not play the file
once you played the file you are responsible for sending audio to the
channel, think of it like a gui where the runtime loop takes input and you
must not do anything blocking in that loop and you can't play the file and
do something else at the same time without more threads which is not easy to
do from an embedded script.



On Mon, Jan 18, 2010 at 1:11 PM, Troy Anderson <troy at tlainvestments.com>wrote:

> I think there may be other ways to do this, but if I could understand how
> to do it from lua, I think it would help me understand more about what's
> going on and make the mod_lua even more valuable to me and anyone else
> struggling with this kind of issue.
>
> I am trying to answer a call in lua, exchange a bit of information with the
> caller, then I want to originate a call to another endpoint, and depending
> on what their response to some audio questions is, connect the original
> caller to them, or send the original caller away.  The problem is that I
> want the original caller to her ring tones or music on hold while they're
> waiting.  In addition to the way presented here, I've tried parking the
> caller using uuid_park, but still can't figure out how to play
> music/ringback for them.
>
> session:answer();
> session:sleep(1000);
> session:execute("playback","pleasehold.wav")
>
> local targetEndpoint = "1100 at default" -- or wherever
>
> -- ringback works only AFTER the bridge line, below.  How do I get it to
> start immediately?
> local destSession = freeswitch.Session("{ringback='myringback.wav'}"..
> targetEndpoint)  -- this "originates" a call to targetEndpoint
>
> local digit
> destSession:playAndGetDigits(1,1,3,3000,"#","instructions.wav","[1-3]")
>
> if (digit == "1") then
>        freeswitch.bridge(session,destSession)
> else
>        session:execute("playback","goodbye.wav")
>        session:hangup()
>        destSession:hangup()
> end
>
>
> I appreciate any help on this.
>
> -Troy
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Mon Jan 18 11:45:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 18 Jan 2010 13:45:51 -0600
Subject: [Freeswitch-users] Takes 5 minutes for FS to start (weird
	Test-Average-Step messages)
In-Reply-To: <1b46b4e81001181138g37157922sfb4f5ddf5f91dae8@mail.gmail.com>
References: <1b46b4e81001181138g37157922sfb4f5ddf5f91dae8@mail.gmail.com>
Message-ID: <191c3a031001181145v34600633uacdee6c9ebc12276@mail.gmail.com>

I dont think you have latest trunk based on what you are reporting.
you should try to get the up to the minuted latest and try again.


On Mon, Jan 18, 2010 at 1:38 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> I got a server on Rackspace Cloud (www.rackspacecloud.com). I compiled
> FreeSWITCH from svn trunk and once it starts it works fine, no problem at
> all. But, it takes FreeSWITCH about 5 minutes to start. It prints,
> literally, hundreds of messages like the ones below, until the value of
> "Test:" reaches 0. I looked for info about this on the wiki and the mailing
> list but couldn't find anything. Is there a way to suppress this? Why does
> it happen? Is it because the server is a virtual machine? Thanks!
>
>
> 2010-01-18 19:12:00.373937 [CONSOLE] switch_time.c:188 Test: 611 Average:
> 10000 Step: 13
> 2010-01-18 19:12:01.373937 [CONSOLE] switch_time.c:188 Test: 598 Average:
> 10000 Step: 12
> 2010-01-18 19:12:02.373937 [CONSOLE] switch_time.c:188 Test: 586 Average:
> 10000 Step: 11
> 2010-01-18 19:12:03.373937 [CONSOLE] switch_time.c:188 Test: 575 Average:
> 10000 Step: 10
> 2010-01-18 19:12:04.373937 [CONSOLE] switch_time.c:188 Test: 565 Average:
> 10000 Step: 9
> 2010-01-18 19:12:05.373937 [CONSOLE] switch_time.c:188 Test: 556 Average:
> 10000 Step: 8
> 2010-01-18 19:12:06.373937 [CONSOLE] switch_time.c:188 Test: 548 Average:
> 10000 Step: 7
> 2010-01-18 19:12:07.373937 [CONSOLE] switch_time.c:188 Test: 541 Average:
> 10000 Step: 6
> 2010-01-18 19:12:08.373937 [CONSOLE] switch_time.c:188 Test: 535 Average:
> 10000 Step: 5
> 2010-01-18 19:12:09.373937 [CONSOLE] switch_time.c:188 Test: 530 Average:
> 10000 Step: 4
> 2010-01-18 19:12:10.373937 [CONSOLE] switch_time.c:188 Test: 526 Average:
> 10000 Step: 3
> 2010-01-18 19:12:11.373937 [CONSOLE] switch_time.c:188 Test: 523 Average:
> 10000 Step: 2
> 2010-01-18 19:12:12.373937 [CONSOLE] switch_time.c:188 Test: 521 Average:
> 10000 Step: 1
> 2010-01-18 19:12:13.373937 [CONSOLE] switch_time.c:188 Test: 520 Average:
> 10000 Step: 1
> 2010-01-18 19:12:14.373937 [CONSOLE] switch_time.c:188 Test: 519 Average:
> 10000 Step: 1
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jmesquita at freeswitch.org  Mon Jan 18 11:51:12 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 18 Jan 2010 17:51:12 -0200
Subject: [Freeswitch-users] Call Manager
In-Reply-To: <4B54B55C.3090402@acsol.net>
References: <4B54B55C.3090402@acsol.net>
Message-ID: <ebbc2d551001181151j4a41bb34s4ba7771d0a874eb7@mail.gmail.com>

FSComm will have the ability to do some of it when the plugins infraestrucre
is developed. It shouldn't take too long for that to happen and of course
that the first big plugin would be ESL goodies.

Regards,
Jo?o Mesquita
FSComm Developer


On Mon, Jan 18, 2010 at 5:24 PM, John <john at acsol.net> wrote:

> Is there an active project for a client call control software for
> Freeswitch, such as ShoreTel's or Cisco's Call Manager? Looking for the
> ability for a Windows or Mac user to be able to transfer calls, See
> presence, conference calls etc.
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Mon Jan 18 11:55:27 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 18 Jan 2010 20:55:27 +0100
Subject: [Freeswitch-users] How to register from FS to Huawei SoftX3000
In-Reply-To: <8ccbff061001160917m22edfed8n291ea8a07518e4f7@mail.gmail.com>
References: <8ccbff061001160917m22edfed8n291ea8a07518e4f7@mail.gmail.com>
Message-ID: <65d96fc81001181155y631d03bdr697360e0bff28ba@mail.gmail.com>

hi Dome,

well,  SOFTX3000 is a great switch :)


anyhow... there are 2 ways you can provide services via SIP to your
customers:


1. MSBR ( Multimedia subscriber)
2. SIP trunk



with MSBR you actually create an account (MMTE - multimedia terminal) and
bind a phone number to that account... When using MSBR you MUST register in
order to place/receive calls and of course you are limited to only 1
simultaneous calls on that account. From my oppinion this is not to be used
with FS because of the limultaneous calls limit per account!

with SIP trunk ... it is simple, this is just number analisys. The important
thing is that SoftX3000 doesn't support registration on SIP Trunk! You must
not register. Also, it uses SIP OPTIONS for keepalive and you should respond
to that messages as well... otherwise it will bring the trunk down.


regarding call limitation ... well, this is something realy unlikelly...
there is no such feature on SX to filter by user-agent.


Anyhow, did you try to register with any softphone? ... I've tested x-lite,
jsphone, ekiga, 3cx, ... and everyone is working!



can you send us the wireshark capture ?

or you can send me the accounts to check that out... of course off the list
:)


T.

On Sat, Jan 16, 2010 at 6:17 PM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear sir,
>          I found some provider use Huawei SoftX3000 and can limit use
> call from they softphone only. (use eyeball SDK).
> They can limit some account can register and call by sip server like
> an FS and Asterisk. but some account can't. (register and call by
> softphone). and i don't know how they can do that.
> So i try to use wireshark to debug sip headeder when use softphone
> with both account type. it's nothing diferent.
>          I want to use both account work by FS register to  Huawei
> SoftX3000. Can someone help me. i can give you softphone and both
> account type for test.
>
>
> Best Regards.
>
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Jan 18 12:09:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Jan 2010 12:09:26 -0800
Subject: [Freeswitch-users] How do I invite group to join existing call?
In-Reply-To: <4B5412C6020000E1000003D6@mail.fribert.dk>
References: <4B5412C6020000E1000003D6@mail.fribert.dk>
Message-ID: <87f2f3b91001181209y7a0aa68fs8a580712484c7a11@mail.gmail.com>

On Sun, Jan 17, 2010 at 10:50 PM, mailinglist <mailinglist at fribert.dk>wrote:

> Hi All
>
> I would like to be able to invite a group / global to join an existing
> call, but how do I accomplish this, can it be done?
>

Malfunction! Need Input!

Could you give us a few more details on what you're trying to accomplish?
What's the big picture? I'm curious what problem you're trying to solve. I'm
sure the gang here will have thoughts to pass along. :)

-MC
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From pete at privateconnect.com  Mon Jan 18 12:22:53 2010
From: pete at privateconnect.com (Pete Mueller)
Date: Mon, 18 Jan 2010 13:22:53 -0700
Subject: [Freeswitch-users] More playing with sessions in lua
Message-ID: <20100118132253.2ad02225396a31c9de30536f2e338977.df3ad2f537.wbe@email04.secureserver.net>

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From nicolas at medularis.com  Mon Jan 18 12:25:53 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 18 Jan 2010 17:25:53 -0300
Subject: [Freeswitch-users] Takes 5 minutes for FS to start (weird
	Test-Average-Step messages)
In-Reply-To: <191c3a031001181145v34600633uacdee6c9ebc12276@mail.gmail.com>
References: <1b46b4e81001181138g37157922sfb4f5ddf5f91dae8@mail.gmail.com>
	<191c3a031001181145v34600633uacdee6c9ebc12276@mail.gmail.com>
Message-ID: <1b46b4e81001181225j39c61895y3360d01685963973@mail.gmail.com>

You are right, I got the source code a few days ago. I just updated and
compiled again. Now I'm not getting those messages, only this:

2010-01-18 20:23:50.386188 [CONSOLE] switch_time.c:959 Calibrating timer,
please wait...
2010-01-18 20:23:50.386188 [WARNING] switch_time.c:190 Timer resolution of
10000 microseconds detected!
Do you have your kernel timer set to higher than 1 kHz? You may experience
audio problems.
2010-01-18 20:23:55.396188 [CONSOLE] switch_loadable_module.c:890
Successfully Loaded [CORE_SOFTTIMER_MODULE]
2010-01-18 20:23:55.396188 [NOTICE] switch_loadable_module.c:229 Adding
Timer 'soft'


Thanks!



On Mon, Jan 18, 2010 at 4:45 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I dont think you have latest trunk based on what you are reporting.
> you should try to get the up to the minuted latest and try again.
>
>
> On Mon, Jan 18, 2010 at 1:38 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> I got a server on Rackspace Cloud (www.rackspacecloud.com). I compiled
>> FreeSWITCH from svn trunk and once it starts it works fine, no problem at
>> all. But, it takes FreeSWITCH about 5 minutes to start. It prints,
>> literally, hundreds of messages like the ones below, until the value of
>> "Test:" reaches 0. I looked for info about this on the wiki and the mailing
>> list but couldn't find anything. Is there a way to suppress this? Why does
>> it happen? Is it because the server is a virtual machine? Thanks!
>>
>>
>> 2010-01-18 19:12:00.373937 [CONSOLE] switch_time.c:188 Test: 611 Average:
>> 10000 Step: 13
>> 2010-01-18 19:12:01.373937 [CONSOLE] switch_time.c:188 Test: 598 Average:
>> 10000 Step: 12
>> 2010-01-18 19:12:02.373937 [CONSOLE] switch_time.c:188 Test: 586 Average:
>> 10000 Step: 11
>> 2010-01-18 19:12:03.373937 [CONSOLE] switch_time.c:188 Test: 575 Average:
>> 10000 Step: 10
>> 2010-01-18 19:12:04.373937 [CONSOLE] switch_time.c:188 Test: 565 Average:
>> 10000 Step: 9
>> 2010-01-18 19:12:05.373937 [CONSOLE] switch_time.c:188 Test: 556 Average:
>> 10000 Step: 8
>> 2010-01-18 19:12:06.373937 [CONSOLE] switch_time.c:188 Test: 548 Average:
>> 10000 Step: 7
>> 2010-01-18 19:12:07.373937 [CONSOLE] switch_time.c:188 Test: 541 Average:
>> 10000 Step: 6
>> 2010-01-18 19:12:08.373937 [CONSOLE] switch_time.c:188 Test: 535 Average:
>> 10000 Step: 5
>> 2010-01-18 19:12:09.373937 [CONSOLE] switch_time.c:188 Test: 530 Average:
>> 10000 Step: 4
>> 2010-01-18 19:12:10.373937 [CONSOLE] switch_time.c:188 Test: 526 Average:
>> 10000 Step: 3
>> 2010-01-18 19:12:11.373937 [CONSOLE] switch_time.c:188 Test: 523 Average:
>> 10000 Step: 2
>> 2010-01-18 19:12:12.373937 [CONSOLE] switch_time.c:188 Test: 521 Average:
>> 10000 Step: 1
>> 2010-01-18 19:12:13.373937 [CONSOLE] switch_time.c:188 Test: 520 Average:
>> 10000 Step: 1
>> 2010-01-18 19:12:14.373937 [CONSOLE] switch_time.c:188 Test: 519 Average:
>> 10000 Step: 1
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Jan 18 13:18:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Jan 2010 13:18:50 -0800
Subject: [Freeswitch-users] Call for help - content ideas for freeswitch.org
Message-ID: <87f2f3b91001181318n98af6beuf6e50ebbcbd4ba5c@mail.gmail.com>

Hello all,

I'd like to ask everyone to think about things that we can put up on
freeswitch.org. We are interested in anything related to FreeSWITCH,
certainly, but also any VoIP or telecom news/articles on other sites that
are of interest to FreeSWITCH users. (See
http://www.freeswitch.org/node/229as an example.)

Please email me off list if you have ideas about content that would be
appropriate for our main page. Likewise, if you would like to be the author
of stories and/or blog posts on the main page then definitely let me know.
We would love to have some fresh perspectives represented.

Thanks!
-Michael
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From djbinter at gmail.com  Mon Jan 18 13:27:49 2010
From: djbinter at gmail.com (DJB INTERNATIONAL)
Date: Mon, 18 Jan 2010 13:27:49 -0800
Subject: [Freeswitch-users] Dial Plan bridge did not return a correct
	variable from reqular expression
Message-ID: <94f7dfb11001181327p5a807b45ld3d44a7238d2f3ee@mail.gmail.com>

Does anyone experience this problem?

SVN:   16395

Dialplan:

   <extension name="tollfree">
     <condition field="${toll_allow}" expression="tollfree"/>
     <condition field="destination_number"
expression="^(\+1|1)?(8(00|88|77|66)[2-9]\d{6})$">
      <action application="set" data="bypass_media=true"/>
       <action application="bridge" data="sofia/external/1$
2 at tollfreetollfree.com"/>
     </condition>
   </extension>

Possible Error:
Dialplan: sofia/internal/2132345567 at 204.110.15.190 Regex (PASS) [tollfree]
destination_number(18005551212) =~ /^(\+1|1)?(8(00|88|77|66)[2-9]\d{6})$/
break=on-false
Dialplan: sofia/internal/2132345567 at 204.110.15.190 Action
set(bypass_media=true)
Dialplan: sofia/internal/2132345567 at 204.110.15.190 Action
bridge(sofia/external/1 at tollfreetollfree.com)   ******  (It should return $2
as 8005551212)


Thank you,
Dorn B. (djbinter)
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From nicolas at medularis.com  Mon Jan 18 13:58:59 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 18 Jan 2010 18:58:59 -0300
Subject: [Freeswitch-users] Call for help - content ideas for
	freeswitch.org
In-Reply-To: <87f2f3b91001181318n98af6beuf6e50ebbcbd4ba5c@mail.gmail.com>
References: <87f2f3b91001181318n98af6beuf6e50ebbcbd4ba5c@mail.gmail.com>
Message-ID: <1b46b4e81001181358m3ebb56eeke827a64ea33debf6@mail.gmail.com>

Here's a little post on getting started with FreeSWITCH to create a small
click to call app:

- http://www.guayal.com/how-to-bridge-two-calls-with-freeswitch

At least 2 more parts to come.


On Mon, Jan 18, 2010 at 6:18 PM, Michael Collins <msc at freeswitch.org> wrote:

> Hello all,
>
> I'd like to ask everyone to think about things that we can put up on
> freeswitch.org. We are interested in anything related to FreeSWITCH,
> certainly, but also any VoIP or telecom news/articles on other sites that
> are of interest to FreeSWITCH users. (See
> http://www.freeswitch.org/node/229 as an example.)
>
> Please email me off list if you have ideas about content that would be
> appropriate for our main page. Likewise, if you would like to be the author
> of stories and/or blog posts on the main page then definitely let me know.
> We would love to have some fresh perspectives represented.
>
> Thanks!
> -Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lists at redbonez.net  Mon Jan 18 14:43:23 2010
From: lists at redbonez.net (Adam Ford)
Date: Mon, 18 Jan 2010 15:43:23 -0700
Subject: [Freeswitch-users] Bria softphone registration issue (NAT/rport
	issue?)
Message-ID: <00f301ca988f$a54eca70$efec5f50$@net>

I don't know if any of you would be able to help with this, but I figured I
would ask.  

 

After a few hours of inactivity, it seems my Bria softphones lose connection
with FreeSWITCH.  They are able to call out, but when calling in I just get
silence until it goes to voicemail.  It sounds like a classic NAT issue,
except the NAT is completely handled by a Soincwall that has VoIP features
to support SIP transformations (SonicOS Enhanced 5.x).  

 

It seems as though after a period of time the port in which the Sonicwall is
assigning for the SIP UA and the port that FreeSWITCH has registered get out
of sync, despite the Bria re-registering every 5 minutes.  Could this be
caused by 'Use rport' being enabled by default in the Bria softphone?

 

Setup -

FreeSWITCH 1.0.4

Bria 2.5.4

Sonicwall E5500 with SIP transformations and Consistent NAT enabled

 

 

Thanks in advance for any help,

 

-Adam

 

 

 

 

 

 

 

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From brian at freeswitch.org  Mon Jan 18 14:55:11 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 Jan 2010 16:55:11 -0600
Subject: [Freeswitch-users] Bria softphone registration issue (NAT/rport
	issue?)
In-Reply-To: <00f301ca988f$a54eca70$efec5f50$@net>
References: <00f301ca988f$a54eca70$efec5f50$@net>
Message-ID: <A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>

1.0.4 is not supported anymore please update to latest.  http://latest.freeswitch.org

/b

On Jan 18, 2010, at 4:43 PM, Adam Ford wrote:

> FreeSWITCH 1.0.4

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From lists at redbonez.net  Mon Jan 18 16:14:30 2010
From: lists at redbonez.net (Adam Ford)
Date: Mon, 18 Jan 2010 17:14:30 -0700
Subject: [Freeswitch-users] Bria softphone registration issue
	(NAT/rport	issue?)
In-Reply-To: <A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>
References: <00f301ca988f$a54eca70$efec5f50$@net>
	<A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>
Message-ID: <010101ca989c$5fb5f1c0$1f21d540$@net>

This is more of a Bria configuration question than FreeSWITCH.  I wouldn't
think the version of FreeSWITCH would matter much, but thanks for the
suggestion.

 

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, January 18, 2010 3:55 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bria softphone registration issue (NAT/rport
issue?)

 

1.0.4 is not supported anymore please update to latest.
http://latest.freeswitch.org

 

/b

 

On Jan 18, 2010, at 4:43 PM, Adam Ford wrote:





FreeSWITCH 1.0.4

 

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From brian at freeswitch.org  Mon Jan 18 16:17:17 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 Jan 2010 18:17:17 -0600
Subject: [Freeswitch-users] Bria softphone registration issue (NAT/rport
	issue?)
In-Reply-To: <010101ca989c$5fb5f1c0$1f21d540$@net>
References: <00f301ca988f$a54eca70$efec5f50$@net>
	<A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>
	<010101ca989c$5fb5f1c0$1f21d540$@net>
Message-ID: <C64D1DDC-2846-4A9F-BB82-F45D59CA99FF@freeswitch.org>

Is the phone behind the nat with FreeSWITCH?

/b

On Jan 18, 2010, at 6:14 PM, Adam Ford wrote:

> This is more of a Bria configuration question than FreeSWITCH.  I wouldn?t think the version of FreeSWITCH would matter much, but thanks for the suggestion.
>  
> -Adam
>  

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From lists at redbonez.net  Mon Jan 18 16:28:16 2010
From: lists at redbonez.net (Adam Ford)
Date: Mon, 18 Jan 2010 17:28:16 -0700
Subject: [Freeswitch-users] Bria softphone registration issue
	(NAT/rport	issue?)
In-Reply-To: <C64D1DDC-2846-4A9F-BB82-F45D59CA99FF@freeswitch.org>
References: <00f301ca988f$a54eca70$efec5f50$@net>	<A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>	<010101ca989c$5fb5f1c0$1f21d540$@net>
	<C64D1DDC-2846-4A9F-BB82-F45D59CA99FF@freeswitch.org>
Message-ID: <012101ca989e$4c6877d0$e5396770$@net>

Yes, but the nat is handled by the Sonicwall, which works perfectly for the
Polycom desk phones.   It is only the Bria softphones that are behaving
strangely.  I think it is just caused by them attempting to use rport, which
is sort of doing to work of the Sonicwall twice, causing them to lose sync
with the FreeSWITCH registration.  I just wanted the opinion of some other
professionals if this sound likely, before I consider the matter resolved.

 

Thank you,

 

-Adam

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, January 18, 2010 5:17 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bria softphone registration issue (NAT/rport
issue?)

 

Is the phone behind the nat with FreeSWITCH?

 

/b

 

On Jan 18, 2010, at 6:14 PM, Adam Ford wrote:





This is more of a Bria configuration question than FreeSWITCH.  I wouldn't
think the version of FreeSWITCH would matter much, but thanks for the
suggestion.

 

-Adam

 

 

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From brian at freeswitch.org  Mon Jan 18 16:47:05 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 Jan 2010 18:47:05 -0600
Subject: [Freeswitch-users] Bria softphone registration issue (NAT/rport
	issue?)
In-Reply-To: <012101ca989e$4c6877d0$e5396770$@net>
References: <00f301ca988f$a54eca70$efec5f50$@net>	<A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>	<010101ca989c$5fb5f1c0$1f21d540$@net>
	<C64D1DDC-2846-4A9F-BB82-F45D59CA99FF@freeswitch.org>
	<012101ca989e$4c6877d0$e5396770$@net>
Message-ID: <3FBC790C-EDC4-4323-935E-F5CA957F3BD7@freeswitch.org>

Well if the Bria Phone is behind nat with FS why is the sonic wall even involved? (Btw... I just realized they can't spell Brian... Bria... who calls a phone Bria?)

/b

On Jan 18, 2010, at 6:28 PM, Adam Ford wrote:

> Yes, but the nat is handled by the Sonicwall, which works perfectly for the Polycom desk phones.   It is only the Bria softphones that are behaving strangely.  I think it is just caused by them attempting to use rport, which is sort of doing to work of the Sonicwall twice, causing them to lose sync with the FreeSWITCH registration.  I just wanted the opinion of some other professionals if this sound likely, before I consider the matter resolved.
>  
> Thank you,
>  
> -Adam

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From lists at redbonez.net  Mon Jan 18 17:08:17 2010
From: lists at redbonez.net (Adam Ford)
Date: Mon, 18 Jan 2010 18:08:17 -0700
Subject: [Freeswitch-users] Bria softphone registration issue
	(NAT/rport	issue?)
In-Reply-To: <3FBC790C-EDC4-4323-935E-F5CA957F3BD7@freeswitch.org>
References: <00f301ca988f$a54eca70$efec5f50$@net>	<A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>	<010101ca989c$5fb5f1c0$1f21d540$@net>	<C64D1DDC-2846-4A9F-BB82-F45D59CA99FF@freeswitch.org>	<012101ca989e$4c6877d0$e5396770$@net>
	<3FBC790C-EDC4-4323-935E-F5CA957F3BD7@freeswitch.org>
Message-ID: <013201ca98a3$e3cca010$ab65e030$@net>

I guess I misunderstood your question.  The phones aren't behind a nat WITH
the FreeSWITCH.  FreeSWITCH is in collocation site, the phones are in my
office, and connecting to FreeSWITCH through the nat.

 

Maybe they wanted to call it Brian but then someone complained that it would
be to common ;)

 

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, January 18, 2010 5:47 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bria softphone registration issue (NAT/rport
issue?)

 

Well if the Bria Phone is behind nat with FS why is the sonic wall even
involved? (Btw... I just realized they can't spell Brian... Bria... who
calls a phone Bria?)

 

/b

 

On Jan 18, 2010, at 6:28 PM, Adam Ford wrote:





Yes, but the nat is handled by the Sonicwall, which works perfectly for the
Polycom desk phones.   It is only the Bria softphones that are behaving
strangely.  I think it is just caused by them attempting to use rport, which
is sort of doing to work of the Sonicwall twice, causing them to lose sync
with the FreeSWITCH registration.  I just wanted the opinion of some other
professionals if this sound likely, before I consider the matter resolved.

 

Thank you,

 

-Adam

 

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From brian at freeswitch.org  Mon Jan 18 17:12:40 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 Jan 2010 19:12:40 -0600
Subject: [Freeswitch-users] Bria softphone registration issue (NAT/rport
	issue?)
In-Reply-To: <013201ca98a3$e3cca010$ab65e030$@net>
References: <00f301ca988f$a54eca70$efec5f50$@net>	<A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>	<010101ca989c$5fb5f1c0$1f21d540$@net>	<C64D1DDC-2846-4A9F-BB82-F45D59CA99FF@freeswitch.org>	<012101ca989e$4c6877d0$e5396770$@net>
	<3FBC790C-EDC4-4323-935E-F5CA957F3BD7@freeswitch.org>
	<013201ca98a3$e3cca010$ab65e030$@net>
Message-ID: <AF55B055-5A95-446F-B700-19A81F2496ED@freeswitch.org>

Try 1.0.5

/b

On Jan 18, 2010, at 7:08 PM, Adam Ford wrote:

> I guess I misunderstood your question.  The phones aren?t behind a nat WITH the FreeSWITCH.  FreeSWITCH is in collocation site, the phones are in my office, and connecting to FreeSWITCH through the nat.
>  
> Maybe they wanted to call it Brian but then someone complained that it would be to common ;)
>  
> -Adam

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From mcampbellsmith at gmail.com  Mon Jan 18 20:52:05 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 19 Jan 2010 15:52:05 +1100
Subject: [Freeswitch-users] Bria softphone registration issue (NAT/rport
	issue?)
In-Reply-To: <AF55B055-5A95-446F-B700-19A81F2496ED@freeswitch.org>
References: <00f301ca988f$a54eca70$efec5f50$@net>
	<A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org>
	<010101ca989c$5fb5f1c0$1f21d540$@net>
	<C64D1DDC-2846-4A9F-BB82-F45D59CA99FF@freeswitch.org>
	<012101ca989e$4c6877d0$e5396770$@net>
	<3FBC790C-EDC4-4323-935E-F5CA957F3BD7@freeswitch.org>
	<013201ca98a3$e3cca010$ab65e030$@net>
	<AF55B055-5A95-446F-B700-19A81F2496ED@freeswitch.org>
Message-ID: <33c87fa31001182052n18602400jafe5090e5189da8d@mail.gmail.com>

Is there a NAT keep alive option in Bria?  Look under SIP Account
Properties, Advanced.

On Tue, Jan 19, 2010 at 12:12 PM, Brian West <brian at freeswitch.org> wrote:
> Try 1.0.5
> /b
> On Jan 18, 2010, at 7:08 PM, Adam Ford wrote:
>
> I guess I misunderstood your question.? The phones aren?t behind a nat WITH
> the FreeSWITCH.? FreeSWITCH is in collocation site, the phones are in my
> office, and connecting to FreeSWITCH through the nat.
>
> Maybe they wanted to call it Brian but then someone complained that it would
> be to common ;)
>
> -Adam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From magesh.freeswitch at gmail.com  Mon Jan 18 23:27:35 2010
From: magesh.freeswitch at gmail.com (Magesh R)
Date: Tue, 19 Jan 2010 02:27:35 -0500
Subject: [Freeswitch-users] Error in finding OpenZAP span id
Message-ID: <369c72d81001182327u38772291pa42ec950c11a1055@mail.gmail.com>

Dear All,
   I have installed Sangoma PRI card and installed wanpipe drivers. The
wanrouter process started sucessfully.
    I had the following configurations,


openzap.conf:
==========
[span wanpipe PRI_1]
name => OpenZAP
number => 1
trunk_type => e1
b-channel => 1:1-15
d-channel => 1:16
b-channel => 1:17-31


[span wanpipe PRI_2]
name => OpenZAP
number => 2
trunk_type => e1
b-channel => 2:1-15
d-channel => 2:16
b-channel => 2:17-31

openzap.conf.xml:
===========
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="1"/>
  </settings>
   <libpri_spans>
     <span name="PRI_1">
        <param name="node" value="cpe"/>
        <param name="switch" value="ni2"/>
        <param name="dialplan" value="XML"/>
        <param name="context" value="public"/>
      </span>
   </libpri_spans>
</configuration>

 When I started the freeswitch I have received the following error,

2010-01-19 12:41:22.693212 [ERR] mod_openzap.c:2039 Error finding OpenZAP
span id: name:PRI_1

Any one please tell me a way to solve this problem...

Thanks,
Mag.
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From lists at redbonez.net  Mon Jan 18 23:33:12 2010
From: lists at redbonez.net (Adam Ford)
Date: Tue, 19 Jan 2010 00:33:12 -0700
Subject: [Freeswitch-users] Bria softphone registration issue
	(NAT/rportissue?)
In-Reply-To: <33c87fa31001182052n18602400jafe5090e5189da8d@mail.gmail.com>
References: <00f301ca988f$a54eca70$efec5f50$@net><A0F25F92-D344-4A96-BAA5-AED83E4CC841@freeswitch.org><010101ca989c$5fb5f1c0$1f21d540$@net><C64D1DDC-2846-4A9F-BB82-F45D59CA99FF@freeswitch.org><012101ca989e$4c6877d0$e5396770$@net><3FBC790C-EDC4-4323-935E-F5CA957F3BD7@freeswitch.org><013201ca98a3$e3cca010$ab65e030$@net><AF55B055-5A95-446F-B700-19A81F2496ED@freeswitch.org>
	<33c87fa31001182052n18602400jafe5090e5189da8d@mail.gmail.com>
Message-ID: <C9E790B9434246C199F31D9F79218097@redbonez>

Yes, that is enabled as well.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark
Campbell-Smith
Sent: Monday, January 18, 2010 9:52 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bria softphone registration issue
(NAT/rportissue?)

Is there a NAT keep alive option in Bria?  Look under SIP Account
Properties, Advanced.

On Tue, Jan 19, 2010 at 12:12 PM, Brian West <brian at freeswitch.org> wrote:
> Try 1.0.5
> /b
> On Jan 18, 2010, at 7:08 PM, Adam Ford wrote:
>
> I guess I misunderstood your question.? The phones aren?t behind a nat
WITH
> the FreeSWITCH.? FreeSWITCH is in collocation site, the phones are in my
> office, and connecting to FreeSWITCH through the nat.
>
> Maybe they wanted to call it Brian but then someone complained that it
would
> be to common ;)
>
> -Adam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From scottferri09 at gmail.com  Tue Jan 19 01:07:06 2010
From: scottferri09 at gmail.com (Scott Fernandez)
Date: Tue, 19 Jan 2010 14:37:06 +0530
Subject: [Freeswitch-users] Establishing a Call from .Net based application
Message-ID: <df97b9c51001190107x39d94eaai7400635f0ac03f86@mail.gmail.com>

Hi,

Is there any API modules available for me to initiate a call from .Net based
application?.

The idea is to include the API modules if any with the .NET base classes so
that the API commands will be made available on it. I know it is doable when
I use socket programming in .NET in which Telnet session is created.
However, this would potentially hamper the performance of the application
because of multiple sessions that will be created for each call.

Other than that, Is there any Freeswitch API modules (like plug-ins)
available in order to include it into the .Net classes and start building
the customized application?

Any help from any one is highly appreciated.

Thanks,
Scott
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From lakindia89 at gmail.com  Tue Jan 19 01:07:34 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 19 Jan 2010 14:37:34 +0530
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <191c3a031001180824q6e364c72g3f789892597e9469@mail.gmail.com>
References: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
	<191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>
	<7d79b3931001132357i36bbb482jdf2bbdd3aea2a583@mail.gmail.com>
	<7d79b3931001180322p74a1227qe0c2199a77cbfe2@mail.gmail.com>
	<191c3a031001180824q6e364c72g3f789892597e9469@mail.gmail.com>
Message-ID: <7d79b3931001190107o200ec04dredd76689dd235588@mail.gmail.com>

I tried with SIGUSR1, but no progress. I got SERVER_DISCONNECTED.
Output:

CHILD 3814: Received USR1
EVENT [SERVER_DISCONNECTED]

In esl.c, in esl_recv_event() function, line no: 824
if (rrval < 0) {
                        strerror_r(handle->errnum, handle->err,
sizeof(handle->err));
                        goto fail;
}
When the program is blocked under receive, I passed the signal. So recv
returns -1, and in fail: it call esl_disconnect(handle).

Is it because of this??? If so, whether it should be fixed or not???



On Mon, Jan 18, 2010 at 9:54 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try a less famous signal like SIGUSR1 it's possible something in perl still
> reacts to SIGINT
>
>
>
> On Mon, Jan 18, 2010 at 5:22 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Here is the result
>>
>> Program:
>>
>> require ESL;
>> use IO::Socket::INET;
>> use Data::Dumper;
>>
>> my $ip = "192.168.1.222";
>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
>> Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>> die "Could not create socket: $!\n" unless $sock;
>>
>>
>> for(;;) {
>>     my $new_sock = $sock->accept();
>>     next if (not defined ($new_sock));
>>     my $pid = fork();
>>     if ($pid) {
>>         close($new_sock);
>>         next;
>>     }
>>     print "CHILD PID: $$\n";
>>     &register_Signals_Child();
>>     sub register_Signals_Child() {
>>         foreach ( keys %SIG ) {
>>             $SIG{$_} = 'Handler';
>>         }
>>     }
>>
>>     sub Handler() {
>>
>>         my $handle=$_[0];
>>         if($handle eq "INT") {
>>             print "CHILD $$: SIGNAL SIG$handle is generated\n";
>>         }
>>         else
>>         {
>>             print "CHILD $$: Received $handle\n";
>>
>>         }
>>     }
>>     my $host = $new_sock->sockhost();
>>     my $fd = fileno($new_sock);
>>
>>     my $con = new ESL::ESLconnection($fd);
>>     my $info = $con->getInfo();
>>
>>     my $uuid = $info->getHeader("unique-id");
>>
>>     printf "Connected call %s, from %s\n", $uuid,
>> $info->getHeader("caller-caller-id-number");
>>     my $r=$con->execute("answer");
>>     print Dumper $r;
>>     $con->events("plain","all");
>>     my
>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>     while($con->connected()) {
>>         my $e = $con->recvEvent();
>>
>>         if ($e) {
>>             my $name = $e->getHeader("event-name");
>>             print "EVENT [$name]\n";
>>             if ($name eq "DTMF") {
>>                 my $digit = $e->getHeader("dtmf-digit");
>>                 print "$digit\n";
>>             }
>>         }
>>     }
>>     close($new_sock);
>> }
>>
>> I executed the program and the following things were printed
>>
>> CHILD PID: 6778
>> Connected call e0d1001a-03f4-11df-b002-db488337e0ea, from 1001
>> $VAR1 = 0;
>> EVENT [CHANNEL_EXECUTE]
>> EVENT [CHANNEL_ANSWER]
>> EVENT [CHANNEL_EXECUTE_COMPLETE]
>> EVENT [COMMAND]
>> EVENT [CHANNEL_EXECUTE]
>> EVENT [HEARTBEAT]
>> EVENT [RE_SCHEDULE]
>> EVENT [CHANNEL_EXECUTE_COMPLETE]
>>
>> Then from another shell I executed kill -2 6778, the result is follows
>> CHILD 6778: SIGNAL SIGINT is generated
>> EVENT [SERVER_DISCONNECTED]
>>
>> But the child process is still running as expected.
>> But  I don't know why I received SERVER_DISCONNECTED from freeswitch???
>>
>>
>>
>>
>>
>>
>> On Thu, Jan 14, 2010 at 1:27 PM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>> I taught the signal handler will be inherited by the child process. It
>>> also does like that.
>>> After making a call, If I press ctrl + c, the above program printed
>>> PARENT PID: Signal SIGINT is generated
>>> CHILD PID: Signal SIGINT is generated.
>>>
>>> So I think the sigal handlers will be inherited to the child.
>>> Anyway I'll also try registering signal handlers in child also, and then
>>> I'll come back with that result.
>>>
>>> Thanks....
>>> On Wed, Jan 13, 2010 at 9:48 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> you would have to register signals in your child process too
>>>>
>>>>   On Wed, Jan 13, 2010 at 3:13 AM, lakshmanan ganapathy <
>>>> lakindia89 at gmail.com> wrote:
>>>>
>>>>>  Hi all,
>>>>>
>>>>> I've done a sample program (In perl ESL) , which play a file to the
>>>>> caller and then it will call recvEvent() and print the event name. I've
>>>>> handled signals also.
>>>>>
>>>>> When I send SIGINT to my program (Perl), the signal handler is called
>>>>> and I can see the print output. But in the same time,  I received
>>>>> SERVER_DISCONNECTED from freeswitch as event.
>>>>>
>>>>> I don't know why I received SERVER_DISCONNECTED from freeswitch. Is it
>>>>> because, the recvEvent() from perl internally  calls the recvevent function
>>>>> in the Esl.c and when it waits to receive the information from socket, the
>>>>> signal occurred???
>>>>>
>>>>> Please clarify me!!
>>>>>
>>>>> Here is my program
>>>>> require ESL;
>>>>> use IO::Socket::INET;
>>>>> use Data::Dumper;
>>>>>
>>>>> my $ip = "192.168.1.222";
>>>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>>>> die "Could not create socket: $!\n" unless $sock;
>>>>> &register_Signals();
>>>>>
>>>>> for(;;) {
>>>>>     my $new_sock = $sock->accept();
>>>>>     next if (not defined ($new_sock));
>>>>>     my $pid = fork();
>>>>>     if ($pid) {
>>>>>         close($new_sock);
>>>>>         next;
>>>>>     }
>>>>>     print "CHILD PID: $$\n";
>>>>>     my $host = $new_sock->sockhost();
>>>>>     my $fd = fileno($new_sock);
>>>>>
>>>>>     my $con = new ESL::ESLconnection($fd);
>>>>>     my $info = $con->getInfo();
>>>>>
>>>>>     my $uuid = $info->getHeader("unique-id");
>>>>>
>>>>>     printf "Connected call %s, from %s\n", $uuid,
>>>>> $info->getHeader("caller-caller-id-number");
>>>>>     my $r=$con->execute("answer");
>>>>>     print Dumper $r;
>>>>>     $con->events("plain","all");
>>>>>     my
>>>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>>>
>>>>>     while($con->connected()) {
>>>>>         my $e = $con->recvEvent();
>>>>>
>>>>>         if ($e) {
>>>>>             my $name = $e->getHeader("event-name");
>>>>>             print "EVENT [$name]\n";
>>>>>             if ($name eq "DTMF") {
>>>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>>>                 print "$digit\n";
>>>>>             }
>>>>>         }
>>>>>     }
>>>>>     close($new_sock);
>>>>> }
>>>>> sub register_Signals() {
>>>>>     foreach ( keys %SIG ) {
>>>>>         $SIG{$_} = 'sig_Handler';
>>>>>     }
>>>>> }
>>>>>
>>>>> sub sig_Handler() {
>>>>>     my $handle=$_[0];
>>>>>     if($handle eq "INT") {
>>>>>         print "$$: SIGNAL SIG$handle is generated\n";
>>>>>     }
>>>>> }
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yehavi.bourvine at gmail.com  Tue Jan 19 02:26:38 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 19 Jan 2010 12:26:38 +0200
Subject: [Freeswitch-users] Polycom configuration for FreeSWITCH
In-Reply-To: <003701ca964b$3241b100$96c51300$@net>
References: <003701ca964b$3241b100$96c51300$@net>
Message-ID: <d361ca7b1001190226y70f6dfa2o3cbf4bcffff6a887@mail.gmail.com>

I've documented my Polycom's setup in the wiki under "Polycom
configuration".

                  Regards, __Yehavi:


 2010/1/16 Adam Ford <lists at redbonez.net>

>   Does anyone who has successfully implemented Polycom IP301/501s, with
> FreeSWITCH, have a configuration they can share with me as an example? I
> have read the Polycom Admin Guide several times and understand what the
> settings are/do, I am just not sure which FreeSWITCH supports, which it
> doesn?t, and which need special configuration to work with FreeSWITCH.
>
>
>
> Thanks in advance,
>
> -Adam
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From linux4michelle at tamay-dogan.net  Tue Jan 19 02:28:54 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Tue, 19 Jan 2010 11:28:54 +0100
Subject: [Freeswitch-users] Which port I have to open?
Message-ID: <20100119102854.GQ4767@tamay-dogan.net>

Hello *,

my FreeSWITCH was in a DMZ but now I have installed it  on  a  dedicated
server (TI OMAP3530) and nothing as working... :-/

Please can someone tell me which ports I have to forward from my  router
fo Freeswitch and reverse?

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From linux4michelle at tamay-dogan.net  Tue Jan 19 02:31:39 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Tue, 19 Jan 2010 11:31:39 +0100
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
Message-ID: <20100119103139.GR4767@tamay-dogan.net>

Hi *,

I loss my last nerv, compiling all the time FreeSWITCH from source...

Is there someone providing a Debian Package from a repository?

Also it would be nice if FreeSWITCH go into the Debian distribution.

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From Russell.Mosemann at cune.org  Tue Jan 19 04:14:56 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Tue, 19 Jan 2010 06:14:56 -0600
Subject: [Freeswitch-users] Error in finding OpenZAP span id
In-Reply-To: <369c72d81001182327u38772291pa42ec950c11a1055@mail.gmail.com>
References: <369c72d81001182327u38772291pa42ec950c11a1055@mail.gmail.com>
Message-ID: <56DA8BCA12C2488A9CD829A12C87C81E@cune.pri>

>  When I started the freeswitch I have received the following error,
> 
> 2010-01-19 12:41:22.693212 [ERR] mod_openzap.c:2039 Error finding OpenZAP
> span id: name:PRI_1

You defined PRI_1 inside libpri_spans. Did you compile OpenZAP with libpri?

--
Russell Mosemann




From kond at nstel.ru  Tue Jan 19 05:22:12 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Tue, 19 Jan 2010 16:22:12 +0300
Subject: [Freeswitch-users] Which port I have to open?
In-Reply-To: <20100119102854.GQ4767@tamay-dogan.net>
Message-ID: <20100119132212.D20B011F53@mail.nstel.ru>

Did you take a look at
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Firewall?


> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Michelle Konzack
> Sent: Tuesday, January 19, 2010 1:29 PM
> To: FreeSWITCH Users
> Subject: [Freeswitch-users] Which port I have to open?
> 
> Hello *,
> 
> my FreeSWITCH was in a DMZ but now I have installed it  on  a  dedicated
> server (TI OMAP3530) and nothing as working... :-/
> 
> Please can someone tell me which ports I have to forward from my  router
> fo Freeswitch and reverse?
> 
> Thanks, Greetings and nice Day/Evening
>     Michelle Konzack
>     Systemadministrator
>     Electronic Engineer
>     Tamay Dogan Network
>     Debian GNU/Linux Consultant
> 
> --
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
> ##################### Debian GNU/Linux Consultant #####################
> <http://www.tamay-dogan.net/>                 Michelle Konzack
> <http://www.can4linux.org/>                   Apt. 917
> <http://www.flexray4linux.org/>               50, rue de Soultz
> Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
> IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
> ICQ    #328449886                             Tel. FR: +33  6  61925193




From stevendt at primrosebank.net  Tue Jan 19 05:31:42 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Tue, 19 Jan 2010 13:31:42 -0000
Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors
Message-ID: <2CCD94BEC92D4DE493B6FA65A554F286@bp1.ad.bp.com>

Hi,

I have just upgraded to the latest SVN (16400) and see a couple of errors when FS starts, although the basic functionality seems to be working.
Some DLLs have not been built, should I be worried about these errors and/or how do I correct them ?

2010-01-19 12:51:33.280718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_cluechoo.dll
**dll open error [126l]

2010-01-19 12:51:33.905718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_file_string.dll
**dll open error [126l]

2010-01-19 12:51:34.108843 [ERR] mod_spidermonkey.c:930 Error Loading module C:\FreeSWITCH\Debug\mod\mod_spidermonkey_core_db.dll
**dll open error [126l]


There is nothing in the Build log on the first two, but there is an error against the last one

88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.pdb'.
88>Make sure that the file is not open by another process and is not write-protected.
88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.idb'.
88>Make sure that the file is not open by another process and is not write-protected

regards
Dave

WindowsXP
VC++ Express 2008
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From linux4michelle at tamay-dogan.net  Tue Jan 19 05:59:20 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Tue, 19 Jan 2010 14:59:20 +0100
Subject: [Freeswitch-users] Which port I have to open?
In-Reply-To: <20100119132212.D20B011F53@mail.nstel.ru>
References: <20100119102854.GQ4767@tamay-dogan.net>
	<20100119132212.D20B011F53@mail.nstel.ru>
Message-ID: <20100119135919.GU4767@tamay-dogan.net>

Hello,

Am 2010-01-19 16:22:12, schrieb Nikolay Kondratyev:
> Did you take a look at
> http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Firewall?

Hmmm, -- do I have to forward incoming 16384-32768 ports to
my FreeSwitch box?  This would kick off my entired network.

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From rupa at rupa.com  Tue Jan 19 06:49:56 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Jan 2010 08:49:56 -0600
Subject: [Freeswitch-users] Which port I have to open?
In-Reply-To: <20100119135919.GU4767@tamay-dogan.net>
References: <20100119102854.GQ4767@tamay-dogan.net>
	<20100119132212.D20B011F53@mail.nstel.ru>
	<20100119135919.GU4767@tamay-dogan.net>
Message-ID: <b39273531001190649j4a510c6fu4cc6dcf3826c8d59@mail.gmail.com>

Does your router support NAT-PMP or UPNP?  If so, FS will do all the port
forwarding configuration automatically.

On Tue, Jan 19, 2010 at 7:59 AM, Michelle Konzack <
linux4michelle at tamay-dogan.net> wrote:

> Hello,
>
> Am 2010-01-19 16:22:12, schrieb Nikolay Kondratyev:
> > Did you take a look at
> > http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Firewall?
>
> Hmmm, -- do I have to forward incoming 16384-32768 ports to
> my FreeSwitch box?  This would kick off my entired network.
>
> Thanks, Greetings and nice Day/Evening
>    Michelle Konzack
>    Systemadministrator
>    Electronic Engineer
>    Tamay Dogan Network
>    Debian GNU/Linux Consultant
>
> --
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
> ##################### Debian GNU/Linux Consultant #####################
> <http://www.tamay-dogan.net/>                 Michelle Konzack
> <http://www.can4linux.org/>                   Apt. 917
> <http://www.flexray4linux.org/>               50, rue de Soultz
> Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
> IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
> ICQ    #328449886                             Tel. FR: +33  6  61925193
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From linux4michelle at tamay-dogan.net  Tue Jan 19 07:39:40 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Tue, 19 Jan 2010 16:39:40 +0100
Subject: [Freeswitch-users] Which port I have to open?
In-Reply-To: <b39273531001190649j4a510c6fu4cc6dcf3826c8d59@mail.gmail.com>
References: <20100119102854.GQ4767@tamay-dogan.net>
	<20100119132212.D20B011F53@mail.nstel.ru>
	<20100119135919.GU4767@tamay-dogan.net>
	<b39273531001190649j4a510c6fu4cc6dcf3826c8d59@mail.gmail.com>
Message-ID: <20100119153940.GW4767@tamay-dogan.net>

Hello,

Am 2010-01-19 08:49:56, schrieb Rupa Schomaker:
> Does your router support NAT-PMP or UPNP?  If so, FS will do all the port
> forwarding configuration automatically.

My router is the Freebox from my ISP <http://www.free.fr>  and  it  does
UPNP.  Also I have complete config datd to connect FreeSWITCH to my  SIP
account from <free.fr> and to my SIP-Povider <sipgate.de>

Since I have a bunch of problems with the Freebox, I  like  to  put  the
FreeBox into bridge mode and use a "Debian GNU/Linux Lenny/Squeeze"  box
to do the routing.

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From help at pdscc.com  Tue Jan 19 07:41:30 2010
From: help at pdscc.com (Harondel J. Sibble)
Date: Tue, 19 Jan 2010 07:41:30 -0800
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4 (fixed)
In-Reply-To: <8C20E0B1-0E56-4892-9D36-4D6DF3331244@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20100105065356.AEE0612F5@sinclaire.sibble.net>,
	<8C20E0B1-0E56-4892-9D36-4D6DF3331244@freeswitch.org>
Message-ID: <20100119154135.E51471DB501@sinclaire.sibble.net>

Okay, all I've done so far is make current from SVN and now have latest 
installed

2010-01-19 07:36:12.591283 [CONSOLE] switch_core.c:1565
FreeSWITCH Version 1.0.trunk (16400) Started.

Do I need to rebuild libzrtp for Freeswitch? Or just try it as is?


On 7 Jan 2010 at 17:30, Brian West wrote:

> Harondel,
> 	Please update your FreeSWITCH source rev 16204 fixes the SAS passing in all
> cases from my testing.  You'll need the latest ZRTP Lib and zfone
> application to make this work... I'm not too sure Tiviphone does this yet as
> I don't have one to test with.  This also fixes the issue when both sides
> are enrolled.  Next we will fix the video portion so both video and audio
> will go thru zrtp.
> 
> Please try it and let me know.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From jeff at jefflenk.com  Tue Jan 19 07:44:41 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 19 Jan 2010 09:44:41 -0600
Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors
In-Reply-To: <2CCD94BEC92D4DE493B6FA65A554F286@bp1.ad.bp.com>
References: <2CCD94BEC92D4DE493B6FA65A554F286@bp1.ad.bp.com>
Message-ID: <COL117-W6424736CC849832D6D9DBFB9650@phx.gbl>


Hi Dave,

 

mod_cluechoo is not built for Windows and is not needed.

The other files should build fine. Do they exist in your Debug folder?

 

- Jeff
 


From: stevendt at primrosebank.net
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 19 Jan 2010 13:31:42 +0000
Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors




Hi,
 
I have just upgraded to the latest SVN (16400) and see a couple of errors when FS starts, although the basic functionality seems to be working.
Some DLLs have not been built, should I be worried about these errors and/or how do I correct them ?
 
2010-01-19 12:51:33.280718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_cluechoo.dll
**dll open error [126l]

2010-01-19 12:51:33.905718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_file_string.dll
**dll open error [126l]

2010-01-19 12:51:34.108843 [ERR] mod_spidermonkey.c:930 Error Loading module C:\FreeSWITCH\Debug\mod\mod_spidermonkey_core_db.dll
**dll open error [126l]

 
There is nothing in the Build log on the first two, but there is an error against the last one
 
88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.pdb'.
88>Make sure that the file is not open by another process and is not write-protected.
88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.idb'.
88>Make sure that the file is not open by another process and is not write-protected
 
regards
Dave
 
WindowsXP
VC++ Express 2008 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with powerful SPAM protection.
http://clk.atdmt.com/GBL/go/196390707/direct/01/
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From john at acsol.net  Tue Jan 19 08:02:30 2010
From: john at acsol.net (John)
Date: Tue, 19 Jan 2010 09:02:30 -0700
Subject: [Freeswitch-users] Voicemail to email problems
Message-ID: <4B55D796.6070001@acsol.net>

Voicemail to email gets a broken pipe error. I believe this is happening 
while writing the file to /tmp? The mail.xxxxxxxxxxxx file is never 
written. Any ideas? Thanks





From anthony.minessale at gmail.com  Tue Jan 19 08:04:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Jan 2010 10:04:45 -0600
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <7d79b3931001190107o200ec04dredd76689dd235588@mail.gmail.com>
References: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
	<191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>
	<7d79b3931001132357i36bbb482jdf2bbdd3aea2a583@mail.gmail.com>
	<7d79b3931001180322p74a1227qe0c2199a77cbfe2@mail.gmail.com>
	<191c3a031001180824q6e364c72g3f789892597e9469@mail.gmail.com>
	<7d79b3931001190107o200ec04dredd76689dd235588@mail.gmail.com>
Message-ID: <191c3a031001190804m692f63acsaf852b8809db09d1@mail.gmail.com>

Its nothing we can fix, that is what you must do on a failed read syscall.
you can do a non blocking read instead and take your chances.


On Tue, Jan 19, 2010 at 3:07 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> I tried with SIGUSR1, but no progress. I got SERVER_DISCONNECTED.
> Output:
>
> CHILD 3814: Received USR1
> EVENT [SERVER_DISCONNECTED]
>
> In esl.c, in esl_recv_event() function, line no: 824
> if (rrval < 0) {
>                         strerror_r(handle->errnum, handle->err,
> sizeof(handle->err));
>                         goto fail;
> }
> When the program is blocked under receive, I passed the signal. So recv
> returns -1, and in fail: it call esl_disconnect(handle).
>
> Is it because of this??? If so, whether it should be fixed or not???
>
>
>
>
> On Mon, Jan 18, 2010 at 9:54 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> try a less famous signal like SIGUSR1 it's possible something in perl
>> still reacts to SIGINT
>>
>>
>>
>> On Mon, Jan 18, 2010 at 5:22 AM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>> Here is the result
>>>
>>> Program:
>>>
>>> require ESL;
>>> use IO::Socket::INET;
>>> use Data::Dumper;
>>>
>>> my $ip = "192.168.1.222";
>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>> die "Could not create socket: $!\n" unless $sock;
>>>
>>>
>>> for(;;) {
>>>     my $new_sock = $sock->accept();
>>>     next if (not defined ($new_sock));
>>>     my $pid = fork();
>>>     if ($pid) {
>>>         close($new_sock);
>>>         next;
>>>     }
>>>     print "CHILD PID: $$\n";
>>>     &register_Signals_Child();
>>>     sub register_Signals_Child() {
>>>         foreach ( keys %SIG ) {
>>>             $SIG{$_} = 'Handler';
>>>         }
>>>     }
>>>
>>>     sub Handler() {
>>>
>>>         my $handle=$_[0];
>>>         if($handle eq "INT") {
>>>             print "CHILD $$: SIGNAL SIG$handle is generated\n";
>>>         }
>>>         else
>>>         {
>>>             print "CHILD $$: Received $handle\n";
>>>
>>>         }
>>>     }
>>>     my $host = $new_sock->sockhost();
>>>     my $fd = fileno($new_sock);
>>>
>>>     my $con = new ESL::ESLconnection($fd);
>>>     my $info = $con->getInfo();
>>>
>>>     my $uuid = $info->getHeader("unique-id");
>>>
>>>     printf "Connected call %s, from %s\n", $uuid,
>>> $info->getHeader("caller-caller-id-number");
>>>     my $r=$con->execute("answer");
>>>     print Dumper $r;
>>>     $con->events("plain","all");
>>>     my
>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>     while($con->connected()) {
>>>         my $e = $con->recvEvent();
>>>
>>>         if ($e) {
>>>             my $name = $e->getHeader("event-name");
>>>             print "EVENT [$name]\n";
>>>             if ($name eq "DTMF") {
>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>                 print "$digit\n";
>>>             }
>>>         }
>>>     }
>>>     close($new_sock);
>>> }
>>>
>>> I executed the program and the following things were printed
>>>
>>> CHILD PID: 6778
>>> Connected call e0d1001a-03f4-11df-b002-db488337e0ea, from 1001
>>> $VAR1 = 0;
>>> EVENT [CHANNEL_EXECUTE]
>>> EVENT [CHANNEL_ANSWER]
>>> EVENT [CHANNEL_EXECUTE_COMPLETE]
>>> EVENT [COMMAND]
>>> EVENT [CHANNEL_EXECUTE]
>>> EVENT [HEARTBEAT]
>>> EVENT [RE_SCHEDULE]
>>> EVENT [CHANNEL_EXECUTE_COMPLETE]
>>>
>>> Then from another shell I executed kill -2 6778, the result is follows
>>> CHILD 6778: SIGNAL SIGINT is generated
>>> EVENT [SERVER_DISCONNECTED]
>>>
>>> But the child process is still running as expected.
>>> But  I don't know why I received SERVER_DISCONNECTED from freeswitch???
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Thu, Jan 14, 2010 at 1:27 PM, lakshmanan ganapathy <
>>> lakindia89 at gmail.com> wrote:
>>>
>>>> I taught the signal handler will be inherited by the child process. It
>>>> also does like that.
>>>> After making a call, If I press ctrl + c, the above program printed
>>>> PARENT PID: Signal SIGINT is generated
>>>> CHILD PID: Signal SIGINT is generated.
>>>>
>>>> So I think the sigal handlers will be inherited to the child.
>>>> Anyway I'll also try registering signal handlers in child also, and then
>>>> I'll come back with that result.
>>>>
>>>> Thanks....
>>>> On Wed, Jan 13, 2010 at 9:48 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> you would have to register signals in your child process too
>>>>>
>>>>>   On Wed, Jan 13, 2010 at 3:13 AM, lakshmanan ganapathy <
>>>>> lakindia89 at gmail.com> wrote:
>>>>>
>>>>>>  Hi all,
>>>>>>
>>>>>> I've done a sample program (In perl ESL) , which play a file to the
>>>>>> caller and then it will call recvEvent() and print the event name. I've
>>>>>> handled signals also.
>>>>>>
>>>>>> When I send SIGINT to my program (Perl), the signal handler is called
>>>>>> and I can see the print output. But in the same time,  I received
>>>>>> SERVER_DISCONNECTED from freeswitch as event.
>>>>>>
>>>>>> I don't know why I received SERVER_DISCONNECTED from freeswitch. Is it
>>>>>> because, the recvEvent() from perl internally  calls the recvevent function
>>>>>> in the Esl.c and when it waits to receive the information from socket, the
>>>>>> signal occurred???
>>>>>>
>>>>>> Please clarify me!!
>>>>>>
>>>>>> Here is my program
>>>>>> require ESL;
>>>>>> use IO::Socket::INET;
>>>>>> use Data::Dumper;
>>>>>>
>>>>>> my $ip = "192.168.1.222";
>>>>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>>>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>>>>> die "Could not create socket: $!\n" unless $sock;
>>>>>> &register_Signals();
>>>>>>
>>>>>> for(;;) {
>>>>>>     my $new_sock = $sock->accept();
>>>>>>     next if (not defined ($new_sock));
>>>>>>     my $pid = fork();
>>>>>>     if ($pid) {
>>>>>>         close($new_sock);
>>>>>>         next;
>>>>>>     }
>>>>>>     print "CHILD PID: $$\n";
>>>>>>     my $host = $new_sock->sockhost();
>>>>>>     my $fd = fileno($new_sock);
>>>>>>
>>>>>>     my $con = new ESL::ESLconnection($fd);
>>>>>>     my $info = $con->getInfo();
>>>>>>
>>>>>>     my $uuid = $info->getHeader("unique-id");
>>>>>>
>>>>>>     printf "Connected call %s, from %s\n", $uuid,
>>>>>> $info->getHeader("caller-caller-id-number");
>>>>>>     my $r=$con->execute("answer");
>>>>>>     print Dumper $r;
>>>>>>     $con->events("plain","all");
>>>>>>     my
>>>>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>>>>
>>>>>>     while($con->connected()) {
>>>>>>         my $e = $con->recvEvent();
>>>>>>
>>>>>>         if ($e) {
>>>>>>             my $name = $e->getHeader("event-name");
>>>>>>             print "EVENT [$name]\n";
>>>>>>             if ($name eq "DTMF") {
>>>>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>>>>                 print "$digit\n";
>>>>>>             }
>>>>>>         }
>>>>>>     }
>>>>>>     close($new_sock);
>>>>>> }
>>>>>> sub register_Signals() {
>>>>>>     foreach ( keys %SIG ) {
>>>>>>         $SIG{$_} = 'sig_Handler';
>>>>>>     }
>>>>>> }
>>>>>>
>>>>>> sub sig_Handler() {
>>>>>>     my $handle=$_[0];
>>>>>>     if($handle eq "INT") {
>>>>>>         print "$$: SIGNAL SIG$handle is generated\n";
>>>>>>     }
>>>>>> }
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From stevendt at primrosebank.net  Tue Jan 19 08:13:44 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Tue, 19 Jan 2010 16:13:44 -0000
Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors
References: <2CCD94BEC92D4DE493B6FA65A554F286@bp1.ad.bp.com>
	<COL117-W6424736CC849832D6D9DBFB9650@phx.gbl>
Message-ID: <E5A67EA005F34E68B5501C860A9B96BC@bp1.ad.bp.com>

Hi Jeff,

the DLLs are not in the Debug folders pointed to by the error messages ?

regards
Dave

  ----- Original Message ----- 
  From: Jeff Lenk 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, January 19, 2010 3:44 PM
  Subject: Re: [Freeswitch-users] SVN 16400 - Windows Build Errors


  Hi Dave,
   
  mod_cluechoo is not built for Windows and is not needed.
  The other files should build fine. Do they exist in your Debug folder?
   
  - Jeff
   

------------------------------------------------------------------------------
  From: stevendt at primrosebank.net
  To: freeswitch-users at lists.freeswitch.org
  Date: Tue, 19 Jan 2010 13:31:42 +0000
  Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors


  Hi,

  I have just upgraded to the latest SVN (16400) and see a couple of errors when FS starts, although the basic functionality seems to be working.
  Some DLLs have not been built, should I be worried about these errors and/or how do I correct them ?

  2010-01-19 12:51:33.280718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_cluechoo.dll
  **dll open error [126l]

  2010-01-19 12:51:33.905718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_file_string.dll
  **dll open error [126l]

  2010-01-19 12:51:34.108843 [ERR] mod_spidermonkey.c:930 Error Loading module C:\FreeSWITCH\Debug\mod\mod_spidermonkey_core_db.dll
  **dll open error [126l]


  There is nothing in the Build log on the first two, but there is an error against the last one

  88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.pdb'.
  88>Make sure that the file is not open by another process and is not write-protected.
  88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.idb'.
  88>Make sure that the file is not open by another process and is not write-protected

  regards
  Dave

  WindowsXP
  VC++ Express 2008


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From devel at thom.fr.eu.org  Tue Jan 19 08:31:45 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Tue, 19 Jan 2010 17:31:45 +0100
Subject: [Freeswitch-users] Voicemail to email problems
Message-ID: <ab161f4749d2cf203ec197497f2ce22e@thom.fr.eu.org>

This is not the error I get.

I modified my config so that the command executed to send the email is cat
/tmp/mail.xxxxxxx | /usr/local/freeswitch/email.sh
with email.sh teeing the stdin to a file and to sendmail.

syslog reports sendmail segfault.

Fran?ois

On Tue, 19 Jan 2010 09:02:30 -0700, John <john at acsol.net> wrote:
> Voicemail to email gets a broken pipe error. I believe this is happening

> while writing the file to /tmp? The mail.xxxxxxxxxxxx file is never 
> written. Any ideas? Thanks
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From a.alalousi at gmail.com  Tue Jan 19 08:31:45 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Tue, 19 Jan 2010 16:31:45 +0000
Subject: [Freeswitch-users] G729 coded issues
Message-ID: <cd93b3701001190831u4147214cn9832c6f469c6d17e@mail.gmail.com>

Hi everyone,

I have the following scenario and a major customer-affecting issue thereof.

Here is the scenario: customer traffic encoded as G.729 from a cisco gateway
-> our FS (G729 passthrough) -> remote end gw (G729)

Calls were failing at an alarming rate, so I looked at the debug logs. It
transpired that the Cisco is offering G729 annex b, while the remote end can
only do G729a.

Besides changing source or destination preferences, is there a way to ensure
that G729a is used end-end ?

Thanks in advance.



-- 
Ahmed Naji
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From brian at freeswitch.org  Tue Jan 19 08:36:32 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Jan 2010 10:36:32 -0600
Subject: [Freeswitch-users] G729 coded issues
In-Reply-To: <cd93b3701001190831u4147214cn9832c6f469c6d17e@mail.gmail.com>
References: <cd93b3701001190831u4147214cn9832c6f469c6d17e@mail.gmail.com>
Message-ID: <6D67C7A9-42CD-4F5F-AD18-E9568098FB71@freeswitch.org>

g729a is 100% INVALID in the sdp fix the param in your cisco SPA or your Linksys SPA phone and it will stop doing that.  Hopefully they'll fix this "bug" soon in the cisco phones to not include the a in the sdp.  The fmtp is the proper way to specify annex a or any other options for g729.

/b

On Jan 19, 2010, at 10:31 AM, Ahmed Naji wrote:

> Hi everyone,
> 
> I have the following scenario and a major customer-affecting issue thereof.
> 
> Here is the scenario: customer traffic encoded as G.729 from a cisco gateway
> -> our FS (G729 passthrough) -> remote end gw (G729)
> 
> Calls were failing at an alarming rate, so I looked at the debug logs. It
> transpired that the Cisco is offering G729 annex b, while the remote end can
> only do G729a.
> 
> Besides changing source or destination preferences, is there a way to ensure
> that G729a is used end-end ?
> 
> Thanks in advance.




From mailinglist at fribert.dk  Tue Jan 19 08:56:01 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Tue, 19 Jan 2010 17:56:01 +0100
Subject: [Freeswitch-users] Svar: Re: How do I invite group to join	
	existing call?
In-Reply-To: <87f2f3b91001181209y7a0aa68fs8a580712484c7a11@mail.gmail.com>
References: <4B5412C6020000E1000003D6@mail.fribert.dk>
	<87f2f3b91001181209y7a0aa68fs8a580712484c7a11@mail.gmail.com>
Message-ID: <4B55F231020000E1000003DB@mail.fribert.dk>

GRIN, ok, I'll see if I can punder a bit more on the subject.
 
I have a home setup, I have a couple of phones attached, as well as a couple of computers.
The private line sends a call to the private group.
The phones attached to the private group rings
My wife picks it up
She wants me to join the conversation, so she presses *11 or something :-)
Now the phones in the private groups rings again, and I pick it up, and we have a conference with a calling party, and two local phones.
 
Any ideas from you guru's?
 
BR
Fribse

>>> 18-01-2010 kl. 21:09 skrev Michael Collins <msc at freeswitch.org> i meddelelsen <87f2f3b91001181209y7a0aa68fs8a580712484c7a11 at mail.gmail.com>:



On Sun, Jan 17, 2010 at 10:50 PM, mailinglist <mailinglist at fribert.dk> wrote:


Hi All

I would like to be able to invite a group / global to join an existing call, but how do I accomplish this, can it be done?


Malfunction! Need Input!

Could you give us a few more details on what you're trying to accomplish? What's the big picture? I'm curious what problem you're trying to solve. I'm sure the gang here will have thoughts to pass along. :)

-MC

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From lists at redbonez.net  Tue Jan 19 08:59:37 2010
From: lists at redbonez.net (Adam Ford)
Date: Tue, 19 Jan 2010 09:59:37 -0700
Subject: [Freeswitch-users] Polycom configuration for FreeSWITCH
In-Reply-To: <d361ca7b1001190226y70f6dfa2o3cbf4bcffff6a887@mail.gmail.com>
References: <003701ca964b$3241b100$96c51300$@net>
	<d361ca7b1001190226y70f6dfa2o3cbf4bcffff6a887@mail.gmail.com>
Message-ID: <015901ca9928$c98e2cc0$5caa8640$@net>

Thanks Yehavi!

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yehavi
Bourvine
Sent: Tuesday, January 19, 2010 3:27 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom configuration for FreeSWITCH

 

I've documented my Polycom's setup in the wiki under "Polycom
configuration".

 

                  Regards, __Yehavi:


 

2010/1/16 Adam Ford <lists at redbonez.net>

Does anyone who has successfully implemented Polycom IP301/501s, with
FreeSWITCH, have a configuration they can share with me as an example? I
have read the Polycom Admin Guide several times and understand what the
settings are/do, I am just not sure which FreeSWITCH supports, which it
doesn't, and which need special configuration to work with FreeSWITCH.

 

Thanks in advance,

-Adam

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Tue Jan 19 09:01:08 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Jan 2010 11:01:08 -0600
Subject: [Freeswitch-users] Svar: Re: How do I invite group to join
	existing call?
In-Reply-To: <4B55F231020000E1000003DB@mail.fribert.dk>
References: <4B5412C6020000E1000003D6@mail.fribert.dk>
	<87f2f3b91001181209y7a0aa68fs8a580712484c7a11@mail.gmail.com>
	<4B55F231020000E1000003DB@mail.fribert.dk>
Message-ID: <3D153366-FE85-4A7A-9351-63D5FBE3F287@freeswitch.org>

What kind of phones?

/b

On Jan 19, 2010, at 10:56 AM, mailinglist wrote:

> GRIN, ok, I'll see if I can punder a bit more on the subject.
>  
> I have a home setup, I have a couple of phones attached, as well as a couple of computers.
> The private line sends a call to the private group.
> The phones attached to the private group rings
> My wife picks it up
> She wants me to join the conversation, so she presses *11 or something :-)
> Now the phones in the private groups rings again, and I pick it up, and we have a conference with a calling party, and two local phones.
>  
> Any ideas from you guru's?
>  
> BR
> Fribse

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From mailinglist at fribert.dk  Tue Jan 19 09:02:08 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Tue, 19 Jan 2010 18:02:08 +0100
Subject: [Freeswitch-users] Home setup with home company
Message-ID: <4B55F3A0020000E1000003E0@mail.fribert.dk>

I have a very small one man constultancy company that has the occasional call, unfortunately we are getting more spam calls after hours than real calls during work hours, so I would like to set up a TOD system.
 
First step for me is just playing with the TOD example, I've gotten this far:
 
<extension name="CompanyTOD" continue="true">
   <condition field="destination_number" expression="^(8203)$"/>
      <!-- man strftime - M-F, 9AM to 5PM -->
      <condition field="${strftime(%w)}" expression="^([1-5])$"/>
      <condition field="${strftime(%H%M)}" expression="^((09|1[0-6])[0-5][0-9]|1700)$">
         <action application="set" data="call_timeout=45"/>
         <action application="set" data="domain_name=10.11.12.25"/>
         <action application="set" data="continue_on_fail=true"/>
         <action application="set" data="hangup_after_bridge=true"/>
         <action application="bridge" data="group/company@${domain_name}"/ ( 'mailto:group/company@$domain_name"/' )>
         <action application="answer" data=""/>
         <action application="sleep" data="1000"/>
         <action application="voicemail" data="default 10.11.12.25 10199"/>
         <anti-action application="voicemail" data="default 10.11.12.25 10199"/>
      </condition>
</extension>
My idea with this, was that in the time 9-17 mon-fri, it rings, the rest of the time it's sent directly to the voicemail.
I would of course like to have it not take messages outside work hours, but that's just refining :-)
 
But it picks up the call, and then nothing...
 
BR
Fribse
 
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From steveu at coppice.org  Tue Jan 19 09:10:44 2010
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 20 Jan 2010 01:10:44 +0800
Subject: [Freeswitch-users] G729 coded issues
In-Reply-To: <6D67C7A9-42CD-4F5F-AD18-E9568098FB71@freeswitch.org>
References: <cd93b3701001190831u4147214cn9832c6f469c6d17e@mail.gmail.com>
	<6D67C7A9-42CD-4F5F-AD18-E9568098FB71@freeswitch.org>
Message-ID: <4B55E794.6020909@coppice.org>

On 01/20/2010 12:36 AM, Brian West wrote:
> g729a is 100% INVALID in the sdp fix the param in your cisco SPA or your Linksys SPA phone and it will stop doing that.  Hopefully they'll fix this "bug" soon in the cisco phones to not include the a in the sdp.  The fmtp is the proper way to specify annex a or any other options for g729.
>    
Annex A only affects the inner workings of the codec. There is 
absolutely no difference whatsoever between G.729 and G.729A on the 
wire. The SDP has no reason to mention it, and the standards say it 
shouldn't.
> /b
>
> On Jan 19, 2010, at 10:31 AM, Ahmed Naji wrote:
>
>    
>> Hi everyone,
>>
>> I have the following scenario and a major customer-affecting issue thereof.
>>
>> Here is the scenario: customer traffic encoded as G.729 from a cisco gateway
>> ->  our FS (G729 passthrough) ->  remote end gw (G729)
>>
>> Calls were failing at an alarming rate, so I looked at the debug logs. It
>> transpired that the Cisco is offering G729 annex b, while the remote end can
>> only do G729a.
>>
>> Besides changing source or destination preferences, is there a way to ensure
>> that G729a is used end-end ?
>>
>> Thanks in advance.
>>      
>
Steve




From earlpinkerton at gmail.com  Tue Jan 19 00:33:04 2010
From: earlpinkerton at gmail.com (Earl Pinkerton)
Date: Tue, 19 Jan 2010 00:33:04 -0800
Subject: [Freeswitch-users] IVR returning info via event socket
Message-ID: <432c6c2b1001190033x2d4e07e7i985ef24bd6f52c7c@mail.gmail.com>

Hi All,

I am trying to call an IVR remotely over the event socket port.  I am
testing using telnet.  I call originate (similar to the following):

api originate
{origination_caller_id_number=8885551111,ignore_early_media=true}sofia/gateway/teliax/18885552222
&ivr(demo_ivr)

This works fine (places the call, runs the IVR, accepts DTMF tones to move
between menus).

My problem is that I would like the remote app to send a PIN as part of the
originate call, then have the callie enter the PIN via DTMF and indicate
whether there was a match or not back to the calling app (which is a remote
web program).  We are trying to keep it simple and have the remote app just
call the IVR and get the result, without having to control the IVR through
asynchronous events.

Is there any way for the IVR to check for a match, then somehow send info
back over the port to the calling app?

By the way, I tried using menu-exec-api inside ivr.conf.xml (to see if I
might be able to use the echo api call or something similar), but I got the
following error:

  2010-01-19 01:22:55.796658 [WARNING] switch_ivr_menu.c:704 Invalid Action
[menu-exec-api]

Thanks.
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From ahmed.ajmal at gmail.com  Tue Jan 19 07:52:22 2010
From: ahmed.ajmal at gmail.com (Ahmed Bhaila)
Date: Tue, 19 Jan 2010 20:52:22 +0500
Subject: [Freeswitch-users] Help with CDR fields:Other-Leg-Network-Addr and
	Other-Leg-Destination-Number
Message-ID: <9d22cc171001190752j797fdf39radc404510128718c@mail.gmail.com>

Hello

I am using a template for Master.csv and having problems with getting values
for the following fields: Other-Leg-Network-Addr and
Other-Leg-Destination-Number. All I am doing is just enclosing these fields
in ${} in my template definition and they always turn out to be empty but
they arent supposed to be as I am seeing values in the freeswitch console.so
I am not sure what is wrong here I have noticed that some of the channel
variables(start_stamp, end_stamp) have "variable_" prepended to them when I
see them in the console so when defining the template I omit the variable_
part and can get the values. Is this something similar? It seems like the
problem is with all field names that start with Other. Any help be greatly
appreciated.


Thanks
AB
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From brian at freeswitch.org  Tue Jan 19 09:36:36 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Jan 2010 11:36:36 -0600
Subject: [Freeswitch-users] Changes for deb and rpm files.
Message-ID: <5CA0D17B-6BD4-4BC6-B7F9-4A117A896C86@freeswitch.org>

Heads up.

http://jira.freeswitch.org/browse/FSCONFIG-17

Someone will need to update the configs for the debs and rpm's  

/b




From mailinglist at fribert.dk  Tue Jan 19 09:46:00 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Tue, 19 Jan 2010 18:46:00 +0100
Subject: [Freeswitch-users] Svar: Re: How do I invite group to join	
	existing call?
In-Reply-To: <3D153366-FE85-4A7A-9351-63D5FBE3F287@freeswitch.org>
References: <4B5412C6020000E1000003D6@mail.fribert.dk>
	<87f2f3b91001181209y7a0aa68fs8a580712484c7a11@mail.gmail.com>
	<4B55F231020000E1000003DB@mail.fribert.dk>
	<3D153366-FE85-4A7A-9351-63D5FBE3F287@freeswitch.org>
Message-ID: <4B55FDE8020000E1000003E5@mail.fribert.dk>

Hi B
 
Well, three are on siemens gigaset, then there's a linksys spa 901, then some computers with x-lite.
The gigaset can have two SIP calls established at the same time, so if she picks up a gigaset or the spa 901, shouldn't make any difference.
 
best regards
Fribse

>>> 19-01-2010 kl. 18:01 skrev Brian West <brian at freeswitch.org> i meddelelsen <3D153366-FE85-4A7A-9351-63D5FBE3F287 at freeswitch.org>:

What kind of phones?

/b

On Jan 19, 2010, at 10:56 AM, mailinglist wrote:




GRIN, ok, I'll see if I can punder a bit more on the subject.

 
I have a home setup, I have a couple of phones attached, as well as a couple of computers.
The private line sends a call to the private group.
The phones attached to the private group rings
My wife picks it up
She wants me to join the conversation, so she presses *11 or something :-)
Now the phones in the private groups rings again, and I pick it up, and we have a conference with a calling party, and two local phones.
 
Any ideas from you guru's?
 
BR
Fribse


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From lists at infosecurity.ch  Tue Jan 19 09:58:20 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Tue, 19 Jan 2010 18:58:20 +0100
Subject: [Freeswitch-users] Is possible from dialplan to detect if called
	party is online?
Message-ID: <A707BD2C-EC69-4379-9A71-4F3666221D09@infosecurity.ch>

There some condition that let to check if the called party is  
currently registered and online, before giving the called a return code?

We need to manage a condition where the called number is an iPhone and  
is usually "always offline" and we need to wakeup the voip client with  
a "push notifcation".

But before doing the push notification we should check if the user is  
already registered/online or not.

Don't know which path to follow

Fabio



From stevendt at primrosebank.net  Tue Jan 19 10:07:16 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Tue, 19 Jan 2010 18:07:16 -0000
Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors
References: <2CCD94BEC92D4DE493B6FA65A554F286@bp1.ad.bp.com><COL117-W6424736CC849832D6D9DBFB9650@phx.gbl>
	<E5A67EA005F34E68B5501C860A9B96BC@bp1.ad.bp.com>
Message-ID: <549C9A85D40D466B900D0415E7998B78@bp1.ad.bp.com>

Jeff,

just looked at the Tortoise SVN log for the 2008.Express.sln file, the last change was on the 17th with the comment . . . .

"move mod_spidermoney build to automake, fix spidermoneky dependencies (I think this really fixes -j builds), move mod_spidermonkey sub modules all under the same source directory and bundle their build together as one" (mikej). Perhaps a "gremlin" has slipped in there for the Spidemonkey build ?

(The only thing that I changed from the default configuration was to download and build the 16k and 32k music and sounds )

regards
Dave

  ----- Original Message ----- 
  From: Dave Stevenson 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, January 19, 2010 4:13 PM
  Subject: Re: [Freeswitch-users] SVN 16400 - Windows Build Errors


  Hi Jeff,

  the DLLs are not in the Debug folders pointed to by the error messages ?

  regards
  Dave

    ----- Original Message ----- 
    From: Jeff Lenk 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Tuesday, January 19, 2010 3:44 PM
    Subject: Re: [Freeswitch-users] SVN 16400 - Windows Build Errors


    Hi Dave,
     
    mod_cluechoo is not built for Windows and is not needed.
    The other files should build fine. Do they exist in your Debug folder?
     
    - Jeff
     

----------------------------------------------------------------------------
    From: stevendt at primrosebank.net
    To: freeswitch-users at lists.freeswitch.org
    Date: Tue, 19 Jan 2010 13:31:42 +0000
    Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors


    Hi,

    I have just upgraded to the latest SVN (16400) and see a couple of errors when FS starts, although the basic functionality seems to be working.
    Some DLLs have not been built, should I be worried about these errors and/or how do I correct them ?

    2010-01-19 12:51:33.280718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_cluechoo.dll
    **dll open error [126l]

    2010-01-19 12:51:33.905718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_file_string.dll
    **dll open error [126l]

    2010-01-19 12:51:34.108843 [ERR] mod_spidermonkey.c:930 Error Loading module C:\FreeSWITCH\Debug\mod\mod_spidermonkey_core_db.dll
    **dll open error [126l]


    There is nothing in the Build log on the first two, but there is an error against the last one

    88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.pdb'.
    88>Make sure that the file is not open by another process and is not write-protected.
    88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.idb'.
    88>Make sure that the file is not open by another process and is not write-protected

    regards
    Dave

    WindowsXP
    VC++ Express 2008


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From anthony.minessale at gmail.com  Tue Jan 19 11:37:58 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Jan 2010 13:37:58 -0600
Subject: [Freeswitch-users] Help with CDR fields:Other-Leg-Network-Addr
	and Other-Leg-Destination-Number
In-Reply-To: <9d22cc171001190752j797fdf39radc404510128718c@mail.gmail.com>
References: <9d22cc171001190752j797fdf39radc404510128718c@mail.gmail.com>
Message-ID: <191c3a031001191137k72530079t402ff54d03b07756@mail.gmail.com>

those are not variables only variables expand.

try bleg_network_addr
and bleg_destination_number


On Tue, Jan 19, 2010 at 9:52 AM, Ahmed Bhaila <ahmed.ajmal at gmail.com> wrote:

> Hello
>
> I am using a template for Master.csv and having problems with getting
> values for the following fields: Other-Leg-Network-Addr and
> Other-Leg-Destination-Number. All I am doing is just enclosing these fields
> in ${} in my template definition and they always turn out to be empty but
> they arent supposed to be as I am seeing values in the freeswitch console.so
> I am not sure what is wrong here I have noticed that some of the channel
> variables(start_stamp, end_stamp) have "variable_" prepended to them when I
> see them in the console so when defining the template I omit the variable_
> part and can get the values. Is this something similar? It seems like the
> problem is with all field names that start with Other. Any help be greatly
> appreciated.
>
>
> Thanks
> AB
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From john at acsol.net  Tue Jan 19 11:38:46 2010
From: john at acsol.net (John)
Date: Tue, 19 Jan 2010 12:38:46 -0700
Subject: [Freeswitch-users] Voicemail to email problems
In-Reply-To: <ab161f4749d2cf203ec197497f2ce22e@thom.fr.eu.org>
References: <ab161f4749d2cf203ec197497f2ce22e@thom.fr.eu.org>
Message-ID: <4B560A46.90000@acsol.net>

I am getting a segfault as well; however I use exim4 on Debian. Email is 
never being written to /tmp. I too can run the exim command by hand and 
it works. Any body have an idea or direction to look at?
Here is the error.
/bin/cat: write error: Broken pipe
sh: line 1:  7102 Done(1)                 /bin/cat /tmp/mail.12639191242a9f
       7103 Segmentation fault

Thanks



On 1/19/2010 9:31 AM, Fran?ois Legal wrote:
> This is not the error I get.
>
> I modified my config so that the command executed to send the email is cat
> /tmp/mail.xxxxxxx | /usr/local/freeswitch/email.sh
> with email.sh teeing the stdin to a file and to sendmail.
>
> syslog reports sendmail segfault.
>
> Fran?ois
>
> On Tue, 19 Jan 2010 09:02:30 -0700, John<john at acsol.net>  wrote:
>    
>> Voicemail to email gets a broken pipe error. I believe this is happening
>>      
>    
>> while writing the file to /tmp? The mail.xxxxxxxxxxxx file is never
>> written. Any ideas? Thanks
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>      
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    




From anthony.minessale at gmail.com  Tue Jan 19 11:43:12 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Jan 2010 13:43:12 -0600
Subject: [Freeswitch-users] Is possible from dialplan to detect if
	called party is online?
In-Reply-To: <A707BD2C-EC69-4379-9A71-4F3666221D09@infosecurity.ch>
References: <A707BD2C-EC69-4379-9A71-4F3666221D09@infosecurity.ch>
Message-ID: <191c3a031001191143w6220e6a4va3f71537133cd798@mail.gmail.com>

sofia_contact api function returns the contact addr of a registered user so
if it's blank they are not registered.




On Tue, Jan 19, 2010 at 11:58 AM, Fabio Pietrosanti (naif) <
lists at infosecurity.ch> wrote:

> There some condition that let to check if the called party is
> currently registered and online, before giving the called a return code?
>
> We need to manage a condition where the called number is an iPhone and
> is usually "always offline" and we need to wakeup the voip client with
> a "push notifcation".
>
> But before doing the push notification we should check if the user is
> already registered/online or not.
>
> Don't know which path to follow
>
> Fabio
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Tue Jan 19 11:46:04 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Jan 2010 13:46:04 -0600
Subject: [Freeswitch-users] Voicemail to email problems
In-Reply-To: <4B560A46.90000@acsol.net>
References: <ab161f4749d2cf203ec197497f2ce22e@thom.fr.eu.org>
	<4B560A46.90000@acsol.net>
Message-ID: <191c3a031001191146r257d34afq98f921c0ff07f2fc@mail.gmail.com>

sounds like exim pretending to be sendmail and not doing it very well.
I think there is a wiki page somewhere that tells you how to config it
properly.


On Tue, Jan 19, 2010 at 1:38 PM, John <john at acsol.net> wrote:

> I am getting a segfault as well; however I use exim4 on Debian. Email is
> never being written to /tmp. I too can run the exim command by hand and
> it works. Any body have an idea or direction to look at?
> Here is the error.
> /bin/cat: write error: Broken pipe
> sh: line 1:  7102 Done(1)                 /bin/cat /tmp/mail.12639191242a9f
>       7103 Segmentation fault
>
> Thanks
>
>
>
> On 1/19/2010 9:31 AM, Fran?ois Legal wrote:
> > This is not the error I get.
> >
> > I modified my config so that the command executed to send the email is
> cat
> > /tmp/mail.xxxxxxx | /usr/local/freeswitch/email.sh
> > with email.sh teeing the stdin to a file and to sendmail.
> >
> > syslog reports sendmail segfault.
> >
> > Fran?ois
> >
> > On Tue, 19 Jan 2010 09:02:30 -0700, John<john at acsol.net>  wrote:
> >
> >> Voicemail to email gets a broken pipe error. I believe this is happening
> >>
> >
> >> while writing the file to /tmp? The mail.xxxxxxxxxxxx file is never
> >> written. Any ideas? Thanks
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From help at pdscc.com  Tue Jan 19 11:50:40 2010
From: help at pdscc.com (Harondel J. Sibble)
Date: Tue, 19 Jan 2010 11:50:40 -0800
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4 (fixed)
In-Reply-To: <20100119154135.E51471DB501@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<8C20E0B1-0E56-4892-9D36-4D6DF3331244@freeswitch.org>,
	<20100119154135.E51471DB501@sinclaire.sibble.net>
Message-ID: <20100119195043.18DF01DB501@sinclaire.sibble.net>

Brian

Okay, so far so good, I'm getting consistent SAS's between devices, I'll test 
some more and report back.

I didn't rebuild libzrtp in this case.

On 19 Jan 2010 at 7:41, Harondel J. Sibble wrote:

> Okay, all I've done so far is make current from SVN and now have latest 
> installed
> 
> 2010-01-19 07:36:12.591283 [CONSOLE] switch_core.c:1565
> FreeSWITCH Version 1.0.trunk (16400) Started.
> 
> Do I need to rebuild libzrtp for Freeswitch? Or just try it as is?

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From brian at freeswitch.org  Tue Jan 19 12:14:18 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Jan 2010 14:14:18 -0600
Subject: [Freeswitch-users] Voicemail to email problems
In-Reply-To: <191c3a031001191146r257d34afq98f921c0ff07f2fc@mail.gmail.com>
References: <ab161f4749d2cf203ec197497f2ce22e@thom.fr.eu.org>
	<4B560A46.90000@acsol.net>
	<191c3a031001191146r257d34afq98f921c0ff07f2fc@mail.gmail.com>
Message-ID: <4E823F6B-75DC-44D9-BD18-C7773D77EAC9@freeswitch.org>

http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings

/b

On Jan 19, 2010, at 1:46 PM, Anthony Minessale wrote:

> sounds like exim pretending to be sendmail and not doing it very well.
> I think there is a wiki page somewhere that tells you how to config it properly.
> 
> 
> On Tue, Jan 19, 2010 at 1:38 PM, John <john at acsol.net> wrote:
> I am getting a segfault as well; however I use exim4 on Debian. Email is
> never being written to /tmp. I too can run the exim command by hand and
> it works. Any body have an idea or direction to look at?
> Here is the error.
> /bin/cat: write error: Broken pipe
> sh: line 1:  7102 Done(1)                 /bin/cat /tmp/mail.12639191242a9f
>       7103 Segmentation fault
> 
> Thanks
> 
> 
> 
> On 1/19/2010 9:31 AM, Fran?ois Legal wrote:
> > This is not the error I get.
> >
> > I modified my config so that the command executed to send the email is cat
> > /tmp/mail.xxxxxxx | /usr/local/freeswitch/email.sh
> > with email.sh teeing the stdin to a file and to sendmail.
> >
> > syslog reports sendmail segfault.
> >
> > Fran?ois
> >
> > On Tue, 19 Jan 2010 09:02:30 -0700, John<john at acsol.net>  wrote:
> >
> >> Voicemail to email gets a broken pipe error. I believe this is happening
> >>
> >
> >> while writing the file to /tmp? The mail.xxxxxxxxxxxx file is never
> >> written. Any ideas? Thanks
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jeff at jefflenk.com  Tue Jan 19 12:26:33 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 19 Jan 2010 14:26:33 -0600
Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors
In-Reply-To: <E5A67EA005F34E68B5501C860A9B96BC@bp1.ad.bp.com>
References: <2CCD94BEC92D4DE493B6FA65A554F286@bp1.ad.bp.com>,
	<COL117-W6424736CC849832D6D9DBFB9650@phx.gbl>,
	<E5A67EA005F34E68B5501C860A9B96BC@bp1.ad.bp.com>
Message-ID: <COL117-W15B08477F1FFA512F0F0B4B9650@phx.gbl>


Do a rebuild all from the solution and you should not see any errors.

 


From: stevendt at primrosebank.net
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 19 Jan 2010 16:13:44 +0000
Subject: Re: [Freeswitch-users] SVN 16400 - Windows Build Errors




Hi Jeff,
 
the DLLs are not in the Debug folders pointed to by the error messages ?
 
regards
Dave
 

----- Original Message ----- 
From: Jeff Lenk 
To: freeswitch-users at lists.freeswitch.org 
Sent: Tuesday, January 19, 2010 3:44 PM
Subject: Re: [Freeswitch-users] SVN 16400 - Windows Build Errors

Hi Dave,
 
mod_cluechoo is not built for Windows and is not needed.
The other files should build fine. Do they exist in your Debug folder?
 
- Jeff
 


From: stevendt at primrosebank.net
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 19 Jan 2010 13:31:42 +0000
Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors




Hi,
 
I have just upgraded to the latest SVN (16400) and see a couple of errors when FS starts, although the basic functionality seems to be working.
Some DLLs have not been built, should I be worried about these errors and/or how do I correct them ?
 
2010-01-19 12:51:33.280718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_cluechoo.dll
**dll open error [126l]

2010-01-19 12:51:33.905718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_file_string.dll
**dll open error [126l]

2010-01-19 12:51:34.108843 [ERR] mod_spidermonkey.c:930 Error Loading module C:\FreeSWITCH\Debug\mod\mod_spidermonkey_core_db.dll
**dll open error [126l]

 
There is nothing in the Build log on the first two, but there is an error against the last one
 
88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.pdb'.
88>Make sure that the file is not open by another process and is not write-protected.
88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.idb'.
88>Make sure that the file is not open by another process and is not write-protected
 
regards
Dave
 
WindowsXP
VC++ Express 2008


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From jerry.richards at teotech.com  Tue Jan 19 07:50:12 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 19 Jan 2010 07:50:12 -0800
Subject: [Freeswitch-users] Bypass Media True Disables MOH
In-Reply-To: <9BF355CF-C633-4BF5-BB8B-642DD81936D1@freeswitch.org>
References: <AAA8629BCAF448659E9B91C20E75CCC6@greyhawk.tonecommander.com>
	<9BF355CF-C633-4BF5-BB8B-642DD81936D1@freeswitch.org>
Message-ID: <E55CF9FDA2AC406B8799A0081BCBA4DD@greyhawk.tonecommander.com>


We are willing to pay a bounty for this.  What amount would you suggest?  We
would like the media to normally go directly between the endpoints, but when
a call is put on-hold, we would like the other end should hear MOH.

Thanks,
Jerry
  

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Tuesday, December 29, 2009 1:12 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bypass Media True Disables MOH

<param name="media-option" value="resume-media-on-hold"/>

But it doesn't go back to bypass after....   Maybe you can post a bounty for
that functionality.

/b

On Dec 29, 2009, at 2:42 PM, Jerry Richards wrote:

> 
> When I uncomment the following tag, internally held calls no longer 
> hear MOH.
> 
> <param name="inbound-bypass-media" value="true"/>
> 
> Is there a way to have the above uncommented and still provide MOH to 
> held calls?
> 
> Best Regards,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org







From freeswitch-users at digitaldan.com  Tue Jan 19 13:03:05 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Tue, 19 Jan 2010 14:03:05 -0700 (MST)
Subject: [Freeswitch-users] Lua: io.popen/read blocking in other incoming
	calls
In-Reply-To: <13515873.38.1263934963058.JavaMail.root@zimbra>
Message-ID: <19409906.41.1263934985398.JavaMail.root@zimbra>

My lua script is calling wget through lua's io.popen to send and receive data from a web service. While the f:read to wget is running, other incoming calls will block on the same io.popen call until the first call closes the pipe (with f:close()). 


I had assumed every incoming call was on its own thread and that each had its own lua instance. Is there a global lock happening here? Below is the runCommand call and the two start and stop methods that are getting called in my script when the call begins and ends (notice they even talk to different hosts, so its not the web server hanging). I have put in debugging statements and its definitely hanging trying to call io.popen and not on the f:read. I noticed on the mod_python page (http://wiki.freeswitch.org/wiki/Mod_python#Known_Bugs) that it had read issues that were fixed in 12958 , could this be related? I'm on svn trunk version 16272 on 32bit debian etch. 


Thanks, 
Dan- 
function runCommand(command)
    local f = io.popen(command) -- runs command
    local l = f:read("*a") -- read output of command
    f:close()
    return l
end function notifyStart(id)
    local url = "http://host1/start?id=" .. id
    
    local wget = "/usr/bin/wget " .. url local out = runCommand(wget) return out; end 
function notifyStop(id)
    local url = "http://host2/start?id=" .. id
    
    local wget = "/usr/bin/wget " .. url local out = runCommand(wget) return out; end 
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From stevendt at primrosebank.net  Tue Jan 19 13:03:41 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Tue, 19 Jan 2010 21:03:41 -0000
Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors
References: <2CCD94BEC92D4DE493B6FA65A554F286@bp1.ad.bp.com>,
	<COL117-W6424736CC849832D6D9DBFB9650@phx.gbl>,
	<E5A67EA005F34E68B5501C860A9B96BC@bp1.ad.bp.com>
	<COL117-W15B08477F1FFA512F0F0B4B9650@phx.gbl>
Message-ID: <FB07BA81F28C43B68925CB0F52B22ECD@bp1.ad.bp.com>

Jeff,

I had already tried to rebuild but have just done it again with the same errors on starting up FreeSwitch, i.e., the DLLs are still not there ?

regards
Dave
  ----- Original Message ----- 
  From: Jeff Lenk 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, January 19, 2010 8:26 PM
  Subject: Re: [Freeswitch-users] SVN 16400 - Windows Build Errors


  Do a rebuild all from the solution and you should not see any errors.
   

------------------------------------------------------------------------------
  From: stevendt at primrosebank.net
  To: freeswitch-users at lists.freeswitch.org
  Date: Tue, 19 Jan 2010 16:13:44 +0000
  Subject: Re: [Freeswitch-users] SVN 16400 - Windows Build Errors


  Hi Jeff,

  the DLLs are not in the Debug folders pointed to by the error messages ?

  regards
  Dave

    ----- Original Message ----- 
    From: Jeff Lenk 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Tuesday, January 19, 2010 3:44 PM
    Subject: Re: [Freeswitch-users] SVN 16400 - Windows Build Errors


    Hi Dave,
     
    mod_cluechoo is not built for Windows and is not needed.
    The other files should build fine. Do they exist in your Debug folder?
     
    - Jeff
     

----------------------------------------------------------------------------
    From: stevendt at primrosebank.net
    To: freeswitch-users at lists.freeswitch.org
    Date: Tue, 19 Jan 2010 13:31:42 +0000
    Subject: [Freeswitch-users] SVN 16400 - Windows Build Errors


    Hi,

    I have just upgraded to the latest SVN (16400) and see a couple of errors when FS starts, although the basic functionality seems to be working.
    Some DLLs have not been built, should I be worried about these errors and/or how do I correct them ?

    2010-01-19 12:51:33.280718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_cluechoo.dll
    **dll open error [126l]

    2010-01-19 12:51:33.905718 [CRIT] switch_loadable_module.c:872 Error Loading module C:\FreeSWITCH\Debug\mod\mod_file_string.dll
    **dll open error [126l]

    2010-01-19 12:51:34.108843 [ERR] mod_spidermonkey.c:930 Error Loading module C:\FreeSWITCH\Debug\mod\mod_spidermonkey_core_db.dll
    **dll open error [126l]


    There is nothing in the Build log on the first two, but there is an error against the last one

    88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.pdb'.
    88>Make sure that the file is not open by another process and is not write-protected.
    88>mod_spidermonkey_core_db : error PRJ0008 : Could not delete file 'c:\FreeSWITCH\src\mod\languages\mod_spidermonkey\Win32\Debug\vc90.idb'.
    88>Make sure that the file is not open by another process and is not write-protected

    regards
    Dave

    WindowsXP
    VC++ Express 2008


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From john at acsol.net  Tue Jan 19 14:12:51 2010
From: john at acsol.net (John)
Date: Tue, 19 Jan 2010 15:12:51 -0700
Subject: [Freeswitch-users] Voicemail to email problems
In-Reply-To: <4E823F6B-75DC-44D9-BD18-C7773D77EAC9@freeswitch.org>
References: <ab161f4749d2cf203ec197497f2ce22e@thom.fr.eu.org>	<4B560A46.90000@acsol.net>	<191c3a031001191146r257d34afq98f921c0ff07f2fc@mail.gmail.com>
	<4E823F6B-75DC-44D9-BD18-C7773D77EAC9@freeswitch.org>
Message-ID: <4B562E63.7000407@acsol.net>

I have modified the Exim configuration as per the wiki. I am still 
getting the same message, in addition, the /tmp/mail.xxxxxxxx file is 
not getting created at any point. Is it possible the problem is that 
there is no file to send, so it errors out? Thank you both for your help!

/bin/cat: write error: Broken pipe
sh: line 1: 21176 Done(1)                 /bin/cat /tmp/mail.1263940067ade7
      21177 Segmentation fault      (core dumped) | exim4 -f 
1004 at voip.server.net -t jhart at server.net


On 1/19/2010 1:14 PM, Brian West wrote:
> http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings
>
> /b
>
> On Jan 19, 2010, at 1:46 PM, Anthony Minessale wrote:
>
>> sounds like exim pretending to be sendmail and not doing it very well.
>> I think there is a wiki page somewhere that tells you how to config 
>> it properly.
>>
>>
>> On Tue, Jan 19, 2010 at 1:38 PM, John <john at acsol.net 
>> <mailto:john at acsol.net>> wrote:
>>
>>     I am getting a segfault as well; however I use exim4 on Debian.
>>     Email is
>>     never being written to /tmp. I too can run the exim command by
>>     hand and
>>     it works. Any body have an idea or direction to look at?
>>     Here is the error.
>>     /bin/cat: write error: Broken pipe
>>     sh: line 1:  7102 Done(1)                 /bin/cat
>>     /tmp/mail.12639191242a9f
>>           7103 Segmentation fault
>>
>>     Thanks
>>
>>
>>
>>     On 1/19/2010 9:31 AM, Fran?ois Legal wrote:
>>     > This is not the error I get.
>>     >
>>     > I modified my config so that the command executed to send the
>>     email is cat
>>     > /tmp/mail.xxxxxxx | /usr/local/freeswitch/email.sh
>>     > with email.sh teeing the stdin to a file and to sendmail.
>>     >
>>     > syslog reports sendmail segfault.
>>     >
>>     > Fran?ois
>>     >
>>     > On Tue, 19 Jan 2010 09:02:30 -0700, John<john at acsol.net
>>     <mailto:john at acsol.net>>  wrote:
>>     >
>>     >> Voicemail to email gets a broken pipe error. I believe this is
>>     happening
>>     >>
>>     >
>>     >> while writing the file to /tmp? The mail.xxxxxxxxxxxx file is
>>     never
>>     >> written. Any ideas? Thanks
>>     >>
>>     >>
>>     >>
>>     >> _______________________________________________
>>     >> FreeSWITCH-users mailing list
>>     >> FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     >>
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     >> http://www.freeswitch.org <http://www.freeswitch.org/>
>>     >>
>>     > _______________________________________________
>>     > FreeSWITCH-users mailing list
>>     > FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     >
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     > http://www.freeswitch.org <http://www.freeswitch.org/>
>>     >
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com 
>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net <http://irc.freenode.net/> #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org 
>> <mailto:sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888 
>> <http://iax:guest at conference.freeswitch.org/888>
>> googletalk:conf+888 at conference.freeswitch.org 
>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From anthony.minessale at gmail.com  Tue Jan 19 14:43:59 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Jan 2010 16:43:59 -0600
Subject: [Freeswitch-users] Lua: io.popen/read blocking in other
	incoming calls
In-Reply-To: <19409906.41.1263934985398.JavaMail.root@zimbra>
References: <13515873.38.1263934963058.JavaMail.root@zimbra>
	<19409906.41.1263934985398.JavaMail.root@zimbra>
Message-ID: <191c3a031001191443o6184a6f4nd3f5a849ad609d5f@mail.gmail.com>

if there is any global lock it would be in the lua lib.  I know that would
not happen in C so you may want to step it in the debugger and look for any
evidence of a global mutex in the lua lib.


On Tue, Jan 19, 2010 at 3:03 PM, Dan <freeswitch-users at digitaldan.com>wrote:

> My lua script is calling wget through lua's io.popen to send
> and receive data from a web service. While the f:read to wget  is running,
> other incoming calls will block on the same io.popen call until the first
> call closes the pipe (with f:close()).
>
>  I had assumed every incoming call was on its own thread and that each had
> its own lua instance.  Is there a global lock happening here? Below is the
> runCommand call and the two start and stop methods that are getting called
> in my script when the call begins and ends (notice they even talk to
> different hosts, so its not the web server hanging).  I have put in
> debugging statements and its definitely  hanging  trying to call io.popen
> and not on the f:read.  I noticed on the mod_python page  (
> http://wiki.freeswitch.org/wiki/Mod_python#Known_Bugs) that it had read
> issues that were fixed in 12958, could this be related?  I'm on svn trunk
> version 16272 on 32bit debian etch.
>
> Thanks,
> Dan-
>
> function runCommand(command)
>     local f = io.popen(command) -- runs command
>     local l = f:read("*a") -- read output of command
>     f:close()
>     return l
> end
>
> function notifyStart(id)
>     local url = "http://host1/start?id=" .. id
>
>     local wget = "/usr/bin/wget " .. url
>
>     local out = runCommand(wget)
>
>     return out;
>
> end
>
>
> function notifyStop(id)
>     local url = "http://host2/start?id=" .. id
>
>     local wget = "/usr/bin/wget " .. url
>
>     local out = runCommand(wget)
>
>     return out;
>
> end
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From kees at mroffice.org  Tue Jan 19 14:58:15 2010
From: kees at mroffice.org (Kees Varekamp)
Date: Wed, 20 Jan 2010 11:58:15 +1300
Subject: [Freeswitch-users] SIP for Skype
Message-ID: <98d38dcf1001191458g4766d028r9540cce9d2c6f6ef@mail.gmail.com>

Hello,

I am trying to hook up my freeswitch server to SIP for skype but skype keeps
sending me 403 forbidden. I'm pretty sure that the account is OK, so I'm
wondering if they are blocking me for other reasons. Has anybody had any
success with SIP for Skype?

Thanks, Kees

Siptrace:

   ------------------------------------------------------------------------
send 878 bytes to udp/[204.9.161.164]:5060 at 22:38:31.115019:
   ------------------------------------------------------------------------
   REGISTER sip:sip.skype.com SIP/2.0
   Via: SIP/2.0/UDP 203.109.207.110;rport;branch=z9hG4bK7Da0S0XSeypHm
   Max-Forwards: 70
   From: <sip:xxxxxx at sip.skype.com <sip%3Axxxxxx at sip.skype.com>
;transport=udp>;tag=6vNF7NSDX1t7H
   To: <sip:xxxxxx at sip.skype.com <sip%3Axxxxxx at sip.skype.com>;transport=udp>
   Call-ID: 38262ed8-054b-11df-899c-d96406a83851
   CSeq: 125852020 REGISTER
   Contact: <sip:gw+sip.skype.com at 203.109.207.110:5060;transport=udp;gw=
sip.skype.com>
   Expires: 3600
   User-Agent: FreeSWITCH-mod_sofia/1.0.5-20100119-0400-16400M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Authorization: Digest username="xxxxxx", realm="sip.skype.com",
nonce="xxxxx", algorithm=MD5, uri="sip:sip.skype.com", response="xxxxx"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 378 bytes from udp/[204.9.161.164]:5060 at 22:38:31.470978:
   ------------------------------------------------------------------------
   SIP/2.0 403 Forbidden
   From: <sip:xxxxx at sip.skype.com <sip%3Axxxxx at sip.skype.com>
;transport=udp>;tag=6vNF7NSDX1t7H
   To: <sip:xxxxxx at sip.skype.com <sip%3Axxxxxx at sip.skype.com>
;transport=udp>;tag=05aed4eb43523e287156e2da6464d890.13a5
   Call-ID: 38262ed8-054b-11df-899c-d96406a83851
   CSeq: 125852020 REGISTER
   Via: SIP/2.0/UDP 203.109.207.110;rport=5060;branch=z9hG4bK7Da0S0XSeypHm
   Server: OpenSIPS
   Content-Length: 0

   ------------------------------------------------------------------------
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From brian at freeswitch.org  Tue Jan 19 15:01:15 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Jan 2010 17:01:15 -0600
Subject: [Freeswitch-users] SIP for Skype
In-Reply-To: <98d38dcf1001191458g4766d028r9540cce9d2c6f6ef@mail.gmail.com>
References: <98d38dcf1001191458g4766d028r9540cce9d2c6f6ef@mail.gmail.com>
Message-ID: <A85FC509-D282-4A8F-A0B8-1AA3E14C402B@freeswitch.org>

Yes, Please contact your provider.

/b

On Jan 19, 2010, at 4:58 PM, Kees Varekamp wrote:

> Hello,
> 
> I am trying to hook up my freeswitch server to SIP for skype but skype keeps sending me 403 forbidden. I'm pretty sure that the account is OK, so I'm wondering if they are blocking me for other reasons. Has anybody had any success with SIP for Skype?
> 
> Thanks, Kees
> 
> Siptrace:
> 




From rupa at rupa.com  Tue Jan 19 15:06:36 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Jan 2010 17:06:36 -0600
Subject: [Freeswitch-users] Lua: io.popen/read blocking in other
	incoming calls
In-Reply-To: <19409906.41.1263934985398.JavaMail.root@zimbra>
References: <13515873.38.1263934963058.JavaMail.root@zimbra>
	<19409906.41.1263934985398.JavaMail.root@zimbra>
Message-ID: <b39273531001191506v17e34e0er99d6c4098d960950@mail.gmail.com>

On Tue, Jan 19, 2010 at 3:03 PM, Dan <freeswitch-users at digitaldan.com>wrote:

> My lua script is calling wget through lua's io.popen to send
> and receive data from a web service. While the f:read to wget  is running,
> other incoming calls will block on the same io.popen call until the first
> call closes the pipe (with f:close()).
>

You might want to look at the api that mod_curl exposes to do what you want.
 No need to do an expensive system call just to call a webservice.

-- 
-Rupa
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From kees at mroffice.org  Tue Jan 19 15:14:03 2010
From: kees at mroffice.org (Kees Varekamp)
Date: Wed, 20 Jan 2010 12:14:03 +1300
Subject: [Freeswitch-users] SIP for Skype
In-Reply-To: <A85FC509-D282-4A8F-A0B8-1AA3E14C402B@freeswitch.org>
References: <98d38dcf1001191458g4766d028r9540cce9d2c6f6ef@mail.gmail.com>
	<A85FC509-D282-4A8F-A0B8-1AA3E14C402B@freeswitch.org>
Message-ID: <98d38dcf1001191514w681862baod098c21f4ae4ec9d@mail.gmail.com>

? You mean Internet provider or Skype?

Would you mind sending me an example <gateway> config?

Thanks, Kees

On Wed, Jan 20, 2010 at 12:01, Brian West <brian at freeswitch.org> wrote:

> Yes, Please contact your provider.
>
> /b
>
> On Jan 19, 2010, at 4:58 PM, Kees Varekamp wrote:
>
> > Hello,
> >
> > I am trying to hook up my freeswitch server to SIP for skype but skype
> keeps sending me 403 forbidden. I'm pretty sure that the account is OK, so
> I'm wondering if they are blocking me for other reasons. Has anybody had any
> success with SIP for Skype?
> >
> > Thanks, Kees
> >
> > Siptrace:
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch-users at digitaldan.com  Tue Jan 19 15:14:04 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Tue, 19 Jan 2010 16:14:04 -0700 (MST)
Subject: [Freeswitch-users] Lua: io.popen/read blocking in
	other	incoming calls
In-Reply-To: <191c3a031001191443o6184a6f4nd3f5a849ad609d5f@mail.gmail.com>
Message-ID: <31453864.51.1263942843967.JavaMail.root@zimbra>

Thanks for your response, I'm looking through the lua source right now. Am I correct in assuming that each incoming call has its own thread and therefore its own lua vm instance? So the only blocking that would be possible among threads is if they were calling something blocking on the core switch module. That's definitely not happening here, so I'll keep looking around. 




Thanks 
Dan- 
----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Tuesday, January 19, 2010 3:43:59 PM 
Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls 

if there is any global lock it would be in the lua lib. I know that would not happen in C so you may want to step it in the debugger and look for any evidence of a global mutex in the lua lib. 



On Tue, Jan 19, 2010 at 3:03 PM, Dan < freeswitch-users at digitaldan.com > wrote: 




My lua script is calling wget through lua's io.popen to send and receive data from a web service. While the f:read to wget is running, other incoming calls will block on the same io.popen call until the first call closes the pipe (with f:close()). 


I had assumed every incoming call was on its own thread and that each had its own lua instance. Is there a global lock happening here? Below is the runCommand call and the two start and stop methods that are getting called in my script when the call begins and ends (notice they even talk to different hosts, so its not the web server hanging). I have put in debugging statements and its definitely hanging trying to call io.popen and not on the f:read. I noticed on the mod_python page ( http://wiki.freeswitch.org/wiki/Mod_python#Known_Bugs ) that it had read issues that were fixed in 12958, could this be related? I'm on svn trunk version 16272 on 32bit debian etch. 


Thanks, 
Dan- 
function runCommand(command)
    local f = io.popen(command) -- runs command
    local l = f:read("*a") -- read output of command
    f:close()
    return l
end function notifyStart(id)
    local url = " http://host1/start?id= " .. id
    
    local wget = "/usr/bin/wget " .. url local out = runCommand(wget) return out; end 
function notifyStop(id)
    local url = " http://host2/start?id= " .. id
    
    local wget = "/usr/bin/wget " .. url local out = runCommand(wget) return out; end 
_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 




-- 
Anthony Minessale II 

FreeSWITCH http://www.freeswitch.org/ 
ClueCon http://www.cluecon.com/ 
Twitter: http://twitter.com/FreeSWITCH_wire 

AIM: anthm 
MSN:anthony_minessale at hotmail.com 
GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com 
IRC: irc.freenode.net #freeswitch 

FreeSWITCH Developer Conference 
sip:888 at conference.freeswitch.org 
iax:guest at conference.freeswitch.org/888 
googletalk:conf+888 at conference.freeswitch.org 
pstn:+19193869900 

_______________________________________________ 
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FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
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From freeswitch-users at digitaldan.com  Tue Jan 19 15:17:37 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Tue, 19 Jan 2010 16:17:37 -0700 (MST)
Subject: [Freeswitch-users] Lua: io.popen/read blocking in
	other	incoming calls
In-Reply-To: <7967781.55.1263942960863.JavaMail.root@zimbra>
Message-ID: <11387916.58.1263943057280.JavaMail.root@zimbra>

I would, but I need to post a a wav file that gets recorded, I didn't see a way to supply the location of a file to use as the post data. It looks like you have to url encode the data in the script and pass it all in the call. 


Thanks 
Dan- 
----- Original Message ----- 
From: "Rupa Schomaker" <rupa at rupa.com> 
To: "freeswitch-users" <freeswitch-users at lists.freeswitch.org> 
Sent: Tuesday, January 19, 2010 4:06:36 PM 
Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls 




On Tue, Jan 19, 2010 at 3:03 PM, Dan < freeswitch-users at digitaldan.com > wrote: 




My lua script is calling wget through lua's io.popen to send and receive data from a web service. While the f:read to wget is running, other incoming calls will block on the same io.popen call until the first call closes the pipe (with f:close()). 


You might want to look at the api that mod_curl exposes to do what you want. No need to do an expensive system call just to call a webservice. 
-- 
-Rupa 

_______________________________________________ 
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
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From brian at freeswitch.org  Tue Jan 19 15:18:27 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Jan 2010 17:18:27 -0600
Subject: [Freeswitch-users] SIP for Skype
In-Reply-To: <98d38dcf1001191514w681862baod098c21f4ae4ec9d@mail.gmail.com>
References: <98d38dcf1001191458g4766d028r9540cce9d2c6f6ef@mail.gmail.com>
	<A85FC509-D282-4A8F-A0B8-1AA3E14C402B@freeswitch.org>
	<98d38dcf1001191514w681862baod098c21f4ae4ec9d@mail.gmail.com>
Message-ID: <5B84A27E-E79D-4670-A525-074A4B1FEE43@freeswitch.org>

Skype.

/b

On Jan 19, 2010, at 5:14 PM, Kees Varekamp wrote:

> ? You mean Internet provider or Skype?
> 
> Would you mind sending me an example <gateway> config?
> 
> Thanks, Kees
>  




From msc at freeswitch.org  Tue Jan 19 15:21:55 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Jan 2010 15:21:55 -0800
Subject: [Freeswitch-users] NOTICE: mod_iax slated for unsupported on Feb. 5
Message-ID: <87f2f3b91001191521y3aec5aadp6ebee73945bb7e0c@mail.gmail.com>

To everyone in the FreeSWITCH community:

We would like to put a call out and let everyone know that unless we find
someone who wants to be the maintainer for mod_iax it will be moved to
unsupported status as of Friday February 5th. Anyone who wishes to maintain
the module will need to be in charge of all bug reports, updates, patches,
and technical support questions. If are in a position to maintain mod_iax
then please contact Brian West offlist: brian at freeswitch.org.

-Michael
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From freeswitch-users at digitaldan.com  Tue Jan 19 15:27:52 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Tue, 19 Jan 2010 16:27:52 -0700 (MST)
Subject: [Freeswitch-users] Lua: io.popen/read blocking
	in	other	incoming calls
In-Reply-To: <29464896.62.1263943497225.JavaMail.root@zimbra>
Message-ID: <22214819.65.1263943672393.JavaMail.root@zimbra>

One more question, popen on linux forks a new child process to execute a shell in, could the freeswitch environment have any influence on this? 


Thanks 
Dan- 


----- Original Message ----- 
From: "Dan" <freeswitch-users at digitaldan.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Tuesday, January 19, 2010 4:14:04 PM 
Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls 


Thanks for your response, I'm looking through the lua source right now. Am I correct in assuming that each incoming call has its own thread and therefore its own lua vm instance? So the only blocking that would be possible among threads is if they were calling something blocking on the core switch module. That's definitely not happening here, so I'll keep looking around. 




Thanks 
Dan- 
----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Tuesday, January 19, 2010 3:43:59 PM 
Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls 

if there is any global lock it would be in the lua lib. I know that would not happen in C so you may want to step it in the debugger and look for any evidence of a global mutex in the lua lib. 



On Tue, Jan 19, 2010 at 3:03 PM, Dan < freeswitch-users at digitaldan.com > wrote: 




My lua script is calling wget through lua's io.popen to send and receive data from a web service. While the f:read to wget is running, other incoming calls will block on the same io.popen call until the first call closes the pipe (with f:close()). 


I had assumed every incoming call was on its own thread and that each had its own lua instance. Is there a global lock happening here? Below is the runCommand call and the two start and stop methods that are getting called in my script when the call begins and ends (notice they even talk to different hosts, so its not the web server hanging). I have put in debugging statements and its definitely hanging trying to call io.popen and not on the f:read. I noticed on the mod_python page ( http://wiki.freeswitch.org/wiki/Mod_python#Known_Bugs ) that it had read issues that were fixed in 12958, could this be related? I'm on svn trunk version 16272 on 32bit debian etch. 


Thanks, 
Dan- 
function runCommand(command)
    local f = io.popen(command) -- runs command
    local l = f:read("*a") -- read output of command
    f:close()
    return l
end function notifyStart(id)
    local url = " http://host1/start?id= " .. id
    
    local wget = "/usr/bin/wget " .. url local out = runCommand(wget) return out; end 
function notifyStop(id)
    local url = " http://host2/start?id= " .. id
    
    local wget = "/usr/bin/wget " .. url local out = runCommand(wget) return out; end 
_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 




-- 
Anthony Minessale II 

FreeSWITCH http://www.freeswitch.org/ 
ClueCon http://www.cluecon.com/ 
Twitter: http://twitter.com/FreeSWITCH_wire 

AIM: anthm 
MSN:anthony_minessale at hotmail.com 
GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com 
IRC: irc.freenode.net #freeswitch 

FreeSWITCH Developer Conference 
sip:888 at conference.freeswitch.org 
iax:guest at conference.freeswitch.org/888 
googletalk:conf+888 at conference.freeswitch.org 
pstn:+19193869900 

_______________________________________________ 
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 

_______________________________________________ 
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FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
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From rupa at rupa.com  Tue Jan 19 15:32:57 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Jan 2010 17:32:57 -0600
Subject: [Freeswitch-users] Lua: io.popen/read blocking in other
	incoming calls
In-Reply-To: <11387916.58.1263943057280.JavaMail.root@zimbra>
References: <7967781.55.1263942960863.JavaMail.root@zimbra>
	<11387916.58.1263943057280.JavaMail.root@zimbra>
Message-ID: <b39273531001191532s7d2ad182oc6751eedc1089d30@mail.gmail.com>

Ah... yes you do.  Patches / Bounty to implement that accepted.

On Tue, Jan 19, 2010 at 5:17 PM, Dan <freeswitch-users at digitaldan.com>wrote:

> I would, but I need to post a a wav file that gets recorded, I didn't see a
> way to supply the location of a file to use as the post data.  It looks like
> you have to url encode the data in the script and pass it all in the call.
>
> Thanks
> Dan-
>
> ----- Original Message -----
> From: "Rupa Schomaker" <rupa at rupa.com>
> To: "freeswitch-users" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, January 19, 2010 4:06:36 PM
> Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in
> other        incoming calls
>
>
>
> On Tue, Jan 19, 2010 at 3:03 PM, Dan <freeswitch-users at digitaldan.com>wrote:
>
>> My lua script is calling wget through lua's io.popen to send
>> and receive data from a web service. While the f:read to wget  is running,
>> other incoming calls will block on the same io.popen call until the first
>> call closes the pipe (with f:close()).
>>
>
> You might want to look at the api that mod_curl exposes to do what you
> want.  No need to do an expensive system call just to call a webservice.
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From msc at freeswitch.org  Tue Jan 19 16:41:40 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Jan 2010 16:41:40 -0800
Subject: [Freeswitch-users] Home setup with home company
In-Reply-To: <4B55F3A0020000E1000003E0@mail.fribert.dk>
References: <4B55F3A0020000E1000003E0@mail.fribert.dk>
Message-ID: <87f2f3b91001191641p1fea3318l3d166fe3555e760d@mail.gmail.com>

On Tue, Jan 19, 2010 at 9:02 AM, mailinglist <mailinglist at fribert.dk> wrote:

>  I have a very small one man constultancy company that has the occasional
> call, unfortunately we are getting more spam calls after hours than real
> calls during work hours, so I would like to set up a TOD system.
>
> First step for me is just playing with the TOD example, I've gotten this
> far:
>
> <extension name="CompanyTOD" continue="true">
>    <condition field="destination_number" expression="^(8203)$"/>
>       <!-- man strftime - M-F, 9AM to 5PM -->
>       <condition field="${strftime(%w)}" expression="^([1-5])$"/>
>       <condition field="${strftime(%H%M)}"
> expression="^((09|1[0-6])[0-5][0-9]|1700)$">
>          <action application="set" data="call_timeout=45"/>
>          <action application="set" data="domain_name=10.11.12.25"/>
>          <action application="set" data="continue_on_fail=true"/>
>          <action application="set" data="hangup_after_bridge=true"/>
>          <action application="bridge" data="group/company@${domain_name}"/<group/company@$%7Bdomain_name%7D%22/>
> >
>          <action application="answer" data=""/>
>          <action application="sleep" data="1000"/>
>          <action application="voicemail" data="default 10.11.12.25 10199"/>
>          <anti-action application="voicemail" data="default 10.11.12.25
> 10199"/>
>       </condition>
> </extension>
> My idea with this, was that in the time 9-17 mon-fri, it rings, the rest of
> the time it's sent directly to the voicemail.
> I would of course like to have it not take messages outside work hours, but
> that's just refining :-)
>
> But it picks up the call, and then nothing...
>
>
We have a much cleaner way of doing TOD and DOW handling. You'll need to get
to the latest FreeSWITCH version. Look in conf/dialplan/default.xml for this
example:

    <!-- Example of doing things based on time of day.

     year = 4 digit year. Example year="2009"
     yday = 1-365
     mon = 1-12
     mday = 1-31
     week = 1-52
     mweek= 1-6
     wday = 1-7
     hour = 0-23
     minute = 0-59
     minute-of-day = 1-1440

     Example:
     <condition minute-of-day="540-1080"> (9am to 6pm EVERY day)
     do something ...
     </condition>
    -->
    <extension name="tod_example" continue="true">
      <condition wday="2-6" hour="9-18">
        <action application="set" data="open=true"/>
      </condition>
    </extension>


Use that condition instead of the two conditions you're now using and see if
you have better success.

-MC
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From lakindia89 at gmail.com  Tue Jan 19 20:49:43 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 20 Jan 2010 10:19:43 +0530
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <191c3a031001190804m692f63acsaf852b8809db09d1@mail.gmail.com>
References: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
	<191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>
	<7d79b3931001132357i36bbb482jdf2bbdd3aea2a583@mail.gmail.com>
	<7d79b3931001180322p74a1227qe0c2199a77cbfe2@mail.gmail.com>
	<191c3a031001180824q6e364c72g3f789892597e9469@mail.gmail.com>
	<7d79b3931001190107o200ec04dredd76689dd235588@mail.gmail.com>
	<191c3a031001190804m692f63acsaf852b8809db09d1@mail.gmail.com>
Message-ID: <7d79b3931001192049h7e4a1a9ex1676ec415e6fd49f@mail.gmail.com>

Thanks for all your reply's.
Will give a try to on non-blocking.


On Tue, Jan 19, 2010 at 9:34 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Its nothing we can fix, that is what you must do on a failed read syscall.
> you can do a non blocking read instead and take your chances.
>
>
>
> On Tue, Jan 19, 2010 at 3:07 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> I tried with SIGUSR1, but no progress. I got SERVER_DISCONNECTED.
>> Output:
>>
>> CHILD 3814: Received USR1
>> EVENT [SERVER_DISCONNECTED]
>>
>> In esl.c, in esl_recv_event() function, line no: 824
>> if (rrval < 0) {
>>                         strerror_r(handle->errnum, handle->err,
>> sizeof(handle->err));
>>                         goto fail;
>> }
>> When the program is blocked under receive, I passed the signal. So recv
>> returns -1, and in fail: it call esl_disconnect(handle).
>>
>> Is it because of this??? If so, whether it should be fixed or not???
>>
>>
>>
>>
>> On Mon, Jan 18, 2010 at 9:54 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> try a less famous signal like SIGUSR1 it's possible something in perl
>>> still reacts to SIGINT
>>>
>>>
>>>
>>> On Mon, Jan 18, 2010 at 5:22 AM, lakshmanan ganapathy <
>>> lakindia89 at gmail.com> wrote:
>>>
>>>> Here is the result
>>>>
>>>> Program:
>>>>
>>>> require ESL;
>>>> use IO::Socket::INET;
>>>> use Data::Dumper;
>>>>
>>>> my $ip = "192.168.1.222";
>>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>>> die "Could not create socket: $!\n" unless $sock;
>>>>
>>>>
>>>> for(;;) {
>>>>     my $new_sock = $sock->accept();
>>>>     next if (not defined ($new_sock));
>>>>     my $pid = fork();
>>>>     if ($pid) {
>>>>         close($new_sock);
>>>>         next;
>>>>     }
>>>>     print "CHILD PID: $$\n";
>>>>     &register_Signals_Child();
>>>>     sub register_Signals_Child() {
>>>>         foreach ( keys %SIG ) {
>>>>             $SIG{$_} = 'Handler';
>>>>         }
>>>>     }
>>>>
>>>>     sub Handler() {
>>>>
>>>>         my $handle=$_[0];
>>>>         if($handle eq "INT") {
>>>>             print "CHILD $$: SIGNAL SIG$handle is generated\n";
>>>>         }
>>>>         else
>>>>         {
>>>>             print "CHILD $$: Received $handle\n";
>>>>
>>>>         }
>>>>     }
>>>>     my $host = $new_sock->sockhost();
>>>>     my $fd = fileno($new_sock);
>>>>
>>>>     my $con = new ESL::ESLconnection($fd);
>>>>     my $info = $con->getInfo();
>>>>
>>>>     my $uuid = $info->getHeader("unique-id");
>>>>
>>>>     printf "Connected call %s, from %s\n", $uuid,
>>>> $info->getHeader("caller-caller-id-number");
>>>>     my $r=$con->execute("answer");
>>>>     print Dumper $r;
>>>>     $con->events("plain","all");
>>>>     my
>>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>>     while($con->connected()) {
>>>>         my $e = $con->recvEvent();
>>>>
>>>>         if ($e) {
>>>>             my $name = $e->getHeader("event-name");
>>>>             print "EVENT [$name]\n";
>>>>             if ($name eq "DTMF") {
>>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>>                 print "$digit\n";
>>>>             }
>>>>         }
>>>>     }
>>>>     close($new_sock);
>>>> }
>>>>
>>>> I executed the program and the following things were printed
>>>>
>>>> CHILD PID: 6778
>>>> Connected call e0d1001a-03f4-11df-b002-db488337e0ea, from 1001
>>>> $VAR1 = 0;
>>>> EVENT [CHANNEL_EXECUTE]
>>>> EVENT [CHANNEL_ANSWER]
>>>> EVENT [CHANNEL_EXECUTE_COMPLETE]
>>>> EVENT [COMMAND]
>>>> EVENT [CHANNEL_EXECUTE]
>>>> EVENT [HEARTBEAT]
>>>> EVENT [RE_SCHEDULE]
>>>> EVENT [CHANNEL_EXECUTE_COMPLETE]
>>>>
>>>> Then from another shell I executed kill -2 6778, the result is follows
>>>> CHILD 6778: SIGNAL SIGINT is generated
>>>> EVENT [SERVER_DISCONNECTED]
>>>>
>>>> But the child process is still running as expected.
>>>> But  I don't know why I received SERVER_DISCONNECTED from freeswitch???
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Thu, Jan 14, 2010 at 1:27 PM, lakshmanan ganapathy <
>>>> lakindia89 at gmail.com> wrote:
>>>>
>>>>> I taught the signal handler will be inherited by the child process. It
>>>>> also does like that.
>>>>> After making a call, If I press ctrl + c, the above program printed
>>>>> PARENT PID: Signal SIGINT is generated
>>>>> CHILD PID: Signal SIGINT is generated.
>>>>>
>>>>> So I think the sigal handlers will be inherited to the child.
>>>>> Anyway I'll also try registering signal handlers in child also, and
>>>>> then I'll come back with that result.
>>>>>
>>>>> Thanks....
>>>>> On Wed, Jan 13, 2010 at 9:48 PM, Anthony Minessale <
>>>>> anthony.minessale at gmail.com> wrote:
>>>>>
>>>>>> you would have to register signals in your child process too
>>>>>>
>>>>>>   On Wed, Jan 13, 2010 at 3:13 AM, lakshmanan ganapathy <
>>>>>> lakindia89 at gmail.com> wrote:
>>>>>>
>>>>>>>  Hi all,
>>>>>>>
>>>>>>> I've done a sample program (In perl ESL) , which play a file to the
>>>>>>> caller and then it will call recvEvent() and print the event name. I've
>>>>>>> handled signals also.
>>>>>>>
>>>>>>> When I send SIGINT to my program (Perl), the signal handler is called
>>>>>>> and I can see the print output. But in the same time,  I received
>>>>>>> SERVER_DISCONNECTED from freeswitch as event.
>>>>>>>
>>>>>>> I don't know why I received SERVER_DISCONNECTED from freeswitch. Is
>>>>>>> it because, the recvEvent() from perl internally  calls the recvevent
>>>>>>> function in the Esl.c and when it waits to receive the information from
>>>>>>> socket, the signal occurred???
>>>>>>>
>>>>>>> Please clarify me!!
>>>>>>>
>>>>>>> Here is my program
>>>>>>> require ESL;
>>>>>>> use IO::Socket::INET;
>>>>>>> use Data::Dumper;
>>>>>>>
>>>>>>> my $ip = "192.168.1.222";
>>>>>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>>>>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>>>>>> die "Could not create socket: $!\n" unless $sock;
>>>>>>> &register_Signals();
>>>>>>>
>>>>>>> for(;;) {
>>>>>>>     my $new_sock = $sock->accept();
>>>>>>>     next if (not defined ($new_sock));
>>>>>>>     my $pid = fork();
>>>>>>>     if ($pid) {
>>>>>>>         close($new_sock);
>>>>>>>         next;
>>>>>>>     }
>>>>>>>     print "CHILD PID: $$\n";
>>>>>>>     my $host = $new_sock->sockhost();
>>>>>>>     my $fd = fileno($new_sock);
>>>>>>>
>>>>>>>     my $con = new ESL::ESLconnection($fd);
>>>>>>>     my $info = $con->getInfo();
>>>>>>>
>>>>>>>     my $uuid = $info->getHeader("unique-id");
>>>>>>>
>>>>>>>     printf "Connected call %s, from %s\n", $uuid,
>>>>>>> $info->getHeader("caller-caller-id-number");
>>>>>>>     my $r=$con->execute("answer");
>>>>>>>     print Dumper $r;
>>>>>>>     $con->events("plain","all");
>>>>>>>     my
>>>>>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>>>>>
>>>>>>>     while($con->connected()) {
>>>>>>>         my $e = $con->recvEvent();
>>>>>>>
>>>>>>>         if ($e) {
>>>>>>>             my $name = $e->getHeader("event-name");
>>>>>>>             print "EVENT [$name]\n";
>>>>>>>             if ($name eq "DTMF") {
>>>>>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>>>>>                 print "$digit\n";
>>>>>>>             }
>>>>>>>         }
>>>>>>>     }
>>>>>>>     close($new_sock);
>>>>>>> }
>>>>>>> sub register_Signals() {
>>>>>>>     foreach ( keys %SIG ) {
>>>>>>>         $SIG{$_} = 'sig_Handler';
>>>>>>>     }
>>>>>>> }
>>>>>>>
>>>>>>> sub sig_Handler() {
>>>>>>>     my $handle=$_[0];
>>>>>>>     if($handle eq "INT") {
>>>>>>>         print "$$: SIGNAL SIG$handle is generated\n";
>>>>>>>     }
>>>>>>> }
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dome at tel.co.th  Tue Jan 19 21:32:26 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 20 Jan 2010 12:32:26 +0700
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <df97b9c51001190107x39d94eaai7400635f0ac03f86@mail.gmail.com>
References: <df97b9c51001190107x39d94eaai7400635f0ac03f86@mail.gmail.com>
Message-ID: <8ccbff061001192132h344a61a4m58d2973391c8182a@mail.gmail.com>

Please try http://wiki.freeswitch.org/wiki/Webapi
you can create class and map to webapi.

Dome C.

2010/1/19 Scott Fernandez <scottferri09 at gmail.com>:
> Hi,
>
> Is there any API modules available for me to initiate a call from .Net based
> application?.
>
> The idea is to include the API modules if any with the .NET base classes so
> that the API commands will be made available on it. I know it is doable when
> I use socket programming in .NET in which Telnet session is created.
> However, this would potentially hamper the performance of the application
> because of multiple sessions that will be created for each call.
>
> Other than that, Is there any Freeswitch API modules (like plug-ins)
> available in order to include it into the .Net classes and start building
> the customized application?
>
> Any help from any one is highly appreciated.
>
> Thanks,
> Scott
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From thangappan143 at gmail.com  Tue Jan 19 21:55:33 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Wed, 20 Jan 2010 11:25:33 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma card
Message-ID: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>

Dear all,

         I have successfully configured wanpipe with freeswitch. When I was
the running wancfg_fs script the following files openzap.conf ,
autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
are created.

         I started the wanrouter command then executed the freeswitch.
         When I was executing freeswitch mod_openzap.c said the error as
"Error  for finding the span id. name:PRI_1".
         But in the openzap.conf and openzap.conf.xml files the span name is
smg_prid.

         Why the freeswitch is referring the span name as PRI_1 ?
         Whether this has to configured in anywhere?

         In the freeswitch CLI using oz command I tried to dump the PRI_1
span id but it said te error  as "PRI_1 is not found".  When I was trying
the command 'oz dump smg_prid'  all the channel states and details shown.

        It seems that smg_prid span configured in openzap perfectly (Its my
assumption). Then Why freeswitch is referring the span name as PRI_1.

DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?

Could anyone please help me?

REFERENCE:

openzap.conf
[span wanpipe smg_prid]
name => smg_prid
trunk_type =>e1
b-channel => 1:1-15
b-channel => 1:17-31


openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
  <boost_spans>
        <span id="smg_prid">
        <!--<param name="hold-music" value="$${moh_uri}"/>-->
        <param name="dialplan" value="XML"/>
        <param name="context" value="default"/>
         <!-- regex to stop dialing when it matches -->
        <!--<param name="dial-regex" value="5555"/>-->
        <!-- regex to stop dialing when it does not match -->
        <!--<param name="fail-dial-regex" value="^5"/>-->
        </span>
 </boost_spans>
 <analog_spans>
 </analog_spans>
</configuration>






-- 
Regards,
Thangappan.M
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From ahmed.ajmal at gmail.com  Tue Jan 19 22:12:51 2010
From: ahmed.ajmal at gmail.com (Ahmed Bhaila)
Date: Wed, 20 Jan 2010 11:12:51 +0500
Subject: [Freeswitch-users] Help with CDR fields:Other-Leg-Network-Addr
	and Other-Leg-Destination-Number
In-Reply-To: <191c3a031001191137k72530079t402ff54d03b07756@mail.gmail.com>
References: <9d22cc171001190752j797fdf39radc404510128718c@mail.gmail.com>
	<191c3a031001191137k72530079t402ff54d03b07756@mail.gmail.com>
Message-ID: <9d22cc171001192212rb7e617aq7f593b64ac59cd9f@mail.gmail.com>

Thanks Anthony. That worked!!!

- AB

On Wed, Jan 20, 2010 at 12:37 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> those are not variables only variables expand.
>
> try bleg_network_addr
> and bleg_destination_number
>
>
> On Tue, Jan 19, 2010 at 9:52 AM, Ahmed Bhaila <ahmed.ajmal at gmail.com>wrote:
>
>> Hello
>>
>> I am using a template for Master.csv and having problems with getting
>> values for the following fields: Other-Leg-Network-Addr and
>> Other-Leg-Destination-Number. All I am doing is just enclosing these fields
>> in ${} in my template definition and they always turn out to be empty but
>> they arent supposed to be as I am seeing values in the freeswitch console.so
>> I am not sure what is wrong here I have noticed that some of the channel
>> variables(start_stamp, end_stamp) have "variable_" prepended to them when I
>> see them in the console so when defining the template I omit the variable_
>> part and can get the values. Is this something similar? It seems like the
>> problem is with all field names that start with Other. Any help be greatly
>> appreciated.
>>
>>
>> Thanks
>> AB
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From troy at tlainvestments.com  Tue Jan 19 22:15:36 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Tue, 19 Jan 2010 23:15:36 -0700
Subject: [Freeswitch-users] Call Screening Example Broken?
In-Reply-To: <8ccbff061001192132h344a61a4m58d2973391c8182a@mail.gmail.com>
References: <df97b9c51001190107x39d94eaai7400635f0ac03f86@mail.gmail.com>
	<8ccbff061001192132h344a61a4m58d2973391c8182a@mail.gmail.com>
Message-ID: <8BE9BCF3-C766-42CE-98B9-AC5F836327FC@tlainvestments.com>

I've implemented the Example 13: Call Screening from http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_13:_Call_Screening (below) and, while the file plays fine (over and over), fs is reporting an error from switch_ivr_originate.  

[ERR] switch_ivr_originate.c:202 sofia/internal/sip:1000 at 10.0.1.100 Error Playing File!

If I remove the group_confirm_file line, it works as expected.  Could it be that the DTMF (pressing 1 to accept the call) is interrupting the playback of the file?

 <extension name="screen">
   <condition field="destination_number" expression="^(\d{4})$">
     <action application="set" data="call_screen_filename=/tmp/${caller_id_number}-name.wav"/>
     <action application="answer"/>
     <action application="sleep" data="1000"/>
     <action application="phrase" data="voicemail_record_name"/>
     <action application="playback" data="tone_stream://%(500, 0, 640)"/>
     <action application="set" data="playback_terminators=#*0123456789"/>
     <action application="record" data="${call_screen_filename} 7 200 2"/>
     <action application="set" data="group_confirm_key=1"/>
     <action application="set" data="fail_on_single_reject=true"/>
     <action application="set" data="group_confirm_file=${call_screen_filename}"/>
     <action application="set" data="continue_on_fail=true"/>
     <action application="bridge" data="user/$1"/>
     <action application="voicemail" data="default $${domain} $1"/>
     <action application="hangup"/>
   </condition>
 </extension>
-Troy
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From mike at jerris.com  Tue Jan 19 22:16:45 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 Jan 2010 01:16:45 -0500
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
Message-ID: <A6200F19-3547-408C-8EB4-BAD6BE6A9DD0@jerris.com>

grep will tell you the answer.

On Jan 20, 2010, at 12:55 AM, Thangappan.M wrote:

> Dear all,
> 
>          I have successfully configured wanpipe with freeswitch. When I was the running wancfg_fs script the following files openzap.conf , autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf are created.
> 
>          I started the wanrouter command then executed the freeswitch.
>          When I was executing freeswitch mod_openzap.c said the error as  "Error  for finding the span id. name:PRI_1".
>          But in the openzap.conf and openzap.conf.xml files the span name is smg_prid.
>  
>          Why the freeswitch is referring the span name as PRI_1 ?  
>          Whether this has to configured in anywhere? 
> 
>          In the freeswitch CLI using oz command I tried to dump the PRI_1 span id but it said te error  as "PRI_1 is not found".  When I was trying the command 'oz dump smg_prid'  all the channel states and details shown.
>  
>         It seems that smg_prid span configured in openzap perfectly (Its my assumption). Then Why freeswitch is referring the span name as PRI_1. 
> 
> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
> 




From mike at jerris.com  Tue Jan 19 22:22:11 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 Jan 2010 01:22:11 -0500
Subject: [Freeswitch-users] Voicemail to email problems
In-Reply-To: <4B562E63.7000407@acsol.net>
References: <ab161f4749d2cf203ec197497f2ce22e@thom.fr.eu.org>	<4B560A46.90000@acsol.net>	<191c3a031001191146r257d34afq98f921c0ff07f2fc@mail.gmail.com>
	<4E823F6B-75DC-44D9-BD18-C7773D77EAC9@freeswitch.org>
	<4B562E63.7000407@acsol.net>
Message-ID: <47DF4005-415B-467E-A618-E477BD519AEC@jerris.com>

if you didn't have the file it would look more like:

<9>:cat fish | more
cat: fish: No such file or directory

Broken pipe is because the thing it was piped too suddenly disappeared, like it segfaulted or something like that.

Mike


On Jan 19, 2010, at 5:12 PM, John wrote:

> I have modified the Exim configuration as per the wiki. I am still getting the same message, in addition, the /tmp/mail.xxxxxxxx file is not getting created at any point. Is it possible the problem is that there is no file to send, so it errors out? Thank you both for your help! 
> 
> /bin/cat: write error: Broken pipe
> sh: line 1: 21176 Done(1)                 /bin/cat /tmp/mail.1263940067ade7
>      21177 Segmentation fault      (core dumped) | exim4 -f 1004 at voip.server.net -t jhart at server.net
> 
> 
> On 1/19/2010 1:14 PM, Brian West wrote:
>> 
>> http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings
>> 
>> /b
>> 
>> On Jan 19, 2010, at 1:46 PM, Anthony Minessale wrote:
>> 
>>> sounds like exim pretending to be sendmail and not doing it very well.
>>> I think there is a wiki page somewhere that tells you how to config it properly.
>>> 

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From mike at jerris.com  Tue Jan 19 22:28:40 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 Jan 2010 01:28:40 -0500
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
In-Reply-To: <20100119103139.GR4767@tamay-dogan.net>
References: <20100119103139.GR4767@tamay-dogan.net>
Message-ID: <2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>

There is a debian dir in tree for our packages.  I suspect that our packages are quite far from meeting the requirements of pretty much any distro so for now, we will at least have packages for the next release available soon after release.  We don't yet have a box for debian instances for the build farm so we do not build the svn snapshot pacakges for any deb distros.

Mike

On Jan 19, 2010, at 5:31 AM, Michelle Konzack wrote:

> Hi *,
> 
> I loss my last nerv, compiling all the time FreeSWITCH from source...
> 
> Is there someone providing a Debian Package from a repository?
> 
> Also it would be nice if FreeSWITCH go into the Debian distribution.




From thangappan143 at gmail.com  Tue Jan 19 23:13:37 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Wed, 20 Jan 2010 12:43:37 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
Message-ID: <7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>

I found the error in it. The file name is used as openzap.conf.xml (
smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
openzap.conf.wiki.xml  file.

Now the another problem is raised here.
When I was using oz list command , the details of the smg_prid shown. When I
was using 'oz dump smg_prid' command it shows all the channels' details. But
all the channels' states are DOWN. why? How can I make it the states to UP?

When I was making the call , the number is busy message was get. The call
was not  at all landed to the freeswitch.

Dial plan Example:
-------------------------------
 <extension name="PRI_Testing">
        <condition field="destination_number" expression="^39114600$">
                <action application="playback"
data="ivr-welcome_to_freeswitch"/>
        </condition>
  </extension>

Please help me...........

*Output Reference:*
http://pastebin.org/79074

On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Dear all,
>
>          I have successfully configured wanpipe with freeswitch. When I was
> the running wancfg_fs script the following files openzap.conf ,
> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
> are created.
>
>          I started the wanrouter command then executed the freeswitch.
>          When I was executing freeswitch mod_openzap.c said the error as
> "Error  for finding the span id. name:PRI_1".
>          But in the openzap.conf and openzap.conf.xml files the span name
> is smg_prid.
>
>          Why the freeswitch is referring the span name as PRI_1 ?
>          Whether this has to configured in anywhere?
>
>          In the freeswitch CLI using oz command I tried to dump the PRI_1
> span id but it said te error  as "PRI_1 is not found".  When I was trying
> the command 'oz dump smg_prid'  all the channel states and details shown.
>
>         It seems that smg_prid span configured in openzap perfectly (Its my
> assumption). Then Why freeswitch is referring the span name as PRI_1.
>
> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>
> Could anyone please help me?
>
> REFERENCE:
>
> openzap.conf
> [span wanpipe smg_prid]
> name => smg_prid
> trunk_type =>e1
> b-channel => 1:1-15
> b-channel => 1:17-31
>
>
> openzap.conf.xml
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>   <settings>
>     <param name="debug" value="0"/>
>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>     <!--<param name="enable-analog-option" value="3-way"/>-->
>   </settings>
>   <boost_spans>
>         <span id="smg_prid">
>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>         <param name="dialplan" value="XML"/>
>         <param name="context" value="default"/>
>          <!-- regex to stop dialing when it matches -->
>         <!--<param name="dial-regex" value="5555"/>-->
>         <!-- regex to stop dialing when it does not match -->
>         <!--<param name="fail-dial-regex" value="^5"/>-->
>         </span>
>  </boost_spans>
>  <analog_spans>
>  </analog_spans>
> </configuration>
>
>
>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From scottferri09 at gmail.com  Tue Jan 19 23:17:50 2010
From: scottferri09 at gmail.com (Scott Fernandez)
Date: Wed, 20 Jan 2010 12:47:50 +0530
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <8ccbff061001192132h344a61a4m58d2973391c8182a@mail.gmail.com>
References: <df97b9c51001190107x39d94eaai7400635f0ac03f86@mail.gmail.com>
	<8ccbff061001192132h344a61a4m58d2973391c8182a@mail.gmail.com>
Message-ID: <df97b9c51001192317p52824e3aq368215cca75f909@mail.gmail.com>

Thanks Dome. Will try it out and get back to you if I come across any
issues.

Regards,
Scott.

On Wed, Jan 20, 2010 at 11:02 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> Please try http://wiki.freeswitch.org/wiki/Webapi
> you can create class and map to webapi.
>
> Dome C.
>
> 2010/1/19 Scott Fernandez <scottferri09 at gmail.com>:
> > Hi,
> >
> > Is there any API modules available for me to initiate a call from .Net
> based
> > application?.
> >
> > The idea is to include the API modules if any with the .NET base classes
> so
> > that the API commands will be made available on it. I know it is doable
> when
> > I use socket programming in .NET in which Telnet session is created.
> > However, this would potentially hamper the performance of the application
> > because of multiple sessions that will be created for each call.
> >
> > Other than that, Is there any Freeswitch API modules (like plug-ins)
> > available in order to include it into the .Net classes and start building
> > the customized application?
> >
> > Any help from any one is highly appreciated.
> >
> > Thanks,
> > Scott
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Wed Jan 20 00:23:43 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 Jan 2010 03:23:43 -0500
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
Message-ID: <073DBFA8-E2A0-43A6-B898-524C8AEAB296@jerris.com>

Down isn't bad, it just means no one is on that channel

On Jan 20, 2010, at 2:13 AM, "Thangappan.M" <thangappan143 at gmail.com>  
wrote:

> I found the error in it. The file name is used as openzap.conf.xml  
> ( smg_prid is specified here) and another file name as  
> openzap.conf.wiki.xml ( PRI_1 span is specified here ). FreeSWITCH  
> referred the PRI_1 span from openzap.conf.wiki.xml  file.
>
> Now the another problem is raised here.
> When I was using oz list command , the details of the smg_prid  
> shown. When I was using 'oz dump smg_prid' command it shows all the  
> channels' details. But all the channels' states are DOWN. why? How  
> can I make it the states to UP?
>
> When I was making the call , the number is busy message was get. The  
> call was not  at all landed to the freeswitch.
>
> Dial plan Example:
> -------------------------------
>  <extension name="PRI_Testing">
>         <condition field="destination_number" expression="^39114600$">
>                 <action application="playback" data="ivr- 
> welcome_to_freeswitch"/>
>         </condition>
>   </extension>
>
> Please help me...........
>
> Output Reference:
> http://pastebin.org/79074
>
> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <thangappan143 at gmail.com 
> > wrote:
> Dear all,
>
>          I have successfully configured wanpipe with freeswitch.  
> When I was the running wancfg_fs script the following files  
> openzap.conf , autoload_confg/openzap.conf.xml , /etc/wanpipe/ 
> wanpipe1.xml, smg_pri.conf are created.
>
>          I started the wanrouter command then executed the freeswitch.
>          When I was executing freeswitch mod_openzap.c said the  
> error as  "Error  for finding the span id. name:PRI_1".
>          But in the openzap.conf and openzap.conf.xml files the span  
> name is smg_prid.
>
>          Why the freeswitch is referring the span name as PRI_1 ?
>          Whether this has to configured in anywhere?
>
>          In the freeswitch CLI using oz command I tried to dump the  
> PRI_1 span id but it said te error  as "PRI_1 is not found".  When I  
> was trying the command 'oz dump smg_prid'  all the channel states  
> and details shown.
>
>         It seems that smg_prid span configured in openzap perfectly  
> (Its my assumption). Then Why freeswitch is referring the span name  
> as PRI_1.
>
> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>
> Could anyone please help me?
>
> REFERENCE:
>
> openzap.conf
> [span wanpipe smg_prid]
> name => smg_prid
> trunk_type =>e1
> b-channel => 1:1-15
> b-channel => 1:17-31
>
>
> openzap.conf.xml
> <configuration name="openzap.conf" description="OpenZAP  
> Configuration">
>   <settings>
>     <param name="debug" value="0"/>
>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>     <!--<param name="enable-analog-option" value="3-way"/>-->
>   </settings>
>   <boost_spans>
>         <span id="smg_prid">
>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>         <param name="dialplan" value="XML"/>
>         <param name="context" value="default"/>
>          <!-- regex to stop dialing when it matches -->
>         <!--<param name="dial-regex" value="5555"/>-->
>         <!-- regex to stop dialing when it does not match -->
>         <!--<param name="fail-dial-regex" value="^5"/>-->
>         </span>
>  </boost_spans>
>  <analog_spans>
>  </analog_spans>
> </configuration>
>
>
>
>
>
>
> -- 
> Regards,
> Thangappan.M
>
>
>
> -- 
> Regards,
> Thangappan.M
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From devel at thom.fr.eu.org  Wed Jan 20 00:42:01 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Wed, 20 Jan 2010 09:42:01 +0100
Subject: [Freeswitch-users] Home setup with home company
In-Reply-To: <4B55F3A0020000E1000003E0@mail.fribert.dk>
References: <4B55F3A0020000E1000003E0@mail.fribert.dk>
Message-ID: <a934cd637ee34ecf507beb261b4442cd@thom.fr.eu.org>



I think you should remove the "continue=true" in the extension
definition, as FS will continue to process the other extensions even after
this one matches, so I you have another "less restrictive" extension that
could match the call and do answer and/or bridge, then it may be processed
instead of this extension. 

Fran?ois 

On Tue, 19 Jan 2010 18:02:08 +0100,
"mailinglist"  wrote:  I have a very small one man constultancy company
that has the occasional call, unfortunately we are getting more spam calls
after hours than real calls during work hours, so I would like to set up a
TOD system.   First step for me is just playing with the TOD example, I've
gotten this far:   

 group/company@${domain_name}"/>

 My idea with this,
was that in the time 9-17 mon-fri, it rings, the rest of the time it's sent
directly to the voicemail. I would of course like to have it not take
messages outside work hours, but that's just refining :-)   But it picks up
the call, and then nothing...   BR Fribse    

 
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From mailinglist at fribert.dk  Wed Jan 20 02:45:27 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Wed, 20 Jan 2010 11:45:27 +0100
Subject: [Freeswitch-users] Svar: Re:  Home setup with home company
Message-ID: <4B56ECD7020000E1000003F1@mail.fribert.dk>

Hi Michael

It's running on pfsense, so it's kinda locked to the version it currently is.
Looks very nice though.
Looking beyond that, is the action / anti-action list corrent?

Best regards
Fribse


>>> Michael Collins  20-01-10 1:53 >>>


On Tue, Jan 19, 2010 at 9:02 AM, mailinglist <mailinglist at fribert.dk> wrote:
I have a very small one man constultancy company that has the occasional call, unfortunately we are getting more spam calls after hours than real calls during work hours, so I would like to set up a TOD system.
 
First step for me is just playing with the TOD example, I've gotten this far:
 

   
      
      
      
         
         
         
         
         group/company@${domain_name}"/>
         
         
         
         
      


My idea with this, was that in the time 9-17 mon-fri, it rings, the rest of the time it's sent directly to the voicemail.
I would of course like to have it not take messages outside work hours, but that's just refining :-)
 
But it picks up the call, and then nothing...
 

We have a much cleaner way of doing TOD and DOW handling. You'll need to get to the latest FreeSWITCH version. Look in conf/dialplan/default.xml for this example:



    
    
      
        
      
    


Use that condition instead of the two conditions you're now using and see if you have better success.

-MC

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From max.bridgewater at gmail.com  Wed Jan 20 03:23:35 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Wed, 20 Jan 2010 06:23:35 -0500
Subject: [Freeswitch-users] Port question again
Message-ID: <c55d46951001200323t18cd8c7bmc1d3df9163bbcb26@mail.gmail.com>

Hey Guys,

Thought the port question was asked a number of times, I couldn't find an
answer to this. So please bear with me. I have a Freeswitch box that is on
the Internet without any sort of NAT. I want to block as much ports as
possible on this box while still allowing Freeswitch to 1) receive calls
from Voip providers and  2) send calls to other VoIP providers.

What port can I block and what ports do I need to let open?

I know 5080 needs to be open. But can I restrict the RTP ports to, say, only
20000?

Thanks so much.
Max.
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From elihayun at gmail.com  Wed Jan 20 04:03:05 2010
From: elihayun at gmail.com (Eli Hayun)
Date: Wed, 20 Jan 2010 14:03:05 +0200
Subject: [Freeswitch-users] Module multicast fail
Message-ID: <4B56F0F9.9090808@savion.huji.ac.il>

Hi

I am trying to load module event_multicast.

I enabled it in modules and compile it. When I run FS I got this error
(ver 1.0.5pre9)


36m2010-01-20 13:26:47.194141 [NOTICE] switch_loadable_module.c:248
Adding Application 'enum'^M
^[[m^[[36m2010-01-20 13:26:47.195106 [NOTICE]
switch_loadable_module.c:270 Adding API Function 'enum'^M
^[[m^[[36m2010-01-20 13:26:47.195356 [NOTICE]
switch_loadable_module.c:270 Adding API Function 'enum_auto'^M
^[[m^[[m2010-01-20 13:26:47.204768 [CONSOLE]
switch_loadable_module.c:889 Successfully Loaded [mod_cdr_csv]^M
^[[m^[[31m2010-01-20 13:26:47.217277 [ERR] mod_event_multicast.c:410
Multicast Error^M
^[[m^[[31m2010-01-20 13:26:47.217345 [CRIT] switch_loadable_module.c:871
Error Loading module /freeswitch-1.0.5/mod/mod_event_multicast.so^M
**Module load routine returned an error**^M
^[[m^[[m2010-01-20 13:26:47.235551 [CONSOLE]
switch_loadable_module.c:889 Successfully Loaded [mod_event_socket]^M
^[[m^[[36m2010-01-20 13:26:47.235613 [NOTICE]
switch_loadable_module.c:248 Adding Application 'socket'^M
^[[m^[[36m2010-01-20 13:26:47.236229 [NOTICE]
switch_loadable_module.c:270 Adding API Function 'event_sink'^M
^[[m^[[36m2010-01-20 13:26:47.335268 [NOTICE] sofia.c:3274 Started
Profile external [sofia_reg_external]^M
^[[m^[[36m2010-01-20 13:26:47.336713 [NOTICE] sofia.c:3274 Started
Profile internal-ipv6 [sofia_reg_internal-ipv6]^M
^[[m^[[36m2010-01-20 13:26:47.339561 [NOTICE] sofia.c:1804 Adding Alias
[132.64.3.86] for profile [internal]^M



Any idea?


I tried to compile that latest trunk too and  got the same error


Thanks


Eli




From a.alalousi at gmail.com  Wed Jan 20 05:14:16 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Wed, 20 Jan 2010 13:14:16 +0000
Subject: [Freeswitch-users] G729 coded issues
In-Reply-To: <4B55E794.6020909@coppice.org>
References: <cd93b3701001190831u4147214cn9832c6f469c6d17e@mail.gmail.com>
	<6D67C7A9-42CD-4F5F-AD18-E9568098FB71@freeswitch.org>
	<4B55E794.6020909@coppice.org>
Message-ID: <cd93b3701001200514p4acc5a5bk26a463164f841ad2@mail.gmail.com>

Gents,

Thanks for the responses.

Now gotten Howler to send me a custom build that will not offer G729b in any
shape or form, as well as the customer to switch off G729b.

Cisco gateways are negotiating correctly by specifying a=fmtp:18 annexb=yes.
If this line is missing, or has an annexb=no, then they will negotiate
G729a.

Now the million dollar question: ${switch_r_sdp} will get you the SDP for
the remote leg. It's returned as a single string in that variable from what
I could tell.

What sort of regex is allowed in the match condition ? can I, for example,
simply use the Perl s/<expression>/<replacement> syntax to search for the
annexb string and get it rewritten ?

My experiments have so far failed in this regard, i.e. to rewrite the
string. Can anyone provide an example ?

I would like to handle codecs in the following way:


   1. receive inbound call
   2. return ringing tone through 3pcc and ring_back
   3. set continue_on_fail=true
   4. set hangup_after_bridge=true
   5. bridge
   6. on failure, transfer to an extension that will look at outcome of
   codec negotiation
   7. rewrite sdp of the A-leg so that remote end point will successfully
   accept it
   8. bridge the call again with the SDP appropriately written
   9. If we fail, then hangup with NORMAL_CIRCUIT_CONGESTION, otherwise just
   let the call be

Regards, and thanks once more.

Ahmed.

2010/1/19 Steve Underwood <steveu at coppice.org>

> On 01/20/2010 12:36 AM, Brian West wrote:
> > g729a is 100% INVALID in the sdp fix the param in your cisco SPA or your
> Linksys SPA phone and it will stop doing that.  Hopefully they'll fix this
> "bug" soon in the cisco phones to not include the a in the sdp.  The fmtp is
> the proper way to specify annex a or any other options for g729.
> >
> Annex A only affects the inner workings of the codec. There is
> absolutely no difference whatsoever between G.729 and G.729A on the
> wire. The SDP has no reason to mention it, and the standards say it
> shouldn't.
> > /b
> >
> > On Jan 19, 2010, at 10:31 AM, Ahmed Naji wrote:
> >
> >
> >> Hi everyone,
> >>
> >> I have the following scenario and a major customer-affecting issue
> thereof.
> >>
> >> Here is the scenario: customer traffic encoded as G.729 from a cisco
> gateway
> >> ->  our FS (G729 passthrough) ->  remote end gw (G729)
> >>
> >> Calls were failing at an alarming rate, so I looked at the debug logs.
> It
> >> transpired that the Cisco is offering G729 annex b, while the remote end
> can
> >> only do G729a.
> >>
> >> Besides changing source or destination preferences, is there a way to
> ensure
> >> that G729a is used end-end ?
> >>
> >> Thanks in advance.
> >>
> >
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From thangappan143 at gmail.com  Wed Jan 20 06:10:12 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Wed, 20 Jan 2010 19:40:12 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
Message-ID: <7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>

      I noticed the 'oz list' output  in that span type is 'ss7 (boost)'.
How can I change this to isdn?


On Wed, Jan 20, 2010 at 12:43 PM, Thangappan.M <thangappan143 at gmail.com>wrote:

> I found the error in it. The file name is used as openzap.conf.xml (
> smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
> PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
> openzap.conf.wiki.xml  file.
>
> Now the another problem is raised here.
> When I was using oz list command , the details of the smg_prid shown. When
> I was using 'oz dump smg_prid' command it shows all the channels' details.
> But all the channels' states are DOWN. why? How can I make it the states to
> UP?
>
> When I was making the call , the number is busy message was get. The call
> was not  at all landed to the freeswitch.
>
> Dial plan Example:
> -------------------------------
>  <extension name="PRI_Testing">
>         <condition field="destination_number" expression="^39114600$">
>                 <action application="playback"
> data="ivr-welcome_to_freeswitch"/>
>         </condition>
>   </extension>
>
> Please help me...........
>
> *Output Reference:*
> http://pastebin.org/79074
>
>
> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>> Dear all,
>>
>>          I have successfully configured wanpipe with freeswitch. When I
>> was the running wancfg_fs script the following files openzap.conf ,
>> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
>> are created.
>>
>>          I started the wanrouter command then executed the freeswitch.
>>          When I was executing freeswitch mod_openzap.c said the error as
>> "Error  for finding the span id. name:PRI_1".
>>          But in the openzap.conf and openzap.conf.xml files the span name
>> is smg_prid.
>>
>>          Why the freeswitch is referring the span name as PRI_1 ?
>>          Whether this has to configured in anywhere?
>>
>>          In the freeswitch CLI using oz command I tried to dump the PRI_1
>> span id but it said te error  as "PRI_1 is not found".  When I was trying
>> the command 'oz dump smg_prid'  all the channel states and details shown.
>>
>>         It seems that smg_prid span configured in openzap perfectly (Its
>> my assumption). Then Why freeswitch is referring the span name as PRI_1.
>>
>> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>>
>> Could anyone please help me?
>>
>> REFERENCE:
>>
>> openzap.conf
>> [span wanpipe smg_prid]
>> name => smg_prid
>> trunk_type =>e1
>> b-channel => 1:1-15
>> b-channel => 1:17-31
>>
>>
>> openzap.conf.xml
>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>   <settings>
>>     <param name="debug" value="0"/>
>>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>>     <!--<param name="enable-analog-option" value="3-way"/>-->
>>   </settings>
>>   <boost_spans>
>>         <span id="smg_prid">
>>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>         <param name="dialplan" value="XML"/>
>>         <param name="context" value="default"/>
>>          <!-- regex to stop dialing when it matches -->
>>         <!--<param name="dial-regex" value="5555"/>-->
>>         <!-- regex to stop dialing when it does not match -->
>>         <!--<param name="fail-dial-regex" value="^5"/>-->
>>         </span>
>>  </boost_spans>
>>  <analog_spans>
>>  </analog_spans>
>> </configuration>
>>
>>
>>
>>
>>
>>
>> --
>> Regards,
>> Thangappan.M
>>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From santiagosoares at gmail.com  Wed Jan 20 06:18:25 2010
From: santiagosoares at gmail.com (Santiago Soares)
Date: Wed, 20 Jan 2010 12:18:25 -0200
Subject: [Freeswitch-users] Port question again
In-Reply-To: <c55d46951001200323t18cd8c7bmc1d3df9163bbcb26@mail.gmail.com>
References: <c55d46951001200323t18cd8c7bmc1d3df9163bbcb26@mail.gmail.com>
Message-ID: <8ea223c01001200618v49a3ade3q621376c084e82f99@mail.gmail.com>

You can use this rule to allow media from any IP:

-A INPUT -m multiport -p udp --dport 16384:32768 -j ACCEPT

And this one to allow signaling:

-A INPUT -s aaa.bbb.ccc.ddd -p udp --dport 5080 -j ACCEPT

Where aaa.bbb.ccc.ddd is the IP address of your VoIP provider.

Santiago Soares


On Wed, Jan 20, 2010 at 9:23 AM, Max Bridgewater
<max.bridgewater at gmail.com> wrote:
> Hey Guys,
>
> Thought the port question was asked a number of times, I couldn't find an
> answer to this. So please bear with me. I have a Freeswitch box that is on
> the Internet without any sort of NAT. I want to block as much ports as
> possible on this box while still allowing Freeswitch to 1) receive calls
> from Voip providers and? 2) send calls to other VoIP providers.
>
> What port can I block and what ports do I need to let open?
>
> I know 5080 needs to be open. But can I restrict the RTP ports to, say, only
> 20000?
>
> Thanks so much.
> Max.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From moises.silva at gmail.com  Wed Jan 20 07:08:50 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 20 Jan 2010 10:08:50 -0500
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
Message-ID: <c4d05cbe1001200708s7f9e37eco750e66eabcd20c18@mail.gmail.com>

Hi

On Wed, Jan 20, 2010 at 9:10 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

>
>       I noticed the 'oz list' output  in that span type is 'ss7 (boost)'.
> How can I change this to isdn?


Ignore the bad name, as long as you run sangoma_prid siganling binary you
get pri signaling, the openzap side does not really know the details of the
signaling and the ss7 boost name is just a misleading name (for historic and
lame reasons).

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From brian at freeswitch.org  Wed Jan 20 07:17:07 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Jan 2010 09:17:07 -0600
Subject: [Freeswitch-users] G729 coded issues
In-Reply-To: <cd93b3701001200514p4acc5a5bk26a463164f841ad2@mail.gmail.com>
References: <cd93b3701001190831u4147214cn9832c6f469c6d17e@mail.gmail.com>
	<6D67C7A9-42CD-4F5F-AD18-E9568098FB71@freeswitch.org>
	<4B55E794.6020909@coppice.org>
	<cd93b3701001200514p4acc5a5bk26a463164f841ad2@mail.gmail.com>
Message-ID: <537338B1-B582-463E-8EFE-7BBED8165D2B@freeswitch.org>

Just an FYI.  Howler doesn't support the FreeSWITCH project in any way, shape or form.  They do not donate any proceeds or help the project at all.  That said.  We have our officially supported G729 coming out soon that will support the project.  I have it in beta if anyone is really interested in testing it please feel free to email me offlist.  We are currently working out how we want to package the lib, binary and module to make installation easy.

Thanks,
/b

On Jan 20, 2010, at 7:14 AM, Ahmed Naji wrote:

> Howler 




From dftoro at yahoo.com  Wed Jan 20 07:17:51 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Wed, 20 Jan 2010 07:17:51 -0800 (PST)
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
Message-ID: <874941.17255.qm@web33502.mail.mud.yahoo.com>

Hi, the answer is yes, you can to use mod_managed wich offer C# managed class to call control http://wiki.freeswitch.org/wiki/Mod_managed. Or using managed ESL (libs/esl/managed) which offer C# managed class to receive and send events and commands to FreeSwitch.
 
Diego Toro
http://lacarretade.blogspot.com/


--- On Wed, 1/20/10, Scott Fernandez <scottferri09 at gmail.com> wrote:

> From: Scott Fernandez <scottferri09 at gmail.com>
> Subject: Re: [Freeswitch-users] Establishing a Call from .Net based application
> To: freeswitch-users at lists.freeswitch.org
> Date: Wednesday, January 20, 2010, 2:17 AM
> Thanks Dome. Will try it out and get back to
> you if I come across any issues.
> 
> Regards,
> Scott.
> 
> On Wed, Jan 20, 2010 at 11:02 AM,
> Dome Charoenyost <dome at tel.co.th>
> wrote:
> 
> Please try http://wiki.freeswitch.org/wiki/Webapi
> 
> 
> you can create class and map to webapi.
> 
> 
> 
> Dome C.
> 
> 
> 
> 2010/1/19 Scott Fernandez <scottferri09 at gmail.com>:
> 
> > Hi,
> 
> >
> 
> > Is there any API modules available for me to initiate
> a call from .Net based
> 
> > application?.
> 
> >
> 
> > The idea is to include the API modules if any with the
> .NET base classes so
> 
> > that the API commands will be made available on it. I
> know it is doable when
> 
> > I use socket programming in .NET in which Telnet
> session is created.
> 
> > However, this would potentially hamper the performance
> of the application
> 
> > because of multiple sessions that will be created for
> each call.
> 
> >
> 
> > Other than that, Is there any Freeswitch API modules
> (like plug-ins)
> 
> > available in order to include it into the .Net classes
> and start building
> 
> > the customized application?
> 
> >
> 
> > Any help from any one is highly appreciated.
> 
> >
> 
> > Thanks,
> 
> > Scott
> 
> >
> 
> >
> _______________________________________________
> 
> > FreeSWITCH-users mailing list
> 
> > FreeSWITCH-users at lists.freeswitch.org
> 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> > http://www.freeswitch.org
> 
> >
> 
> >
> 
> 
> 
> _______________________________________________
> 
> FreeSWITCH-users mailing list
> 
> FreeSWITCH-users at lists.freeswitch.org
> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> http://www.freeswitch.org
> 
> 
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



      



From brian at freeswitch.org  Wed Jan 20 07:18:36 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Jan 2010 09:18:36 -0600
Subject: [Freeswitch-users] Module multicast fail
In-Reply-To: <4B56F0F9.9090808@savion.huji.ac.il>
References: <4B56F0F9.9090808@savion.huji.ac.il>
Message-ID: <2A8BE7A9-EC50-47CD-9FE8-172BF0F97DB6@freeswitch.org>

Please visit http://latest.freeswitch.org  and update to the latest ;)  Its the best you can get to date!  All the preX releases are gone from the download site.

/b

On Jan 20, 2010, at 6:03 AM, Eli Hayun wrote:

> (ver 1.0.5pre9)




From ecasarero at gmail.com  Wed Jan 20 07:10:43 2010
From: ecasarero at gmail.com (Eduardo Casarero)
Date: Wed, 20 Jan 2010 12:10:43 -0300
Subject: [Freeswitch-users] Port question again
In-Reply-To: <8ea223c01001200618v49a3ade3q621376c084e82f99@mail.gmail.com>
References: <c55d46951001200323t18cd8c7bmc1d3df9163bbcb26@mail.gmail.com> 
	<8ea223c01001200618v49a3ade3q621376c084e82f99@mail.gmail.com>
Message-ID: <7d9b3cf21001200710v47bf7b59pf7fc47705396abbb@mail.gmail.com>

Hi list, i'm a brand new freeswitch user (without previous asterisk/voip
experience), after reading all wiki pages, google searchs, etc i need some
help to solve a problem.

configuration:

Freeswitch -> Firewall (nat) -> internet -> Sip Provider

In my current configuration the gateway is REGED and inbound calls (from
provider to freeswitch) works ok with good audio quality. However outbound
calls don't. When i call through the gateway the destination phone rings,
and when is answered there is no audio.

I've check with "show channels" in fs_cli and i cant see any codec in the
read_codec write_codec part, they are blank. I've reviewed all sip profiles
configuration, but obviously i'm missing something.

I will really appreciate any comment,guidance,help,etc. (if someone is in
Buenos Aires/Argentina i can also offer a free beer!)

Thanks in advance.

Eduardo.
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From ecasarero at gmail.com  Wed Jan 20 07:11:55 2010
From: ecasarero at gmail.com (Eduardo Casarero)
Date: Wed, 20 Jan 2010 12:11:55 -0300
Subject: [Freeswitch-users] Problem with outbound calls
Message-ID: <7d9b3cf21001200711h6ce5eda3v1609a1487ff7dc2@mail.gmail.com>

Hi list, i'm a brand new freeswitch user (without previous asterisk/voip
experience), after reading all wiki pages, google searchs, etc i need some
help to solve a problem.

configuration:

Freeswitch -> Firewall (nat) -> internet -> Sip Provider

In my current configuration the gateway is REGED and inbound calls (from
provider to freeswitch) works ok with good audio quality. However outbound
calls don't. When i call through the gateway the destination phone rings,
and when is answered there is no audio.

I've check with "show channels" in fs_cli and i cant see any codec in the
read_codec write_codec part, they are blank. I've reviewed all sip profiles
configuration, but obviously i'm missing something.

I will really appreciate any comment,guidance,help,etc. (if someone is in
Buenos Aires/Argentina i can also offer a free beer!)

Thanks in advance.

Eduardo.
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From stevendt at primrosebank.net  Wed Jan 20 08:17:53 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 20 Jan 2010 16:17:53 -0000
Subject: [Freeswitch-users] Configuration Preservation through Trunk Updates
Message-ID: <21A169E64A904A188DC34E67B1A17651@bp1.ad.bp.com>

Hi,

What is the philosophy/technique for preserving user configuration when doing updates to the latest SVN ?

For example, under Windows, using Visual C++ 2008 Express, the program is built under the "\FreeSwitch\.\Debug" directory with all other FreeSwitch directories below that. When FreeSwitch is installed, the configuration directories, including conf\autoload_configs, conf\dialplan, conf\directory etc, are copied from the distro.

What is the best way of preserving user configuration through future rebuilds, i.e., dialplans, extensions etc. which may have been modified from the defaults ?

Can previously configs just be copied back into the appropriate directory ?

Is compatibility of configs "guaranteed" to be preserved between releases, e.g., 1.0.4 to 1.0.5 or even between SVNs of the same release ?

How do others manage this ?

Regards
Dave
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From brian at freeswitch.org  Wed Jan 20 08:28:09 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Jan 2010 10:28:09 -0600
Subject: [Freeswitch-users] Configuration Preservation through Trunk
	Updates
In-Reply-To: <21A169E64A904A188DC34E67B1A17651@bp1.ad.bp.com>
References: <21A169E64A904A188DC34E67B1A17651@bp1.ad.bp.com>
Message-ID: <C3424C68-4E69-4CA1-A494-2B143C0E8E8F@freeswitch.org>

The best bet is to never touch the installed configs.  And thats what we don on linux.

/b

On Jan 20, 2010, at 10:17 AM, Dave Stevenson wrote:

> What is the philosophy/technique for preserving user configuration when doing updates to the latest SVN ?

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From fdelawarde at wirelessmundi.com  Wed Jan 20 08:29:23 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Wed, 20 Jan 2010 17:29:23 +0100
Subject: [Freeswitch-users] Port question again
In-Reply-To: <8ea223c01001200618v49a3ade3q621376c084e82f99@mail.gmail.com>
References: <c55d46951001200323t18cd8c7bmc1d3df9163bbcb26@mail.gmail.com>
	<8ea223c01001200618v49a3ade3q621376c084e82f99@mail.gmail.com>
Message-ID: <1264004963.14614.30.camel@luna.tc.commsmundi.com>

On Wed, 2010-01-20 at 12:18 -0200, Santiago Soares wrote:
> You can use this rule to allow media from any IP:
> 
> -A INPUT -m multiport -p udp --dport 16384:32768 -j ACCEPT
> 
> And this one to allow signaling:
> 
> -A INPUT -s aaa.bbb.ccc.ddd -p udp --dport 5080 -j ACCEPT

No need to load multiport in that case:

-A INPUT -p udp --dport 5080 -j ACCEPT
-A INPUT -p udp --dport 16384:32768 -j ACCEPT

Equivalent with multiport:

-A INPUT -p udp -m multiport --dports 5080,16384:32768 -j ACCEPT

Fran?ois.




From brian at freeswitch.org  Wed Jan 20 08:32:28 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Jan 2010 10:32:28 -0600
Subject: [Freeswitch-users] Port question again
In-Reply-To: <1264004963.14614.30.camel@luna.tc.commsmundi.com>
References: <c55d46951001200323t18cd8c7bmc1d3df9163bbcb26@mail.gmail.com>
	<8ea223c01001200618v49a3ade3q621376c084e82f99@mail.gmail.com>
	<1264004963.14614.30.camel@luna.tc.commsmundi.com>
Message-ID: <256CB704-0659-46AF-B14A-E48311B17EB2@freeswitch.org>

I'm going to point out that you should open up tcp on 5080 also.  As we actually DO support TCP!

/b

On Jan 20, 2010, at 10:29 AM, Fran?ois Delawarde wrote:

> No need to load multiport in that case:
> 
> -A INPUT -p udp --dport 5080 -j ACCEPT
> -A INPUT -p udp --dport 16384:32768 -j ACCEPT
> 
> Equivalent with multiport:
> 
> -A INPUT -p udp -m multiport --dports 5080,16384:32768 -j ACCEPT
> 
> Fran?ois.




From jcasale at activenetwerx.com  Wed Jan 20 08:36:35 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 20 Jan 2010 16:36:35 +0000
Subject: [Freeswitch-users] Sip video intercom
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9083ACB@prato.activenetwerx.local>

I need to get an intercom integrated into the voip system of a highend home.
That being said, I am looking for a nice looking discrete panel to mount
outside by the front door. Anyone have any experience with these and know of
a model they recommend?

Thanks!
jlc



From wchao at yahoo.com  Wed Jan 20 08:48:08 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Wed, 20 Jan 2010 11:48:08 -0500 (EST)
Subject: [Freeswitch-users] Eavesdrop when using simring
Message-ID: <alpine.LRH.2.00.1001201139480.13391@osgood.microoffice.us>

I have eavesdrop working fine on outbound calls and also inbound calls 
where there is a single DID per IP phone. When I have a DID that rings 
multiple extensions simultaneously, what is the best way to obtain 
information about which extension has picked up the call and store that 
using hash? I can set a variable before I issue the bridge action, like 
so:

<action application="hash" data="insert/${domain_name}-spymap/646xxxyyyy-1000/${uuid}"/>
<action application="bridge" data="user/1001@${domain_name},user/1002@${domain_name},user/1003@${domain_name},user/1004@${domain_name}"/>

However, that doesn't tell me who actually picked up, so at best I can 
allow users to eavesdrop on the last incoming call to the main DID, not 
the last incoming call to a particular extension. Is there something I can 
do in the bridge that will cause it to set a variable once it knows which 
extension has picked up the call?



From jerry.richards at teotech.com  Wed Jan 20 08:59:40 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 20 Jan 2010 08:59:40 -0800
Subject: [Freeswitch-users] Freeswitch Test Tool
Message-ID: <94D82278DEC141DD9B4B951C67848708@greyhawk.tonecommander.com>


Can anyone recommend a good SIP Test Tool to test the Freeswitch PBX?  I
would like one capable of both load tests and also various call scenarios.

Thanks,
Jerry




From gmaruzz at celliax.org  Wed Jan 20 09:16:30 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 20 Jan 2010 18:16:30 +0100
Subject: [Freeswitch-users] Freeswitch Test Tool
In-Reply-To: <94D82278DEC141DD9B4B951C67848708@greyhawk.tonecommander.com>
References: <94D82278DEC141DD9B4B951C67848708@greyhawk.tonecommander.com>
Message-ID: <7b197bef1001200916yaf64a47j74b8fa7577fac53d@mail.gmail.com>

Sipp (http://sipp.sourceforge.net/)?



On Wed, Jan 20, 2010 at 5:59 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:
>
> Can anyone recommend a good SIP Test Tool to test the Freeswitch PBX? ?I
> would like one capable of both load tests and also various call scenarios.
>
> Thanks,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From frank at carmickle.com  Wed Jan 20 09:27:26 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Wed, 20 Jan 2010 12:27:26 -0500
Subject: [Freeswitch-users] Port question again
In-Reply-To: <7d9b3cf21001200710v47bf7b59pf7fc47705396abbb@mail.gmail.com>
References: <c55d46951001200323t18cd8c7bmc1d3df9163bbcb26@mail.gmail.com>
	<8ea223c01001200618v49a3ade3q621376c084e82f99@mail.gmail.com>
	<7d9b3cf21001200710v47bf7b59pf7fc47705396abbb@mail.gmail.com>
Message-ID: <20100120172725.GD7006@base.carmickle.com>

Hello

On Wed, Jan 20, Eduardo Casarero wrote:
> Hi list, i'm a brand new freeswitch user (without previous asterisk/voip
> experience), after reading all wiki pages, google searchs, etc i need some
> help to solve a problem.
> 
> configuration:
> 
> Freeswitch -> Firewall (nat) -> internet -> Sip Provider
> 
> In my current configuration the gateway is REGED and inbound calls (from
> provider to freeswitch) works ok with good audio quality. However outbound
> calls don't. When i call through the gateway the destination phone rings,
> and when is answered there is no audio.
> 
> I've check with "show channels" in fs_cli and i cant see any codec in the
> read_codec write_codec part, they are blank. I've reviewed all sip profiles
> configuration, but obviously i'm missing something.

Sounds to me like your firewall is blocking outbound ports.  If it's a linux machine you'll want something like

-A OUTPUT -p udp --dport 16384:32768 -j ACCEPT

> 
> I will really appreciate any comment,guidance,help,etc. (if someone is in
> Buenos Aires/Argentina i can also offer a free beer!)

I am not but I'd sure love to try your local beer.

HTH
--FC



From stevendt at primrosebank.net  Wed Jan 20 09:29:03 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 20 Jan 2010 17:29:03 -0000
Subject: [Freeswitch-users] Configuration Preservation through
	TrunkUpdates
References: <21A169E64A904A188DC34E67B1A17651@bp1.ad.bp.com>
	<C3424C68-4E69-4CA1-A494-2B143C0E8E8F@freeswitch.org>
Message-ID: <5A041CBBECB441D094B86C935CC6C792@bp1.ad.bp.com>

Hi Thanks Brian,

OK, but I'm sure everyone has their own requirements for dialplan actions, extensions etc. If I understand you right, you're saying that I should leave all the defaults (dialplans, extensions etc.) in place and do anything specific to my installation in separate files ?

I think that I can see how that would work :-

Provided that I use extensions not already defined in dialplan\default.xml. then any extensions that I add won't be touched by FS.
What if I redefined a pre-defined extension, would it take precedence over the data in dialpan\default.xml ?

Any user dialplan actions would go into nn_xxxx.xml files in the dialplan\defaults dir and be processed after dialpan\default.xml ?

If I understand correctly, I'll head off and put things right !

regards
Dave


  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, January 20, 2010 4:28 PM
  Subject: Re: [Freeswitch-users] Configuration Preservation through TrunkUpdates


  The best bet is to never touch the installed configs.  And thats what we don on linux.


  /b


  On Jan 20, 2010, at 10:17 AM, Dave Stevenson wrote:


    What is the philosophy/technique for preserving user configuration when doing updates to the latest SVN ?




------------------------------------------------------------------------------


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From brian at freeswitch.org  Wed Jan 20 09:32:42 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Jan 2010 11:32:42 -0600
Subject: [Freeswitch-users] Port question again
In-Reply-To: <20100120172725.GD7006@base.carmickle.com>
References: <c55d46951001200323t18cd8c7bmc1d3df9163bbcb26@mail.gmail.com>
	<8ea223c01001200618v49a3ade3q621376c084e82f99@mail.gmail.com>
	<7d9b3cf21001200710v47bf7b59pf7fc47705396abbb@mail.gmail.com>
	<20100120172725.GD7006@base.carmickle.com>
Message-ID: <95B92BEF-14BA-4274-9F0C-F6DD1F18E665@freeswitch.org>

I have an even better solution:

install MiniUPnP daemon.

/b

On Jan 20, 2010, at 11:27 AM, Frank Carmickle wrote:

> Hello




From brian at freeswitch.org  Wed Jan 20 09:34:39 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Jan 2010 11:34:39 -0600
Subject: [Freeswitch-users] Configuration Preservation through
	TrunkUpdates
In-Reply-To: <5A041CBBECB441D094B86C935CC6C792@bp1.ad.bp.com>
References: <21A169E64A904A188DC34E67B1A17651@bp1.ad.bp.com>
	<C3424C68-4E69-4CA1-A494-2B143C0E8E8F@freeswitch.org>
	<5A041CBBECB441D094B86C935CC6C792@bp1.ad.bp.com>
Message-ID: <1F2B2E59-6C17-4DB8-A197-E57D2A804132@freeswitch.org>

You should NEVER install anything into a config folder if one already exists.  But you're free to do what you want locally but we will NEVER allow the install process to install extra files or overwrite existing configs its bad behavior to do so.

/b

On Jan 20, 2010, at 11:29 AM, Dave Stevenson wrote:

> Hi Thanks Brian,
>  
> OK, but I'm sure everyone has their own requirements for dialplan actions, extensions etc. If I understand you right, you're saying that I should leave all the defaults (dialplans, extensions etc.) in place and do anything specific to my installation in separate files ?
>  
> I think that I can see how that would work :-
>  
> Provided that I use extensions not already defined in dialplan\default.xml. then any extensions that I add won't be touched by FS.
> What if I redefined a pre-defined extension, would it take precedence over the data in dialpan\default.xml ?
>  
> Any user dialplan actions would go into nn_xxxx.xml files in the dialplan\defaults dir and be processed after dialpan\default.xml ?
>  
> If I understand correctly, I'll head off and put things right !
>  
> regards
> Dave
>  

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From anthony.minessale at gmail.com  Wed Jan 20 09:50:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 20 Jan 2010 11:50:24 -0600
Subject: [Freeswitch-users] Bypass Media True Disables MOH
In-Reply-To: <E55CF9FDA2AC406B8799A0081BCBA4DD@greyhawk.tonecommander.com>
References: <AAA8629BCAF448659E9B91C20E75CCC6@greyhawk.tonecommander.com>
	<9BF355CF-C633-4BF5-BB8B-642DD81936D1@freeswitch.org>
	<E55CF9FDA2AC406B8799A0081BCBA4DD@greyhawk.tonecommander.com>
Message-ID: <191c3a031001200950i228b9198o78cfaa7185fd7eb0@mail.gmail.com>

I would say $500 bounty to make it go back to bypassing after the hold is
over.
contact me directly if you wish to proceed.


On Tue, Jan 19, 2010 at 9:50 AM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> We are willing to pay a bounty for this.  What amount would you suggest?
>  We
> would like the media to normally go directly between the endpoints, but
> when
> a call is put on-hold, we would like the other end should hear MOH.
>
> Thanks,
> Jerry
>
>
> -----Original Message-----
> From: Brian West [mailto:brian at freeswitch.org]
> Sent: Tuesday, December 29, 2009 1:12 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Bypass Media True Disables MOH
>
> <param name="media-option" value="resume-media-on-hold"/>
>
> But it doesn't go back to bypass after....   Maybe you can post a bounty
> for
> that functionality.
>
> /b
>
> On Dec 29, 2009, at 2:42 PM, Jerry Richards wrote:
>
> >
> > When I uncomment the following tag, internally held calls no longer
> > hear MOH.
> >
> > <param name="inbound-bypass-media" value="true"/>
> >
> > Is there a way to have the above uncommented and still provide MOH to
> > held calls?
> >
> > Best Regards,
> > Jerry
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From linux4michelle at tamay-dogan.net  Wed Jan 20 10:07:20 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Wed, 20 Jan 2010 19:07:20 +0100
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
In-Reply-To: <2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>
References: <20100119103139.GR4767@tamay-dogan.net>
	<2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>
Message-ID: <20100120180720.GH4767@tamay-dogan.net>

Hello,

Am 2010-01-20 01:28:40, schrieb Michael Jerris:
> There is a debian dir in tree for our packages.  I suspect that our
> packages are quite far from meeting the requirements of pretty much
> any distro so for now, we will at least have packages for the next
> release available soon after release.  We don't yet have a box for
> debian instances for the build farm so we do not build the svn
> snapshot pacakges for any deb distros.

Hmmm, currently I have only one fixed IP and some VServers and  PBuilder
runing on i386 and ARM (only a small Ti Sitara AM3517 with 256 MB memory
plus SATA drive)...

Also I am trying to relocate back to Germany...
Maybe I could do the Job for Debian (i386 ARMEL) and Ubuntu (i386)

My Website is currently a backup fro 2008/12 and 2009/03 because  I  was
offine since 2009-07-23 du to my fuckingbusiness partner...

Maybe it work:
    http://www.debian.tamay-dogan.net/

I will try to reinstall the PBuilder interface which  allow  uploads  of
sources/configs and autobuilding.  Also  I  like  to  include  an  auto-
checkout from "svn" and "git" so, the author of the software should give
a signal to my interface and the build is done automaticaly.

Also I am Package Maintainer of some Debian packages...

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From stevendt at primrosebank.net  Wed Jan 20 11:44:09 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 20 Jan 2010 19:44:09 -0000
Subject: [Freeswitch-users] Configuration Preservation
	throughTrunkUpdates
References: <21A169E64A904A188DC34E67B1A17651@bp1.ad.bp.com><C3424C68-4E69-4CA1-A494-2B143C0E8E8F@freeswitch.org><5A041CBBECB441D094B86C935CC6C792@bp1.ad.bp.com>
	<1F2B2E59-6C17-4DB8-A197-E57D2A804132@freeswitch.org>
Message-ID: <DF8D6745E8E1492A9D2E345A6B42D768@bp1.ad.bp.com>

Brian,

Following on from before, there's one item that I can't see how to do outside of modifying the directory\default.xml file, and that is setting up call groups.

I thought that I'd perhaps be able to do something similar to creating user dial plans and create a new file in directory\default\  which would be loaded before the other extensions, i.e., called something like 00_groups.xml and have the call group created there. (My test file is shown below). That did not seem to work, am I on the right lines or should custom groups get created somewhere else ?

regards
Dave


<!-- Group Configuration File -->

<include>
    <groups>
      <group name="primrose">
         <users>
           <user id="101" type="pointer"/>
           <user id="111" type="pointer"/>
         </users>
      </group>

    </groups>

</include>



  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, January 20, 2010 5:34 PM
  Subject: Re: [Freeswitch-users] Configuration Preservation throughTrunkUpdates


  You should NEVER install anything into a config folder if one already exists.  But you're free to do what you want locally but we will NEVER allow the install process to install extra files or overwrite existing configs its bad behavior to do so.


  /b


  On Jan 20, 2010, at 11:29 AM, Dave Stevenson wrote:


    Hi Thanks Brian,

    OK, but I'm sure everyone has their own requirements for dialplan actions, extensions etc. If I understand you right, you're saying that I should leave all the defaults (dialplans, extensions etc.) in place and do anything specific to my installation in separate files ?

    I think that I can see how that would work :-

    Provided that I use extensions not already defined in dialplan\default.xml. then any extensions that I add won't be touched by FS.
    What if I redefined a pre-defined extension, would it take precedence over the data in dialpan\default.xml ?

    Any user dialplan actions would go into nn_xxxx.xml files in the dialplan\defaults dir and be processed after dialpan\default.xml ?

    If I understand correctly, I'll head off and put things right !

    regards
    Dave





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Wed Jan 20 12:00:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 20 Jan 2010 14:00:07 -0600
Subject: [Freeswitch-users] Freeswitch Test Tool
In-Reply-To: <94D82278DEC141DD9B4B951C67848708@greyhawk.tonecommander.com>
References: <94D82278DEC141DD9B4B951C67848708@greyhawk.tonecommander.com>
Message-ID: <191c3a031001201200t9bf3505o45bb22de368ddfe2@mail.gmail.com>

another FS box

On Wed, Jan 20, 2010 at 10:59 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

>
> Can anyone recommend a good SIP Test Tool to test the Freeswitch PBX?  I
> would like one capable of both load tests and also various call scenarios.
>
> Thanks,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Jan 20 12:09:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 20 Jan 2010 14:09:48 -0600
Subject: [Freeswitch-users] Eavesdrop when using simring
In-Reply-To: <alpine.LRH.2.00.1001201139480.13391@osgood.microoffice.us>
References: <alpine.LRH.2.00.1001201139480.13391@osgood.microoffice.us>
Message-ID: <191c3a031001201209y53d68c83t8ba066b63ff4b723@mail.gmail.com>

maybe api_on_answer var?

On Wed, Jan 20, 2010 at 10:48 AM, Wellie Chao <wchao at yahoo.com> wrote:

> I have eavesdrop working fine on outbound calls and also inbound calls
> where there is a single DID per IP phone. When I have a DID that rings
> multiple extensions simultaneously, what is the best way to obtain
> information about which extension has picked up the call and store that
> using hash? I can set a variable before I issue the bridge action, like
> so:
>
> <action application="hash"
> data="insert/${domain_name}-spymap/646xxxyyyy-1000/${uuid}"/>
> <action application="bridge" data="user/1001@${domain_name},user/1002@
> ${domain_name},user/1003@${domain_name},user/1004@${domain_name}"/>
>
> However, that doesn't tell me who actually picked up, so at best I can
> allow users to eavesdrop on the last incoming call to the main DID, not
> the last incoming call to a particular extension. Is there something I can
> do in the bridge that will cause it to set a variable once it knows which
> extension has picked up the call?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From larclap at yahoo.com  Wed Jan 20 13:02:45 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 20 Jan 2010 13:02:45 -0800
Subject: [Freeswitch-users] Debug message on console?
Message-ID: <00a801ca9a13$ea603470$bf209d50$@com>

For about the last few weeks I've noticed the following message on the
console:

 

[DEBUG] sofia_reg.c:1815 adding X-Real-IP => 192.168.10.104 to xml_curl
request

 

I am sorry but cannot understand the code. The endpoint is a SNOM 320 at
7.3.14. It is registered for two extensions. The message is output twice
every 5 minutes.

 

Does this message indicate a problem? If so, how can I correct it?

 

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

FreeSWITCH v16385

 

Thanks, Lars

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From mike at van.lammeren.net  Wed Jan 20 13:18:44 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Wed, 20 Jan 2010 16:18:44 -0500
Subject: [Freeswitch-users] Question about Lua script: How do I detect when
	someone picks up the phone?
Message-ID: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>

Hello!

Some day, I'll probably laugh at asking this question, but today I can't
figure it out.

I've written a Lua script that listens for a call, then dials a phone number
to a second person. It plays a message, then prompts the second person to
hit pound to connect. If the second person hits pound, then it bridges the
two calls together.

All that works great, but I can't figure out how to get the session for the
second person to wait until that person answers.

I'm using FreeSWITCH 1.0.4, and although there is a *getState* function
documented in the wiki, it doesn't seem to exist for me.

Any help would be appreciated!


Mike van Lammeren
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From rob4manhere at gmail.com  Wed Jan 20 13:32:18 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Wed, 20 Jan 2010 15:32:18 -0600
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
Message-ID: <e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>

Hi Mike,

I don't think v1.0.4 is supported any longer.   You'll have better luck
getting assistance by upgrading to trunk or the latest tar and reporting
back.

Good luck!
Rob

On Wed, Jan 20, 2010 at 3:18 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Hello!
>
> Some day, I'll probably laugh at asking this question, but today I can't
> figure it out.
>
> I've written a Lua script that listens for a call, then dials a phone
> number to a second person. It plays a message, then prompts the second
> person to hit pound to connect. If the second person hits pound, then it
> bridges the two calls together.
>
> All that works great, but I can't figure out how to get the session for the
> second person to wait until that person answers.
>
> I'm using FreeSWITCH 1.0.4, and although there is a *getState* function
> documented in the wiki, it doesn't seem to exist for me.
>
> Any help would be appreciated!
>
>
> Mike van Lammeren
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at van.lammeren.net  Wed Jan 20 13:45:23 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Wed, 20 Jan 2010 16:45:23 -0500
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
	<e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
Message-ID: <5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>

Hi Rob!

Unfortunately, I have the next few weeks to complete this part of the
project. Without a stable release of 1.0.5, I have to stick with 1.0.4. I'm
sure that 1.0.4 can detect and report when a phone is picked up. It's just
that I can't figure out how to get that information! Either that, or I have
something mis-configured.

Mike van Lammeren


On Wed, Jan 20, 2010 at 4:32 PM, Rob Forman <rob4manhere at gmail.com> wrote:

> Hi Mike,
>
> I don't think v1.0.4 is supported any longer.   You'll have better luck
> getting assistance by upgrading to trunk or the latest tar and reporting
> back.
>
> Good luck!
> Rob
>
> On Wed, Jan 20, 2010 at 3:18 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:
>
>> Hello!
>>
>> Some day, I'll probably laugh at asking this question, but today I can't
>> figure it out.
>>
>> I've written a Lua script that listens for a call, then dials a phone
>> number to a second person. It plays a message, then prompts the second
>> person to hit pound to connect. If the second person hits pound, then it
>> bridges the two calls together.
>>
>> All that works great, but I can't figure out how to get the session for
>> the second person to wait until that person answers.
>>
>> I'm using FreeSWITCH 1.0.4, and although there is a *getState* function
>> documented in the wiki, it doesn't seem to exist for me.
>>
>> Any help would be appreciated!
>>
>>
>> Mike van Lammeren
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at van.lammeren.net  Wed Jan 20 14:06:38 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Wed, 20 Jan 2010 17:06:38 -0500
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
	<e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
	<5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>
Message-ID: <5d2828f1001201406p4154b98ald1af1c5c25f59337@mail.gmail.com>

So, I've been reading about early media in the wiki, and have made a little
progress, which leads to more questions.

I understand now why a call is considered connected before one person has
picked up the phone. I am also able to get my script to wait for the phone
to be picked up, by setting the ignore_early_media variable when starting a
new session, like this:

customerSession =
freeswitch.Session("{ignore_early_media=true}sofia/gateway/example.com/" ..
customerPhoneNumber)


After that line, the script waits for the other phone to be picked up.

However, now I wonder what to do with calls that don't complete, get busy
signals, etc.

What do people do in this case? The only related example I can find on the
web is for a javascript dialer, which doesn't address any of these cases.

Early Media: http://wiki.freeswitch.org/wiki/Early_media
ignore_early_media variable:
http://wiki.freeswitch.org/wiki/Variable_ignore_early_media
<http://wiki.freeswitch.org/wiki/Variable_ignore_early_media>javascript
dialer: http://alexn.org/docs/dialer.html


<http://alexn.org/docs/dialer.html>Mike van Lammeren

On Wed, Jan 20, 2010 at 4:45 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Hi Rob!
>
> Unfortunately, I have the next few weeks to complete this part of the
> project. Without a stable release of 1.0.5, I have to stick with 1.0.4. I'm
> sure that 1.0.4 can detect and report when a phone is picked up. It's just
> that I can't figure out how to get that information! Either that, or I have
> something mis-configured.
>
> Mike van Lammeren
>
>
> On Wed, Jan 20, 2010 at 4:32 PM, Rob Forman <rob4manhere at gmail.com> wrote:
>
>> Hi Mike,
>>
>> I don't think v1.0.4 is supported any longer.   You'll have better luck
>> getting assistance by upgrading to trunk or the latest tar and reporting
>> back.
>>
>> Good luck!
>> Rob
>>
>> On Wed, Jan 20, 2010 at 3:18 PM, Mike van Lammeren <mike at van.lammeren.net
>> > wrote:
>>
>>>  Hello!
>>>
>>> Some day, I'll probably laugh at asking this question, but today I can't
>>> figure it out.
>>>
>>> I've written a Lua script that listens for a call, then dials a phone
>>> number to a second person. It plays a message, then prompts the second
>>> person to hit pound to connect. If the second person hits pound, then it
>>> bridges the two calls together.
>>>
>>> All that works great, but I can't figure out how to get the session for
>>> the second person to wait until that person answers.
>>>
>>> I'm using FreeSWITCH 1.0.4, and although there is a *getState* function
>>> documented in the wiki, it doesn't seem to exist for me.
>>>
>>> Any help would be appreciated!
>>>
>>>
>>> Mike van Lammeren
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From larclap at yahoo.com  Wed Jan 20 14:15:38 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 20 Jan 2010 14:15:38 -0800
Subject: [Freeswitch-users] Additional endpoints
Message-ID: <00d701ca9a1e$18afa0e0$4a0ee2a0$@com>

I know this has been answered before, but I cannot find it.

 

How do I setup more than the default 20 endpoints (1000-1019)?  Do I extend
the definition in dialplan/public.xml (public_extensions) and add the extra
in directory/default?

 

Thanks, Lars

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From anthony.minessale at gmail.com  Wed Jan 20 14:28:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 20 Jan 2010 16:28:37 -0600
Subject: [Freeswitch-users] Debug message on console?
In-Reply-To: <00a801ca9a13$ea603470$bf209d50$@com>
References: <00a801ca9a13$ea603470$bf209d50$@com>
Message-ID: <191c3a031001201428m72bcad24u6a29b0014d00f6a0@mail.gmail.com>

no it's overly chatty, i will move it up to debug level 10 so you won't see
it.


On Wed, Jan 20, 2010 at 3:02 PM, Lars Zeb <larclap at yahoo.com> wrote:

>  For about the last few weeks I?ve noticed the following message on the
> console:
>
>
>
> [DEBUG] sofia_reg.c:1815 adding X-Real-IP => 192.168.10.104 to xml_curl
> request
>
>
>
> I am sorry but cannot understand the code. The endpoint is a SNOM 320 at
> 7.3.14. It is registered for two extensions. The message is output twice
> every 5 minutes.
>
>
>
> Does this message indicate a problem? If so, how can I correct it?
>
>
>
> Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
> i386 GNU/Linux
>
> FreeSWITCH v16385
>
>
>
> Thanks, Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From rob4manhere at gmail.com  Wed Jan 20 14:29:00 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Wed, 20 Jan 2010 16:29:00 -0600
Subject: [Freeswitch-users] Additional endpoints
In-Reply-To: <00d701ca9a1e$18afa0e0$4a0ee2a0$@com>
References: <00d701ca9a1e$18afa0e0$4a0ee2a0$@com>
Message-ID: <e15d328c1001201429m6130a132vccc8e373d65cd1f6@mail.gmail.com>

Hi Lars,

For endpoint authentication, yes, copy and add more entries to
./conf/directory/default/.

For internal dialing, you'd need to change the regex expression under
"Local_Extension" in conf/dialplan/default.xml, or add additional extensions
under conf/dialplan/default/.

Rob

On Wed, Jan 20, 2010 at 4:15 PM, Lars Zeb <larclap at yahoo.com> wrote:

>  I know this has been answered before, but I cannot find it.
>
>
>
> How do I setup more than the default 20 endpoints (1000-1019)?  Do I extend
> the definition in dialplan/public.xml (public_extensions) and add the extra
> in directory/default?
>
>
>
> Thanks, Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jerry.richards at teotech.com  Wed Jan 20 14:44:24 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 20 Jan 2010 14:44:24 -0800
Subject: [Freeswitch-users] Presence Not Working After Version 1.0.5pre9?
In-Reply-To: <mailman.5334.1264017641.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.5334.1264017641.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <C726F91949324BDFA34596256B6E102A@greyhawk.tonecommander.com>

Does anyone know why I do not see NOTIFY messages with presence status being
sent out from FS for two Bria softphones?  It used to work in my old version
1.0.5pre9.  I upgraded Freeswitch on Jan 12 and Jan 20 and edited in my XML
configurations, but I do not see the NOTIFY messages since version
1.0.5pre9.

I have mostly default configuration and I added the manage-presence=true
setting.  I posted a pastebin at http://pastebin.freeswitch.org/11867.

Can anyone tell why this isn't working?

Best Regards,
Jerry




From brian at freeswitch.org  Wed Jan 20 14:50:52 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Jan 2010 16:50:52 -0600
Subject: [Freeswitch-users] Presence Not Working After Version 1.0.5pre9?
In-Reply-To: <C726F91949324BDFA34596256B6E102A@greyhawk.tonecommander.com>
References: <mailman.5334.1264017641.2495.freeswitch-users@lists.freeswitch.org>
	<C726F91949324BDFA34596256B6E102A@greyhawk.tonecommander.com>
Message-ID: <4CFA4AF5-6E5A-40AB-816F-25EE1426F3A0@freeswitch.org>

PRE9 is no longer supported please use LATEST.  http://latest.freeswitch.org

/b

On Jan 20, 2010, at 4:44 PM, Jerry Richards wrote:

> Does anyone know why I do not see NOTIFY messages with presence status being
> sent out from FS for two Bria softphones?  It used to work in my old version
> 1.0.5pre9.  I upgraded Freeswitch on Jan 12 and Jan 20 and edited in my XML
> configurations, but I do not see the NOTIFY messages since version
> 1.0.5pre9.
> 
> I have mostly default configuration and I added the manage-presence=true
> setting.  I posted a pastebin at http://pastebin.freeswitch.org/11867.
> 
> Can anyone tell why this isn't working?
> 
> Best Regards,
> Jerry

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From dftoro at yahoo.com  Wed Jan 20 15:31:35 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Wed, 20 Jan 2010 15:31:35 -0800 (PST)
Subject: [Freeswitch-users] mutiple playback files (unescape_char) Windows
Message-ID: <24068.48012.qm@web33507.mail.mud.yahoo.com>

Greetings

I have a next section dial plan:

<action application="set" data="sound_prefix=$${base_dir}\sounds\es\co\callie\" />
<action application="set" data="playback_delimiter=!"/>

<action application="playback" data="misc\8000\serror.wav!misc\8000\provide_reference_number.wav!digits\8000\5.wav" />  
 
....

I have a problem using multiple playback files on Windows, the path  misc\8000\serror.wav is changed by misc\8000 serror.wav.  I check C code on switch_utils.c, cleanup_separated_string function call to unescape_char function which change \s by ' '. This is correct, but on Windows '\' is the path separator, so is not possible to use  '\s', '\n'... into path file. I think this is possible to fix it.

Thanks

Diego Toro
http://lacarretade.blogspot.com/


      



From anthony.minessale at gmail.com  Wed Jan 20 15:31:53 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 20 Jan 2010 17:31:53 -0600
Subject: [Freeswitch-users] Presence Not Working After Version 1.0.5pre9?
In-Reply-To: <C726F91949324BDFA34596256B6E102A@greyhawk.tonecommander.com>
References: <mailman.5334.1264017641.2495.freeswitch-users@lists.freeswitch.org>
	<C726F91949324BDFA34596256B6E102A@greyhawk.tonecommander.com>
Message-ID: <191c3a031001201531g7c78cb9fw1ed1a2ba07f5773c@mail.gmail.com>

try turning on sip debug and console loglevel debug

sofia loglevel all 9
console loglevel debug

Did you try manually running the same sql stmts from the sqlite3 app?
maybe you have something misconfigured.


On Wed, Jan 20, 2010 at 4:44 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> Does anyone know why I do not see NOTIFY messages with presence status
> being
> sent out from FS for two Bria softphones?  It used to work in my old
> version
> 1.0.5pre9.  I upgraded Freeswitch on Jan 12 and Jan 20 and edited in my XML
> configurations, but I do not see the NOTIFY messages since version
> 1.0.5pre9.
>
> I have mostly default configuration and I added the manage-presence=true
> setting.  I posted a pastebin at http://pastebin.freeswitch.org/11867.
>
> Can anyone tell why this isn't working?
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From tom at tomcarlson.com  Wed Jan 20 17:38:04 2010
From: tom at tomcarlson.com (Tom Carlson)
Date: Wed, 20 Jan 2010 17:38:04 -0800
Subject: [Freeswitch-users] luacurl vs io.popen("curl ...") vs
	api:executeString( "curl ..")
Message-ID: <21e9d36c1001201738i2b2f0106w175edf4d31494f69@mail.gmail.com>

I was hoping that those knowledgeable in these things could tell me,
strategically, which curl method would be most cpu friendly for me to use to
access my restful web service.


   1. Use freeSWITCH's mod_curl and issue an api:executeString( "curl ..")
   command
   2. Use the luacurl library from http://luaforge.net/projects/luacurl/
   3. Use linux's curl through an io.popen() command


I was intending to use the luacurl library, but it occured to me that
mod_curl might have been specially engineered to provide better efficiency.

Thanks.
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From rupa at rupa.com  Wed Jan 20 18:03:39 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 20 Jan 2010 20:03:39 -0600
Subject: [Freeswitch-users] luacurl vs io.popen("curl ...") vs
	api:executeString( "curl ..")
In-Reply-To: <21e9d36c1001201738i2b2f0106w175edf4d31494f69@mail.gmail.com>
References: <21e9d36c1001201738i2b2f0106w175edf4d31494f69@mail.gmail.com>
Message-ID: <b39273531001201803g7aae7227x19245abfde31f802@mail.gmail.com>

1 and 2  both use libcurl.  2 might be better for you IF it doesn't
introduce memory leaks or other instabilities.  mod_curl is just a simple
wrapper around libcurl so nothing special.

On Wed, Jan 20, 2010 at 7:38 PM, Tom Carlson <tom at tomcarlson.com> wrote:

> I was hoping that those knowledgeable in these things could tell me,
> strategically, which curl method would be most cpu friendly for me to use to
> access my restful web service.
>
>
>    1. Use freeSWITCH's mod_curl and issue an api:executeString( "curl ..")
>    command
>    2. Use the luacurl library from http://luaforge.net/projects/luacurl/
>    3. Use linux's curl through an io.popen() command
>
>
> I was intending to use the luacurl library, but it occured to me that
> mod_curl might have been specially engineered to provide better efficiency.
>
> Thanks.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From larclap at yahoo.com  Wed Jan 20 20:48:15 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 20 Jan 2010 20:48:15 -0800
Subject: [Freeswitch-users] Debug message on console?
In-Reply-To: <191c3a031001201428m72bcad24u6a29b0014d00f6a0@mail.gmail.com>
References: <00a801ca9a13$ea603470$bf209d50$@com>
	<191c3a031001201428m72bcad24u6a29b0014d00f6a0@mail.gmail.com>
Message-ID: <019501ca9a54$f23af180$d6b0d480$@com>

Thanks for changing the message, Anthony.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, January 20, 2010 2:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Debug message on console?

 

no it's overly chatty, i will move it up to debug level 10 so you won't see
it.



On Wed, Jan 20, 2010 at 3:02 PM, Lars Zeb <larclap at yahoo.com> wrote:

For about the last few weeks I've noticed the following message on the
console:

 

[DEBUG] sofia_reg.c:1815 adding X-Real-IP => 192.168.10.104 to xml_curl
request

 

I am sorry but cannot understand the code. The endpoint is a SNOM 320 at
7.3.14. It is registered for two extensions. The message is output twice
every 5 minutes.

 

Does this message indicate a problem? If so, how can I correct it?

 

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

FreeSWITCH v16385

 

Thanks, Lars


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From thangappan143 at gmail.com  Wed Jan 20 21:04:21 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Thu, 21 Jan 2010 10:34:21 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
Message-ID: <7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>

OpenZap is loading the ss7 signalling type. As per your concern openzap does
not know the details of the signalling then how it is loading the ss7_boost
libraries?

FreeSWITCH log:
-----------------------------
2010-01-21 10:10:46.707844 [INFO] zap_io.c:2374 Configured 30 channel(s)
2010-01-21 10:10:46.708600 [INFO] zap_io.c:2468 Loading SIG from
/usr/local/freeswitch/mod/ozmod_ss7_boost.so
2010-01-21 10:10:46.709031 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost'
2010-01-21 10:10:46.709466 [DEBUG] ss7_boost_client.c:124 Creating L=
127.0.0.65:53000 R=127.0.0.66:53000
2010-01-21 10:10:46.709834 [DEBUG] ss7_boost_client.c:124 Creating L=
127.0.0.65:53001 R=127.0.0.66:53001
2010-01-21 10:10:46.710424 [WARNING] ss7_boost_client.c:244 TX EVENT (P):
SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0

The signalling type might be anything but when I used the oz list command it
showed the span details. But I am unable to make a inbound and outbound
call.

Outbound call result:
============
>  originate openzap/smg_prid/a/9940464753 openzap/smg_prid/a/9843171457
-ERR NORMAL_CIRCUIT_CONGESTION

2010-01-21 10:26:14.304816 [CRIT] ozmod_ss7_boost.c:244 SPAN is not online.
freeswitch at internal> 2010-01-21 10:26:14.304816 [ERR] mod_openzap.c:1043 No
channels available
2010-01-21 10:26:14.304816 [ERR] switch_ivr_originate.c:1510 Cannot create
outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
2010-01-21 10:26:14.304816 [DEBUG] switch_ivr_originate.c:2138 Originate
Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]

Inbound call result:
-----------------------------

     I made incoming call for the dial plan which is specified in the
earlier post at that time it said the number is busy. We did the packet
capture using the following command.

wanpipemon -i w1g1 -pcap -pcap file isdn.pcap -port ISDN -full -systime -c
trd

Here I attached the pcap file for that.


Where I did mistake or Did I miss any thing to do?
Please help me.......


On Wed, Jan 20, 2010 at 7:40 PM, Thangappan.M <thangappan143 at gmail.com>wrote:

>
>       I noticed the 'oz list' output  in that span type is 'ss7 (boost)'.
> How can I change this to isdn?
>
>
>
> On Wed, Jan 20, 2010 at 12:43 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>> I found the error in it. The file name is used as openzap.conf.xml (
>> smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
>> PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
>> openzap.conf.wiki.xml  file.
>>
>> Now the another problem is raised here.
>> When I was using oz list command , the details of the smg_prid shown. When
>> I was using 'oz dump smg_prid' command it shows all the channels' details.
>> But all the channels' states are DOWN. why? How can I make it the states to
>> UP?
>>
>> When I was making the call , the number is busy message was get. The call
>> was not  at all landed to the freeswitch.
>>
>> Dial plan Example:
>> -------------------------------
>>  <extension name="PRI_Testing">
>>         <condition field="destination_number" expression="^39114600$">
>>                 <action application="playback"
>> data="ivr-welcome_to_freeswitch"/>
>>         </condition>
>>   </extension>
>>
>> Please help me...........
>>
>> *Output Reference:*
>> http://pastebin.org/79074
>>
>>
>> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>
>>> Dear all,
>>>
>>>          I have successfully configured wanpipe with freeswitch. When I
>>> was the running wancfg_fs script the following files openzap.conf ,
>>> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
>>> are created.
>>>
>>>          I started the wanrouter command then executed the freeswitch.
>>>          When I was executing freeswitch mod_openzap.c said the error as
>>> "Error  for finding the span id. name:PRI_1".
>>>          But in the openzap.conf and openzap.conf.xml files the span name
>>> is smg_prid.
>>>
>>>          Why the freeswitch is referring the span name as PRI_1 ?
>>>          Whether this has to configured in anywhere?
>>>
>>>          In the freeswitch CLI using oz command I tried to dump the PRI_1
>>> span id but it said te error  as "PRI_1 is not found".  When I was trying
>>> the command 'oz dump smg_prid'  all the channel states and details shown.
>>>
>>>         It seems that smg_prid span configured in openzap perfectly (Its
>>> my assumption). Then Why freeswitch is referring the span name as PRI_1.
>>>
>>> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>>>
>>> Could anyone please help me?
>>>
>>> REFERENCE:
>>>
>>> openzap.conf
>>> [span wanpipe smg_prid]
>>> name => smg_prid
>>> trunk_type =>e1
>>> b-channel => 1:1-15
>>> b-channel => 1:17-31
>>>
>>>
>>> openzap.conf.xml
>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>   <settings>
>>>     <param name="debug" value="0"/>
>>>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>     <!--<param name="enable-analog-option" value="3-way"/>-->
>>>   </settings>
>>>   <boost_spans>
>>>         <span id="smg_prid">
>>>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>         <param name="dialplan" value="XML"/>
>>>         <param name="context" value="default"/>
>>>          <!-- regex to stop dialing when it matches -->
>>>         <!--<param name="dial-regex" value="5555"/>-->
>>>         <!-- regex to stop dialing when it does not match -->
>>>         <!--<param name="fail-dial-regex" value="^5"/>-->
>>>         </span>
>>>  </boost_spans>
>>>  <analog_spans>
>>>  </analog_spans>
>>> </configuration>
>>>
>>>
>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Thangappan.M
>>>
>>
>>
>>
>> --
>> Regards,
>> Thangappan.M
>>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From larclap at yahoo.com  Wed Jan 20 21:17:57 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 20 Jan 2010 21:17:57 -0800
Subject: [Freeswitch-users] Can't register Polycom
Message-ID: <01a301ca9a59$17e45fd0$47ad1f70$@com>

I am having trouble registering a Polycom 550. From the siptrace it looks
like there is no username coming from the Polycom. I configured the Polycom
via the web interface. I have inserted "1008" in the "Display Name",
"Address" and "Auth User ID" fields.

 

In conf/dialplan/default/1008.xml  the first line is <user id="1008"
cidr="192.168.10.105/24">.

 

What am I missing?

 

Thanks, Lars

 

http://pastebin.freeswitch.org/11875

 

 

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From irmatov at gmail.com  Wed Jan 20 21:59:38 2010
From: irmatov at gmail.com (Timur Irmatov)
Date: Thu, 21 Jan 2010 10:59:38 +0500
Subject: [Freeswitch-users] mod_erlang_event: disconnects
Message-ID: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>

Hi!


We have build a small and simple call center using FreeSWITCH and
mod_erlang_event. My erlang process keeps track of available agents
and routes incoming calls to them. Calls are sent to my application
via:

<action application="erlang" data="switch_event:! erlswitch at localhost" />

switch_event is a registered process, which spawns a new process for
each incoming call and returns new pid when it receives {get_pid,
UUID, Ref, From} message from FreeSWITCH.

The problem is, that pretty frequently processes which handle incoming
calls receive messages like {'EXIT', <5406.48.0>, noconnection} from
FreeSWITCH. As I understand from googling, this happens when remote C
node disconnects (and I see TCP connections from FreeSWITCH to epmd
daemon being torn down and reestablished). FreeSWITCH drops calls at
that moment.

Have anyone seen this? Is there any fix/ advice?

My system is Debian Lenny (5.0.3), 64-bit system, erlang installed
from Debian packages, no backports.


-- 
Timur Irmatov, xmpp:irmatov at jabber.ru



From mike at jerris.com  Wed Jan 20 22:48:42 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 21 Jan 2010 01:48:42 -0500
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
In-Reply-To: <20100120180720.GH4767@tamay-dogan.net>
References: <20100119103139.GR4767@tamay-dogan.net>
	<2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>
	<20100120180720.GH4767@tamay-dogan.net>
Message-ID: <BB497725-7989-44D6-8321-27B424861945@jerris.com>

We have a hudson instance doing builds, we just need boxes that can be the build drones.  And someone with a little time to set it up and bandwidth to handle it.  we build every 30 min that there is a change to svn, which is depending how long the builds take, usually 20+ times a day and upload the build results to the hudson server.

Mike

On Jan 20, 2010, at 1:07 PM, Michelle Konzack wrote:

> Hello,
> 
> Am 2010-01-20 01:28:40, schrieb Michael Jerris:
>> There is a debian dir in tree for our packages.  I suspect that our
>> packages are quite far from meeting the requirements of pretty much
>> any distro so for now, we will at least have packages for the next
>> release available soon after release.  We don't yet have a box for
>> debian instances for the build farm so we do not build the svn
>> snapshot pacakges for any deb distros.
> 
> Hmmm, currently I have only one fixed IP and some VServers and  PBuilder
> runing on i386 and ARM (only a small Ti Sitara AM3517 with 256 MB memory
> plus SATA drive)...
> 
> Also I am trying to relocate back to Germany...
> Maybe I could do the Job for Debian (i386 ARMEL) and Ubuntu (i386)
> 
> My Website is currently a backup fro 2008/12 and 2009/03 because  I  was
> offine since 2009-07-23 du to my fuckingbusiness partner...
> 
> Maybe it work:
>    http://www.debian.tamay-dogan.net/
> 
> I will try to reinstall the PBuilder interface which  allow  uploads  of
> sources/configs and autobuilding.  Also  I  like  to  include  an  auto-
> checkout from "svn" and "git" so, the author of the software should give
> a signal to my interface and the build is done automaticaly.
> 
> Also I am Package Maintainer of some Debian packages...
> 
> Thanks, Greetings and nice Day/Evening
>    Michelle Konzack
>    Systemadministrator
>    Electronic Engineer
>    Tamay Dogan Network
>    Debian GNU/Linux Consultant
> 
> -- 
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
> ##################### Debian GNU/Linux Consultant #####################
> <http://www.tamay-dogan.net/>                 Michelle Konzack
> <http://www.can4linux.org/>                   Apt. 917
> <http://www.flexray4linux.org/>               50, rue de Soultz
> Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
> IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
> ICQ    #328449886                             Tel. FR: +33  6  61925193
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Wed Jan 20 23:12:46 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 21 Jan 2010 02:12:46 -0500
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <24068.48012.qm@web33507.mail.mud.yahoo.com>
References: <24068.48012.qm@web33507.mail.mud.yahoo.com>
Message-ID: <DA570C12-70CA-43C0-B014-4B3CCBA7A7EF@jerris.com>


On Jan 20, 2010, at 6:31 PM, Diego Toro wrote:

> Greetings
> 
> I have a next section dial plan:
> 
> <action application="set" data="sound_prefix=$${base_dir}\sounds\es\co\callie\" />
> <action application="set" data="playback_delimiter=!"/>
> 
> <action application="playback" data="misc\8000\serror.wav!misc\8000\provide_reference_number.wav!digits\8000\5.wav" />  
> 
> 

<action application="playback" data="misc\\8000\\serror.wav!misc\\8000\\provide_reference_number.wav!digits\\8000\\5.wav" />  

??

> ....
> 
> I have a problem using multiple playback files on Windows, the path  misc\8000\serror.wav is changed by misc\8000 serror.wav.  I check C code on switch_utils.c, cleanup_separated_string function call to unescape_char function which change \s by ' '. This is correct, but on Windows '\' is the path separator, so is not possible to use  '\s', '\n'... into path file. I think this is possible to fix it.

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From msc at freeswitch.org  Thu Jan 21 00:25:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Jan 2010 00:25:32 -0800
Subject: [Freeswitch-users] Svar: Re: Home setup with home company
In-Reply-To: <4B56ECD7020000E1000003F1@mail.fribert.dk>
References: <4B56ECD7020000E1000003F1@mail.fribert.dk>
Message-ID: <87f2f3b91001210025t38bd679cu647b4935bef509c@mail.gmail.com>

On Wed, Jan 20, 2010 at 2:45 AM, mailinglist <mailinglist at fribert.dk> wrote:

> Hi Michael
>
> It's running on pfsense, so it's kinda locked to the version it currently
> is.
> Looks very nice though.
> Looking beyond that, is the action / anti-action list corrent?
>

I would say that you need to add an anti-action under the day of week check
and go to vm if it does not match. Right now if the DOW is 0 or 6 then the
entire extension will "fail" and the dialplan will just move on. Remember
that if any conditions fail then the entire thing extension "fails" unless
you are doing interesting things with the break= parameter. See the dialplan
page on the wiki for examples of how to use break in your conditions.
-MC


> Best regards
> Fribse
>
>
> >>> Michael Collins 20-01-10 1:53 >>>
>
>
> On Tue, Jan 19, 2010 at 9:02 AM, mailinglist <mailinglist at fribert.dk>wrote:
>
>>  I have a very small one man constultancy company that has the occasional
>> call, unfortunately we are getting more spam calls after hours than real
>> calls during work hours, so I would like to set up a TOD system.
>>
>> First step for me is just playing with the TOD example, I've gotten this
>> far:
>>
>> <extension name="CompanyTOD" continue="true">
>>    <condition field="destination_number" expression="^(8203)$"/>
>>       <!-- man strftime - M-F, 9AM to 5PM -->
>>       <condition field="${strftime(%w)}" expression="^([1-5])$"/>
>>       <condition field="${strftime(%H%M)}"
>> expression="^((09|1[0-6])[0-5][0-9]|1700)$">
>>          <action application="set" data="call_timeout=45"/>
>>          <action application="set" data="domain_name=10.11.12.25"/>
>>          <action application="set" data="continue_on_fail=true"/>
>>          <action application="set" data="hangup_after_bridge=true"/>
>>          <action application="bridge" data="
>> group/company@${domain_name}"/ <group/company@$%7Bdomain_name%7D%22/>>
>>          <action application="answer" data=""/>
>>          <action application="sleep" data="1000"/>
>>          <action application="voicemail" data="default 10.11.12.25
>> 10199"/>
>>          <anti-action application="voicemail" data="default 10.11.12.25
>> 10199"/>
>>       </condition>
>> </extension>
>> My idea with this, was that in the time 9-17 mon-fri, it rings, the rest
>> of the time it's sent directly to the voicemail.
>> I would of course like to have it not take messages outside work hours,
>> but that's just refining :-)
>>
>> But it picks up the call, and then nothing...
>>
>>
> We have a much cleaner way of doing TOD and DOW handling. You'll need to
> get to the latest FreeSWITCH version. Look in conf/dialplan/default.xml for
> this example:
>
>     <!-- Example of doing things based on time of day.
>
>      year = 4 digit year. Example year="2009"
>      yday = 1-365
>      mon = 1-12
>      mday = 1-31
>      week = 1-52
>      mweek= 1-6
>      wday = 1-7
>      hour = 0-23
>      minute = 0-59
>      minute-of-day = 1-1440
>
>      Example:
>      <condition minute-of-day="540-1080"> (9am to 6pm EVERY day)
>      do something ...
>      </condition>
>     -->
>     <extension name="tod_example" continue="true">
>       <condition wday="2-6" hour="9-18">
>         <action application="set" data="open=true"/>
>       </condition>
>     </extension>
>
>
> Use that condition instead of the two conditions you're now using and see
> if you have better success.
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From linux4michelle at tamay-dogan.net  Thu Jan 21 01:13:46 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Thu, 21 Jan 2010 10:13:46 +0100
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
In-Reply-To: <BB497725-7989-44D6-8321-27B424861945@jerris.com>
References: <20100119103139.GR4767@tamay-dogan.net>
	<2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>
	<20100120180720.GH4767@tamay-dogan.net>
	<BB497725-7989-44D6-8321-27B424861945@jerris.com>
Message-ID: <20100121091346.GF4767@tamay-dogan.net>

Hello,

Am 2010-01-21 01:48:42, schrieb Michael Jerris:
> We have a hudson instance doing builds, we just need boxes that can be
> the build drones.  And someone with a little time to set it up and
> bandwidth to handle it.  we build every 30 min that there is a change
> to svn, which is depending how long the builds take, usually 20+ times
> a day and upload the build results to the hudson server.

If I a installed in Frankfurt/Germany I will  try  to  get  as  fast  as
possibel an E3 (34Mbit) or FTTB (100Mbit) and  bandwidth  should  be  no
problem.

How many MByte is one SVN checkout?

1 Mbit bandwitdh (~320 GiB/month) cost me arround 50 Euro additional  to
the base-price.  I do bot think it wil kill my finacial resources.

My current bandwidth usage is nearly 400 kBit  because  I  have  only  a
1 Mbit access.  :-(  and I can not get a 100/50 Mbit FTTH where  I  live
otherwise you could have one of my spare Sun Fire X4100M2 (3 x 76  GByte
SAS in Raid-1 with Hotfix) as  Build-Daemon

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strabourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From jingwei.yang at gmail.com  Thu Jan 21 01:22:12 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 21 Jan 2010 17:22:12 +0800
Subject: [Freeswitch-users]  Is this queue flow correct?
Message-ID: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>

Hi All,

Please advise whether the following flow makes sense.

1. Client A calls in and parked in Queue 1
2. Originate calls to several consumers simultaneously and park them in
Queue 2
3. Intercept A's call to the first consumer of Queue 2

Basically I want A's call picked up by the first among many consumers with
no errors. Please let me know whether I'm on the right track.

Thanks and best regards,
-Jingwei
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From a.afzali2003 at gmail.com  Thu Jan 21 01:44:27 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Thu, 21 Jan 2010 13:14:27 +0330
Subject: [Freeswitch-users] Managing Presence on Gateways
Message-ID: <debff5a81001210144n4b32c19cic36415155ee52231@mail.gmail.com>

Hi Guys,

In the external profile (as in the internal) there is an option to enable
presence functionality (with setting it to passive). My question is how does
it mean presence functionality for a gateway which interfaces home domain to
another one? Does it mean that the gateway could subscribe itself for some
presence information in that domain in behaves of local users and relays
them?

Appreciate all comments,
-- afshin
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From oscav at hotmail.fr  Thu Jan 21 01:47:53 2010
From: oscav at hotmail.fr (Oscav)
Date: Thu, 21 Jan 2010 01:47:53 -0800 (PST)
Subject: [Freeswitch-users]  All channels are frozen while receiving DTMF
Message-ID: <27255181.post@talk.nabble.com>


Hi,

I'm running a script that gets some DTMF from caller. I found that when a
caller is entering DTMF , all the others channels are frozen until all the
DTMF  are received. In the logs I see that each DTMF takes 1 second. It
means that if the caller enters 10 digits then all the other running scripts
are paused for 10 seconds. The problem is exponential with traffic load.

Anyone have an idea ??

Thanks
-- 
View this message in context: http://old.nabble.com/All-channels-are-frozen-while-receiving-DTMF-tp27255181p27255181.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From oscav at hotmail.fr  Thu Jan 21 01:49:48 2010
From: oscav at hotmail.fr (Oscav)
Date: Thu, 21 Jan 2010 01:49:48 -0800 (PST)
Subject: [Freeswitch-users] Failed to connect to a SKYPE API
In-Reply-To: <27078464.post@talk.nabble.com>
References: <27062783.post@talk.nabble.com> <27078464.post@talk.nabble.com>
Message-ID: <27255195.post@talk.nabble.com>


I solved the problem. It was due to the logon session.


Oscav wrote:
> 
> Im' running FS on windows server 2003 64bits
> 
> 
> Oscav wrote:
>> 
>> Hi,
>> 
>> I'm trying to use to SkypeIAX. When I load the mod_skypiax, I got the
>> following error :
>> 
>>  Failed to connect to a SKYPE API for interface_id=1, no SKYPE client
>> running, please (re)start Skype client. Skypiax exiting
>> 
>> Skype is running with the correct account and skypiax.conf use the same
>> account. I was expecting a permission request from the Skype user but
>> nothing happens.
>> 
>> Somebody knows how I can solve this ??
>> 
>> Many thanks.
>> 
> 
> 

-- 
View this message in context: http://old.nabble.com/Failed-to-connect-to-a-SKYPE-API-tp27062783p27255195.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From a.alalousi at gmail.com  Thu Jan 21 02:10:59 2010
From: a.alalousi at gmail.com (Ahmed Naji)
Date: Thu, 21 Jan 2010 10:10:59 +0000
Subject: [Freeswitch-users] G729 coded issues
In-Reply-To: <537338B1-B582-463E-8EFE-7BBED8165D2B@freeswitch.org>
References: <cd93b3701001190831u4147214cn9832c6f469c6d17e@mail.gmail.com>
	<6D67C7A9-42CD-4F5F-AD18-E9568098FB71@freeswitch.org>
	<4B55E794.6020909@coppice.org>
	<cd93b3701001200514p4acc5a5bk26a463164f841ad2@mail.gmail.com>
	<537338B1-B582-463E-8EFE-7BBED8165D2B@freeswitch.org>
Message-ID: <cd93b3701001210210v2ddba8afwc6a310540d2e5180@mail.gmail.com>

Hi Brian,

All for it, so ye, let me have the beta.

The only reason I went Howler is because of a pressing need.

Moreover, I'm willing to put mine and my teams resources into the project. I
and others cut code in C++/C/Perl/....etc. As an organisation, we are
actively involved in telecoms consultancy and software development, and I
really would like to put our backs into pushing the case for FS wherever
possible.

Ping me offline on the subscription e-mail for this account, and let's
exchange details if there is interest your side.

Support ? we are all sold on FS down here believe me, so you can count on
it.

Regards,

Ahmed.


2010/1/20 Brian West <brian at freeswitch.org>

> Just an FYI.  Howler doesn't support the FreeSWITCH project in any way,
> shape or form.  They do not donate any proceeds or help the project at all.
>  That said.  We have our officially supported G729 coming out soon that will
> support the project.  I have it in beta if anyone is really interested in
> testing it please feel free to email me offlist.  We are currently working
> out how we want to package the lib, binary and module to make installation
> easy.
>
> Thanks,
> /b
>
> On Jan 20, 2010, at 7:14 AM, Ahmed Naji wrote:
>
> > Howler
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ahmed Naji
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From nicolas at medularis.com  Thu Jan 21 03:35:07 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Thu, 21 Jan 2010 08:35:07 -0300
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <5d2828f1001201406p4154b98ald1af1c5c25f59337@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
	<e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
	<5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>
	<5d2828f1001201406p4154b98ald1af1c5c25f59337@mail.gmail.com>
Message-ID: <1b46b4e81001210335l42baef16r8a2952aa5b92f6e6@mail.gmail.com>

On Wed, Jan 20, 2010 at 7:06 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> So, I've been reading about early media in the wiki, and have made a little
> progress, which leads to more questions.
>
> I understand now why a call is considered connected before one person has
> picked up the phone. I am also able to get my script to wait for the phone
> to be picked up, by setting the ignore_early_media variable when starting a
> new session, like this:
>
> customerSession =
> freeswitch.Session("{ignore_early_media=true}sofia/gateway/example.com/"
> .. customerPhoneNumber)
>
>
> After that line, the script waits for the other phone to be picked up.
>
> However, now I wonder what to do with calls that don't complete, get busy
> signals, etc.
>
> What do people do in this case? The only related example I can find on the
> web is for a javascript dialer, which doesn't address any of these cases.
>


I guess it depends on what you want to do. For example I have a lua script
very similar to what you describe, although there is no confirmation
involved. Depending on the hangup cause the session gets, it might try
redialing with a different gateway, try again or just hangup.

Take a look here http://wiki.freeswitch.org/wiki/Hangup_causes to see what
each hangup cause means. You don't need to have a special case for all of
them, only the ones you are interested in.

Here's an example in code which retries a call depending on the hangup
cause. It retries max_retries1 times and alternates between 2 different
gateways:

session1 = null;
max_retries1 = 3;
retries = 0;
ostr = "";
repeat
        retries = retries + 1;
        if (retries % 2) then ostr = originate_str1;
        else ostr = originate_str12; end
        freeswitch.consoleLog("notice", "*********** Dialing Leg1: " .. ostr
.. " - Try: "..retries.." ***********\n");
        session1 = freeswitch.Session(ostr);
        local hcause = session1:hangupCause();
        freeswitch.consoleLog("notice", "*********** Leg1: " .. hcause .. "
- Try: "..retries.." ***********\n");
until not ((hcause == 'NO_ROUTE_DESTINATION' or hcause ==
'RECOVERY_ON_TIMER_EXPIRE' or hcause == 'INCOMPATIBLE_DESTINATION' or hcause
== 'CALL_REJECTED' or hcause == 'NORMAL_TEMPORARY_FAILURE') and (retries <
max_retriesl1))


Note: originate_str1 and originate_str2 are two different dial strings for 2
different gateways.



>
> Early Media: http://wiki.freeswitch.org/wiki/Early_media
> ignore_early_media variable:
> http://wiki.freeswitch.org/wiki/Variable_ignore_early_media
> <http://wiki.freeswitch.org/wiki/Variable_ignore_early_media>javascript
> dialer: http://alexn.org/docs/dialer.html
>
>
> <http://alexn.org/docs/dialer.html>Mike van Lammeren
>
>
> On Wed, Jan 20, 2010 at 4:45 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:
>
>> Hi Rob!
>>
>> Unfortunately, I have the next few weeks to complete this part of the
>> project. Without a stable release of 1.0.5, I have to stick with 1.0.4. I'm
>> sure that 1.0.4 can detect and report when a phone is picked up. It's just
>> that I can't figure out how to get that information! Either that, or I have
>> something mis-configured.
>>
>> Mike van Lammeren
>>
>>
>> On Wed, Jan 20, 2010 at 4:32 PM, Rob Forman <rob4manhere at gmail.com>wrote:
>>
>>> Hi Mike,
>>>
>>> I don't think v1.0.4 is supported any longer.   You'll have better luck
>>> getting assistance by upgrading to trunk or the latest tar and reporting
>>> back.
>>>
>>> Good luck!
>>> Rob
>>>
>>> On Wed, Jan 20, 2010 at 3:18 PM, Mike van Lammeren <
>>> mike at van.lammeren.net> wrote:
>>>
>>>>  Hello!
>>>>
>>>> Some day, I'll probably laugh at asking this question, but today I can't
>>>> figure it out.
>>>>
>>>> I've written a Lua script that listens for a call, then dials a phone
>>>> number to a second person. It plays a message, then prompts the second
>>>> person to hit pound to connect. If the second person hits pound, then it
>>>> bridges the two calls together.
>>>>
>>>> All that works great, but I can't figure out how to get the session for
>>>> the second person to wait until that person answers.
>>>>
>>>> I'm using FreeSWITCH 1.0.4, and although there is a *getState* function
>>>> documented in the wiki, it doesn't seem to exist for me.
>>>>
>>>> Any help would be appreciated!
>>>>
>>>>
>>>> Mike van Lammeren
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Thu Jan 21 03:41:40 2010
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 21 Jan 2010 19:41:40 +0800
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
Message-ID: <23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>

what no errors mean? how do you originate calls to consumers? I don't
understand your scenario.

2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
> Hi All,
>
> Please advise whether the following flow makes sense.
>
> 1. Client A calls in and parked in Queue 1
> 2. Originate calls to several consumers simultaneously and park them in
> Queue 2
> 3. Intercept A's call to the first consumer of Queue 2
>
> Basically I want A's call picked up by the first among many consumers with
> no errors. Please let me know whether I'm on the right track.
>
> Thanks and best regards,
> -Jingwei
>



From mikael at bjerkeland.com  Thu Jan 21 04:20:28 2010
From: mikael at bjerkeland.com (Mikael Bjerkeland)
Date: Thu, 21 Jan 2010 13:20:28 +0100
Subject: [Freeswitch-users] Sip video intercom
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9083ACB@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E9083ACB@prato.activenetwerx.local>
Message-ID: <cb79a141001210420o196ae36ck615e7a5c69791bb@mail.gmail.com>

http://www.2n.cz/products/door-lift-phones/door-entry-systems/ip-communication.html

2010/1/20 Joseph L. Casale <jcasale at activenetwerx.com>

> I need to get an intercom integrated into the voip system of a highend
> home.
> That being said, I am looking for a nice looking discrete panel to mount
> outside by the front door. Anyone have any experience with these and know
> of
> a model they recommend?
>
> Thanks!
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dftoro at yahoo.com  Thu Jan 21 05:02:40 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Thu, 21 Jan 2010 05:02:40 -0800 (PST)
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <DA570C12-70CA-43C0-B014-4B3CCBA7A7EF@jerris.com>
Message-ID: <37012.20543.qm@web33507.mail.mud.yahoo.com>

Hi MikeJ, using '\\' the behavior is the same, '\\s' is replaced by ' '.

<action application="playback" data="misc\\8000\\serror.wav!misc\\8000\\provide_reference_number.wav!digits\\8000\\5.wav"/>  

Console output error is:

[ERR] mod_sndfile.c:194 Error Opening File [d:\fs\fs_trunk_20100118\Debug\sounds\es\co\callie\misc\8000 error.wav]


S.O.: Windows 7
FreeSwitch: Trunk (svn latest version)

Thank you


Diego Toro
http://lacarretade.blogspot.com/


--- On Thu, 1/21/10, Michael Jerris <mike at jerris.com> wrote:

> From: Michael Jerris <mike at jerris.com>
> Subject: Re: [Freeswitch-users] mutiple playback files (unescape_char) Windows
> To: freeswitch-users at lists.freeswitch.org
> Date: Thursday, January 21, 2010, 2:12 AM
> 
> On Jan 20, 2010, at 6:31 PM, Diego Toro
> wrote:
> Greetings
> 
> I have a next section dial plan:
> 
> <action application="set"
> data="sound_prefix=$${base_dir}\sounds\es\co\callie\"
> />
> <action application="set"
> data="playback_delimiter=!"/>
> 
> <action application="playback"
> data="misc\8000\serror.wav!misc\8000\provide_reference_number.wav!digits\8000\5.wav"
> /> ?
> 
> 
> 
> <action
> application="playback"
> data="misc\\8000\\serror.wav!misc\\8000\\provide_reference_number.wav!digits\\8000\\5.wav"
> /> ?
> ??
> ....
> 
> I have a problem using multiple playback files on Windows,
> the path ?misc\8000\serror.wav is changed by
> misc\8000 serror.wav. ?I check C code on
> switch_utils.c, cleanup_separated_string function call to
> unescape_char function which change \s by ' '.
> This is correct, but on Windows '\' is the path
> separator, so is not possible to use ?'\s',
> '\n'... into path file. I think this is possible
> to fix it.
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From andrew at hijacked.us  Thu Jan 21 05:42:41 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 21 Jan 2010 08:42:41 -0500
Subject: [Freeswitch-users] mod_erlang_event: disconnects
In-Reply-To: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>
References: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>
Message-ID: <20100121134241.GD1036@hijacked.us>

On Thu, Jan 21, 2010 at 10:59:38AM +0500, Timur Irmatov wrote:
> We have build a small and simple call center using FreeSWITCH and
> mod_erlang_event. My erlang process keeps track of available agents
> and routes incoming calls to them. Calls are sent to my application
> via:
> 
> <action application="erlang" data="switch_event:! erlswitch at localhost" />
> 
> switch_event is a registered process, which spawns a new process for
> each incoming call and returns new pid when it receives {get_pid,
> UUID, Ref, From} message from FreeSWITCH.

That all looks fine.
> 
> The problem is, that pretty frequently processes which handle incoming
> calls receive messages like {'EXIT', <5406.48.0>, noconnection} from
> FreeSWITCH. As I understand from googling, this happens when remote C
> node disconnects (and I see TCP connections from FreeSWITCH to epmd
> daemon being torn down and reestablished). FreeSWITCH drops calls at
> that moment.

Does it drop ALL calls being handled in erlang, or just that one?
> 
> Have anyone seen this? Is there any fix/ advice?

I haven't seen this before, how many calls are involved? I'm willing to
help you troubleshoot though. Is there anything relevant in the logs
(even at DEBUG)?

> 
> My system is Debian Lenny (5.0.3), 64-bit system, erlang installed
> from Debian packages, no backports.
> 
What OTP release does that equate to, R12 or R13?

Andrew (mod_erlang_event author)



From andrew at hijacked.us  Thu Jan 21 05:44:51 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 21 Jan 2010 08:44:51 -0500
Subject: [Freeswitch-users] mod_erlang_event: disconnects
In-Reply-To: <20100121134241.GD1036@hijacked.us>
References: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>
	<20100121134241.GD1036@hijacked.us>
Message-ID: <20100121134451.GE1036@hijacked.us>

Also, what FS version are you running?

Andrew



From Russell.Mosemann at cune.org  Thu Jan 21 06:28:17 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 21 Jan 2010 14:28:17 -0000
Subject: [Freeswitch-users] Can't register Polycom
In-Reply-To: <01a301ca9a59$17e45fd0$47ad1f70$@com>
Message-ID: <20100121142817.CDAB42A865A@cuneorg-email.cune.pri>

Lars Zeb <larclap at yahoo.com> said:

> cidr="192.168.10.105/24"

The IP address and mask don't make sense. Either it needs to be
192.168.10.0/24 or 192.168.10.105.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From Claudio.Cavalera at italtel.it  Thu Jan 21 07:17:49 2010
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Thu, 21 Jan 2010 16:17:49 +0100
Subject: [Freeswitch-users] playing with sessions in lua
In-Reply-To: <191c3a031001140837m7cffcdd5w71886d6c8ba1dafe@mail.gmail.com>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0F36308@BESONE.corp.dom>

Anthony,
I've noticed that you did not mention javascript, is it the exception?
Is the C++ file with the api
src/include/switch_swigable_cpp.h ?
Thx,
Claudio
 


________________________________

	From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
	Sent: Thursday, January 14, 2010 5:38 PM
	To: freeswitch-users at lists.freeswitch.org
	Subject: Re: [Freeswitch-users] playing with sessions in lua
	
	
	Don't forget that lua,perl,python,managed,java all share the
same exact C++ source file with swig so the same exact api applies to
all of the above.
	
	


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From brian at freeswitch.org  Thu Jan 21 07:26:09 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 21 Jan 2010 09:26:09 -0600
Subject: [Freeswitch-users] Can't register Polycom
In-Reply-To: <20100121142817.CDAB42A865A@cuneorg-email.cune.pri>
References: <20100121142817.CDAB42A865A@cuneorg-email.cune.pri>
Message-ID: <77267B4B-6A0C-422F-ACA5-34240FCCDD37@freeswitch.org>

It will still take that mask and make it work. 

/b

On Jan 21, 2010, at 8:28 AM, <Russell.Mosemann at cune.org> <Russell.Mosemann at cune.org> wrote:

> Lars Zeb <larclap at yahoo.com> said:
> 
>> cidr="192.168.10.105/24"
> 
> The IP address and mask don't make sense. Either it needs to be
> 192.168.10.0/24 or 192.168.10.105.
> 
> -- 
> Russell Mosemann

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From jingwei.yang at gmail.com  Thu Jan 21 07:39:07 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 21 Jan 2010 23:39:07 +0800
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
Message-ID: <13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>

Sorry about the confusion, I'm just trying to think over all the abnormal
situations before the implementation and hope the flow has no design flaws.

Client A is parked in Queue 1, multiple consumers will be ringed to answer
him. And once the first one is connected, all the rest will hang up. This is
the targeted function.

To achieve this, I'm considering to originate a call to each consumer and
put the calls in Queue 2. Then intercept client A to the first element of
Queue 2.

I'm not sure if it's the usual or the best way. If you feel not, please
don't hesitate to correct me. Any thoughts are warmly appreciated.

On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com> wrote:

> what no errors mean? how do you originate calls to consumers? I don't
> understand your scenario.
>
> 2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
> > Hi All,
> >
> > Please advise whether the following flow makes sense.
> >
> > 1. Client A calls in and parked in Queue 1
> > 2. Originate calls to several consumers simultaneously and park them in
> > Queue 2
> > 3. Intercept A's call to the first consumer of Queue 2
> >
> > Basically I want A's call picked up by the first among many consumers
> with
> > no errors. Please let me know whether I'm on the right track.
> >
> > Thanks and best regards,
> > -Jingwei
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fdelawarde at wirelessmundi.com  Thu Jan 21 08:50:20 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Thu, 21 Jan 2010 17:50:20 +0100
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
Message-ID: <1264092620.14614.73.camel@luna.tc.commsmundi.com>

Why do you need 2 fifos? You could have callback agents connected to the
fifo and send incoming calls there, mod_fifo will do the rest.

To configure agents for callback:
http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback

To place a call into a fifo:
http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue

Fran?ois.

On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
> Sorry about the confusion, I'm just trying to think over all the
> abnormal situations before the implementation and hope the flow has no
> design flaws.
> 
> Client A is parked in Queue 1, multiple consumers will be ringed to
> answer him. And once the first one is connected, all the rest will
> hang up. This is the targeted function. 
> 
> To achieve this, I'm considering to originate a call to each consumer
> and put the calls in Queue 2. Then intercept client A to the first
> element of Queue 2. 
> 
> I'm not sure if it's the usual or the best way. If you feel not,
> please don't hesitate to correct me. Any thoughts are warmly
> appreciated.
> 
> On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com> wrote:
>         what no errors mean? how do you originate calls to consumers?
>         I don't
>         understand your scenario.
>         
>         2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
>         
>         > Hi All,
>         >
>         > Please advise whether the following flow makes sense.
>         >
>         > 1. Client A calls in and parked in Queue 1
>         > 2. Originate calls to several consumers simultaneously and
>         park them in
>         > Queue 2
>         > 3. Intercept A's call to the first consumer of Queue 2
>         >
>         > Basically I want A's call picked up by the first among many
>         consumers with
>         > no errors. Please let me know whether I'm on the right
>         track.
>         >
>         > Thanks and best regards,
>         > -Jingwei
>         >
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From larclap at yahoo.com  Thu Jan 21 09:01:25 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 21 Jan 2010 09:01:25 -0800
Subject: [Freeswitch-users] Can't register Polycom
In-Reply-To: <77267B4B-6A0C-422F-ACA5-34240FCCDD37@freeswitch.org>
References: <20100121142817.CDAB42A865A@cuneorg-email.cune.pri>
	<77267B4B-6A0C-422F-ACA5-34240FCCDD37@freeswitch.org>
Message-ID: <014501ca9abb$5ddae970$1990bc50$@com>

I removed the /24 from the cidr and still 403 Forbidden.

 

I configured this extension on the Polycom on Line 4. I removed this
definition and put it on Line 3. It then registered. 

 

There must be something wrong with the phone.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, January 21, 2010 7:26 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't register Polycom

 

It will still take that mask and make it work. 

 

/b

 

On Jan 21, 2010, at 8:28 AM, <Russell.Mosemann at cune.org>
<Russell.Mosemann at cune.org> wrote:





Lars Zeb <larclap at yahoo.com> said:




cidr="192.168.10.105/24"


The IP address and mask don't make sense. Either it needs to be
192.168.10.0/24 or 192.168.10.105.

-- 
Russell Mosemann



 

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From mike at van.lammeren.net  Thu Jan 21 09:03:08 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 21 Jan 2010 12:03:08 -0500
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <1b46b4e81001210335l42baef16r8a2952aa5b92f6e6@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
	<e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
	<5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>
	<5d2828f1001201406p4154b98ald1af1c5c25f59337@mail.gmail.com>
	<1b46b4e81001210335l42baef16r8a2952aa5b92f6e6@mail.gmail.com>
Message-ID: <5d2828f1001210903r2e5ec264q44945e17b48dda50@mail.gmail.com>

Awesome example code, Nicolas! Thanks!

On Thu, Jan 21, 2010 at 6:35 AM, Nicolas Brenner <nicolas at medularis.com>wrote:

>
> On Wed, Jan 20, 2010 at 7:06 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:
>
>> So, I've been reading about early media in the wiki, and have made a
>> little progress, which leads to more questions.
>>
>> I understand now why a call is considered connected before one person has
>> picked up the phone. I am also able to get my script to wait for the phone
>> to be picked up, by setting the ignore_early_media variable when starting a
>> new session, like this:
>>
>> customerSession =
>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/example.com/"
>> .. customerPhoneNumber)
>>
>>
>> After that line, the script waits for the other phone to be picked up.
>>
>> However, now I wonder what to do with calls that don't complete, get busy
>> signals, etc.
>>
>> What do people do in this case? The only related example I can find on the
>> web is for a javascript dialer, which doesn't address any of these cases.
>>
>
>
> I guess it depends on what you want to do. For example I have a lua script
> very similar to what you describe, although there is no confirmation
> involved. Depending on the hangup cause the session gets, it might try
> redialing with a different gateway, try again or just hangup.
>
> Take a look here http://wiki.freeswitch.org/wiki/Hangup_causes to see what
> each hangup cause means. You don't need to have a special case for all of
> them, only the ones you are interested in.
>
> Here's an example in code which retries a call depending on the hangup
> cause. It retries max_retries1 times and alternates between 2 different
> gateways:
>
> session1 = null;
> max_retries1 = 3;
> retries = 0;
> ostr = "";
> repeat
>         retries = retries + 1;
>         if (retries % 2) then ostr = originate_str1;
>         else ostr = originate_str12; end
>         freeswitch.consoleLog("notice", "*********** Dialing Leg1: " ..
> ostr .. " - Try: "..retries.." ***********\n");
>         session1 = freeswitch.Session(ostr);
>         local hcause = session1:hangupCause();
>         freeswitch.consoleLog("notice", "*********** Leg1: " .. hcause .. "
> - Try: "..retries.." ***********\n");
> until not ((hcause == 'NO_ROUTE_DESTINATION' or hcause ==
> 'RECOVERY_ON_TIMER_EXPIRE' or hcause == 'INCOMPATIBLE_DESTINATION' or hcause
> == 'CALL_REJECTED' or hcause == 'NORMAL_TEMPORARY_FAILURE') and (retries <
> max_retriesl1))
>
>
> Note: originate_str1 and originate_str2 are two different dial strings for
> 2 different gateways.
>
>
>
>>
>> Early Media: http://wiki.freeswitch.org/wiki/Early_media
>> ignore_early_media variable:
>> http://wiki.freeswitch.org/wiki/Variable_ignore_early_media
>> <http://wiki.freeswitch.org/wiki/Variable_ignore_early_media>javascript
>> dialer: http://alexn.org/docs/dialer.html
>>
>>
>> <http://alexn.org/docs/dialer.html>Mike van Lammeren
>>
>>
>> On Wed, Jan 20, 2010 at 4:45 PM, Mike van Lammeren <mike at van.lammeren.net
>> > wrote:
>>
>>> Hi Rob!
>>>
>>> Unfortunately, I have the next few weeks to complete this part of the
>>> project. Without a stable release of 1.0.5, I have to stick with 1.0.4. I'm
>>> sure that 1.0.4 can detect and report when a phone is picked up. It's just
>>> that I can't figure out how to get that information! Either that, or I have
>>> something mis-configured.
>>>
>>> Mike van Lammeren
>>>
>>>
>>> On Wed, Jan 20, 2010 at 4:32 PM, Rob Forman <rob4manhere at gmail.com>wrote:
>>>
>>>> Hi Mike,
>>>>
>>>> I don't think v1.0.4 is supported any longer.   You'll have better luck
>>>> getting assistance by upgrading to trunk or the latest tar and reporting
>>>> back.
>>>>
>>>> Good luck!
>>>> Rob
>>>>
>>>> On Wed, Jan 20, 2010 at 3:18 PM, Mike van Lammeren <
>>>> mike at van.lammeren.net> wrote:
>>>>
>>>>>  Hello!
>>>>>
>>>>> Some day, I'll probably laugh at asking this question, but today I
>>>>> can't figure it out.
>>>>>
>>>>> I've written a Lua script that listens for a call, then dials a phone
>>>>> number to a second person. It plays a message, then prompts the second
>>>>> person to hit pound to connect. If the second person hits pound, then it
>>>>> bridges the two calls together.
>>>>>
>>>>> All that works great, but I can't figure out how to get the session for
>>>>> the second person to wait until that person answers.
>>>>>
>>>>> I'm using FreeSWITCH 1.0.4, and although there is a *getState*function documented in the wiki, it doesn't seem to exist for me.
>>>>>
>>>>> Any help would be appreciated!
>>>>>
>>>>>
>>>>> Mike van Lammeren
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tayeb.meftah at gmail.com  Thu Jan 21 09:09:35 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Thu, 21 Jan 2010 18:09:35 +0100
Subject: [Freeswitch-users] Can't register Polycom
In-Reply-To: <01a301ca9a59$17e45fd0$47ad1f70$@com>
References: <01a301ca9a59$17e45fd0$47ad1f70$@com>
Message-ID: <4B588A4F.5060303@gmail.com>

hi,
make sure that you are using the user: 1008 and pass: 1008
don't forgot to add autorisation user: 1008
the dialplan don't have any relation with the user/pass acording to your 
post, you are montioning the dialplan path but you need to edit the 
directory (conf/directory/default/1008.xml)
chedck your freeswitch ip and configure your policom ip acording to your 
fs ip
good luck
Le 21/01/2010 06:17, Lars Zeb a ?crit :
>
> I am having trouble registering a Polycom 550. From the siptrace it 
> looks like there is no username coming from the Polycom. I configured 
> the Polycom via the web interface. I have inserted "1008" in the 
> "Display Name", "Address" and "Auth User ID" fields.
>
> In conf/dialplan/default/1008.xml  the first line is <user id="1008" 
> cidr="192.168.10.105/24">.
>
> What am I missing?
>
> Thanks, Lars
>
> http://pastebin.freeswitch.org/11875
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From regs at kinetix.gr  Thu Jan 21 10:18:22 2010
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Thu, 21 Jan 2010 20:18:22 +0200
Subject: [Freeswitch-users] reloadxml/rescan profile  and  killgw
Message-ID: <4B589A6E.8010205@kinetix.gr>

Wouldn't it be nice a "reloadxml"/"rescan profile" to delete all 
nonexistent gateways (when they are absent in the xml config) without 
having to issue a "sofia profile xxxxx killgw yyyyyy" command?

I always seem to find forgotten gateway's in the profile because of 
this... Any thoughts?


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
-------------------------------------------



From sos at sokhapkin.dyndns.org  Thu Jan 21 10:41:09 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 21 Jan 2010 13:41:09 -0500
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION?
Message-ID: <201001211341.09739.sos@sokhapkin.dyndns.org>

I often see in FS log the following problem (bypass_media=true), SVN r16340:

SDP sent out to gateway (INVITE):

   v=0
   o=bandx-msw3 0 0 IN IP4 213.166.9.4
   s=sip call
   c=IN IP4 213.166.9.6
   t=0 0
   m=audio 56032 RTP/AVP 0 8 18
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=yes
   a=silenceSupp:on - - - -


SDP response from gateway (183 Session Progress):

   v=0
   o=- 3473087019 3473087037 IN IP4 67.203.64.182
   s=-
   c=IN IP4 67.203.64.182
   t=0 0
   m=audio 14116 RTP/AVP 0
   a=sendrecv
   a=ptime:20
   a=rtpmap:0 PCMU/8000

Freeswitch hangs up the call with hangup cause INCOMPATIBLE_DESTINATION. Where 
is incompatibility? There is common codec 0.



From anthony.minessale at gmail.com  Thu Jan 21 10:53:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Jan 2010 12:53:23 -0600
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION?
In-Reply-To: <201001211341.09739.sos@sokhapkin.dyndns.org>
References: <201001211341.09739.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031001211053i6670cec3l8b8f317dac4c6072@mail.gmail.com>

if you use bypass_media=true from the dialplan without late-negotiation set
in the profile, it still tries to match the codecs locally on the inbound
leg and the variable does not work if the call has established media before
making the outbound leg.

It's hard to tell you the exact answer without a console trace on debug
level.


On Thu, Jan 21, 2010 at 12:41 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> I often see in FS log the following problem (bypass_media=true), SVN
> r16340:
>
> SDP sent out to gateway (INVITE):
>
>   v=0
>   o=bandx-msw3 0 0 IN IP4 213.166.9.4
>   s=sip call
>   c=IN IP4 213.166.9.6
>   t=0 0
>   m=audio 56032 RTP/AVP 0 8 18
>   a=rtpmap:18 G729/8000
>   a=fmtp:18 annexb=yes
>   a=silenceSupp:on - - - -
>
>
> SDP response from gateway (183 Session Progress):
>
>   v=0
>   o=- 3473087019 3473087037 IN IP4 67.203.64.182
>   s=-
>   c=IN IP4 67.203.64.182
>   t=0 0
>   m=audio 14116 RTP/AVP 0
>   a=sendrecv
>   a=ptime:20
>   a=rtpmap:0 PCMU/8000
>
> Freeswitch hangs up the call with hangup cause INCOMPATIBLE_DESTINATION.
> Where
> is incompatibility? There is common codec 0.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From sos at sokhapkin.dyndns.org  Thu Jan 21 11:05:19 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 21 Jan 2010 14:05:19 -0500
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION?
In-Reply-To: <191c3a031001211053i6670cec3l8b8f317dac4c6072@mail.gmail.com>
References: <201001211341.09739.sos@sokhapkin.dyndns.org>
	<191c3a031001211053i6670cec3l8b8f317dac4c6072@mail.gmail.com>
Message-ID: <201001211405.19271.sos@sokhapkin.dyndns.org>

Late negotiation is set. I will try to enable debug when the traffic will be 
low and open a problem on jira.

On Thursday 21 January 2010, Anthony Minessale wrote:
> if you use bypass_media=true from the dialplan without late-negotiation set
> in the profile, it still tries to match the codecs locally on the inbound
> leg and the variable does not work if the call has established media before
> making the outbound leg.
>
> It's hard to tell you the exact answer without a console trace on debug
> level.
>
>
> On Thu, Jan 21, 2010 at 12:41 PM, Sergey Okhapkin
>
> <sos at sokhapkin.dyndns.org>wrote:
> > I often see in FS log the following problem (bypass_media=true), SVN
> > r16340:
> >
> > SDP sent out to gateway (INVITE):
> >
> >   v=0
> >   o=bandx-msw3 0 0 IN IP4 213.166.9.4
> >   s=sip call
> >   c=IN IP4 213.166.9.6
> >   t=0 0
> >   m=audio 56032 RTP/AVP 0 8 18
> >   a=rtpmap:18 G729/8000
> >   a=fmtp:18 annexb=yes
> >   a=silenceSupp:on - - - -
> >
> >
> > SDP response from gateway (183 Session Progress):
> >
> >   v=0
> >   o=- 3473087019 3473087037 IN IP4 67.203.64.182
> >   s=-
> >   c=IN IP4 67.203.64.182
> >   t=0 0
> >   m=audio 14116 RTP/AVP 0
> >   a=sendrecv
> >   a=ptime:20
> >   a=rtpmap:0 PCMU/8000
> >
> > Freeswitch hangs up the call with hangup cause INCOMPATIBLE_DESTINATION.
> > Where
> > is incompatibility? There is common codec 0.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From msc at freeswitch.org  Thu Jan 21 11:20:06 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Jan 2010 11:20:06 -0800
Subject: [Freeswitch-users] Configuration Preservation
	throughTrunkUpdates
In-Reply-To: <DF8D6745E8E1492A9D2E345A6B42D768@bp1.ad.bp.com>
References: <21A169E64A904A188DC34E67B1A17651@bp1.ad.bp.com>
	<C3424C68-4E69-4CA1-A494-2B143C0E8E8F@freeswitch.org>
	<5A041CBBECB441D094B86C935CC6C792@bp1.ad.bp.com>
	<1F2B2E59-6C17-4DB8-A197-E57D2A804132@freeswitch.org>
	<DF8D6745E8E1492A9D2E345A6B42D768@bp1.ad.bp.com>
Message-ID: <87f2f3b91001211120q478a6d71nc8b2afb2799e91ec@mail.gmail.com>

On Wed, Jan 20, 2010 at 11:44 AM, Dave Stevenson
<stevendt at primrosebank.net>wrote:

>  Brian,
>
> Following on from before, there's one item that I can't see how to do
> outside of modifying the directory\default.xml file, and that is setting up
> call groups.
>
> I thought that I'd perhaps be able to do something similar to creating user
> dial plans and create a new file in directory\default\  which would be
> loaded before the other extensions, i.e., called something like
> 00_groups.xml and have the call group created there. (My test file is shown
> below). That did not seem to work, am I on the right lines or should custom
> groups get created somewhere else ?
>
>

This leads to the bigger question of where people should put their
customizations without breaking the defaults. A good example of the issue is
with ivr.conf.xml and the corresponding phrase macros in
conf/lang/en/en.xml. In some cases it is impossible to add customizations
without modifying the default config files.

I will bring this up on tomorrow's conference call. If you want to be heard
on this topic then please join us on the call. I will get the agenda posted
ASAP.

-MC
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From stevendt at primrosebank.net  Thu Jan 21 11:41:48 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Thu, 21 Jan 2010 19:41:48 -0000
Subject: [Freeswitch-users] Configuration PreservationthroughTrunkUpdates
References: <21A169E64A904A188DC34E67B1A17651@bp1.ad.bp.com><C3424C68-4E69-4CA1-A494-2B143C0E8E8F@freeswitch.org><5A041CBBECB441D094B86C935CC6C792@bp1.ad.bp.com><1F2B2E59-6C17-4DB8-A197-E57D2A804132@freeswitch.org><DF8D6745E8E1492A9D2E345A6B42D768@bp1.ad.bp.com>
	<87f2f3b91001211120q478a6d71nc8b2afb2799e91ec@mail.gmail.com>
Message-ID: <EE53C4F234D848EBA490E0A82847257F@bp1.ad.bp.com>

Hi Michael,

thanks for picking this up - I'm glad that it's something that is relevant to a wider audience than just me !

I'll try and join the call tomorrow - if only to listen in,

regards
Dave
  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, January 21, 2010 7:20 PM
  Subject: Re: [Freeswitch-users] Configuration PreservationthroughTrunkUpdates





  On Wed, Jan 20, 2010 at 11:44 AM, Dave Stevenson <stevendt at primrosebank.net> wrote:

    Brian,

    Following on from before, there's one item that I can't see how to do outside of modifying the directory\default.xml file, and that is setting up call groups.

    I thought that I'd perhaps be able to do something similar to creating user dial plans and create a new file in directory\default\  which would be loaded before the other extensions, i.e., called something like 00_groups.xml and have the call group created there. (My test file is shown below). That did not seem to work, am I on the right lines or should custom groups get created somewhere else ?


  This leads to the bigger question of where people should put their customizations without breaking the defaults. A good example of the issue is with ivr.conf.xml and the corresponding phrase macros in conf/lang/en/en.xml. In some cases it is impossible to add customizations without modifying the default config files. 

  I will bring this up on tomorrow's conference call. If you want to be heard on this topic then please join us on the call. I will get the agenda posted ASAP.

  -MC




------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From msc at freeswitch.org  Thu Jan 21 11:59:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Jan 2010 11:59:49 -0800
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
Message-ID: <87f2f3b91001211159x55f9be96qb9278c4516b4c5ec@mail.gmail.com>

Can you do two things:

Get a debug log of openzap loading, put it on pastebin. (you can unload
mod_openzap and then press F8 and type load mod_openzap, capturing the
output. Also do "oz list" and "oz dump 1" and pastebin the output.

Pastebin your openzap.conf.xml, openzap.conf, and smg_prid.conf files.

-MC

On Wed, Jan 20, 2010 at 9:04 PM, Thangappan.M <thangappan143 at gmail.com>wrote:

> OpenZap is loading the ss7 signalling type. As per your concern openzap
> does not know the details of the signalling then how it is loading the
> ss7_boost libraries?
>
> FreeSWITCH log:
> -----------------------------
> 2010-01-21 10:10:46.707844 [INFO] zap_io.c:2374 Configured 30 channel(s)
> 2010-01-21 10:10:46.708600 [INFO] zap_io.c:2468 Loading SIG from
> /usr/local/freeswitch/mod/ozmod_ss7_boost.so
> 2010-01-21 10:10:46.709031 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost'
> 2010-01-21 10:10:46.709466 [DEBUG] ss7_boost_client.c:124 Creating L=
> 127.0.0.65:53000 R=127.0.0.66:53000
> 2010-01-21 10:10:46.709834 [DEBUG] ss7_boost_client.c:124 Creating L=
> 127.0.0.65:53001 R=127.0.0.66:53001
> 2010-01-21 10:10:46.710424 [WARNING] ss7_boost_client.c:244 TX EVENT (P):
> SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>
> The signalling type might be anything but when I used the oz list command
> it showed the span details. But I am unable to make a inbound and outbound
> call.
>
> Outbound call result:
> ============
> >  originate openzap/smg_prid/a/9940464753 openzap/smg_prid/a/9843171457
> -ERR NORMAL_CIRCUIT_CONGESTION
>
> 2010-01-21 10:26:14.304816 [CRIT] ozmod_ss7_boost.c:244 SPAN is not online.
> freeswitch at internal> 2010-01-21 10:26:14.304816 [ERR] mod_openzap.c:1043
> No channels available
> 2010-01-21 10:26:14.304816 [ERR] switch_ivr_originate.c:1510 Cannot create
> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
> 2010-01-21 10:26:14.304816 [DEBUG] switch_ivr_originate.c:2138 Originate
> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>
> Inbound call result:
> -----------------------------
>
>      I made incoming call for the dial plan which is specified in the
> earlier post at that time it said the number is busy. We did the packet
> capture using the following command.
>
> wanpipemon -i w1g1 -pcap -pcap file isdn.pcap -port ISDN -full -systime -c
> trd
>
> Here I attached the pcap file for that.
>
>
> Where I did mistake or Did I miss any thing to do?
> Please help me.......
>
>
>
> On Wed, Jan 20, 2010 at 7:40 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>>
>>       I noticed the 'oz list' output  in that span type is 'ss7 (boost)'.
>> How can I change this to isdn?
>>
>>
>>
>> On Wed, Jan 20, 2010 at 12:43 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>
>>> I found the error in it. The file name is used as openzap.conf.xml (
>>> smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
>>> PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
>>> openzap.conf.wiki.xml  file.
>>>
>>> Now the another problem is raised here.
>>> When I was using oz list command , the details of the smg_prid shown.
>>> When I was using 'oz dump smg_prid' command it shows all the channels'
>>> details. But all the channels' states are DOWN. why? How can I make it the
>>> states to UP?
>>>
>>> When I was making the call , the number is busy message was get. The call
>>> was not  at all landed to the freeswitch.
>>>
>>> Dial plan Example:
>>> -------------------------------
>>>  <extension name="PRI_Testing">
>>>         <condition field="destination_number" expression="^39114600$">
>>>                 <action application="playback"
>>> data="ivr-welcome_to_freeswitch"/>
>>>         </condition>
>>>   </extension>
>>>
>>> Please help me...........
>>>
>>> *Output Reference:*
>>> http://pastebin.org/79074
>>>
>>>
>>> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>>
>>>> Dear all,
>>>>
>>>>          I have successfully configured wanpipe with freeswitch. When I
>>>> was the running wancfg_fs script the following files openzap.conf ,
>>>> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
>>>> are created.
>>>>
>>>>          I started the wanrouter command then executed the freeswitch.
>>>>          When I was executing freeswitch mod_openzap.c said the error
>>>> as  "Error  for finding the span id. name:PRI_1".
>>>>          But in the openzap.conf and openzap.conf.xml files the span
>>>> name is smg_prid.
>>>>
>>>>          Why the freeswitch is referring the span name as PRI_1 ?
>>>>          Whether this has to configured in anywhere?
>>>>
>>>>          In the freeswitch CLI using oz command I tried to dump the
>>>> PRI_1 span id but it said te error  as "PRI_1 is not found".  When I was
>>>> trying the command 'oz dump smg_prid'  all the channel states and details
>>>> shown.
>>>>
>>>>         It seems that smg_prid span configured in openzap perfectly (Its
>>>> my assumption). Then Why freeswitch is referring the span name as PRI_1.
>>>>
>>>> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>>>>
>>>> Could anyone please help me?
>>>>
>>>> REFERENCE:
>>>>
>>>> openzap.conf
>>>> [span wanpipe smg_prid]
>>>> name => smg_prid
>>>> trunk_type =>e1
>>>> b-channel => 1:1-15
>>>> b-channel => 1:17-31
>>>>
>>>>
>>>> openzap.conf.xml
>>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>>   <settings>
>>>>     <param name="debug" value="0"/>
>>>>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>>     <!--<param name="enable-analog-option" value="3-way"/>-->
>>>>   </settings>
>>>>   <boost_spans>
>>>>         <span id="smg_prid">
>>>>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>         <param name="dialplan" value="XML"/>
>>>>         <param name="context" value="default"/>
>>>>          <!-- regex to stop dialing when it matches -->
>>>>         <!--<param name="dial-regex" value="5555"/>-->
>>>>         <!-- regex to stop dialing when it does not match -->
>>>>         <!--<param name="fail-dial-regex" value="^5"/>-->
>>>>         </span>
>>>>  </boost_spans>
>>>>  <analog_spans>
>>>>  </analog_spans>
>>>> </configuration>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Thangappan.M
>>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Thangappan.M
>>>
>>
>>
>>
>> --
>> Regards,
>> Thangappan.M
>>
>
>
>
> --
> Regards,
> Thangappan.M
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Thu Jan 21 12:08:37 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 21 Jan 2010 14:08:37 -0600
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <87f2f3b91001211159x55f9be96qb9278c4516b4c5ec@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
	<87f2f3b91001211159x55f9be96qb9278c4516b4c5ec@mail.gmail.com>
Message-ID: <1873B030-24AA-4B2E-8444-E4678591D2B4@freeswitch.org>

You're running old FreeSWITCH you'll need the very latest.

/b

On Jan 21, 2010, at 1:59 PM, Michael Collins wrote:

> 2010-01-21 10:10:46.710424 [WARNING] ss7_boost_client.c:244 TX EVENT (P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0




From lists at redbonez.net  Thu Jan 21 12:12:30 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 21 Jan 2010 13:12:30 -0700
Subject: [Freeswitch-users] Can't register Polycom
In-Reply-To: <4B588A4F.5060303@gmail.com>
References: <01a301ca9a59$17e45fd0$47ad1f70$@com> <4B588A4F.5060303@gmail.com>
Message-ID: <003601ca9ad6$108bb2b0$31a31810$@net>

I dunno about the 550s, but on my Polycom IP501 I had to configure the auth
username both from the phone interface and the web interface to get it to
stick.

 

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Meftah
Tayeb
Sent: Thursday, January 21, 2010 10:10 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't register Polycom

 

hi,
make sure that you are using the user: 1008 and pass: 1008
don't forgot to add autorisation user: 1008
the dialplan don't have any relation with the user/pass acording to your
post, you are montioning the dialplan path but you need to edit the
directory (conf/directory/default/1008.xml)
chedck your freeswitch ip and configure your policom ip acording to your fs
ip
good luck
Le 21/01/2010 06:17, Lars Zeb a ?crit : 

I am having trouble registering a Polycom 550. >From the siptrace it looks
like there is no username coming from the Polycom. I configured the Polycom
via the web interface. I have inserted ?1008? in the ?Display Name?,
?Address? and ?Auth User ID? fields.

 

In conf/dialplan/default/1008.xml  the first line is <user id="1008"
cidr="192.168.10.105/24">.

 

What am I missing?

 

Thanks, Lars

 

http://pastebin.freeswitch.org/11875

 

 

 
 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  

 

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From msc at freeswitch.org  Thu Jan 21 12:45:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Jan 2010 12:45:18 -0800
Subject: [Freeswitch-users] Additional endpoints
In-Reply-To: <00d701ca9a1e$18afa0e0$4a0ee2a0$@com>
References: <00d701ca9a1e$18afa0e0$4a0ee2a0$@com>
Message-ID: <87f2f3b91001211245l1b724266ibe99f7001d757b33@mail.gmail.com>

On Wed, Jan 20, 2010 at 2:15 PM, Lars Zeb <larclap at yahoo.com> wrote:

>  I know this has been answered before, but I cannot find it.
>
>
>
> How do I setup more than the default 20 endpoints (1000-1019)?  Do I extend
> the definition in dialplan/public.xml (public_extensions) and add the extra
> in directory/default?
>
> This topic is covered in the FreeSWITCH article in Linux Mag:
http://bit.ly/EpVrv

-MC
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From msc at freeswitch.org  Thu Jan 21 12:49:36 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Jan 2010 12:49:36 -0800
Subject: [Freeswitch-users] All channels are frozen while receiving DTMF
In-Reply-To: <27255181.post@talk.nabble.com>
References: <27255181.post@talk.nabble.com>
Message-ID: <87f2f3b91001211249o531eda47m68658b016cacdfef@mail.gmail.com>

On Thu, Jan 21, 2010 at 1:47 AM, Oscav <oscav at hotmail.fr> wrote:

>
> Hi,
>
> I'm running a script that gets some DTMF from caller. I found that when a
> caller is entering DTMF , all the others channels are frozen until all the
> DTMF  are received. In the logs I see that each DTMF takes 1 second. It
> means that if the caller enters 10 digits then all the other running
> scripts
> are paused for 10 seconds. The problem is exponential with traffic load.
>
> Anyone have an idea ??
>
> What version of FS? What OS are you running? Which scripting language?
Please pastebin your script and a debug log of this symptom happening. Of
course, you should be on the latest version of FreeSWITCH first because the
devs frequently fix bugs and add features. :)

-MC
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From msc at freeswitch.org  Thu Jan 21 13:00:34 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Jan 2010 13:00:34 -0800
Subject: [Freeswitch-users] G729 coded issues
In-Reply-To: <cd93b3701001210210v2ddba8afwc6a310540d2e5180@mail.gmail.com>
References: <cd93b3701001190831u4147214cn9832c6f469c6d17e@mail.gmail.com>
	<6D67C7A9-42CD-4F5F-AD18-E9568098FB71@freeswitch.org>
	<4B55E794.6020909@coppice.org>
	<cd93b3701001200514p4acc5a5bk26a463164f841ad2@mail.gmail.com>
	<537338B1-B582-463E-8EFE-7BBED8165D2B@freeswitch.org>
	<cd93b3701001210210v2ddba8afwc6a310540d2e5180@mail.gmail.com>
Message-ID: <87f2f3b91001211300q18449180hfe7e51d2e86b74f9@mail.gmail.com>

On Thu, Jan 21, 2010 at 2:10 AM, Ahmed Naji <a.alalousi at gmail.com> wrote:

> Hi Brian,
>
> All for it, so ye, let me have the beta.
>
> The only reason I went Howler is because of a pressing need.
>
> Moreover, I'm willing to put mine and my teams resources into the project.
> I and others cut code in C++/C/Perl/....etc. As an organisation, we are
> actively involved in telecoms consultancy and software development, and I
> really would like to put our backs into pushing the case for FS wherever
> possible.
>
> Ping me offline on the subscription e-mail for this account, and let's
> exchange details if there is interest your side.
>
> Support ? we are all sold on FS down here believe me, so you can count on
> it.
>
> Regards,
>
> Ahmed.
>
Ahmed, that's great to hear. You may want to join the community conference
calls that we have every Friday. It's 11:00AM CST (or 16:00 GMT) and we'd
love to have new people join up. The info can be found here:
http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_22

-MC
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From msc at freeswitch.org  Thu Jan 21 14:12:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Jan 2010 14:12:23 -0800
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <1b46b4e81001210335l42baef16r8a2952aa5b92f6e6@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
	<e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
	<5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>
	<5d2828f1001201406p4154b98ald1af1c5c25f59337@mail.gmail.com>
	<1b46b4e81001210335l42baef16r8a2952aa5b92f6e6@mail.gmail.com>
Message-ID: <87f2f3b91001211412h7aa0a84ageadb9557b869ca01@mail.gmail.com>

On Thu, Jan 21, 2010 at 3:35 AM, Nicolas Brenner <nicolas at medularis.com>wrote:

>
> On Wed, Jan 20, 2010 at 7:06 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:
>
>> So, I've been reading about early media in the wiki, and have made a
>> little progress, which leads to more questions.
>>
>> I understand now why a call is considered connected before one person has
>> picked up the phone. I am also able to get my script to wait for the phone
>> to be picked up, by setting the ignore_early_media variable when starting a
>> new session, like this:
>>
>> customerSession =
>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/example.com/"
>> .. customerPhoneNumber)
>>
>>
>> After that line, the script waits for the other phone to be picked up.
>>
>> However, now I wonder what to do with calls that don't complete, get busy
>> signals, etc.
>>
>> What do people do in this case? The only related example I can find on the
>> web is for a javascript dialer, which doesn't address any of these cases.
>>
>
>
> I guess it depends on what you want to do. For example I have a lua script
> very similar to what you describe, although there is no confirmation
> involved. Depending on the hangup cause the session gets, it might try
> redialing with a different gateway, try again or just hangup.
>
> Take a look here http://wiki.freeswitch.org/wiki/Hangup_causes to see what
> each hangup cause means. You don't need to have a special case for all of
> them, only the ones you are interested in.
>
> Here's an example in code which retries a call depending on the hangup
> cause. It retries max_retries1 times and alternates between 2 different
> gateways:
>
> session1 = null;
> max_retries1 = 3;
> retries = 0;
> ostr = "";
> repeat
>         retries = retries + 1;
>         if (retries % 2) then ostr = originate_str1;
>         else ostr = originate_str12; end
>         freeswitch.consoleLog("notice", "*********** Dialing Leg1: " ..
> ostr .. " - Try: "..retries.." ***********\n");
>         session1 = freeswitch.Session(ostr);
>         local hcause = session1:hangupCause();
>         freeswitch.consoleLog("notice", "*********** Leg1: " .. hcause .. "
> - Try: "..retries.." ***********\n");
> until not ((hcause == 'NO_ROUTE_DESTINATION' or hcause ==
> 'RECOVERY_ON_TIMER_EXPIRE' or hcause == 'INCOMPATIBLE_DESTINATION' or hcause
> == 'CALL_REJECTED' or hcause == 'NORMAL_TEMPORARY_FAILURE') and (retries <
> max_retriesl1))
>
>
> Note: originate_str1 and originate_str2 are two different dial strings for
> 2 different gateways.
>
>
Nicolas,

This is really nice. Would you be willing to add this script and a brief
explanation to the wiki? You could create a whole new page and just link to
it from here: http://wiki.freeswitch.org/wiki/Mod_lua#More_Samples

If you have any questions please let me know!
-MC
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From jerry.richards at teotech.com  Thu Jan 21 14:39:28 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 21 Jan 2010 14:39:28 -0800
Subject: [Freeswitch-users] Presence Not Working After Version 1.0.5pre9?
In-Reply-To: <191c3a031001201531g7c78cb9fw1ed1a2ba07f5773c@mail.gmail.com>
References: <mailman.5334.1264017641.2495.freeswitch-users@lists.freeswitch.org><C726F91949324BDFA34596256B6E102A@greyhawk.tonecommander.com>
	<191c3a031001201531g7c78cb9fw1ed1a2ba07f5773c@mail.gmail.com>
Message-ID: <B4EB2C4365CD4CA79B54134ECF232EED@greyhawk.tonecommander.com>

Yes you are correct.  The Bria Softphone has a setting under
ContactProfile/Advanced.../Account menu which is required to be the
softphone's extension (not blank and not the extension that is being
subscribed to).  After I set this field to the softphone's extension, FS
starting reporting the the contact's presence status.
 
Thanks and Best Regards,
Jerry
 


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Wednesday, January 20, 2010 3:32 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Not Working After Version
1.0.5pre9?


try turning on sip debug and console loglevel debug

sofia loglevel all 9
console loglevel debug

Did you try manually running the same sql stmts from the sqlite3 app?
maybe you have something misconfigured.



On Wed, Jan 20, 2010 at 4:44 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


Does anyone know why I do not see NOTIFY messages with presence status being
sent out from FS for two Bria softphones?  It used to work in my old version
1.0.5pre9.  I upgraded Freeswitch on Jan 12 and Jan 20 and edited in my XML
configurations, but I do not see the NOTIFY messages since version
1.0.5pre9.

I have mostly default configuration and I added the manage-presence=true
setting.  I posted a pastebin at http://pastebin.freeswitch.org/11867.

Can anyone tell why this isn't working?

Best Regards,
Jerry


_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


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From mike at jerris.com  Thu Jan 21 14:56:21 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 21 Jan 2010 17:56:21 -0500
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <37012.20543.qm@web33507.mail.mud.yahoo.com>
References: <37012.20543.qm@web33507.mail.mud.yahoo.com>
Message-ID: <5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>

How about with svn r16440

On Jan 21, 2010, at 8:02 AM, Diego Toro wrote:

> Hi MikeJ, using '\\' the behavior is the same, '\\s' is replaced by ' '.
> 
> <action application="playback" data="misc\\8000\\serror.wav!misc\\8000\\provide_reference_number.wav!digits\\8000\\5.wav"/>  
> 
> Console output error is:
> 
> [ERR] mod_sndfile.c:194 Error Opening File [d:\fs\fs_trunk_20100118\Debug\sounds\es\co\callie\misc\8000 error.wav]
> 
> 
> S.O.: Windows 7
> FreeSwitch: Trunk (svn latest version)

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From troy at tlainvestments.com  Thu Jan 21 15:21:27 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 21 Jan 2010 16:21:27 -0700
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
In-Reply-To: <5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>
References: <37012.20543.qm@web33507.mail.mud.yahoo.com>
	<5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>
Message-ID: <517DA285-7E9B-4226-8AEF-C2DAB52976CF@tlainvestments.com>

Hello,

I'm on the latest trunk version (16440) and having an issue with Polycom and transferring.  The dial plan is set up so that unanswered calls go to voicemail.  When I answer a call with a polycom phone and then transfer that call to another phone, if the other phone doesn't pick up and the voicemail app starts, then I hit transfer again with the intent of having the caller leave a voicemail, the call is dropped.  If the phone does pick up during the transfer, it works fine.

I also have an Aastra phone, and when I do the same thing, but from the Aastra phone, it works as expected.  Is this known to be a problem with Polycom?

Thanks!
Troy


From larclap at yahoo.com  Thu Jan 21 15:36:19 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Thu, 21 Jan 2010 15:36:19 -0800
Subject: [Freeswitch-users] Can't register Polycom
In-Reply-To: <003601ca9ad6$108bb2b0$31a31810$@net>
References: <01a301ca9a59$17e45fd0$47ad1f70$@com> <4B588A4F.5060303@gmail.com>
	<003601ca9ad6$108bb2b0$31a31810$@net>
Message-ID: <02e101ca9af2$887829e0$99687da0$@com>

Thanks for that hint. In the meantime I just used Line 3. I?ll try your
suggestion soon.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam
Ford
Sent: Thursday, January 21, 2010 12:13 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't register Polycom

 

I dunno about the 550s, but on my Polycom IP501 I had to configure the auth
username both from the phone interface and the web interface to get it to
stick.

 

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Meftah
Tayeb
Sent: Thursday, January 21, 2010 10:10 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't register Polycom

 

hi,
make sure that you are using the user: 1008 and pass: 1008
don't forgot to add autorisation user: 1008
the dialplan don't have any relation with the user/pass acording to your
post, you are montioning the dialplan path but you need to edit the
directory (conf/directory/default/1008.xml)
chedck your freeswitch ip and configure your policom ip acording to your fs
ip
good luck
Le 21/01/2010 06:17, Lars Zeb a ?crit : 

I am having trouble registering a Polycom 550. >From the siptrace it looks
like there is no username coming from the Polycom. I configured the Polycom
via the web interface. I have inserted ?1008? in the ?Display Name?,
?Address? and ?Auth User ID? fields.

 

In conf/dialplan/default/1008.xml  the first line is <user id="1008"
cidr="192.168.10.105/24">.

 

What am I missing?

 

Thanks, Lars

 

http://pastebin.freeswitch.org/11875

 

 

 
 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  

 

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From lists at redbonez.net  Thu Jan 21 15:42:39 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 21 Jan 2010 16:42:39 -0700
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
In-Reply-To: <517DA285-7E9B-4226-8AEF-C2DAB52976CF@tlainvestments.com>
References: <37012.20543.qm@web33507.mail.mud.yahoo.com>	<5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>
	<517DA285-7E9B-4226-8AEF-C2DAB52976CF@tlainvestments.com>
Message-ID: <004e01ca9af3$6c5a2d20$450e8760$@net>

Yes it is a known issue with Polycom phones. Polycom supports a non-standard
transfer method which does not work with FreeSWITCH. 

See this article for further details -
http://www.junctionnetworks.com/knowledgebase/onsip/phones-routers-and-devic
es/phone-configuration/polycom/polycom-disabling-non-standard-transfer-meth

I ran into the same problem, disabling
voIpProt.SIP.allowTransferOnProceeding as suggested in that article resolved
the issue for me.

-Adam

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Troy
Anderson
Sent: Thursday, January 21, 2010 4:21 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail

Hello,

I'm on the latest trunk version (16440) and having an issue with Polycom and
transferring.  The dial plan is set up so that unanswered calls go to
voicemail.  When I answer a call with a polycom phone and then transfer that
call to another phone, if the other phone doesn't pick up and the voicemail
app starts, then I hit transfer again with the intent of having the caller
leave a voicemail, the call is dropped.  If the phone does pick up during
the transfer, it works fine.

I also have an Aastra phone, and when I do the same thing, but from the
Aastra phone, it works as expected.  Is this known to be a problem with
Polycom?

Thanks!
Troy
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From brian at freeswitch.org  Thu Jan 21 15:45:32 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 21 Jan 2010 17:45:32 -0600
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
In-Reply-To: <004e01ca9af3$6c5a2d20$450e8760$@net>
References: <37012.20543.qm@web33507.mail.mud.yahoo.com>	<5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>
	<517DA285-7E9B-4226-8AEF-C2DAB52976CF@tlainvestments.com>
	<004e01ca9af3$6c5a2d20$450e8760$@net>
Message-ID: <84550BFB-81A3-4CF4-8CD0-9CC4F29A66F0@freeswitch.org>

No this works.  But not to an IVR I suspect.

This is when people don't know what blind vs attended means and they could just press one more button and get the same effect.

/b

On Jan 21, 2010, at 5:42 PM, Adam Ford wrote:

> voIpProt.SIP.allowTransferOnProceeding 




From lists at redbonez.net  Thu Jan 21 16:00:06 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 21 Jan 2010 17:00:06 -0700
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
In-Reply-To: <004e01ca9af3$6c5a2d20$450e8760$@net>
References: <37012.20543.qm@web33507.mail.mud.yahoo.com>	<5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>	<517DA285-7E9B-4226-8AEF-C2DAB52976CF@tlainvestments.com>
	<004e01ca9af3$6c5a2d20$450e8760$@net>
Message-ID: <005b01ca9af5$dc759980$9560cc80$@net>

That link didn't come through very well, here is a shortened one -
http://bit.ly/6wDAXD

-Adam

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam
Ford
Sent: Thursday, January 21, 2010 4:43 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom Consultative Transfer and Voicemail

Yes it is a known issue with Polycom phones. Polycom supports a non-standard
transfer method which does not work with FreeSWITCH. 

See this article for further details -
http://www.junctionnetworks.com/knowledgebase/onsip/phones-routers-and-devic
es/phone-configuration/polycom/polycom-disabling-non-standard-transfer-meth

I ran into the same problem, disabling
voIpProt.SIP.allowTransferOnProceeding as suggested in that article resolved
the issue for me.

-Adam

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Troy
Anderson
Sent: Thursday, January 21, 2010 4:21 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail

Hello,

I'm on the latest trunk version (16440) and having an issue with Polycom and
transferring.  The dial plan is set up so that unanswered calls go to
voicemail.  When I answer a call with a polycom phone and then transfer that
call to another phone, if the other phone doesn't pick up and the voicemail
app starts, then I hit transfer again with the intent of having the caller
leave a voicemail, the call is dropped.  If the phone does pick up during
the transfer, it works fine.

I also have an Aastra phone, and when I do the same thing, but from the
Aastra phone, it works as expected.  Is this known to be a problem with
Polycom?

Thanks!
Troy
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From mouncifbb at gmail.com  Thu Jan 21 15:58:37 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Thu, 21 Jan 2010 18:58:37 -0500
Subject: [Freeswitch-users] Javascript self.session.getVariable
Message-ID: <ade741d61001211558h7b84cb96q16754650812a80e2@mail.gmail.com>

I have the following in javascript:

caller_id = self.session.getVariable("caller_id_number")


for some reasons it returns:
ReferenceError: self is not defined, I am following this page:
http://wiki.freeswitch.org/wiki/Session_getVariable


any ideas?
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From anthony.minessale at gmail.com  Thu Jan 21 16:48:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Jan 2010 18:48:17 -0600
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
In-Reply-To: <005b01ca9af5$dc759980$9560cc80$@net>
References: <37012.20543.qm@web33507.mail.mud.yahoo.com>
	<5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>
	<517DA285-7E9B-4226-8AEF-C2DAB52976CF@tlainvestments.com>
	<004e01ca9af3$6c5a2d20$450e8760$@net>
	<005b01ca9af5$dc759980$9560cc80$@net>
Message-ID: <191c3a031001211648i5ed86253k8640c49f29121e0c@mail.gmail.com>

if you used the loopback endpoint to loop around to voicemail or made a
looped sip call back to your own box you could xfer it as desired.


bridge to "loopback/app=voicemail:default ${domain_name}
${dialed_extension}"

That will make the vm app run as a channel instead of an inline app.

This is an undocumented feature because it's not well tested so if it
doesn't work *shrug* =D



On Thu, Jan 21, 2010 at 6:00 PM, Adam Ford <lists at redbonez.net> wrote:

> That link didn't come through very well, here is a shortened one -
> http://bit.ly/6wDAXD
>
> -Adam
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam
> Ford
> Sent: Thursday, January 21, 2010 4:43 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
>
> Yes it is a known issue with Polycom phones. Polycom supports a
> non-standard
> transfer method which does not work with FreeSWITCH.
>
> See this article for further details -
>
> http://www.junctionnetworks.com/knowledgebase/onsip/phones-routers-and-devic
> es/phone-configuration/polycom/polycom-disabling-non-standard-transfer-meth<http://www.junctionnetworks.com/knowledgebase/onsip/phones-routers-and-devic%0Aes/phone-configuration/polycom/polycom-disabling-non-standard-transfer-meth>
>
> I ran into the same problem, disabling
> voIpProt.SIP.allowTransferOnProceeding as suggested in that article
> resolved
> the issue for me.
>
> -Adam
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Troy
> Anderson
> Sent: Thursday, January 21, 2010 4:21 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
>
> Hello,
>
> I'm on the latest trunk version (16440) and having an issue with Polycom
> and
> transferring.  The dial plan is set up so that unanswered calls go to
> voicemail.  When I answer a call with a polycom phone and then transfer
> that
> call to another phone, if the other phone doesn't pick up and the voicemail
> app starts, then I hit transfer again with the intent of having the caller
> leave a voicemail, the call is dropped.  If the phone does pick up during
> the transfer, it works fine.
>
> I also have an Aastra phone, and when I do the same thing, but from the
> Aastra phone, it works as expected.  Is this known to be a problem with
> Polycom?
>
> Thanks!
> Troy
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From styli1 at hotmail.com  Thu Jan 21 16:55:21 2010
From: styli1 at hotmail.com (styli1 at hotmail.com)
Date: Thu, 21 Jan 2010 16:55:21 -0800
Subject: [Freeswitch-users] Vacation reply
In-Reply-To: <191c3a031001211648i5ed86253k8640c49f29121e0c@mail.gmail.com>
Message-ID: <SNT0-MC1-F211A2E5BC7F772D7C5E81A85620@phx.gbl>

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From jingwei.yang at gmail.com  Thu Jan 21 17:05:13 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Fri, 22 Jan 2010 09:05:13 +0800
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <1264092620.14614.73.camel@luna.tc.commsmundi.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
	<1264092620.14614.73.camel@luna.tc.commsmundi.com>
Message-ID: <13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>

Thanks for the reply. All the agents are dynamic and I can't predefine them
in the config file.

Regards,
-Jingwei

On Fri, Jan 22, 2010 at 12:50 AM, Fran?ois Delawarde <
fdelawarde at wirelessmundi.com> wrote:

> Why do you need 2 fifos? You could have callback agents connected to the
> fifo and send incoming calls there, mod_fifo will do the rest.
>
> To configure agents for callback:
> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>
> To place a call into a fifo:
> http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue
>
> Fran?ois.
>
> On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
> > Sorry about the confusion, I'm just trying to think over all the
> > abnormal situations before the implementation and hope the flow has no
> > design flaws.
> >
> > Client A is parked in Queue 1, multiple consumers will be ringed to
> > answer him. And once the first one is connected, all the rest will
> > hang up. This is the targeted function.
> >
> > To achieve this, I'm considering to originate a call to each consumer
> > and put the calls in Queue 2. Then intercept client A to the first
> > element of Queue 2.
> >
> > I'm not sure if it's the usual or the best way. If you feel not,
> > please don't hesitate to correct me. Any thoughts are warmly
> > appreciated.
> >
> > On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com> wrote:
> >         what no errors mean? how do you originate calls to consumers?
> >         I don't
> >         understand your scenario.
> >
> >         2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
> >
> >         > Hi All,
> >         >
> >         > Please advise whether the following flow makes sense.
> >         >
> >         > 1. Client A calls in and parked in Queue 1
> >         > 2. Originate calls to several consumers simultaneously and
> >         park them in
> >         > Queue 2
> >         > 3. Intercept A's call to the first consumer of Queue 2
> >         >
> >         > Basically I want A's call picked up by the first among many
> >         consumers with
> >         > no errors. Please let me know whether I'm on the right
> >         track.
> >         >
> >         > Thanks and best regards,
> >         > -Jingwei
> >         >
> >
> >
> >         _______________________________________________
> >         FreeSWITCH-users mailing list
> >         FreeSWITCH-users at lists.freeswitch.org
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From styli1 at hotmail.com  Thu Jan 21 17:10:58 2010
From: styli1 at hotmail.com (styli1 at hotmail.com)
Date: Thu, 21 Jan 2010 17:10:58 -0800
Subject: [Freeswitch-users] Vacation reply
In-Reply-To: <13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
Message-ID: <COL0-MC4-F33AC87B2CCB44670A656DB85620@phx.gbl>

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From jmesquita at freeswitch.org  Thu Jan 21 17:21:14 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 21 Jan 2010 22:21:14 -0300
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
	<1264092620.14614.73.camel@luna.tc.commsmundi.com>
	<13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
Message-ID: <ebbc2d551001211721n51cb5c54m996e0d494842c383@mail.gmail.com>

Jingwei, check my contrib dir. I think it may help you with one FIFO since
we are able there to sign in and sign off dynamic agents as well as
customize what we do when the FIFO makes a call to them.

Regards,
Jo?o Mesquita
FSComm Developer

On Thu, Jan 21, 2010 at 10:05 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:

> Thanks for the reply. All the agents are dynamic and I can't predefine them
> in the config file.
>
> Regards,
> -Jingwei
>
>
> On Fri, Jan 22, 2010 at 12:50 AM, Fran?ois Delawarde <
> fdelawarde at wirelessmundi.com> wrote:
>
>> Why do you need 2 fifos? You could have callback agents connected to the
>> fifo and send incoming calls there, mod_fifo will do the rest.
>>
>> To configure agents for callback:
>> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>>
>> To place a call into a fifo:
>> http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue
>>
>> Fran?ois.
>>
>> On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
>> > Sorry about the confusion, I'm just trying to think over all the
>> > abnormal situations before the implementation and hope the flow has no
>> > design flaws.
>> >
>> > Client A is parked in Queue 1, multiple consumers will be ringed to
>> > answer him. And once the first one is connected, all the rest will
>> > hang up. This is the targeted function.
>> >
>> > To achieve this, I'm considering to originate a call to each consumer
>> > and put the calls in Queue 2. Then intercept client A to the first
>> > element of Queue 2.
>> >
>> > I'm not sure if it's the usual or the best way. If you feel not,
>> > please don't hesitate to correct me. Any thoughts are warmly
>> > appreciated.
>> >
>> > On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com> wrote:
>> >         what no errors mean? how do you originate calls to consumers?
>> >         I don't
>> >         understand your scenario.
>> >
>> >         2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
>> >
>> >         > Hi All,
>> >         >
>> >         > Please advise whether the following flow makes sense.
>> >         >
>> >         > 1. Client A calls in and parked in Queue 1
>> >         > 2. Originate calls to several consumers simultaneously and
>> >         park them in
>> >         > Queue 2
>> >         > 3. Intercept A's call to the first consumer of Queue 2
>> >         >
>> >         > Basically I want A's call picked up by the first among many
>> >         consumers with
>> >         > no errors. Please let me know whether I'm on the right
>> >         track.
>> >         >
>> >         > Thanks and best regards,
>> >         > -Jingwei
>> >         >
>> >
>> >
>> >         _______________________________________________
>> >         FreeSWITCH-users mailing list
>> >         FreeSWITCH-users at lists.freeswitch.org
>> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >         UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >         http://www.freeswitch.org
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jingwei.yang at gmail.com  Thu Jan 21 18:06:22 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Fri, 22 Jan 2010 10:06:22 +0800
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <ebbc2d551001211721n51cb5c54m996e0d494842c383@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
	<1264092620.14614.73.camel@luna.tc.commsmundi.com>
	<13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
	<ebbc2d551001211721n51cb5c54m996e0d494842c383@mail.gmail.com>
Message-ID: <13529f9d1001211806o35fc3435j93b71bbb2ede3028@mail.gmail.com>

Hi Jo?o, thanks for the reply. I'll try it out.

Regards,
-Jingwei

2010/1/22 Jo?o Mesquita <jmesquita at freeswitch.org>

> Jingwei, check my contrib dir. I think it may help you with one FIFO since
> we are able there to sign in and sign off dynamic agents as well as
> customize what we do when the FIFO makes a call to them.
>
> Regards,
> Jo?o Mesquita
> FSComm Developer
>
>
> On Thu, Jan 21, 2010 at 10:05 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>
>> Thanks for the reply. All the agents are dynamic and I can't predefine
>> them in the config file.
>>
>> Regards,
>> -Jingwei
>>
>>
>> On Fri, Jan 22, 2010 at 12:50 AM, Fran?ois Delawarde <
>> fdelawarde at wirelessmundi.com> wrote:
>>
>>> Why do you need 2 fifos? You could have callback agents connected to the
>>> fifo and send incoming calls there, mod_fifo will do the rest.
>>>
>>> To configure agents for callback:
>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>>>
>>> To place a call into a fifo:
>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue
>>>
>>> Fran?ois.
>>>
>>> On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
>>> > Sorry about the confusion, I'm just trying to think over all the
>>> > abnormal situations before the implementation and hope the flow has no
>>> > design flaws.
>>> >
>>> > Client A is parked in Queue 1, multiple consumers will be ringed to
>>> > answer him. And once the first one is connected, all the rest will
>>> > hang up. This is the targeted function.
>>> >
>>> > To achieve this, I'm considering to originate a call to each consumer
>>> > and put the calls in Queue 2. Then intercept client A to the first
>>> > element of Queue 2.
>>> >
>>> > I'm not sure if it's the usual or the best way. If you feel not,
>>> > please don't hesitate to correct me. Any thoughts are warmly
>>> > appreciated.
>>> >
>>> > On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com> wrote:
>>> >         what no errors mean? how do you originate calls to consumers?
>>> >         I don't
>>> >         understand your scenario.
>>> >
>>> >         2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
>>> >
>>> >         > Hi All,
>>> >         >
>>> >         > Please advise whether the following flow makes sense.
>>> >         >
>>> >         > 1. Client A calls in and parked in Queue 1
>>> >         > 2. Originate calls to several consumers simultaneously and
>>> >         park them in
>>> >         > Queue 2
>>> >         > 3. Intercept A's call to the first consumer of Queue 2
>>> >         >
>>> >         > Basically I want A's call picked up by the first among many
>>> >         consumers with
>>> >         > no errors. Please let me know whether I'm on the right
>>> >         track.
>>> >         >
>>> >         > Thanks and best regards,
>>> >         > -Jingwei
>>> >         >
>>> >
>>> >
>>> >         _______________________________________________
>>> >         FreeSWITCH-users mailing list
>>> >         FreeSWITCH-users at lists.freeswitch.org
>>> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >         UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >         http://www.freeswitch.org
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fvillarroel at yahoo.com  Thu Jan 21 20:26:10 2010
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Thu, 21 Jan 2010 20:26:10 -0800 (PST)
Subject: [Freeswitch-users] CDR Gateways
Message-ID: <956716.34674.qm@web34301.mail.mud.yahoo.com>

Dear All.

I have defined various gateways in ~/sip-profiles/external

My questions is if a gateway named foo that send calls from diferents IP address like x.x.x.x and x.x.x.y. How i can doing a group for both ip address where i can doing later a sql like accountcode=foo?

In this moment if i need know the traffic of customer foo, i should doing two differents cdr, one for every ip address.

It's fine or i can doing of different way?

I hope anyone could me explain how i can doing. 

my gateway foo.xml

<include>
    <gateway name="foo">
      <param name="username" value="foo"/>
      <param name="password" value="foo"/>
      <param name="realm" value="x.x.x.x"/>
      <param name="register" value="false"/>
      <param name="context" value="public"/>
    </gateway>
</include>

foo1.xml

<include>
    <gateway name="foo1">
      <param name="username" value="foo1"/>
      <param name="password" value="foo1"/>
      <param name="realm" value="x.x.x.y"/>
      <param name="register" value="false"/>
      <param name="context" value="public"/>
    </gateway>
</include>

Both gateways foo and foo1 are the same customer

my cdr_csv.conf.xml

<configuration name="cdr_csv.conf" description="CDR CSV Format">
  <settings>
    <!-- 'cdr-csv' will always be appended to log-base -->
    <!--<param name="log-base" value="/var/log"/>-->
    <param name="default-template" value="sql"/>
    <!-- This is like the info app but after the call is hung up -->
    <!--<param name="debug" value="true"/>-->
    <param name="rotate-on-hup" value="false"/>
    <!-- may be a b or ab -->
    <param name="legs" value="ab"/>
  </settings>
  <templates>
        <template name="sql">INSERT INTO v_cdr (v_id, caller_id_name, caller_id_number, destination_number, context, start_stamp, answer_stamp, end_stamp,
duration, billsec, hangup_cause, uuid, bleg_uuid, accountcode, read_codec, write_codec, remote_media_ip, network_addr) VALUES ("{v_id}", "${caller_id_name}"
,"${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}"
,"${uuid}","${bleg_uuid}", "${accountcode}", "${read_codec}", "${write_codec}", "${remote_media_ip}", "${network_addr}");</template>

The argument accountcode on my database is Blank or None for all records of gateways.
Regards.


      



From thangappan143 at gmail.com  Thu Jan 21 20:45:58 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Fri, 22 Jan 2010 10:15:58 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
Message-ID: <7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>

The following link have the openzap.conf,openzap.conf.xml ,smg_prid.conf ,
debug log of mod_openzap , oz list and oz dump 1 output.

http://pastebin.org/80095


On Thu, Jan 21, 2010 at 10:34 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> OpenZap is loading the ss7 signalling type. As per your concern openzap
> does not know the details of the signalling then how it is loading the
> ss7_boost libraries?
>
> FreeSWITCH log:
> -----------------------------
> 2010-01-21 10:10:46.707844 [INFO] zap_io.c:2374 Configured 30 channel(s)
> 2010-01-21 10:10:46.708600 [INFO] zap_io.c:2468 Loading SIG from
> /usr/local/freeswitch/mod/ozmod_ss7_boost.so
> 2010-01-21 10:10:46.709031 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost'
> 2010-01-21 10:10:46.709466 [DEBUG] ss7_boost_client.c:124 Creating L=
> 127.0.0.65:53000 R=127.0.0.66:53000
> 2010-01-21 10:10:46.709834 [DEBUG] ss7_boost_client.c:124 Creating L=
> 127.0.0.65:53001 R=127.0.0.66:53001
> 2010-01-21 10:10:46.710424 [WARNING] ss7_boost_client.c:244 TX EVENT (P):
> SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>
> The signalling type might be anything but when I used the oz list command
> it showed the span details. But I am unable to make a inbound and outbound
> call.
>
> Outbound call result:
> ============
> >  originate openzap/smg_prid/a/9940464753 openzap/smg_prid/a/9843171457
> -ERR NORMAL_CIRCUIT_CONGESTION
>
> 2010-01-21 10:26:14.304816 [CRIT] ozmod_ss7_boost.c:244 SPAN is not online.
> freeswitch at internal> 2010-01-21 10:26:14.304816 [ERR] mod_openzap.c:1043
> No channels available
> 2010-01-21 10:26:14.304816 [ERR] switch_ivr_originate.c:1510 Cannot create
> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
> 2010-01-21 10:26:14.304816 [DEBUG] switch_ivr_originate.c:2138 Originate
> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>
> Inbound call result:
> -----------------------------
>
>      I made incoming call for the dial plan which is specified in the
> earlier post at that time it said the number is busy. We did the packet
> capture using the following command.
>
> wanpipemon -i w1g1 -pcap -pcap file isdn.pcap -port ISDN -full -systime -c
> trd
>
> Here I attached the pcap file for that.
>
>
> Where I did mistake or Did I miss any thing to do?
> Please help me.......
>
>
>
> On Wed, Jan 20, 2010 at 7:40 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>>
>>       I noticed the 'oz list' output  in that span type is 'ss7 (boost)'.
>> How can I change this to isdn?
>>
>>
>>
>> On Wed, Jan 20, 2010 at 12:43 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>
>>> I found the error in it. The file name is used as openzap.conf.xml (
>>> smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
>>> PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
>>> openzap.conf.wiki.xml  file.
>>>
>>> Now the another problem is raised here.
>>> When I was using oz list command , the details of the smg_prid shown.
>>> When I was using 'oz dump smg_prid' command it shows all the channels'
>>> details. But all the channels' states are DOWN. why? How can I make it the
>>> states to UP?
>>>
>>> When I was making the call , the number is busy message was get. The call
>>> was not  at all landed to the freeswitch.
>>>
>>> Dial plan Example:
>>> -------------------------------
>>>  <extension name="PRI_Testing">
>>>         <condition field="destination_number" expression="^39114600$">
>>>                 <action application="playback"
>>> data="ivr-welcome_to_freeswitch"/>
>>>         </condition>
>>>   </extension>
>>>
>>> Please help me...........
>>>
>>> *Output Reference:*
>>> http://pastebin.org/79074
>>>
>>>
>>> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>>
>>>> Dear all,
>>>>
>>>>          I have successfully configured wanpipe with freeswitch. When I
>>>> was the running wancfg_fs script the following files openzap.conf ,
>>>> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
>>>> are created.
>>>>
>>>>          I started the wanrouter command then executed the freeswitch.
>>>>          When I was executing freeswitch mod_openzap.c said the error
>>>> as  "Error  for finding the span id. name:PRI_1".
>>>>          But in the openzap.conf and openzap.conf.xml files the span
>>>> name is smg_prid.
>>>>
>>>>          Why the freeswitch is referring the span name as PRI_1 ?
>>>>          Whether this has to configured in anywhere?
>>>>
>>>>          In the freeswitch CLI using oz command I tried to dump the
>>>> PRI_1 span id but it said te error  as "PRI_1 is not found".  When I was
>>>> trying the command 'oz dump smg_prid'  all the channel states and details
>>>> shown.
>>>>
>>>>         It seems that smg_prid span configured in openzap perfectly (Its
>>>> my assumption). Then Why freeswitch is referring the span name as PRI_1.
>>>>
>>>> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>>>>
>>>> Could anyone please help me?
>>>>
>>>> REFERENCE:
>>>>
>>>> openzap.conf
>>>> [span wanpipe smg_prid]
>>>> name => smg_prid
>>>> trunk_type =>e1
>>>> b-channel => 1:1-15
>>>> b-channel => 1:17-31
>>>>
>>>>
>>>> openzap.conf.xml
>>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>>   <settings>
>>>>     <param name="debug" value="0"/>
>>>>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>>     <!--<param name="enable-analog-option" value="3-way"/>-->
>>>>   </settings>
>>>>   <boost_spans>
>>>>         <span id="smg_prid">
>>>>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>         <param name="dialplan" value="XML"/>
>>>>         <param name="context" value="default"/>
>>>>          <!-- regex to stop dialing when it matches -->
>>>>         <!--<param name="dial-regex" value="5555"/>-->
>>>>         <!-- regex to stop dialing when it does not match -->
>>>>         <!--<param name="fail-dial-regex" value="^5"/>-->
>>>>         </span>
>>>>  </boost_spans>
>>>>  <analog_spans>
>>>>  </analog_spans>
>>>> </configuration>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Thangappan.M
>>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Thangappan.M
>>>
>>
>>
>>
>> --
>> Regards,
>> Thangappan.M
>>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From troy at tlainvestments.com  Thu Jan 21 20:57:49 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 21 Jan 2010 21:57:49 -0700
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
In-Reply-To: <191c3a031001211648i5ed86253k8640c49f29121e0c@mail.gmail.com>
References: <37012.20543.qm@web33507.mail.mud.yahoo.com>
	<5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>
	<517DA285-7E9B-4226-8AEF-C2DAB52976CF@tlainvestments.com>
	<004e01ca9af3$6c5a2d20$450e8760$@net>
	<005b01ca9af5$dc759980$9560cc80$@net>
	<191c3a031001211648i5ed86253k8640c49f29121e0c@mail.gmail.com>
Message-ID: <8589C894-EE61-4FA7-92E1-5CB9C52EBD60@tlainvestments.com>

I get the idea, but can't seem to get it to work.

I tried doing a bridge to "loopback/app=bridge ${sofia_contact(${dialed_user}@${dialed_domain}", but fs complained => Cannot create outgoing channel of type [loopback=app:sofia]
Also, I tried "loopback/app=voicemail:default ${domain_name} ${dialed_extension}" and it did get to voicemail, but it didn't prompt me for any info - just immediately complained that the recording was too short.

I tried the suggestions about modifying the sip.cfg for the phones and that does work (thanks!), but that forces a Consultative Transfer.  It would be nice to get this method working, which would result in the transfer button doing a Consultive Transfer unless you hang up, then it would be like a blind transfer.

Thanks,
Troy

On Jan 21, 2010, at 5:48 PM, Anthony Minessale wrote:

> if you used the loopback endpoint to loop around to voicemail or made a looped sip call back to your own box you could xfer it as desired.
> 
> 
> bridge to "loopback/app=voicemail:default ${domain_name} ${dialed_extension}"
> 
> That will make the vm app run as a channel instead of an inline app.
> 
> This is an undocumented feature because it's not well tested so if it doesn't work *shrug* =D
> 
> 
> 
> On Thu, Jan 21, 2010 at 6:00 PM, Adam Ford <lists at redbonez.net> wrote:
> That link didn't come through very well, here is a shortened one -
> http://bit.ly/6wDAXD
> 
> -Adam
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam
> Ford
> Sent: Thursday, January 21, 2010 4:43 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
> 
> Yes it is a known issue with Polycom phones. Polycom supports a non-standard
> transfer method which does not work with FreeSWITCH.
> 
> See this article for further details -
> http://www.junctionnetworks.com/knowledgebase/onsip/phones-routers-and-devic
> es/phone-configuration/polycom/polycom-disabling-non-standard-transfer-meth
> 
> I ran into the same problem, disabling
> voIpProt.SIP.allowTransferOnProceeding as suggested in that article resolved
> the issue for me.
> 
> -Adam
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Troy
> Anderson
> Sent: Thursday, January 21, 2010 4:21 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
> 
> Hello,
> 
> I'm on the latest trunk version (16440) and having an issue with Polycom and
> transferring.  The dial plan is set up so that unanswered calls go to
> voicemail.  When I answer a call with a polycom phone and then transfer that
> call to another phone, if the other phone doesn't pick up and the voicemail
> app starts, then I hit transfer again with the intent of having the caller
> leave a voicemail, the call is dropped.  If the phone does pick up during
> the transfer, it works fine.
> 
> I also have an Aastra phone, and when I do the same thing, but from the
> Aastra phone, it works as expected.  Is this known to be a problem with
> Polycom?
> 
> Thanks!
> Troy
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lakindia89 at gmail.com  Thu Jan 21 21:50:02 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 22 Jan 2010 11:20:02 +0530
Subject: [Freeswitch-users] Server Disconnected when SIGINT occured
In-Reply-To: <191c3a031001190804m692f63acsaf852b8809db09d1@mail.gmail.com>
References: <7d79b3931001130113t5e65a400h812db7562ced8702@mail.gmail.com>
	<191c3a031001130818v31376c16j68a8e7e6de040319@mail.gmail.com>
	<7d79b3931001132357i36bbb482jdf2bbdd3aea2a583@mail.gmail.com>
	<7d79b3931001180322p74a1227qe0c2199a77cbfe2@mail.gmail.com>
	<191c3a031001180824q6e364c72g3f789892597e9469@mail.gmail.com>
	<7d79b3931001190107o200ec04dredd76689dd235588@mail.gmail.com>
	<191c3a031001190804m692f63acsaf852b8809db09d1@mail.gmail.com>
Message-ID: <7d79b3931001212150q2533f49l1725ee1d9cd5848f@mail.gmail.com>

Hi all,
I've solved that problem by adding.

use POSIX;
POSIX::sigaction( SIGINT, POSIX::SigAction->new( \&Handler, 0,
POSIX::SA_RESTART),);

This will restart the system calls if that is failed because of the SIGINT
signal.

Provided here as an information...
Thanks all...


On Tue, Jan 19, 2010 at 9:34 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Its nothing we can fix, that is what you must do on a failed read syscall.
> you can do a non blocking read instead and take your chances.
>
>
>
> On Tue, Jan 19, 2010 at 3:07 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> I tried with SIGUSR1, but no progress. I got SERVER_DISCONNECTED.
>> Output:
>>
>> CHILD 3814: Received USR1
>> EVENT [SERVER_DISCONNECTED]
>>
>> In esl.c, in esl_recv_event() function, line no: 824
>> if (rrval < 0) {
>>                         strerror_r(handle->errnum, handle->err,
>> sizeof(handle->err));
>>                         goto fail;
>> }
>> When the program is blocked under receive, I passed the signal. So recv
>> returns -1, and in fail: it call esl_disconnect(handle).
>>
>> Is it because of this??? If so, whether it should be fixed or not???
>>
>>
>>
>>
>> On Mon, Jan 18, 2010 at 9:54 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> try a less famous signal like SIGUSR1 it's possible something in perl
>>> still reacts to SIGINT
>>>
>>>
>>>
>>> On Mon, Jan 18, 2010 at 5:22 AM, lakshmanan ganapathy <
>>> lakindia89 at gmail.com> wrote:
>>>
>>>> Here is the result
>>>>
>>>> Program:
>>>>
>>>> require ESL;
>>>> use IO::Socket::INET;
>>>> use Data::Dumper;
>>>>
>>>> my $ip = "192.168.1.222";
>>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>>> die "Could not create socket: $!\n" unless $sock;
>>>>
>>>>
>>>> for(;;) {
>>>>     my $new_sock = $sock->accept();
>>>>     next if (not defined ($new_sock));
>>>>     my $pid = fork();
>>>>     if ($pid) {
>>>>         close($new_sock);
>>>>         next;
>>>>     }
>>>>     print "CHILD PID: $$\n";
>>>>     &register_Signals_Child();
>>>>     sub register_Signals_Child() {
>>>>         foreach ( keys %SIG ) {
>>>>             $SIG{$_} = 'Handler';
>>>>         }
>>>>     }
>>>>
>>>>     sub Handler() {
>>>>
>>>>         my $handle=$_[0];
>>>>         if($handle eq "INT") {
>>>>             print "CHILD $$: SIGNAL SIG$handle is generated\n";
>>>>         }
>>>>         else
>>>>         {
>>>>             print "CHILD $$: Received $handle\n";
>>>>
>>>>         }
>>>>     }
>>>>     my $host = $new_sock->sockhost();
>>>>     my $fd = fileno($new_sock);
>>>>
>>>>     my $con = new ESL::ESLconnection($fd);
>>>>     my $info = $con->getInfo();
>>>>
>>>>     my $uuid = $info->getHeader("unique-id");
>>>>
>>>>     printf "Connected call %s, from %s\n", $uuid,
>>>> $info->getHeader("caller-caller-id-number");
>>>>     my $r=$con->execute("answer");
>>>>     print Dumper $r;
>>>>     $con->events("plain","all");
>>>>     my
>>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>>     while($con->connected()) {
>>>>         my $e = $con->recvEvent();
>>>>
>>>>         if ($e) {
>>>>             my $name = $e->getHeader("event-name");
>>>>             print "EVENT [$name]\n";
>>>>             if ($name eq "DTMF") {
>>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>>                 print "$digit\n";
>>>>             }
>>>>         }
>>>>     }
>>>>     close($new_sock);
>>>> }
>>>>
>>>> I executed the program and the following things were printed
>>>>
>>>> CHILD PID: 6778
>>>> Connected call e0d1001a-03f4-11df-b002-db488337e0ea, from 1001
>>>> $VAR1 = 0;
>>>> EVENT [CHANNEL_EXECUTE]
>>>> EVENT [CHANNEL_ANSWER]
>>>> EVENT [CHANNEL_EXECUTE_COMPLETE]
>>>> EVENT [COMMAND]
>>>> EVENT [CHANNEL_EXECUTE]
>>>> EVENT [HEARTBEAT]
>>>> EVENT [RE_SCHEDULE]
>>>> EVENT [CHANNEL_EXECUTE_COMPLETE]
>>>>
>>>> Then from another shell I executed kill -2 6778, the result is follows
>>>> CHILD 6778: SIGNAL SIGINT is generated
>>>> EVENT [SERVER_DISCONNECTED]
>>>>
>>>> But the child process is still running as expected.
>>>> But  I don't know why I received SERVER_DISCONNECTED from freeswitch???
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Thu, Jan 14, 2010 at 1:27 PM, lakshmanan ganapathy <
>>>> lakindia89 at gmail.com> wrote:
>>>>
>>>>> I taught the signal handler will be inherited by the child process. It
>>>>> also does like that.
>>>>> After making a call, If I press ctrl + c, the above program printed
>>>>> PARENT PID: Signal SIGINT is generated
>>>>> CHILD PID: Signal SIGINT is generated.
>>>>>
>>>>> So I think the sigal handlers will be inherited to the child.
>>>>> Anyway I'll also try registering signal handlers in child also, and
>>>>> then I'll come back with that result.
>>>>>
>>>>> Thanks....
>>>>> On Wed, Jan 13, 2010 at 9:48 PM, Anthony Minessale <
>>>>> anthony.minessale at gmail.com> wrote:
>>>>>
>>>>>> you would have to register signals in your child process too
>>>>>>
>>>>>>   On Wed, Jan 13, 2010 at 3:13 AM, lakshmanan ganapathy <
>>>>>> lakindia89 at gmail.com> wrote:
>>>>>>
>>>>>>>  Hi all,
>>>>>>>
>>>>>>> I've done a sample program (In perl ESL) , which play a file to the
>>>>>>> caller and then it will call recvEvent() and print the event name. I've
>>>>>>> handled signals also.
>>>>>>>
>>>>>>> When I send SIGINT to my program (Perl), the signal handler is called
>>>>>>> and I can see the print output. But in the same time,  I received
>>>>>>> SERVER_DISCONNECTED from freeswitch as event.
>>>>>>>
>>>>>>> I don't know why I received SERVER_DISCONNECTED from freeswitch. Is
>>>>>>> it because, the recvEvent() from perl internally  calls the recvevent
>>>>>>> function in the Esl.c and when it waits to receive the information from
>>>>>>> socket, the signal occurred???
>>>>>>>
>>>>>>> Please clarify me!!
>>>>>>>
>>>>>>> Here is my program
>>>>>>> require ESL;
>>>>>>> use IO::Socket::INET;
>>>>>>> use Data::Dumper;
>>>>>>>
>>>>>>> my $ip = "192.168.1.222";
>>>>>>> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort =>
>>>>>>> '8447',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
>>>>>>> die "Could not create socket: $!\n" unless $sock;
>>>>>>> &register_Signals();
>>>>>>>
>>>>>>> for(;;) {
>>>>>>>     my $new_sock = $sock->accept();
>>>>>>>     next if (not defined ($new_sock));
>>>>>>>     my $pid = fork();
>>>>>>>     if ($pid) {
>>>>>>>         close($new_sock);
>>>>>>>         next;
>>>>>>>     }
>>>>>>>     print "CHILD PID: $$\n";
>>>>>>>     my $host = $new_sock->sockhost();
>>>>>>>     my $fd = fileno($new_sock);
>>>>>>>
>>>>>>>     my $con = new ESL::ESLconnection($fd);
>>>>>>>     my $info = $con->getInfo();
>>>>>>>
>>>>>>>     my $uuid = $info->getHeader("unique-id");
>>>>>>>
>>>>>>>     printf "Connected call %s, from %s\n", $uuid,
>>>>>>> $info->getHeader("caller-caller-id-number");
>>>>>>>     my $r=$con->execute("answer");
>>>>>>>     print Dumper $r;
>>>>>>>     $con->events("plain","all");
>>>>>>>     my
>>>>>>> $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav");
>>>>>>>
>>>>>>>     while($con->connected()) {
>>>>>>>         my $e = $con->recvEvent();
>>>>>>>
>>>>>>>         if ($e) {
>>>>>>>             my $name = $e->getHeader("event-name");
>>>>>>>             print "EVENT [$name]\n";
>>>>>>>             if ($name eq "DTMF") {
>>>>>>>                 my $digit = $e->getHeader("dtmf-digit");
>>>>>>>                 print "$digit\n";
>>>>>>>             }
>>>>>>>         }
>>>>>>>     }
>>>>>>>     close($new_sock);
>>>>>>> }
>>>>>>> sub register_Signals() {
>>>>>>>     foreach ( keys %SIG ) {
>>>>>>>         $SIG{$_} = 'sig_Handler';
>>>>>>>     }
>>>>>>> }
>>>>>>>
>>>>>>> sub sig_Handler() {
>>>>>>>     my $handle=$_[0];
>>>>>>>     if($handle eq "INT") {
>>>>>>>         print "$$: SIGNAL SIG$handle is generated\n";
>>>>>>>     }
>>>>>>> }
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Jan 21 22:00:11 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 22 Jan 2010 00:00:11 -0600
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
In-Reply-To: <191c3a031001212158s4c574324m4c8545f238967907@mail.gmail.com>
References: <37012.20543.qm@web33507.mail.mud.yahoo.com>
	<5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>
	<517DA285-7E9B-4226-8AEF-C2DAB52976CF@tlainvestments.com>
	<004e01ca9af3$6c5a2d20$450e8760$@net>
	<005b01ca9af5$dc759980$9560cc80$@net>
	<191c3a031001211648i5ed86253k8640c49f29121e0c@mail.gmail.com>
	<8589C894-EE61-4FA7-92E1-5CB9C52EBD60@tlainvestments.com>
	<191c3a031001212158s4c574324m4c8545f238967907@mail.gmail.com>
Message-ID: <191c3a031001212200k3744d2bctb0e54eb41897ee74@mail.gmail.com>

You don't need loopback if you were calling over sofia back to your own box,
you just need to use a dest that could be regexed to right to vm when it
comes back around. The loopback to an app should work you probably just have
a syntax err

On Jan 21, 2010 11:03 PM, "Troy Anderson" <troy at tlainvestments.com> wrote:

I get the idea, but can't seem to get it to work.

I tried doing a bridge to
"loopback/app=bridge ${sofia_contact(${dialed_user}@${dialed_domain}", but
fs complained => Cannot create outgoing channel of type [loopback=app:sofia]
Also, I tried "loopback/app=voicemail:default ${domain_name}
${dialed_extension}" and it did get to voicemail, but it didn't prompt me
for any info - just immediately complained that the recording was too short.

I tried the suggestions about modifying the sip.cfg for the phones and that
does work (thanks!), but that forces a Consultative Transfer.  It would be
nice to get this method working, which would result in the transfer button
doing a Consultive Transfer unless you hang up, then it would be like a
blind transfer.

Thanks,
Troy

On Jan 21, 2010, at 5:48 PM, Anthony Minessale wrote: > if you used the
loopback endpoint to loop...

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From mailinglist at fribert.dk  Fri Jan 22 03:14:41 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Fri, 22 Jan 2010 12:14:41 +0100
Subject: [Freeswitch-users] Svar: Re: Svar: Re: Home setup with home	 company
Message-ID: <4B5996B1020000E100000404@mail.fribert.dk>

Ok, I set it up like this:

   
     
      
         
      
         
         
         
         
         
         
         
         
         
      


But now it gives me:

2010-01-22 11:52:08.667564 [NOTICE] switch_channel.c:602 New Channel sofia/external/2680xxxx at 87.54.25.116 [16baec2f-4407-df11-8fb3-000c29b7b4cb]
2010-01-22 11:52:08.800123 [INFO] mod_dialplan_xml.c:252 Processing 2680xxxx->4692xxxx in context public
2010-01-22 11:52:08.830071 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/2680xxxx at 87.54.25.116 to XML[8203 at default]
2010-01-22 11:52:08.830071 [INFO] mod_dialplan_xml.c:252 Processing 2680xxxx->8203 in context default
2010-01-22 11:52:08.830071 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/2680xxxx at 87.54.25.116 to enum[8203 at default]
2010-01-22 11:52:09.163439 [INFO] switch_core_state_machine.c:136 No Route, Aborting

huh


>>> Michael Collins  21-01-10 9:39 >>>


On Wed, Jan 20, 2010 at 2:45 AM, mailinglist <mailinglist at fribert.dk> wrote:
Hi Michael

It's running on pfsense, so it's kinda locked to the version it currently is.
Looks very nice though.
Looking beyond that, is the action / anti-action list corrent?


I would say that you need to add an anti-action under the day of week check and go to vm if it does not match. Right now if the DOW is 0 or 6 then the entire extension will "fail" and the dialplan will just move on. Remember that if any conditions fail then the entire thing extension "fails" unless you are doing interesting things with the break= parameter. See the dialplan page on the wiki for examples of how to use break in your conditions.
-MC



Best regards
Fribse


>>> Michael Collins  20-01-10 1:53 >>>


On Tue, Jan 19, 2010 at 9:02 AM, mailinglist <mailinglist at fribert.dk> wrote:
I have a very small one man constultancy company that has the occasional call, unfortunately we are getting more spam calls after hours than real calls during work hours, so I would like to set up a TOD system.
 
First step for me is just playing with the TOD example, I've gotten this far:
 

   
      
      
      
         
         
         
         
         group/company@${domain_name}"/>
         
         
         
         
      


My idea with this, was that in the time 9-17 mon-fri, it rings, the rest of the time it's sent directly to the voicemail.
I would of course like to have it not take messages outside work hours, but that's just refining :-)
 
But it picks up the call, and then nothing...
 

We have a much cleaner way of doing TOD and DOW handling. You'll need to get to the latest FreeSWITCH version. Look in conf/dialplan/default.xml for this example:



    
    
      
        
      
    


Use that condition instead of the two conditions you're now using and see if you have better success.

-MC


_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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http://www.freeswitch.org




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From mayamatakeshi at gmail.com  Fri Jan 22 03:22:00 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Fri, 22 Jan 2010 20:22:00 +0900
Subject: [Freeswitch-users] Distinguishing Blind REFER from application
	transfer
Message-ID: <15b9404e1001220322h5636c1aat968f515e0601d769@mail.gmail.com>

Hello,
I'm using mod_xml_curl to provide dialplan.
In my application, I need to differentiate if a call has entered the
dialplan again due to uuid_transfer or due to Blind Transfer.
I know I can recognize a Blind Transfer by checking
variable_sip_h_Referred-By and variable_sip_refer_to.
However, these variables are not cleaned up when the call reenters the
dialplan due to application transfer or uuid_transfer.

I realize I can differentiate them by adding some prefix like this:
uuid_transfer TRANSFER,DestinationNumber XML default

Or it is my responsibility to call
application set or uuid_setvar to unset the variable(s) in this case?
it seems to me the bad thing here would be to have to add lots of preventive

 <action application="set" data="sip_refer_to=" />
in my dialplan.

Before doing so I thought in ask if the REFER-related variables being
preserved upon dialplan reentry would not be a bug (well, it could be a
feature useful in some scenarios I suspect).

br,
takeshi
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From mailinglist at fribert.dk  Fri Jan 22 04:12:46 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Fri, 22 Jan 2010 13:12:46 +0100
Subject: [Freeswitch-users] Svar: Re: How do I invite group to join	
	existing call?
Message-ID: <4B59A44E020000E100000413@mail.fribert.dk>

Hi Michael et al.

I found the 'attended xfer' example in the documentation, and it looks like rxactly to what I want to do.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer

The example 2:
Do I add the first part to my dialplan/default.xml, and the second part to my dialplan/features.xml

How does it affect if I try to do a normal local dial if it just reacts on the fact that three digits have been dialed (which I should change to 2, as I use 2 digit local numbers).

Best regards
Fribse

>>> Michael Collins  18-01-10 21:19 >>>


On Sun, Jan 17, 2010 at 10:50 PM, mailinglist <mailinglist at fribert.dk> wrote:
Hi All

I would like to be able to invite a group / global to join an existing call, but how do I accomplish this, can it be done?


Malfunction! Need Input!

Could you give us a few more details on what you're trying to accomplish? What's the big picture? I'm curious what problem you're trying to solve. I'm sure the gang here will have thoughts to pass along. :)

-MC




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From irmatov at gmail.com  Fri Jan 22 05:22:21 2010
From: irmatov at gmail.com (Timur Irmatov)
Date: Fri, 22 Jan 2010 18:22:21 +0500
Subject: [Freeswitch-users] mod_erlang_event: disconnects
In-Reply-To: <20100121134241.GD1036@hijacked.us>
References: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>
	<20100121134241.GD1036@hijacked.us>
Message-ID: <241d382f1001220522g27f67ef5p5ba86ebb3afe63c@mail.gmail.com>

On Thu, Jan 21, 2010 at 6:42 PM, Andrew Thompson <andrew at hijacked.us> wrote:
>> The problem is, that pretty frequently processes which handle incoming
>> calls receive messages like {'EXIT', <5406.48.0>, noconnection} from
>> FreeSWITCH. As I understand from googling, this happens when remote C
>> node disconnects (and I see TCP connections from FreeSWITCH to epmd
>> daemon being torn down and reestablished). FreeSWITCH drops calls at
>> that moment.
>
> Does it drop ALL calls being handled in erlang, or just that one?

It seems at it drops all calls handled in erlang. At the moment we
have only 7 calls maximum, and my applications logs several such exits
happening at exactly same time, up to 7 at once.

>> Have anyone seen this? Is there any fix/ advice?
>
> I haven't seen this before, how many calls are involved? I'm willing to
> help you troubleshoot though. Is there anything relevant in the logs
> (even at DEBUG)?

It is good and bad to know that you haven't seen this before.. :)
Good, because then it seems like local anomaly which, hopefully, can
be debugged and fixed. Bad, because it means we should debug it and
there's no ready fix.

As for the load, it is small. FreeSWITCH has 7 sip registrations to
our upstream, so 7 simultaneous calls is maximum.

I will send you logs offlist. This is small excerpt:

2010-01-22 16:40:00.059202 [DEBUG] mod_sofia.c:293
sofia/external/1504291 at 10.0.2.5 SOFIA DESTROY
2010-01-22 16:40:00.059202 [DEBUG] switch_core_state_machine.c:60
sofia/external/1504291 at 10.0.2.5 Standard DESTROY
2010-01-22 16:40:00.059202 [DEBUG] switch_core_state_machine.c:434
(sofia/external/1504291 at 10.0.2.5) State DESTROY going to sleep
2010-01-22 16:40:00.091201 [WARNING] mod_erlang_event.c:489 Can't
locate session df4f60c0-074a-11df-af37-b1ee9ca9c744
2010-01-22 16:40:00.091201 [DEBUG] mod_erlang_event.c:541 Notifying
new session failed
2010-01-22 16:40:00.091201 [DEBUG] mod_erlang_event.c:842
check_attached_sessions requested exit
2010-01-22 16:40:00.091201 [DEBUG] mod_erlang_event.c:920 Session
complete, waiting for children
2010-01-22 16:40:00.091201 [DEBUG] mod_erlang_event.c:930 Connection Closed
2010-01-22 16:40:00.095844 [DEBUG] mod_erlang_event.c:1723 Launching
listener, connection from node erlswitch at localhost, ip 127.0.0.1
2010-01-22 16:40:00.095844 [DEBUG] mod_erlang_event.c:910 Connection
Open from 127.0.0.1
2010-01-22 16:40:00.095844 [DEBUG] switch_core_session.c:857 Send
signal sofia/internal/sip:106 at 192.168.1.107:5060 [BREAK]
2010-01-22 16:40:00.095844 [DEBUG] switch_core_session.c:857 Send
signal sofia/internal/sip:108 at 192.168.1.108 [BREAK]
2010-01-22 16:40:00.102708 [DEBUG] switch_ivr_bridge.c:315
sofia/internal/sip:108 at 192.168.1.108 receive message [UNBRIDGE]

>> My system is Debian Lenny (5.0.3), 64-bit system, erlang installed
>> from Debian packages, no backports.
> What OTP release does that equate to, R12 or R13?

I guess this corresponds to R12B3:

$ apt-cache show erlang-nox|grep Version
Version: 1:12.b.3-dfsg-4

May be I should try to build latest erlang from source, rebuild
FreeSWITCH and see if it helps..

>Also, what FS version are you running?

'version' output in fs_cli does not reveal it's version:
freeswitch at internal> version
FreeSWITCH Version 1.0.trunk (hacked)

It was build from svn, i guess it is revision 16041.


-- 
Timur Irmatov, xmpp:irmatov at jabber.ru



From brian at freeswitch.org  Fri Jan 22 06:17:45 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 22 Jan 2010 08:17:45 -0600
Subject: [Freeswitch-users] Distinguishing Blind REFER from application
	transfer
In-Reply-To: <15b9404e1001220322h5636c1aat968f515e0601d769@mail.gmail.com>
References: <15b9404e1001220322h5636c1aat968f515e0601d769@mail.gmail.com>
Message-ID: <9F3D655F-BF09-4503-9709-953A839DC65C@freeswitch.org>

Have you done a uuid_dump to see all the variables?

/b

On Jan 22, 2010, at 5:22 AM, mayamatakeshi wrote:

> Before doing so I thought in ask if the REFER-related variables being preserved upon dialplan reentry would not be a bug (well, it could be a feature useful in some scenarios I suspect).




From dftoro at yahoo.com  Fri Jan 22 06:18:18 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Fri, 22 Jan 2010 06:18:18 -0800 (PST)
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <5C35A623-C13D-472C-BAEE-5F53CD2D95B6@jerris.com>
Message-ID: <984278.36075.qm@web33504.mail.mud.yahoo.com>

Hi, with svn r16440 the problem persists, I creted a jira report http://jira.freeswitch.org/browse/LBSNDF-8 this is a minor issue, but activing playback delimiter no audio file can be played. On FS the audio files are placed in the \sound\ directory, building the path on Windows would be \sound '\s' which is replaced by 'ound'.

Thank you

Diego Toro
http://lacarretade.blogspot.com/


--- On Thu, 1/21/10, Michael Jerris <mike at jerris.com> wrote:

> From: Michael Jerris <mike at jerris.com>
> Subject: Re: [Freeswitch-users] mutiple playback files (unescape_char) Windows
> To: freeswitch-users at lists.freeswitch.org
> Date: Thursday, January 21, 2010, 5:56 PM
> How about with svn
> r16440
> On Jan 21, 2010, at 8:02
> AM, Diego Toro wrote:
> Hi MikeJ, using '\\' the
> behavior is the same, '\\s' is replaced by
> ' '.
> 
> <action application="playback"
> data="misc\\8000\\serror.wav!misc\\8000\\provide_reference_number.wav!digits\\8000\\5.wav"/>
> ?
> 
> Console output error is:
> 
> [ERR] mod_sndfile.c:194 Error Opening File
> [d:\fs\fs_trunk_20100118\Debug\sounds\es\co\callie\misc\8000
> error.wav]
> 
> 
> S.O.: Windows 7
> FreeSwitch: Trunk (svn latest version)
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From andrew at hijacked.us  Fri Jan 22 07:46:58 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Fri, 22 Jan 2010 10:46:58 -0500
Subject: [Freeswitch-users] mod_erlang_event: disconnects
In-Reply-To: <241d382f1001220522g27f67ef5p5ba86ebb3afe63c@mail.gmail.com>
References: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>
	<20100121134241.GD1036@hijacked.us>
	<241d382f1001220522g27f67ef5p5ba86ebb3afe63c@mail.gmail.com>
Message-ID: <20100122154658.GC25693@hijacked.us>

Give this patch a shot:

http://eagle.bsd.st/~andrew/erlang_session_fix.diff

And see if it makes a difference.

Andrew



From mayamatakeshi at gmail.com  Fri Jan 22 07:47:34 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Sat, 23 Jan 2010 00:47:34 +0900
Subject: [Freeswitch-users] Distinguishing Blind REFER from application
	transfer
In-Reply-To: <9F3D655F-BF09-4503-9709-953A839DC65C@freeswitch.org>
References: <15b9404e1001220322h5636c1aat968f515e0601d769@mail.gmail.com>
	<9F3D655F-BF09-4503-9709-953A839DC65C@freeswitch.org>
Message-ID: <15b9404e1001220747y79571117qab826152a2df1dc8@mail.gmail.com>

On Fri, Jan 22, 2010 at 11:17 PM, Brian West <brian at freeswitch.org> wrote:

> Have you done a uuid_dump to see all the variables?
>

I just tried that with trunk. I can see the REFER variables stay set till
the end of the call.
They will show up in CHANNEL_HANGUP_COMPLETE:

variable_sip_h_Referred-By: user2 <sip:user2 at test1.com<sip%3Auser2 at test1.com>
>
variable_sip_refer_to: <sip:2000 at test1.com <sip%3A2000 at test1.com>>

I suppose the only thing that will change them is another blind refer. But
they will never be unset.


>
> On Jan 22, 2010, at 5:22 AM, mayamatakeshi wrote:
>
> > Before doing so I thought in ask if the REFER-related variables being
> preserved upon dialplan reentry would not be a bug (well, it could be a
> feature useful in some scenarios I suspect).
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From robert.hadley at teotech.com  Fri Jan 22 08:21:55 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 22 Jan 2010 08:21:55 -0800
Subject: [Freeswitch-users] No external sangoma calls running FS as daemon
Message-ID: <B654371338324F17A504DF26271798B9@greyhawk.tonecommander.com>

I have freeswitch running as daemon as user freeswitch.  Internal calls
work.  I get 503 service unavailable when I make external calls through
sangoma.  If I change the FS daemon to run as root then external sangoma
calls work.  Does anybody know what permissions I need to fix?

 

Thanks,

Robert

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From msc at freeswitch.org  Fri Jan 22 08:29:46 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 22 Jan 2010 08:29:46 -0800
Subject: [Freeswitch-users] FreeSWITCH Weekly Call Starting in 30 min
Message-ID: <87f2f3b91001220829k32f60609kbd76c07c93db435e@mail.gmail.com>

Hello all,

The weekly conference call will be starting in a bit. Here's the agenda:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_22

I'll be on just after 9am my time as I have to get my kids off to school.
Talk to you all soon.

-Michael
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From moises.silva at gmail.com  Fri Jan 22 09:45:54 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 22 Jan 2010 12:45:54 -0500
Subject: [Freeswitch-users] No external sangoma calls running FS as
	daemon
In-Reply-To: <B654371338324F17A504DF26271798B9@greyhawk.tonecommander.com>
References: <B654371338324F17A504DF26271798B9@greyhawk.tonecommander.com>
Message-ID: <c4d05cbe1001220945v265ee803x3969595ece7e218b@mail.gmail.com>

On Fri, Jan 22, 2010 at 11:21 AM, Robert Hadley
<robert.hadley at teotech.com>wrote:

>  I have freeswitch running as daemon as user freeswitch.  Internal calls
> work.  I get 503 service unavailable when I make external calls through
> sangoma.  If I change the FS daemon to run as root then external sangoma
> calls work.  Does anybody know what permissions I need to fix?
>

Check the FS logs and pastebin a call attempt. In any case, my guess is
/dev/wanpipe* devices permissions must be changed using udev.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From oscav at hotmail.fr  Fri Jan 22 09:53:25 2010
From: oscav at hotmail.fr (Oscav)
Date: Fri, 22 Jan 2010 09:53:25 -0800 (PST)
Subject: [Freeswitch-users] Script ends when originate receives
 INVALID_NUMBER_FORMAT
Message-ID: <27277429.post@talk.nabble.com>


Hi,

My script ends when I received a 484 INVALID_NUMBER_FORMAT, and doesn't
continue even the script or a hangup Hook.

Here is the script :

session.setVariable("hangup_after_bridge",false);
...
session.setVariable("continue_on_fail","true");
...
session.preAnswer();
...
new_session = new Session(route,session);
new_session.waitForAnswer(15000);
if (new_session.ready()) {	
  bridge(session, new_session);
}
...
console_log("info","call ended")
...

Is there something missing on my script ??
Thanks.
-- 
View this message in context: http://old.nabble.com/Script-ends-when-originate-receives-INVALID_NUMBER_FORMAT-tp27277429p27277429.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From brian at freeswitch.org  Fri Jan 22 09:54:22 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 22 Jan 2010 11:54:22 -0600
Subject: [Freeswitch-users] IAX2 Support Removed.
Message-ID: <0C2714A2-355A-42C7-B589-F0704D436607@freeswitch.org>

Due to lack of support for the libiax2 being updated to support the newer protocol changes and the lack of interest from anyone willing to actually work on it.  I have moved mod_iax to unsupported where it will stay until someone steps up to rewrite a new IAX2 lib.  

Thanks,
Brian


From jerry.richards at teotech.com  Fri Jan 22 10:17:18 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 22 Jan 2010 10:17:18 -0800
Subject: [Freeswitch-users] Wanpipe Driver Install Before FS Install
In-Reply-To: <mailman.5417.1264166574.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.5417.1264166574.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <2D7783AC2C3A4E2E8E0222572286DE51@greyhawk.tonecommander.com>


Is it true that the Sangoma Wanpipe Driver should be installed before
Freeswitch, because the Freeswitch build will autodetect the wanpipe
drivers?

Thanks,
Jerry




From moises.silva at gmail.com  Fri Jan 22 10:33:15 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 22 Jan 2010 13:33:15 -0500
Subject: [Freeswitch-users] Wanpipe Driver Install Before FS Install
In-Reply-To: <2D7783AC2C3A4E2E8E0222572286DE51@greyhawk.tonecommander.com>
References: <mailman.5417.1264166574.2495.freeswitch-users@lists.freeswitch.org>
	<2D7783AC2C3A4E2E8E0222572286DE51@greyhawk.tonecommander.com>
Message-ID: <c4d05cbe1001221033y73aa99del14194540d818a041@mail.gmail.com>

On Fri, Jan 22, 2010 at 1:17 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> Is it true that the Sangoma Wanpipe Driver should be installed before
> Freeswitch, because the Freeswitch build will autodetect the wanpipe
> drivers?
>

If you want to use mod_openzap ( the endpoint used by FreeSWITCH to make
calls using analog and TDM telephony) and you are using Sangoma cards, yes,
you need to install the Wanpipe drivers before even boostraping (./boostrap)
FreeSWITCH. OpenZAP will look during the ./configure state for libsangoma
(which is installed along with the drivers) and then when compiling it will
need the Wanpipe headers, if not found, ozmod_wanpipe will not be compiled
and FreeSWITCH will not have Sangoma cards support for analog and TDM.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From Prometheus001 at gmx.net  Fri Jan 22 11:06:51 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 22 Jan 2010 20:06:51 +0100
Subject: [Freeswitch-users] skypiax and xml-curl
Message-ID: <4B59F74B.1010201@gmx.net>

Hello,

is there a way to manage skypiax via XML-curl besides the dialplan?

Best regards
Peter




From mailinglist at fribert.dk  Fri Jan 22 12:50:17 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Fri, 22 Jan 2010 21:50:17 +0100
Subject: [Freeswitch-users] Svar: Re: How do I invite group to join		
	existing call?
In-Reply-To: <4B59A44E020000E100000413@mail.fribert.dk>
References: <4B59A44E020000E100000413@mail.fribert.dk>
Message-ID: <4B5A1D99020000E100000418@mail.fribert.dk>

Hmm, I don't get it, it might not do the right thing.
 
The situation is that I receive a call from the outside, answers it on a phone, and then wants to ask a third (local) party to join the conversation.
 
I thought from the example that I should just press *3, and then the extension I want to invite, but nothing happens.
 
I haven't the faintest how I accomplish this :-o


>>> 22-01-2010 kl. 13:12 skrev "mailinglist" <mailinglist at fribert.dk> i meddelelsen <4B59A44E020000E100000413 at mail.fribert.dk>:

Hi Michael et al.

I found the 'attended xfer' example in the documentation, and it looks like rxactly to what I want to do.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer

The example 2:
Do I add the first part to my dialplan/default.xml, and the second part to my dialplan/features.xml

How does it affect if I try to do a normal local dial if it just reacts on the fact that three digits have been dialed (which I should change to 2, as I use 2 digit local numbers).

Best regards
Fribse

>>> Michael Collins 18-01-10 21:19 >>>


On Sun, Jan 17, 2010 at 10:50 PM, mailinglist <mailinglist at fribert.dk> wrote:


Hi All

I would like to be able to invite a group / global to join an existing call, but how do I accomplish this, can it be done?


Malfunction! Need Input!

Could you give us a few more details on what you're trying to accomplish? What's the big picture? I'm curious what problem you're trying to solve. I'm sure the gang here will have thoughts to pass along. :)

-MC

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From jerry.richards at teotech.com  Fri Jan 22 13:05:35 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 22 Jan 2010 13:05:35 -0800
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 43, Issue 187
In-Reply-To: <mailman.5425.1264184283.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.5425.1264184283.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <FABCD242F6D74C3D906C886156FEDFE2@greyhawk.tonecommander.com>


I tried sending this Email earlier in the week but it got kicked back by our
Mail Server, so if this is a duplicate, my apologies...

Are you still working toward a release version 1.0.5?

Best Regards,
Jerry




From nicolas at medularis.com  Fri Jan 22 13:22:46 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 22 Jan 2010 18:22:46 -0300
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <87f2f3b91001211412h7aa0a84ageadb9557b869ca01@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
	<e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
	<5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>
	<5d2828f1001201406p4154b98ald1af1c5c25f59337@mail.gmail.com>
	<1b46b4e81001210335l42baef16r8a2952aa5b92f6e6@mail.gmail.com>
	<87f2f3b91001211412h7aa0a84ageadb9557b869ca01@mail.gmail.com>
Message-ID: <1b46b4e81001221322w5da04799s2ea86000c5c4a9a@mail.gmail.com>

No problem, here it is:

- http://wiki.freeswitch.org/wiki/Call_retry_based_on_hangup_cause

It is linked from your reference (
http://wiki.freeswitch.org/wiki/Mod_lua#More_Samples).

Sorry I didn't do it early, I hadn't seen your email.

I also added another, more complete, example here (also linked):

- http://wiki.freeswitch.org/wiki/Bridging_two_calls_with_retry



On Thu, Jan 21, 2010 at 7:12 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, Jan 21, 2010 at 3:35 AM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>>
>> On Wed, Jan 20, 2010 at 7:06 PM, Mike van Lammeren <mike at van.lammeren.net
>> > wrote:
>>
>>> So, I've been reading about early media in the wiki, and have made a
>>> little progress, which leads to more questions.
>>>
>>> I understand now why a call is considered connected before one person has
>>> picked up the phone. I am also able to get my script to wait for the phone
>>> to be picked up, by setting the ignore_early_media variable when starting a
>>> new session, like this:
>>>
>>> customerSession =
>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/example.com/"
>>> .. customerPhoneNumber)
>>>
>>>
>>> After that line, the script waits for the other phone to be picked up.
>>>
>>> However, now I wonder what to do with calls that don't complete, get busy
>>> signals, etc.
>>>
>>> What do people do in this case? The only related example I can find on
>>> the web is for a javascript dialer, which doesn't address any of these
>>> cases.
>>>
>>
>>
>> I guess it depends on what you want to do. For example I have a lua script
>> very similar to what you describe, although there is no confirmation
>> involved. Depending on the hangup cause the session gets, it might try
>> redialing with a different gateway, try again or just hangup.
>>
>> Take a look here http://wiki.freeswitch.org/wiki/Hangup_causes to see
>> what each hangup cause means. You don't need to have a special case for all
>> of them, only the ones you are interested in.
>>
>> Here's an example in code which retries a call depending on the hangup
>> cause. It retries max_retries1 times and alternates between 2 different
>> gateways:
>>
>> session1 = null;
>> max_retries1 = 3;
>> retries = 0;
>> ostr = "";
>> repeat
>>         retries = retries + 1;
>>         if (retries % 2) then ostr = originate_str1;
>>         else ostr = originate_str12; end
>>         freeswitch.consoleLog("notice", "*********** Dialing Leg1: " ..
>> ostr .. " - Try: "..retries.." ***********\n");
>>         session1 = freeswitch.Session(ostr);
>>         local hcause = session1:hangupCause();
>>         freeswitch.consoleLog("notice", "*********** Leg1: " .. hcause ..
>> " - Try: "..retries.." ***********\n");
>> until not ((hcause == 'NO_ROUTE_DESTINATION' or hcause ==
>> 'RECOVERY_ON_TIMER_EXPIRE' or hcause == 'INCOMPATIBLE_DESTINATION' or hcause
>> == 'CALL_REJECTED' or hcause == 'NORMAL_TEMPORARY_FAILURE') and (retries <
>> max_retriesl1))
>>
>>
>> Note: originate_str1 and originate_str2 are two different dial strings for
>> 2 different gateways.
>>
>>
> Nicolas,
>
> This is really nice. Would you be willing to add this script and a brief
> explanation to the wiki? You could create a whole new page and just link to
> it from here: http://wiki.freeswitch.org/wiki/Mod_lua#More_Samples
>
> If you have any questions please let me know!
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From michal.zubac at comgate.cz  Fri Jan 22 10:20:20 2010
From: michal.zubac at comgate.cz (=?ISO-8859-2?Q?Michal_Zub=E1=E8?=)
Date: Fri, 22 Jan 2010 19:20:20 +0100
Subject: [Freeswitch-users] E1 hangups
Message-ID: <4B59EC64.3080907@comgate.cz>

Hi.

I'm trying to correct this behaviour, but can't figure out, where is the
problem. Here's the scenario:

- we're trying to execute simple dialplan
    * answer
    * play sound
    * wait for 3 seconds
    * hangup

- for SIP caller it works as expected
- problems are when, we try to call into it from E1 line
- for E1 we're using sangoma winpipe & openzap
- dialplan in freeswitch console is done in a moment ending with hangup
- on the E1 line I hear nothing and after 2 seconds it disconnects
- similar problem when there's only bridge to another number (E1)
- it rings (on the destination phone) for a short moment (0.5-1s), but
then hangs up spontaneously

Thanks for any clues.

mZubac
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From tim at communicatefreely.net  Fri Jan 22 12:30:17 2010
From: tim at communicatefreely.net (Tim St. Pierre)
Date: Fri, 22 Jan 2010 15:30:17 -0500
Subject: [Freeswitch-users] Choppy conference audio
Message-ID: <4B5A0AD9.8020700@communicatefreely.net>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello all,

I'm having a tough time sorting out the audio in mod_conference, I'm hoping someone can point me in
the right direction.  I took a good look around the wiki, but couldn't really find what I was
looking for.

I have a test system set up, with three Aastra 9133i, a 6731i, and a 57i.  All register, call fine, etc.

Music on hold audio is perfect.  Calls between phones are perfect.

CPU load on the lab machine is about 8%, so not really busy.

Conference audio is choppy, in the way that things get choppy when there is some sort of timing
issue.  Even with one member in the conference, the prompts and moh sound choppy.

I played around with the rate and interval values, there wasn't any change.  I tried disabling the
monotonic timer, and that didn't change anything either.

I'm running FreeBSD 7, with the latest release of FS downloaded last week.

Thanks for any help.  Let me know if I need to post any other information.

- -Tim


-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.4 (FreeBSD)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iQCVAwUBS1oK2YqVcvNCnHOrAQLriAP/csjIfD/VP0CA3ePyRBKXbDPfxEiqx/cP
iWILZ65F0bxaryKTYT2ZV8W7u+wH6YxMCaoej+H+yd1XG08hZr4kLUcewUaCTMna
S8Zd24ReknEmL8d0AXCMgospf2wmwZZx2pJMFmbPN3IlXup20dKWOESBp/Dru3vL
63GE8s6AdyA=
=ndAy
-----END PGP SIGNATURE-----



From anthony.minessale at gmail.com  Fri Jan 22 14:32:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 22 Jan 2010 16:32:43 -0600
Subject: [Freeswitch-users] Choppy conference audio
In-Reply-To: <4B5A0AD9.8020700@communicatefreely.net>
References: <4B5A0AD9.8020700@communicatefreely.net>
Message-ID: <191c3a031001221432t1d8f9614r1d86e0379f82aa2d@mail.gmail.com>

try combos of -vm and -nocal flags
one or both of each


On Fri, Jan 22, 2010 at 2:30 PM, Tim St. Pierre
<tim at communicatefreely.net>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello all,
>
> I'm having a tough time sorting out the audio in mod_conference, I'm hoping
> someone can point me in
> the right direction.  I took a good look around the wiki, but couldn't
> really find what I was
> looking for.
>
> I have a test system set up, with three Aastra 9133i, a 6731i, and a 57i.
>  All register, call fine, etc.
>
> Music on hold audio is perfect.  Calls between phones are perfect.
>
> CPU load on the lab machine is about 8%, so not really busy.
>
> Conference audio is choppy, in the way that things get choppy when there is
> some sort of timing
> issue.  Even with one member in the conference, the prompts and moh sound
> choppy.
>
> I played around with the rate and interval values, there wasn't any change.
>  I tried disabling the
> monotonic timer, and that didn't change anything either.
>
> I'm running FreeBSD 7, with the latest release of FS downloaded last week.
>
> Thanks for any help.  Let me know if I need to post any other information.
>
> - -Tim
>
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v2.0.4 (FreeBSD)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iQCVAwUBS1oK2YqVcvNCnHOrAQLriAP/csjIfD/VP0CA3ePyRBKXbDPfxEiqx/cP
> iWILZ65F0bxaryKTYT2ZV8W7u+wH6YxMCaoej+H+yd1XG08hZr4kLUcewUaCTMna
> S8Zd24ReknEmL8d0AXCMgospf2wmwZZx2pJMFmbPN3IlXup20dKWOESBp/Dru3vL
> 63GE8s6AdyA=
> =ndAy
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Jan 22 14:36:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 22 Jan 2010 16:36:23 -0600
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 43, Issue 187
In-Reply-To: <FABCD242F6D74C3D906C886156FEDFE2@greyhawk.tonecommander.com>
References: <mailman.5425.1264184283.2495.freeswitch-users@lists.freeswitch.org>
	<FABCD242F6D74C3D906C886156FEDFE2@greyhawk.tonecommander.com>
Message-ID: <191c3a031001221436m2089a333ne0aafe99136845fa@mail.gmail.com>

Current Scheduled release date is the week of Feb 8th


On Fri, Jan 22, 2010 at 3:05 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> I tried sending this Email earlier in the week but it got kicked back by
> our
> Mail Server, so if this is a duplicate, my apologies...
>
> Are you still working toward a release version 1.0.5?
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From thangappan143 at gmail.com  Fri Jan 22 20:32:32 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Sat, 23 Jan 2010 10:02:32 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
	<7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>
Message-ID: <7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>

Updated the latest version of freeswitch ( 1.0.5-20100121-0400) and run the
wanrouter then freeswitch. While executing the freeswtich it said the
following error.

[ERR] zap_io.c:2562 Error
loading/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
[/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
file: No such file or directory]
[ERR] zap_io.c:2722 can't find 'sangoma_boost




Searched about this in freeswitch mailing list and found one post was there
regarding the same problem. Finally found the problem. I missed to install
the SCTP packages. Installed it and compiled the freeswitch again now the
inbound call was landed on freeswitch.

But I am unable to make a outbound call. When I was trying the following was
get.

freeswitch at internal> originate
openzap/smg_prid/a/9940464753 at g1openzap/smg_prid/a/9940464753 at g1
-ERR NORMAL_CIRCUIT_CONGESTION

2010-01-23 10:00:45.688854 [WARNING] ozmod_sangoma_boost.c:348 TX EVENT:
CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[9940464753]
Ci=[0000000000] Rdnis=[]
freeswitch at internal> 2010-01-23 10:00:46.709355 [WARNING]
ozmod_sangoma_boost.c:1373 RX EVENT (N): CALL_START_NACK:(82) [w256g256]
Rc=0 CSid=2 Seq=2
2010-01-23 10:00:46.709355 [WARNING] sangoma_boost_client.c:220 TX EVENT
(N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=2 Seq=3
2010-01-23 10:00:46.709355 [ERR] mod_openzap.c:1162 No channels available
2010-01-23 10:00:46.709355 [ERR] switch_ivr_originate.c:2411 Cannot create
outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
2010-01-23 10:00:46.709355 [DEBUG] switch_ivr_originate.c:3211 Originate
Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]

Please help me...........


On Fri, Jan 22, 2010 at 10:15 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> The following link have the openzap.conf,openzap.conf.xml ,smg_prid.conf ,
> debug log of mod_openzap , oz list and oz dump 1 output.
>
> http://pastebin.org/80095
>
>
>
> On Thu, Jan 21, 2010 at 10:34 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>> OpenZap is loading the ss7 signalling type. As per your concern openzap
>> does not know the details of the signalling then how it is loading the
>> ss7_boost libraries?
>>
>> FreeSWITCH log:
>> -----------------------------
>> 2010-01-21 10:10:46.707844 [INFO] zap_io.c:2374 Configured 30 channel(s)
>> 2010-01-21 10:10:46.708600 [INFO] zap_io.c:2468 Loading SIG from
>> /usr/local/freeswitch/mod/ozmod_ss7_boost.so
>> 2010-01-21 10:10:46.709031 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost'
>> 2010-01-21 10:10:46.709466 [DEBUG] ss7_boost_client.c:124 Creating L=
>> 127.0.0.65:53000 R=127.0.0.66:53000
>> 2010-01-21 10:10:46.709834 [DEBUG] ss7_boost_client.c:124 Creating L=
>> 127.0.0.65:53001 R=127.0.0.66:53001
>> 2010-01-21 10:10:46.710424 [WARNING] ss7_boost_client.c:244 TX EVENT (P):
>> SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>>
>> The signalling type might be anything but when I used the oz list command
>> it showed the span details. But I am unable to make a inbound and outbound
>> call.
>>
>> Outbound call result:
>> ============
>> >  originate openzap/smg_prid/a/9940464753 openzap/smg_prid/a/9843171457
>> -ERR NORMAL_CIRCUIT_CONGESTION
>>
>> 2010-01-21 10:26:14.304816 [CRIT] ozmod_ss7_boost.c:244 SPAN is not
>> online.
>> freeswitch at internal> 2010-01-21 10:26:14.304816 [ERR] mod_openzap.c:1043
>> No channels available
>> 2010-01-21 10:26:14.304816 [ERR] switch_ivr_originate.c:1510 Cannot create
>> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>> 2010-01-21 10:26:14.304816 [DEBUG] switch_ivr_originate.c:2138 Originate
>> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>>
>> Inbound call result:
>> -----------------------------
>>
>>      I made incoming call for the dial plan which is specified in the
>> earlier post at that time it said the number is busy. We did the packet
>> capture using the following command.
>>
>> wanpipemon -i w1g1 -pcap -pcap file isdn.pcap -port ISDN -full -systime -c
>> trd
>>
>> Here I attached the pcap file for that.
>>
>>
>> Where I did mistake or Did I miss any thing to do?
>> Please help me.......
>>
>>
>>
>> On Wed, Jan 20, 2010 at 7:40 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>
>>>
>>>       I noticed the 'oz list' output  in that span type is 'ss7 (boost)'.
>>> How can I change this to isdn?
>>>
>>>
>>>
>>> On Wed, Jan 20, 2010 at 12:43 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>>
>>>> I found the error in it. The file name is used as openzap.conf.xml (
>>>> smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
>>>> PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
>>>> openzap.conf.wiki.xml  file.
>>>>
>>>> Now the another problem is raised here.
>>>> When I was using oz list command , the details of the smg_prid shown.
>>>> When I was using 'oz dump smg_prid' command it shows all the channels'
>>>> details. But all the channels' states are DOWN. why? How can I make it the
>>>> states to UP?
>>>>
>>>> When I was making the call , the number is busy message was get. The
>>>> call was not  at all landed to the freeswitch.
>>>>
>>>> Dial plan Example:
>>>> -------------------------------
>>>>  <extension name="PRI_Testing">
>>>>         <condition field="destination_number" expression="^39114600$">
>>>>                 <action application="playback"
>>>> data="ivr-welcome_to_freeswitch"/>
>>>>         </condition>
>>>>   </extension>
>>>>
>>>> Please help me...........
>>>>
>>>> *Output Reference:*
>>>> http://pastebin.org/79074
>>>>
>>>>
>>>> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <thangappan143 at gmail.com
>>>> > wrote:
>>>>
>>>>> Dear all,
>>>>>
>>>>>          I have successfully configured wanpipe with freeswitch. When I
>>>>> was the running wancfg_fs script the following files openzap.conf ,
>>>>> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
>>>>> are created.
>>>>>
>>>>>          I started the wanrouter command then executed the freeswitch.
>>>>>          When I was executing freeswitch mod_openzap.c said the error
>>>>> as  "Error  for finding the span id. name:PRI_1".
>>>>>          But in the openzap.conf and openzap.conf.xml files the span
>>>>> name is smg_prid.
>>>>>
>>>>>          Why the freeswitch is referring the span name as PRI_1 ?
>>>>>          Whether this has to configured in anywhere?
>>>>>
>>>>>          In the freeswitch CLI using oz command I tried to dump the
>>>>> PRI_1 span id but it said te error  as "PRI_1 is not found".  When I was
>>>>> trying the command 'oz dump smg_prid'  all the channel states and details
>>>>> shown.
>>>>>
>>>>>         It seems that smg_prid span configured in openzap perfectly
>>>>> (Its my assumption). Then Why freeswitch is referring the span name as
>>>>> PRI_1.
>>>>>
>>>>> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>>>>>
>>>>> Could anyone please help me?
>>>>>
>>>>> REFERENCE:
>>>>>
>>>>> openzap.conf
>>>>> [span wanpipe smg_prid]
>>>>> name => smg_prid
>>>>> trunk_type =>e1
>>>>> b-channel => 1:1-15
>>>>> b-channel => 1:17-31
>>>>>
>>>>>
>>>>> openzap.conf.xml
>>>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>>>   <settings>
>>>>>     <param name="debug" value="0"/>
>>>>>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>>>     <!--<param name="enable-analog-option" value="3-way"/>-->
>>>>>   </settings>
>>>>>   <boost_spans>
>>>>>         <span id="smg_prid">
>>>>>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>>         <param name="dialplan" value="XML"/>
>>>>>         <param name="context" value="default"/>
>>>>>          <!-- regex to stop dialing when it matches -->
>>>>>         <!--<param name="dial-regex" value="5555"/>-->
>>>>>         <!-- regex to stop dialing when it does not match -->
>>>>>         <!--<param name="fail-dial-regex" value="^5"/>-->
>>>>>         </span>
>>>>>  </boost_spans>
>>>>>  <analog_spans>
>>>>>  </analog_spans>
>>>>> </configuration>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Regards,
>>>>> Thangappan.M
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Thangappan.M
>>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Thangappan.M
>>>
>>
>>
>>
>> --
>> Regards,
>> Thangappan.M
>>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From nazim.agabekov at gmail.com  Fri Jan 22 15:38:49 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Sat, 23 Jan 2010 03:38:49 +0400
Subject: [Freeswitch-users] CDR Gateways
In-Reply-To: <956716.34674.qm@web34301.mail.mud.yahoo.com>
References: <956716.34674.qm@web34301.mail.mud.yahoo.com>
Message-ID: <4B5A3709.4090304@gmail.com>

Hello Fernando,

I have coded a small xml_cdr FCGI logger for this purpose, it receives
mod_xml_cdr's data and inserts it into mysql table.
Basically it's a FastCgi and Libxml's XPATH hacked together. Example
config file is pretty easy to understand.
Software is of pre-alfa quality, but works. You could get the svn
snapshot from blog.buta-tech.com.

Best Regards,
Nazim Aghabayov


On 01/22/2010 08:26 AM, FERNANDO VILLARROEL wrote:
> Dear All.
>
> I have defined various gateways in ~/sip-profiles/external
>
> My questions is if a gateway named foo that send calls from diferents IP address like x.x.x.x and x.x.x.y. How i can doing a group for both ip address where i can doing later a sql like accountcode=foo?
>
> In this moment if i need know the traffic of customer foo, i should doing two differents cdr, one for every ip address.
>
> It's fine or i can doing of different way?
>
> I hope anyone could me explain how i can doing. 
>
> my gateway foo.xml
>
> <include>
>     <gateway name="foo">
>       <param name="username" value="foo"/>
>       <param name="password" value="foo"/>
>       <param name="realm" value="x.x.x.x"/>
>       <param name="register" value="false"/>
>       <param name="context" value="public"/>
>     </gateway>
> </include>
>
> foo1.xml
>
> <include>
>     <gateway name="foo1">
>       <param name="username" value="foo1"/>
>       <param name="password" value="foo1"/>
>       <param name="realm" value="x.x.x.y"/>
>       <param name="register" value="false"/>
>       <param name="context" value="public"/>
>     </gateway>
> </include>
>
> Both gateways foo and foo1 are the same customer
>
> my cdr_csv.conf.xml
>
> <configuration name="cdr_csv.conf" description="CDR CSV Format">
>   <settings>
>     <!-- 'cdr-csv' will always be appended to log-base -->
>     <!--<param name="log-base" value="/var/log"/>-->
>     <param name="default-template" value="sql"/>
>     <!-- This is like the info app but after the call is hung up -->
>     <!--<param name="debug" value="true"/>-->
>     <param name="rotate-on-hup" value="false"/>
>     <!-- may be a b or ab -->
>     <param name="legs" value="ab"/>
>   </settings>
>   <templates>
>         <template name="sql">INSERT INTO v_cdr (v_id, caller_id_name, caller_id_number, destination_number, context, start_stamp, answer_stamp, end_stamp,
> duration, billsec, hangup_cause, uuid, bleg_uuid, accountcode, read_codec, write_codec, remote_media_ip, network_addr) VALUES ("{v_id}", "${caller_id_name}"
> ,"${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}"
> ,"${uuid}","${bleg_uuid}", "${accountcode}", "${read_codec}", "${write_codec}", "${remote_media_ip}", "${network_addr}");</template>
>
> The argument accountcode on my database is Blank or None for all records of gateways.
> Regards.
>
>
>       
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From jingwei.yang at gmail.com  Fri Jan 22 22:00:36 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Sat, 23 Jan 2010 14:00:36 +0800
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <13529f9d1001211806o35fc3435j93b71bbb2ede3028@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
	<1264092620.14614.73.camel@luna.tc.commsmundi.com>
	<13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
	<ebbc2d551001211721n51cb5c54m996e0d494842c383@mail.gmail.com>
	<13529f9d1001211806o35fc3435j93b71bbb2ede3028@mail.gmail.com>
Message-ID: <13529f9d1001222200p44afd96ema21729d0038e89ba@mail.gmail.com>

Hi Jo?o, do you know how to sign the agent off automatically when either
party hangs up the call?

Here's how I originate the call to the agent and sign him up in ACD1:

originate skypiax/ANY/jingwei.yang 6*1

However, I found the user_name property is empty. May I know how it is set?

Thanks and best regards,
-Jingwei

On Fri, Jan 22, 2010 at 10:06 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:

> Hi Jo?o, thanks for the reply. I'll try it out.
>
> Regards,
> -Jingwei
>
> 2010/1/22 Jo?o Mesquita <jmesquita at freeswitch.org>
>
> Jingwei, check my contrib dir. I think it may help you with one FIFO since
>> we are able there to sign in and sign off dynamic agents as well as
>> customize what we do when the FIFO makes a call to them.
>>
>> Regards,
>> Jo?o Mesquita
>> FSComm Developer
>>
>>
>> On Thu, Jan 21, 2010 at 10:05 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>>
>>> Thanks for the reply. All the agents are dynamic and I can't predefine
>>> them in the config file.
>>>
>>> Regards,
>>> -Jingwei
>>>
>>>
>>> On Fri, Jan 22, 2010 at 12:50 AM, Fran?ois Delawarde <
>>> fdelawarde at wirelessmundi.com> wrote:
>>>
>>>> Why do you need 2 fifos? You could have callback agents connected to the
>>>> fifo and send incoming calls there, mod_fifo will do the rest.
>>>>
>>>> To configure agents for callback:
>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>>>>
>>>> To place a call into a fifo:
>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue
>>>>
>>>> Fran?ois.
>>>>
>>>> On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
>>>> > Sorry about the confusion, I'm just trying to think over all the
>>>> > abnormal situations before the implementation and hope the flow has no
>>>> > design flaws.
>>>> >
>>>> > Client A is parked in Queue 1, multiple consumers will be ringed to
>>>> > answer him. And once the first one is connected, all the rest will
>>>> > hang up. This is the targeted function.
>>>> >
>>>> > To achieve this, I'm considering to originate a call to each consumer
>>>> > and put the calls in Queue 2. Then intercept client A to the first
>>>> > element of Queue 2.
>>>> >
>>>> > I'm not sure if it's the usual or the best way. If you feel not,
>>>> > please don't hesitate to correct me. Any thoughts are warmly
>>>> > appreciated.
>>>> >
>>>> > On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com>
>>>> wrote:
>>>> >         what no errors mean? how do you originate calls to consumers?
>>>> >         I don't
>>>> >         understand your scenario.
>>>> >
>>>> >         2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
>>>> >
>>>> >         > Hi All,
>>>> >         >
>>>> >         > Please advise whether the following flow makes sense.
>>>> >         >
>>>> >         > 1. Client A calls in and parked in Queue 1
>>>> >         > 2. Originate calls to several consumers simultaneously and
>>>> >         park them in
>>>> >         > Queue 2
>>>> >         > 3. Intercept A's call to the first consumer of Queue 2
>>>> >         >
>>>> >         > Basically I want A's call picked up by the first among many
>>>> >         consumers with
>>>> >         > no errors. Please let me know whether I'm on the right
>>>> >         track.
>>>> >         >
>>>> >         > Thanks and best regards,
>>>> >         > -Jingwei
>>>> >         >
>>>> >
>>>> >
>>>> >         _______________________________________________
>>>> >         FreeSWITCH-users mailing list
>>>> >         FreeSWITCH-users at lists.freeswitch.org
>>>> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >         UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >         http://www.freeswitch.org
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From a.afzali2003 at gmail.com  Sat Jan 23 06:49:30 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sat, 23 Jan 2010 18:19:30 +0330
Subject: [Freeswitch-users] Possibly Bug in mod_sofia
Message-ID: <debff5a81001230649v5dcd1127nd46ea112b03b4c1c@mail.gmail.com>

Hi,

In sofia_reg.c line 190 :

nua_handle_bind(gateway_ptr->nh, gateway_ptr->sofia_private);

It seems the right handle be gateway_ptr->sub_nh. It causes every SUBSCRIBE
message's response print "Gateway information missing" error.

-- afshin
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From satish_lx at hotmail.com  Fri Jan 22 23:08:14 2010
From: satish_lx at hotmail.com (satish patel)
Date: Sat, 23 Jan 2010 07:08:14 +0000
Subject: [Freeswitch-users] mod_radius_cdr module load error
Message-ID: <BAY141-W1AF84CAAD730644C7A09790610@phx.gbl>


Hi All,
I am following this wiki http://wiki.freeswitch.org/wiki/Mod_radius_cdr to hook up freeradius with freeswitch but i am getting following error in log
2010-01-23 01:56:25.717201 [ERR] mod_radius_cdr.c:662 Open of mod_radius_cdr.conf failed2010-01-23 01:56:25.717225 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_radius_cdr.so**Module load routine returned an error**

Any idea Team?
Appreciate your help.
Best,
S. 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
http://clk.atdmt.com/GBL/go/196390706/direct/01/
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From jmesquita at freeswitch.org  Sat Jan 23 08:24:39 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 23 Jan 2010 14:24:39 -0200
Subject: [Freeswitch-users] Call for help on FSComm
Message-ID: <ebbc2d551001230824p53f4319dj85bc4b44254c4a18@mail.gmail.com>

FreeSWITCH?ers,

I believe that everyone already heard about FSComm by now. Development is
moving fast and features are being added very fast. Nonetheless, we need
help to get this software the way we want.

If you are a GUI Designer or have any design skills and want to contribute,
please, get in touch with me by email, IM, IRC on #fscomm or even pigeon if
you like. I believe everyone wants a sexy looking softphone with all the
features FreeSWITCH? is able to provide.

Regards,
Jo?o Mesquita
FSComm Developer
GTalk: jmesquita at gmail.com
PayPal: jmesquita at gmail.com
IRC: jmesquita on #fscomm @freenode
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From brian at freeswitch.org  Sat Jan 23 08:29:31 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 23 Jan 2010 10:29:31 -0600
Subject: [Freeswitch-users] Possibly Bug in mod_sofia
In-Reply-To: <debff5a81001230649v5dcd1127nd46ea112b03b4c1c@mail.gmail.com>
References: <debff5a81001230649v5dcd1127nd46ea112b03b4c1c@mail.gmail.com>
Message-ID: <73D97305-56C0-4E68-B452-92FC0D1A438C@freeswitch.org>

No I'm pretty sure its correct. Their are options on gateways to make or take inbound subscriptions and route them its not documented.

Also that line is NOT an error level its a Debug level log.

If it were dangerous it would be CRIT or ERROR level logging.

Look at parse_gateway_subscriptions

/b


On Jan 23, 2010, at 8:49 AM, afshin afzali wrote:

> Hi,
> 
> In sofia_reg.c line 190 : 
> 
> nua_handle_bind(gateway_ptr->nh, gateway_ptr->sofia_private);
> 
> It seems the right handle be gateway_ptr->sub_nh. It causes every SUBSCRIBE message's response print "Gateway information missing" error.
> 
> -- afshin

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From yehavi.bourvine at gmail.com  Sat Jan 23 09:06:12 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 23 Jan 2010 19:06:12 +0200
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
Message-ID: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>

Hello,

  We are running 1.0.5pre10 for a while, and today I tried to move to the
latest tarball (from January 22nd). The software crashes with a core dump
after a few seconds. The core dump. The two relevant lines (to my opinion)
are:

#4  0xb7547d2d in nua_bye_server_report (sr=0xb6c910b0, tags=0x0)
    at nua_session.c:3938
#5  0xb7541bb3 in nua_server_report (sr=0x6) at nua_server.c:643
line 3938 is an assert() statement. Any idea?

                          Thanks! __Yehavi:
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From brian at freeswitch.org  Sat Jan 23 09:28:20 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 23 Jan 2010 11:28:20 -0600
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
Message-ID: <45626378-10E7-4723-A2D3-B05A831CD8E9@freeswitch.org>

Please collect the sip trace, console logs and everything you can up till the crash. Do not make decisions on what you think is relevant collect everything you can and open a jira.  

Also did you do a fresh checkout? or a make current?

Thanks,
Brian

On Jan 23, 2010, at 11:06 AM, Yehavi Bourvine wrote:

> Hello,
>  
>   We are running 1.0.5pre10 for a while, and today I tried to move to the latest tarball (from January 22nd). The software crashes with a core dump after a few seconds. The core dump. The two relevant lines (to my opinion) are:
>  
> #4  0xb7547d2d in nua_bye_server_report (sr=0xb6c910b0, tags=0x0)
>     at nua_session.c:3938
> #5  0xb7541bb3 in nua_server_report (sr=0x6) at nua_server.c:643
> line 3938 is an assert() statement. Any idea?
>  
>                           Thanks! __Yehavi:
>  




From sos at sokhapkin.dyndns.org  Sat Jan 23 09:35:05 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 23 Jan 2010 12:35:05 -0500
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
Message-ID: <201001231235.05300.sos@sokhapkin.dyndns.org>

Could you confirm that you have an issue described in 
http://jira.freeswitch.org/browse/SFSIP-197 ?

Seems like you're not the only unlucky...

On Saturday 23 January 2010, Yehavi Bourvine wrote:
> Hello,
>
>   We are running 1.0.5pre10 for a while, and today I tried to move to the
> latest tarball (from January 22nd). The software crashes with a core dump
> after a few seconds. The core dump. The two relevant lines (to my opinion)
> are:
>
> #4  0xb7547d2d in nua_bye_server_report (sr=0xb6c910b0, tags=0x0)
>     at nua_session.c:3938
> #5  0xb7541bb3 in nua_server_report (sr=0x6) at nua_server.c:643
> line 3938 is an assert() statement. Any idea?
>
>                           Thanks! __Yehavi:





From moises.silva at gmail.com  Sat Jan 23 09:41:53 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 23 Jan 2010 12:41:53 -0500
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
	<7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>
	<7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>
Message-ID: <c4d05cbe1001230941ud426917m77aa774430110a2b@mail.gmail.com>

On Fri, Jan 22, 2010 at 11:32 PM, Thangappan.M <thangappan143 at gmail.com>wrote:

> But I am unable to make a outbound call. When I was trying the following
> was get.
>
> freeswitch at internal> originate openzap/smg_prid/a/9940464753 at g1openzap/smg_prid/a/9940464753 at g1
> -ERR NORMAL_CIRCUIT_CONGESTION
>
> 2010-01-23 10:00:45.688854 [WARNING] ozmod_sangoma_boost.c:348 TX EVENT:
> CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[9940464753]
> Ci=[0000000000] Rdnis=[]
> freeswitch at internal> 2010-01-23 10:00:46.709355 [WARNING]
> ozmod_sangoma_boost.c:1373 RX EVENT (N): CALL_START_NACK:(82) [w256g256]
> Rc=0 CSid=2 Seq=2
> 2010-01-23 10:00:46.709355 [WARNING] sangoma_boost_client.c:220 TX EVENT
> (N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=2 Seq=3
> 2010-01-23 10:00:46.709355 [ERR] mod_openzap.c:1162 No channels available
> 2010-01-23 10:00:46.709355 [ERR] switch_ivr_originate.c:2411 Cannot create
> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
> 2010-01-23 10:00:46.709355 [DEBUG] switch_ivr_originate.c:3211 Originate
> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>
> Please help me...........
>

Did you define group 1 in /etc/wanpipe/smg_prid.conf, pastebin the file plz.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From yehavi.bourvine at gmail.com  Sat Jan 23 10:02:09 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 23 Jan 2010 20:02:09 +0200
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <201001231235.05300.sos@sokhapkin.dyndns.org>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<201001231235.05300.sos@sokhapkin.dyndns.org>
Message-ID: <d361ca7b1001231002q216a23d5j96bbc87e01c01749@mail.gmail.com>

Yes, it looks exactly the same, with the same code in retval. It happens
just when an incoming INVITE arrives.
Since it has already a jira issue opened, do I still have to provide the
traces?

About how I upgrade: I've downloded the tarball of the latest version into a
fresh directory, built it, and in order to install it:
deleted everyhting in bin, mod and lib, and then made "make install".

                     Thanks, __Yehavi:




2010/1/23 Sergey Okhapkin <sos at sokhapkin.dyndns.org>

> Could you confirm that you have an issue described in
> http://jira.freeswitch.org/browse/SFSIP-197 ?
>
> Seems like you're not the only unlucky...
>
> On Saturday 23 January 2010, Yehavi Bourvine wrote:
> > Hello,
> >
> >   We are running 1.0.5pre10 for a while, and today I tried to move to the
> > latest tarball (from January 22nd). The software crashes with a core dump
> > after a few seconds. The core dump. The two relevant lines (to my
> opinion)
> > are:
> >
> > #4  0xb7547d2d in nua_bye_server_report (sr=0xb6c910b0, tags=0x0)
> >     at nua_session.c:3938
> > #5  0xb7541bb3 in nua_server_report (sr=0x6) at nua_server.c:643
> > line 3938 is an assert() statement. Any idea?
> >
> >                           Thanks! __Yehavi:
>
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From a.afzali2003 at gmail.com  Sat Jan 23 10:23:48 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sat, 23 Jan 2010 21:53:48 +0330
Subject: [Freeswitch-users] Possibly Bug in mod_sofia
In-Reply-To: <73D97305-56C0-4E68-B452-92FC0D1A438C@freeswitch.org>
References: <debff5a81001230649v5dcd1127nd46ea112b03b4c1c@mail.gmail.com>
	<73D97305-56C0-4E68-B452-92FC0D1A438C@freeswitch.org>
Message-ID: <debff5a81001231023n7c491282ub878b645ed38b09a@mail.gmail.com>

Thanks to your reply,

I've paid attention to that error because the successful subscription,
immediately does unsubscribe by sending another SUBSCRIBE message. By
inspection in the function which the error does log (sofia_presence : 2198)
it appears that it is a precondition to accept any SUBSCRIBE message
response. and finally I did just this modification : nh -> sub_nh and the
result is the stable subscription operation.

of course you are right  :)
-- afshin

On Sat, Jan 23, 2010 at 7:59 PM, Brian West <brian at freeswitch.org> wrote:

> No I'm pretty sure its correct. Their are options on gateways to make or
> take inbound subscriptions and route them its not documented.
>
> Also that line is NOT an error level its a Debug level log.
>
> If it were dangerous it would be CRIT or ERROR level logging.
>
> Look at parse_gateway_subscriptions
>
> /b
>
>
> On Jan 23, 2010, at 8:49 AM, afshin afzali wrote:
>
> Hi,
>
> In sofia_reg.c line 190 :
>
> nua_handle_bind(gateway_ptr->nh, gateway_ptr->sofia_private);
>
> It seems the right handle be gateway_ptr->sub_nh. It causes every SUBSCRIBE
> message's response print "Gateway information missing" error.
>
> -- afshin
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sos at sokhapkin.dyndns.org  Sat Jan 23 10:25:53 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 23 Jan 2010 13:25:53 -0500
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001231002q216a23d5j96bbc87e01c01749@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<201001231235.05300.sos@sokhapkin.dyndns.org>
	<d361ca7b1001231002q216a23d5j96bbc87e01c01749@mail.gmail.com>
Message-ID: <201001231325.54564.sos@sokhapkin.dyndns.org>

Add your traces and all information you can provide to 
http://jira.freeswitch.org/browse/SFSIP-197 , the more information developers 
will get, the sooner they will be able to find and fix the issue.

I wish my FS crash "in a few seconds" like yours, then I will be able to debug 
myself, but it crashes very rarely...

On Saturday 23 January 2010, Yehavi Bourvine wrote:
> Yes, it looks exactly the same, with the same code in retval. It happens
> just when an incoming INVITE arrives.
> Since it has already a jira issue opened, do I still have to provide the
> traces?
>
> About how I upgrade: I've downloded the tarball of the latest version into
> a fresh directory, built it, and in order to install it:
> deleted everyhting in bin, mod and lib, and then made "make install".
>
>                      Thanks, __Yehavi:
>
>
>
>
> 2010/1/23 Sergey Okhapkin <sos at sokhapkin.dyndns.org>
>
> > Could you confirm that you have an issue described in
> > http://jira.freeswitch.org/browse/SFSIP-197 ?
> >
> > Seems like you're not the only unlucky...
> >
> > On Saturday 23 January 2010, Yehavi Bourvine wrote:
> > > Hello,
> > >
> > >   We are running 1.0.5pre10 for a while, and today I tried to move to
> > > the latest tarball (from January 22nd). The software crashes with a
> > > core dump after a few seconds. The core dump. The two relevant lines
> > > (to my
> >
> > opinion)
> >
> > > are:
> > >
> > > #4  0xb7547d2d in nua_bye_server_report (sr=0xb6c910b0, tags=0x0)
> > >     at nua_session.c:3938
> > > #5  0xb7541bb3 in nua_server_report (sr=0x6) at nua_server.c:643
> > > line 3938 is an assert() statement. Any idea?
> > >
> > >                           Thanks! __Yehavi:
> >
> >  _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From brian at freeswitch.org  Sat Jan 23 10:29:57 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 23 Jan 2010 12:29:57 -0600
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001231002q216a23d5j96bbc87e01c01749@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<201001231235.05300.sos@sokhapkin.dyndns.org>
	<d361ca7b1001231002q216a23d5j96bbc87e01c01749@mail.gmail.com>
Message-ID: <AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>

Yes please provide traces the more information we have the clearer the picture and possibility of fixing it.  

/b

On Jan 23, 2010, at 12:02 PM, Yehavi Bourvine wrote:

> Yes, it looks exactly the same, with the same code in retval. It happens just when an incoming INVITE arrives.
> Since it has already a jira issue opened, do I still have to provide the traces?
>  
> About how I upgrade: I've downloded the tarball of the latest version into a fresh directory, built it, and in order to install it:
> deleted everyhting in bin, mod and lib, and then made "make install".
>  
>                      Thanks, __Yehavi:
>  
> 




From brian at freeswitch.org  Sat Jan 23 10:31:02 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 23 Jan 2010 12:31:02 -0600
Subject: [Freeswitch-users] Possibly Bug in mod_sofia
In-Reply-To: <debff5a81001231023n7c491282ub878b645ed38b09a@mail.gmail.com>
References: <debff5a81001230649v5dcd1127nd46ea112b03b4c1c@mail.gmail.com>
	<73D97305-56C0-4E68-B452-92FC0D1A438C@freeswitch.org>
	<debff5a81001231023n7c491282ub878b645ed38b09a@mail.gmail.com>
Message-ID: <A46ABB96-5355-44DE-9F0E-B098AAB5B44D@freeswitch.org>

If its still a bug or you had to change code please post a jira.. its the correct place to debate and talk about issues so we can track them properly.  ;)

Thanks,
Brian

On Jan 23, 2010, at 12:23 PM, afshin afzali wrote:

> Thanks to your reply,
> 
> I've paid attention to that error because the successful subscription, immediately does unsubscribe by sending another SUBSCRIBE message. By inspection in the function which the error does log (sofia_presence : 2198) it appears that it is a precondition to accept any SUBSCRIBE message response. and finally I did just this modification : nh -> sub_nh and the result is the stable subscription operation.
> 
> of course you are right  :)
> -- afshin     




From yehavi.bourvine at gmail.com  Sat Jan 23 10:38:27 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 23 Jan 2010 20:38:27 +0200
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<201001231235.05300.sos@sokhapkin.dyndns.org>
	<d361ca7b1001231002q216a23d5j96bbc87e01c01749@mail.gmail.com>
	<AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
Message-ID: <d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>

Just uploaded. It happens when I call our voicemail number and disconnects
the call. The voicemail application answers, and I disconnect the call the
Freeswitch crash.

                 Thanks, __Yehavi:

2010/1/23 Brian West <brian at freeswitch.org>

> Yes please provide traces the more information we have the clearer the
> picture and possibility of fixing it.
>
> /b
>
> On Jan 23, 2010, at 12:02 PM, Yehavi Bourvine wrote:
>
> > Yes, it looks exactly the same, with the same code in retval. It happens
> just when an incoming INVITE arrives.
> > Since it has already a jira issue opened, do I still have to provide the
> traces?
> >
> > About how I upgrade: I've downloded the tarball of the latest version
> into a fresh directory, built it, and in order to install it:
> > deleted everyhting in bin, mod and lib, and then made "make install".
> >
> >                      Thanks, __Yehavi:
> >
> >
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sos at sokhapkin.dyndns.org  Sat Jan 23 10:55:30 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 23 Jan 2010 13:55:30 -0500
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
	<d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
Message-ID: <201001231355.31032.sos@sokhapkin.dyndns.org>

GOOD! You have a consistent way to reproduce the problem!

On Saturday 23 January 2010, Yehavi Bourvine wrote:
> Just uploaded. It happens when I call our voicemail number and disconnects
> the call. The voicemail application answers, and I disconnect the call the
> Freeswitch crash.
>
>                  Thanks, __Yehavi:
>
> 2010/1/23 Brian West <brian at freeswitch.org>
>
> > Yes please provide traces the more information we have the clearer the
> > picture and possibility of fixing it.
> >
> > /b
> >
> > On Jan 23, 2010, at 12:02 PM, Yehavi Bourvine wrote:
> > > Yes, it looks exactly the same, with the same code in retval. It
> > > happens
> >
> > just when an incoming INVITE arrives.
> >
> > > Since it has already a jira issue opened, do I still have to provide
> > > the
> >
> > traces?
> >
> > > About how I upgrade: I've downloded the tarball of the latest version
> >
> > into a fresh directory, built it, and in order to install it:
> > > deleted everyhting in bin, mod and lib, and then made "make install".
> > >
> > >                      Thanks, __Yehavi:
> >
> >  _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From brian at freeswitch.org  Sat Jan 23 11:00:13 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 23 Jan 2010 13:00:13 -0600
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<201001231235.05300.sos@sokhapkin.dyndns.org>
	<d361ca7b1001231002q216a23d5j96bbc87e01c01749@mail.gmail.com>
	<AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
	<d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
Message-ID: <FE53682D-179E-4F6D-AF2F-714913C76B6B@freeswitch.org>

Please get a full pcap trace console log and attach it.

/b

On Jan 23, 2010, at 12:38 PM, Yehavi Bourvine wrote:

> Just uploaded. It happens when I call our voicemail number and disconnects the call. The voicemail application answers, and I disconnect the call the Freeswitch crash.
>  
>                  Thanks, __Yehavi:




From brian at freeswitch.org  Sat Jan 23 11:02:53 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 23 Jan 2010 13:02:53 -0600
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <201001231355.31032.sos@sokhapkin.dyndns.org>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
	<d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
	<201001231355.31032.sos@sokhapkin.dyndns.org>
Message-ID: <AC76DB08-C278-4F59-90D0-2C3724CDB39A@freeswitch.org>

Best to find us on IRC when anthm is around and lets get into your box and fix this.

/b

On Jan 23, 2010, at 12:55 PM, Sergey Okhapkin wrote:

> GOOD! You have a consistent way to reproduce the problem!




From camilin2212 at hotmail.com  Sat Jan 23 14:08:57 2010
From: camilin2212 at hotmail.com (juan camilo ospina quintero)
Date: Sat, 23 Jan 2010 17:08:57 -0500
Subject: [Freeswitch-users] FreeSwitch Integration With Sailfin
Message-ID: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>


hi


im having serious trouble trying to integrate freeswitch with sailfin, if someone could help me, taht would be awesome. 
I guess is with a bridge statement in the dialplan , but not sure how to do that, and if that is the right solution.

thanks
 		 	   		  
_________________________________________________________________
Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail?.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009
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From brian at freeswitch.org  Sat Jan 23 14:34:04 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 23 Jan 2010 16:34:04 -0600
Subject: [Freeswitch-users] FreeSwitch Integration With Sailfin
In-Reply-To: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>
References: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>
Message-ID: <CFA0D82A-33B3-4B50-B6AE-BAA61E0862A4@freeswitch.org>

When asking a question on the list you'll need to provide some back story on what exactly you're doing.  Outline any issues you're having and try to include as much info as possible.  Nobody can help you otherwise.  For example what is Sailfin?  What bridge line?  What do your logs say?

Thanks,
/b

On Jan 23, 2010, at 4:08 PM, juan camilo ospina quintero wrote:

> hi
> 
> 
> im having serious trouble trying to integrate freeswitch with sailfin, if someone could help me, taht would be awesome. 
> I guess is with a bridge statement in the dialplan , but not sure how to do that, and if that is the right solution.
> 
> thanks

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From camilin2212 at hotmail.com  Sat Jan 23 14:48:42 2010
From: camilin2212 at hotmail.com (juan camilo ospina quintero)
Date: Sat, 23 Jan 2010 17:48:42 -0500
Subject: [Freeswitch-users] FreeSwitch Integration With Sailfin
In-Reply-To: <CFA0D82A-33B3-4B50-B6AE-BAA61E0862A4@freeswitch.org>
References: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>,
	<CFA0D82A-33B3-4B50-B6AE-BAA61E0862A4@freeswitch.org>
Message-ID: <BAY143-W13760770FF3FCA224AACEA1610@phx.gbl>


OK

sorry, is my first time in the mailing list, well what i need to do, is to integrate freeswitch with sailfin, sailfin is an application server SIP based, i need to forwards the SIP traffic from freeswitch to sailfin. the registration can be done in freeswitch. But i need to redirect the Sip invite methods to sailfin. i have no log because i have not been able to do anything. i have freeswitch installed and runnig, also sailfin. i can make calls between users with freeswitch. thats all. i read that adding a bridge statement to the dial plan would help, but i dont know how to do that, im new in freeswitch

thanks to all

From: brian at freeswitch.org
Date: Sat, 23 Jan 2010 16:34:04 -0600
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSwitch Integration With Sailfin



When asking a question on the list you'll need to provide some back story on what exactly you're doing.  Outline any issues you're having and try to include as much info as possible.  Nobody can help you otherwise.  For example what is Sailfin?  What bridge line?  What do your logs say?
Thanks,/b
On Jan 23, 2010, at 4:08 PM, juan camilo ospina quintero wrote:hi


im having serious trouble trying to integrate freeswitch with sailfin, if someone could help me, taht would be awesome. 
I guess is with a bridge statement in the dialplan , but not sure how to do that, and if that is the right solution.

thanks
 		 	   		  
_________________________________________________________________
Windows Live: Keep your friends up to date with what you do online.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_1:092010
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From edpimentl at gmail.com  Sat Jan 23 15:01:07 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Sat, 23 Jan 2010 18:01:07 -0500
Subject: [Freeswitch-users] FreeSwitch Integration With Sailfin
In-Reply-To: <BAY143-W13760770FF3FCA224AACEA1610@phx.gbl>
References: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>
	<CFA0D82A-33B3-4B50-B6AE-BAA61E0862A4@freeswitch.org> 
	<BAY143-W13760770FF3FCA224AACEA1610@phx.gbl>
Message-ID: <9dc4a1671001231501t65b1281epeeb7e7fa1b0bf5bf@mail.gmail.com>

It would be good if you include the following and then inject your question
http://wiki.glassfish.java.net/Wiki.jsp?page=GlassFishWiki
http://wiki.glassfish.java.net/Wiki.jsp?page=SailFin
https://sailfin.dev.java.net/


-E
http://vCardCloud.com




On Sat, Jan 23, 2010 at 5:48 PM, juan camilo ospina quintero <
camilin2212 at hotmail.com> wrote:

>  OK
>
> sorry, is my first time in the mailing list, well what i need to do, is to
> integrate freeswitch with sailfin, sailfin is an application server SIP
> based, i need to forwards the SIP traffic from freeswitch to sailfin. the
> registration can be done in freeswitch. But i need to redirect the Sip
> invite methods to sailfin. i have no log because i have not been able to do
> anything. i have freeswitch installed and runnig, also sailfin. i can make
> calls between users with freeswitch. thats all. i read that adding a bridge
> statement to the dial plan would help, but i dont know how to do that, im
> new in freeswitch
>
> thanks to all
>
> ------------------------------
> From: brian at freeswitch.org
> Date: Sat, 23 Jan 2010 16:34:04 -0600
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FreeSwitch Integration With Sailfin
>
> When asking a question on the list you'll need to provide some back story
> on what exactly you're doing.  Outline any issues you're having and try to
> include as much info as possible.  Nobody can help you otherwise.  For
> example what is Sailfin?  What bridge line?  What do your logs say?
>
> Thanks,
> /b
>
> On Jan 23, 2010, at 4:08 PM, juan camilo ospina quintero wrote:
>
> hi
>
>
> im having serious trouble trying to integrate freeswitch with sailfin, if
> someone could help me, taht would be awesome.
> I guess is with a bridge statement in the dialplan , but not sure how to do
> that, and if that is the right solution.
>
> thanks
>
>
>
> ------------------------------
> Windows Live: Keep your friends up to date with what you do online.<http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_1:092010>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wchao at yahoo.com  Sat Jan 23 19:11:10 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Sat, 23 Jan 2010 22:11:10 -0500 (EST)
Subject: [Freeswitch-users] Eavesdrop when using simring
In-Reply-To: <191c3a031001201209y53d68c83t8ba066b63ff4b723@mail.gmail.com>
References: <alpine.LRH.2.00.1001201139480.13391@osgood.microoffice.us>
	<191c3a031001201209y53d68c83t8ba066b63ff4b723@mail.gmail.com>
Message-ID: <alpine.LRH.2.00.1001232153410.26382@osgood.microoffice.us>

Thanks. Your answer helped me find execute_on_answer, which worked for me. 
Using execute_on_answer, I was able to get it to work exactly as I wanted, 
but I had to hardcode the variable per bridge target. I am wondering if 
there's a better way to handle it. Here is what I have now (which works):

<action application="bridge" 
data="[execute_on_answer='hash 
insert/${domain_name}-spymap/646xxxyyyy-1000/${uuid}']user/1001@${domain_name},[execute_on_answer='hash 
insert/${domain_name}-spymap/646xxxyyyy-1001/${uuid}']user/1002@${domain_name},[execute_on_answer='hash 
insert/${domain_name}-spymap/646xxxyyyy-1002/${uuid}']user/1003@${domain_name},[execute_on_answer='hash 
insert/${domain_name}-spymap/646xxxyyyy-1003/${uuid}']user/1004@${domain_name}"/>

What I am wondering is whether there's a way to do it like this instead:

<action application="export" data="execute_on_answer='hash 
insert/${domain_name}-spymap/646xxxyyyy-${answered_username}/${uuid}'"/>
<action application="bridge" 
data="user/1001@${domain_name},user/1002@${domain_name},user/1003@${domain_name},user/1004@${domain_name}"/>

So, the two questions are:

[1] How would I get the username of the bridge target that picks up the 
call?

[2] Is there a way to defer variable substitution/evaluation until after 
the call is answered?


Date: Wed, 20 Jan 2010 14:09:48 -0600
From: Anthony Minessale <anthony.minessale at gmail.com>
Reply-To: freeswitch-users at lists.freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Eavesdrop when using simring

maybe api_on_answer var?

On Wed, Jan 20, 2010 at 10:48 AM, Wellie Chao <wchao at yahoo.com> wrote:
      I have eavesdrop working fine on outbound calls and also inbound calls
      where there is a single DID per IP phone. When I have a DID that rings
      multiple extensions simultaneously, what is the best way to obtain
      information about which extension has picked up the call and store that
      using hash? I can set a variable before I issue the bridge action, like
      so:

      <action application="hash" data="insert/${domain_name}-spymap/646xxxyyyy-1000/${uuid}"/>
      <action application="bridge"
      data="user/1001@${domain_name},user/1002@${domain_name},user/1003@${domain_name},user/1004@${domain_name}"/>

      However, that doesn't tell me who actually picked up, so at best I can
      allow users to eavesdrop on the last incoming call to the main DID, not
      the last incoming call to a particular extension. Is there something I can
      do in the bridge that will cause it to set a variable once it knows which
      extension has picked up the call?

      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900




From camilin2212 at hotmail.com  Sat Jan 23 20:48:42 2010
From: camilin2212 at hotmail.com (juan camilo ospina quintero)
Date: Sat, 23 Jan 2010 23:48:42 -0500
Subject: [Freeswitch-users] FreeSwitch Integration With Sailfin
In-Reply-To: <9dc4a1671001231501t65b1281epeeb7e7fa1b0bf5bf@mail.gmail.com>
References: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>,
	<CFA0D82A-33B3-4B50-B6AE-BAA61E0862A4@freeswitch.org>,
	<BAY143-W13760770FF3FCA224AACEA1610@phx.gbl>,
	<9dc4a1671001231501t65b1281epeeb7e7fa1b0bf5bf@mail.gmail.com>
Message-ID: <BAY143-W760D1216FA26D7EFD8E9DA1600@phx.gbl>


Hi 
thanks for fast answers
i already have sailfin installed and runnig, also freeswitch, both on the same machine.
now my quiestion is how to redirect o forward the SIP flow from freeswitch to salifin, the softphones must register in freeswitch, but to make a call the invite should go through sailfin and get back to freeswitch, this is for implementing some services of VoIP, for a project i'm on.

From: edpimentl at gmail.com
Date: Sat, 23 Jan 2010 18:01:07 -0500
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSwitch Integration With Sailfin

It would be good if you include the following and then inject your question
http://wiki.glassfish.java.net/Wiki.jsp?page=GlassFishWiki
http://wiki.glassfish.java.net/Wiki.jsp?page=SailFin


https://sailfin.dev.java.net/

-E
http://vCardCloud.com





On Sat, Jan 23, 2010 at 5:48 PM, juan camilo ospina quintero <camilin2212 at hotmail.com> wrote:







OK

sorry, is my first time in the mailing list, well what i need to do, is to integrate freeswitch with sailfin, sailfin is an application server SIP based, i need to forwards the SIP traffic from freeswitch to sailfin. the registration can be done in freeswitch. But i need to redirect the Sip invite methods to sailfin. i have no log because i have not been able to do anything. i have freeswitch installed and runnig, also sailfin. i can make calls between users with freeswitch. thats all. i read that adding a bridge statement to the dial plan would help, but i dont know how to do that, im new in freeswitch



thanks to all

From: brian at freeswitch.org
Date: Sat, 23 Jan 2010 16:34:04 -0600
To: freeswitch-users at lists.freeswitch.org


Subject: Re: [Freeswitch-users] FreeSwitch Integration With Sailfin



When asking a question on the list you'll need to provide some back story on what exactly you're doing.  Outline any issues you're having and try to include as much info as possible.  Nobody can help you otherwise.  For example what is Sailfin?  What bridge line?  What do your logs say?


Thanks,/b
On Jan 23, 2010, at 4:08 PM, juan camilo ospina quintero wrote:
hi




im having serious trouble trying to integrate freeswitch with sailfin, if someone could help me, taht would be awesome. 
I guess is with a bridge statement in the dialplan , but not sure how to do that, and if that is the right solution.



thanks
 		 	   		  
Windows Live: Keep your friends up to date with what you do online.



_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org



 		 	   		  
_________________________________________________________________
Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail?.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009
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From mike at jerris.com  Sat Jan 23 22:54:50 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 01:54:50 -0500
Subject: [Freeswitch-users] Managing Presence on Gateways
In-Reply-To: <debff5a81001210144n4b32c19cic36415155ee52231@mail.gmail.com>
References: <debff5a81001210144n4b32c19cic36415155ee52231@mail.gmail.com>
Message-ID: <326ADCAA-AC7E-4625-886B-3FFF7E30FAD2@jerris.com>

No, we don't have the functionality to gateway presence.

On Jan 21, 2010, at 4:44 AM, afshin afzali <a.afzali2003 at gmail.com>  
wrote:

> Hi Guys,
>
> In the external profile (as in the internal) there is an option to  
> enable presence functionality (with setting it to passive). My  
> question is how does it mean presence functionality for a gateway  
> which interfaces home domain to another one? Does it mean that the  
> gateway could subscribe itself for some presence information in that  
> domain in behaves of local users and relays them?
>
> Appreciate all comments,
> -- afshin
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From mike at jerris.com  Sun Jan 24 01:48:13 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 04:48:13 -0500
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION?
In-Reply-To: <201001211405.19271.sos@sokhapkin.dyndns.org>
References: <201001211341.09739.sos@sokhapkin.dyndns.org>
	<191c3a031001211053i6670cec3l8b8f317dac4c6072@mail.gmail.com>
	<201001211405.19271.sos@sokhapkin.dyndns.org>
Message-ID: <6206D9B0-EA6B-4CAC-A42A-727BAF6254FC@jerris.com>

Try this out again with current trunk before taking logs and potentially filing a bug.  I suspect this is fixed now if it is what I think it is.

Mike

On Jan 21, 2010, at 2:05 PM, Sergey Okhapkin wrote:

> Late negotiation is set. I will try to enable debug when the traffic will be 
> low and open a problem on jira.
> 
> On Thursday 21 January 2010, Anthony Minessale wrote:
>> if you use bypass_media=true from the dialplan without late-negotiation set
>> in the profile, it still tries to match the codecs locally on the inbound
>> leg and the variable does not work if the call has established media before
>> making the outbound leg.
>> 
>> It's hard to tell you the exact answer without a console trace on debug
>> level.
>> 
>> 
>> On Thu, Jan 21, 2010 at 12:41 PM, Sergey Okhapkin
>> 
>> <sos at sokhapkin.dyndns.org>wrote:
>>> I often see in FS log the following problem (bypass_media=true), SVN
>>> r16340:
>>> 
>>> SDP sent out to gateway (INVITE):
>>> 
>>>  v=0
>>>  o=bandx-msw3 0 0 IN IP4 213.166.9.4
>>>  s=sip call
>>>  c=IN IP4 213.166.9.6
>>>  t=0 0
>>>  m=audio 56032 RTP/AVP 0 8 18
>>>  a=rtpmap:18 G729/8000
>>>  a=fmtp:18 annexb=yes
>>>  a=silenceSupp:on - - - -
>>> 
>>> 
>>> SDP response from gateway (183 Session Progress):
>>> 
>>>  v=0
>>>  o=- 3473087019 3473087037 IN IP4 67.203.64.182
>>>  s=-
>>>  c=IN IP4 67.203.64.182
>>>  t=0 0
>>>  m=audio 14116 RTP/AVP 0
>>>  a=sendrecv
>>>  a=ptime:20
>>>  a=rtpmap:0 PCMU/8000
>>> 
>>> Freeswitch hangs up the call with hangup cause INCOMPATIBLE_DESTINATION.
>>> Where
>>> is incompatibility? There is common codec 0.
>>> 



From mike at jerris.com  Sun Jan 24 01:50:00 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 04:50:00 -0500
Subject: [Freeswitch-users] reloadxml/rescan profile  and  killgw
In-Reply-To: <4B589A6E.8010205@kinetix.gr>
References: <4B589A6E.8010205@kinetix.gr>
Message-ID: <3F378D25-85F8-48B9-B814-7F7979EFE840@jerris.com>

A lot of things would be nice.  Patches for example.  I find them quite nice.  This should probably be a different command or argument.

Mike

On Jan 21, 2010, at 1:18 PM, Apostolos Pantsiopoulos wrote:
  
> Wouldn't it be nice a "reloadxml"/"rescan profile" to delete all 
> nonexistent gateways (when they are absent in the xml config) without 
> having to issue a "sofia profile xxxxx killgw yyyyyy" command?
> 
> I always seem to find forgotten gateway's in the profile because of 
> this... Any thoughts?




From mike at jerris.com  Sun Jan 24 01:53:13 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 04:53:13 -0500
Subject: [Freeswitch-users] Presence Not Working After Version 1.0.5pre9?
In-Reply-To: <B4EB2C4365CD4CA79B54134ECF232EED@greyhawk.tonecommander.com>
References: <mailman.5334.1264017641.2495.freeswitch-users@lists.freeswitch.org><C726F91949324BDFA34596256B6E102A@greyhawk.tonecommander.com>
	<191c3a031001201531g7c78cb9fw1ed1a2ba07f5773c@mail.gmail.com>
	<B4EB2C4365CD4CA79B54134ECF232EED@greyhawk.tonecommander.com>
Message-ID: <75590ED4-F843-48F3-93B8-3EFEE107D411@jerris.com>

If you could document the configuration requirements on the wiki I would appreciate it.

Mike

On Jan 21, 2010, at 5:39 PM, Jerry Richards wrote:

> Yes you are correct.  The Bria Softphone has a setting under ContactProfile/Advanced.../Account menu which is required to be the softphone's extension (not blank and not the extension that is being subscribed to).  After I set this field to the softphone's extension, FS starting reporting the the contact's presence status.
>  
> Thanks and Best Regards,
> Jerry
>  

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From mike at jerris.com  Sun Jan 24 01:57:17 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 04:57:17 -0500
Subject: [Freeswitch-users] Javascript self.session.getVariable
In-Reply-To: <ade741d61001211558h7b84cb96q16754650812a80e2@mail.gmail.com>
References: <ade741d61001211558h7b84cb96q16754650812a80e2@mail.gmail.com>
Message-ID: <ECA755AB-6BE3-46DF-9272-5085271A5F4F@jerris.com>

it should be just session.getVariable

var base_dir = session.getVariable ("base_dir"); 

example taken from http://svn.freeswitch.org/svn/freeswitch/trunk/scripts/javascript/aadir/aadir.js

This of course assumes that you are running as an application and you have a session there.

Mike


On Jan 21, 2010, at 6:58 PM, Mouncif Benniane wrote:

> I have the following in javascript:
> 
> caller_id = self.session.getVariable("caller_id_number")
> 
> 
> for some reasons it returns: 
> ReferenceError: self is not defined, I am following this page:
> 
> http://wiki.freeswitch.org/wiki/Session_getVariable
> 
> 
> any ideas?
> 

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From mike at jerris.com  Sun Jan 24 01:59:06 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 04:59:06 -0500
Subject: [Freeswitch-users] Svar: Re: Svar: Re: Home setup with home	
	company
In-Reply-To: <4B5996B1020000E100000404@mail.fribert.dk>
References: <4B5996B1020000E100000404@mail.fribert.dk>
Message-ID: <DA8197DE-0287-4097-AFE4-00295589DDE3@jerris.com>

Debug logs should help you figure out what is and is not matching in your conditions.

Mike

On Jan 22, 2010, at 6:14 AM, mailinglist wrote:

> Ok, I set it up like this:
> <extension name="CompanyIn">
>    <condition field="destination_number" expression="^(8203)$" continue="true"/>
>      <!-- man strftime - M-F, 9AM to 5PM -->
>       <condition field="${strftime(%w)}" expression="^([1-2])$"/>
>          <anti-action application="voicemail" data="default 10.11.12.25 10199"/>
>       <condition field="${strftime(%H%M)}" expression="^((09|1[0-6])[0-5][0-9]|1700)$">
>          <action application="set" data="call_timeout=45"/>
>          <action application="set" data="domain_name=10.11.12.25"/>
>          <action application="set" data="continue_on_fail=true"/>
>          <action application="set" data="hangup_after_bridge=true"/>
>          <action application="bridge" data="group/company@${domain_name}"/>
>          <action application="answer" data=""/>
>          <action application="sleep" data="1000"/>
>          <action application="voicemail" data="default 10.11.12.25 10199"/>
>          <anti-action application="voicemail" data="default 10.11.12.25 10199"/>
>       </condition>
> </extension>
> 
> But now it gives me:
> 
> 2010-01-22 11:52:08.667564 [NOTICE] switch_channel.c:602 New Channel sofia/external/2680xxxx at 87.54.25.116 [16baec2f-4407-df11-8fb3-000c29b7b4cb]
> 2010-01-22 11:52:08.800123 [INFO] mod_dialplan_xml.c:252 Processing 2680xxxx->4692xxxx in context public
> 2010-01-22 11:52:08.830071 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/2680xxxx at 87.54.25.116 to XML[8203 at default]
> 2010-01-22 11:52:08.830071 [INFO] mod_dialplan_xml.c:252 Processing 2680xxxx->8203 in context default
> 2010-01-22 11:52:08.830071 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/2680xxxx at 87.54.25.116 to enum[8203 at default]
> 2010-01-22 11:52:09.163439 [INFO] switch_core_state_machine.c:136 No Route, Aborting
> 
> huh

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From mike at jerris.com  Sun Jan 24 02:03:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 05:03:18 -0500
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <984278.36075.qm@web33504.mail.mud.yahoo.com>
References: <984278.36075.qm@web33504.mail.mud.yahoo.com>
Message-ID: <9881D312-67D1-40D3-B169-A178202F4E6C@jerris.com>

As noted on that bug, you should be able to either use \\ or / for the path separator there and it should work.

Mike

On Jan 22, 2010, at 9:18 AM, Diego Toro wrote:

> Hi, with svn r16440 the problem persists, I creted a jira report http://jira.freeswitch.org/browse/LBSNDF-8 this is a minor issue, but activing playback delimiter no audio file can be played. On FS the audio files are placed in the \sound\ directory, building the path on Windows would be \sound '\s' which is replaced by 'ound'.
> 
> Thank you
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 




From mike at jerris.com  Sun Jan 24 02:08:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 05:08:20 -0500
Subject: [Freeswitch-users] Distinguishing Blind REFER from application
	transfer
In-Reply-To: <15b9404e1001220747y79571117qab826152a2df1dc8@mail.gmail.com>
References: <15b9404e1001220322h5636c1aat968f515e0601d769@mail.gmail.com>
	<9F3D655F-BF09-4503-9709-953A839DC65C@freeswitch.org>
	<15b9404e1001220747y79571117qab826152a2df1dc8@mail.gmail.com>
Message-ID: <78A051B8-57E4-44ED-BFCC-17AB7C41B366@jerris.com>

I think they are intended to be there as long as you want them to be, up until they can be used in the cdr modules.  If you are looking for this much control, you would have to set them off to other vars, use some scripting language to manipulate, or use some socket based control mechanism.  

Mike


On Jan 22, 2010, at 10:47 AM, mayamatakeshi wrote:

> 
> On Fri, Jan 22, 2010 at 11:17 PM, Brian West <brian at freeswitch.org> wrote:
> Have you done a uuid_dump to see all the variables?
> 
> I just tried that with trunk. I can see the REFER variables stay set till the end of the call.
> They will show up in CHANNEL_HANGUP_COMPLETE:
> 
> variable_sip_h_Referred-By: user2 <sip:user2 at test1.com>
> variable_sip_refer_to: <sip:2000 at test1.com>
> 
> I suppose the only thing that will change them is another blind refer. But they will never be unset.
> 
> 
> 
> On Jan 22, 2010, at 5:22 AM, mayamatakeshi wrote:
> 
> > Before doing so I thought in ask if the REFER-related variables being preserved upon dialplan reentry would not be a bug (well, it could be a feature useful in some scenarios I suspect).
> 
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From mike at jerris.com  Sun Jan 24 02:13:32 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 05:13:32 -0500
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <c4d05cbe1001230941ud426917m77aa774430110a2b@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
	<7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>
	<7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>
	<c4d05cbe1001230941ud426917m77aa774430110a2b@mail.gmail.com>
Message-ID: <AED7091C-7115-4B19-B555-02E8BDBFD1EE@jerris.com>

Also note, we just made a backwards incompatible change to boost in the latest svn..  this will require sangoma_prid version 1.48 or later.  I don't think this is packaged up with wanpipe anywhere yet.  We are doing more validation of the new code, and updated driver packages should be available very soon.

Mike

On Jan 23, 2010, at 12:41 PM, Moises Silva wrote:

> On Fri, Jan 22, 2010 at 11:32 PM, Thangappan.M <thangappan143 at gmail.com> wrote:
> But I am unable to make a outbound call. When I was trying the following was get.
> 
> freeswitch at internal> originate openzap/smg_prid/a/9940464753 at g1 openzap/smg_prid/a/9940464753 at g1
> -ERR NORMAL_CIRCUIT_CONGESTION
> 
> 2010-01-23 10:00:45.688854 [WARNING] ozmod_sangoma_boost.c:348 TX EVENT: CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[9940464753] Ci=[0000000000] Rdnis=[]
> freeswitch at internal> 2010-01-23 10:00:46.709355 [WARNING] ozmod_sangoma_boost.c:1373 RX EVENT (N): CALL_START_NACK:(82) [w256g256] Rc=0 CSid=2 Seq=2
> 2010-01-23 10:00:46.709355 [WARNING] sangoma_boost_client.c:220 TX EVENT (N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=2 Seq=3
> 2010-01-23 10:00:46.709355 [ERR] mod_openzap.c:1162 No channels available
> 2010-01-23 10:00:46.709355 [ERR] switch_ivr_originate.c:2411 Cannot create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
> 2010-01-23 10:00:46.709355 [DEBUG] switch_ivr_originate.c:3211 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
> 
> Please help me...........
> 
> Did you define group 1 in /etc/wanpipe/smg_prid.conf, pastebin the file plz.
> 
> -- 
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Sun Jan 24 02:17:33 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 05:17:33 -0500
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <AC76DB08-C278-4F59-90D0-2C3724CDB39A@freeswitch.org>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
	<d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
	<201001231355.31032.sos@sokhapkin.dyndns.org>
	<AC76DB08-C278-4F59-90D0-2C3724CDB39A@freeswitch.org>
Message-ID: <5F6D7DFA-2432-4542-813D-3E77FFF92DCD@jerris.com>

I have looked into the code and backtrace related to this issue, and the best I can see, its impossible...  We will definitely need access to a box this is reproducible on to fix this issue.

Mike

On Jan 23, 2010, at 2:02 PM, Brian West wrote:

> Best to find us on IRC when anthm is around and lets get into your box and fix this.
> 
> /b
> 
> On Jan 23, 2010, at 12:55 PM, Sergey Okhapkin wrote:
> 
>> GOOD! You have a consistent way to reproduce the problem!
> 




From b_ball_henry at hotmail.com  Sun Jan 24 02:18:04 2010
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Sun, 24 Jan 2010 02:18:04 -0800
Subject: [Freeswitch-users] Accessing Sangoma card inside a openVZ container
Message-ID: <59ad9ca11001240218w3accbb12r462e5c2fa7511024@mail.gmail.com>

I know there are a couple of OpenVZ expert here on the mailing list.

Has anyone of you tried to run freeswitch in a container with access to
physical Sangoma card?
The reason for this is that I would like to create an "all in one box" for
small to medium size company that takes care of the most essential services.
Things like an IP PBX would be a huge plus. I have been playing with OpenVZ
for a while now, and I know how to let containers access devices in the /dev
directory from the physical box. But to be able to use something like
sangoma card, I think there are some things in the /proc directory need to
be made available to the containers. But I don't know how to do this yet.
Maybe some of you know..

Thanks,

-- 
Henry Huang
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From mike at jerris.com  Sun Jan 24 02:20:23 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 05:20:23 -0500
Subject: [Freeswitch-users] FreeSwitch Integration With Sailfin
In-Reply-To: <BAY143-W760D1216FA26D7EFD8E9DA1600@phx.gbl>
References: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>,
	<CFA0D82A-33B3-4B50-B6AE-BAA61E0862A4@freeswitch.org>,
	<BAY143-W13760770FF3FCA224AACEA1610@phx.gbl>,
	<9dc4a1671001231501t65b1281epeeb7e7fa1b0bf5bf@mail.gmail.com>
	<BAY143-W760D1216FA26D7EFD8E9DA1600@phx.gbl>
Message-ID: <84AB66EA-956C-4B42-87C8-42660CBEDCE3@jerris.com>

This is a good place to start reading on how to configure dialplan:

http://wiki.freeswitch.org/wiki/Dialplan
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#dialplan

Mike

On Jan 23, 2010, at 11:48 PM, juan camilo ospina quintero wrote:

> Hi 
> thanks for fast answers
> i already have sailfin installed and runnig, also freeswitch, both on the same machine.
> now my quiestion is how to redirect o forward the SIP flow from freeswitch to salifin, the softphones must register in freeswitch, but to make a call the invite should go through sailfin and get back to freeswitch, this is for implementing some services of VoIP, for a project i'm on.

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From yehavi.bourvine at gmail.com  Sun Jan 24 02:23:51 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 24 Jan 2010 12:23:51 +0200
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <AC76DB08-C278-4F59-90D0-2C3724CDB39A@freeswitch.org>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
	<d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
	<201001231355.31032.sos@sokhapkin.dyndns.org>
	<AC76DB08-C278-4F59-90D0-2C3724CDB39A@freeswitch.org>
Message-ID: <d361ca7b1001240223s6d88d86ardc05b9c573987559@mail.gmail.com>

About getting into my box: It is a production machine with quite tight
access control, so this is close to impossible...
I am trying to reproduce it on a backup system. If it works, then I might be
able to give access to it. I am working on it now...

                          __Yehavi:

2010/1/23 Brian West <brian at freeswitch.org>

> Best to find us on IRC when anthm is around and lets get into your box and
> fix this.
>
> /b
>
> On Jan 23, 2010, at 12:55 PM, Sergey Okhapkin wrote:
>
> > GOOD! You have a consistent way to reproduce the problem!
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mcampbellsmith at gmail.com  Sun Jan 24 02:47:51 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 24 Jan 2010 21:47:51 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
Message-ID: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>

Hi!

Is there any way that a custom NOTIFY message can be built and sent in
FS without cutting code?

I have a bunch of Linksys SPA's and I want to implement the
Resync_From_SIP option, which enables a resync to be triggered via a
SIP NOTIFY message. (ie Event: resync in the NOTIFY message)

Is this at all possible?

Thanks!



From mcampbellsmith at gmail.com  Sun Jan 24 03:28:57 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 24 Jan 2010 22:28:57 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
Message-ID: <33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>

Actually I just found http://wiki.freeswitch.org/wiki/Mod_event_socket

If I telnet to FS as described
http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do I
just need to enter somthing like:

sendevent NOTIFY
profile: internal
event-string: resync
user: 1000
host: 192.168.1.121
content-type: application/simple-message-summary

where 192.168.1.121 is the ip address of one of the Linksys devices?

I don't see any messages sent when I do this.  What am I doing wrong?

Thanks

On Sun, Jan 24, 2010 at 9:47 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Hi!
>
> Is there any way that a custom NOTIFY message can be built and sent in
> FS without cutting code?
>
> I have a bunch of Linksys SPA's and I want to implement the
> Resync_From_SIP option, which enables a resync to be triggered via a
> SIP NOTIFY message. (ie Event: resync in the NOTIFY message)
>
> Is this at all possible?
>
> Thanks!
>



From mcampbellsmith at gmail.com  Sun Jan 24 03:41:26 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 24 Jan 2010 22:41:26 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
Message-ID: <33c87fa31001240341x58f1c953n8a1e958a867591cd@mail.gmail.com>

Sorry for the spam ...

Playing around with this a bit more and I noticed that host should be
the ip address of FS, not the Linksys ATA.

How do I authorize the NOTIFY message?  I see FS tries to Authorize
but uses one of my external sip profiles in the authorization details,
instead of the extension details.

For example the following is sent:

   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bKevv1r7F0SNSUm
   Max-Forwards: 70
   From: <sip:1000 at 192.168.1.120>;tag=ZHS035c6yc4rD
   To: <sip:1000 at 192.168.1.120>
   Call-ID: 488d261f-837f-122d-7ba9-00e04c0312e9
   CSeq: 126048101 NOTIFY
   Contact: <sip:mod_sofia at 192.168.1.120:5060>
   Expires: 3590
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Event: resync
   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Subscription-State: terminated;reason=timeout
   Authorization: Digest username="XXXXXXXXXXXX",
realm="192.168.1.120", nonce="efe79af6",
cnonce="SJIYXoN/Ei2pewDgTAMS6Q", algorithm=MD5,
uri="sip:1000 at 192.168.1.121:5060",
response="9b27c57170fca740df4f538634e2e407", qop=auth, nc=00000001
   Content-Type: application/simple-message-summary
   Content-Length: 0

The username XXXXXXXXXXXX is one of my SIP profiles - I would have
expected FS to use the profile for extension 1000 here.  How can I do
that?

Thanks!

On Sun, Jan 24, 2010 at 10:28 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Actually I just found http://wiki.freeswitch.org/wiki/Mod_event_socket
>
> If I telnet to FS as described
> http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do I
> just need to enter somthing like:
>
> sendevent NOTIFY
> profile: internal
> event-string: resync
> user: 1000
> host: 192.168.1.121
> content-type: application/simple-message-summary
>
> where 192.168.1.121 is the ip address of one of the Linksys devices?
>
> I don't see any messages sent when I do this. ?What am I doing wrong?
>
> Thanks
>
> On Sun, Jan 24, 2010 at 9:47 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>> Hi!
>>
>> Is there any way that a custom NOTIFY message can be built and sent in
>> FS without cutting code?
>>
>> I have a bunch of Linksys SPA's and I want to implement the
>> Resync_From_SIP option, which enables a resync to be triggered via a
>> SIP NOTIFY message. (ie Event: resync in the NOTIFY message)
>>
>> Is this at all possible?
>>
>> Thanks!
>>
>



From tzury.by at reguluslabs.com  Sun Jan 24 04:25:14 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Sun, 24 Jan 2010 14:25:14 +0200
Subject: [Freeswitch-users] strange behavior of openzap/libpri (is that a
	bug?)
Message-ID: <10128ef11001240425x5481679ahcebb94efce745933@mail.gmail.com>

Hi All,

In my environment I have the following components:
* FreeSWITCH Version 1.0.4
* latest libpri
* latest openZAP

I have noticed that when originating calls to a specific number, the
system sometimes reports this as UNALLOCATED_NUMBER and sometimes not.
That is, I am calling a number which certainly exists, however, in
most of the attempts to cal this number I get the UNALLOCATED_NUMBER
error, while every 4th or 5th attempt the line get connected.

I paste-bin the log at http://pastebin.freeswitch.org/11928.
There is one with oz debug output at http://pastebin.freeswitch.org/11929

please advise,
Tzury



From yehavi.bourvine at gmail.com  Sun Jan 24 06:42:23 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 24 Jan 2010 16:42:23 +0200
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001240223s6d88d86ardc05b9c573987559@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
	<d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
	<201001231355.31032.sos@sokhapkin.dyndns.org>
	<AC76DB08-C278-4F59-90D0-2C3724CDB39A@freeswitch.org>
	<d361ca7b1001240223s6d88d86ardc05b9c573987559@mail.gmail.com>
Message-ID: <d361ca7b1001240642s680e770ao749dd9949cc14850@mail.gmail.com>

I've tried printing the value of retval and it is 1 (and not 6...).  BTW,
after adding the printf()'s I cannot examine anymore the value of retval in
gdb as it is optimized away. Maybe this is the reason for a value of 6 which
does not exist in the code that returns it.

nua_base_server_report() returns one as "terminated" variable is false.
Hope this helps.

                           Regards, __Yehavi:
2010/1/24 Yehavi Bourvine <yehavi.bourvine at gmail.com>

> About getting into my box: It is a production machine with quite tight
> access control, so this is close to impossible...
> I am trying to reproduce it on a backup system. If it works, then I might
> be able to give access to it. I am working on it now...
>
>                           __Yehavi:
>
>  2010/1/23 Brian West <brian at freeswitch.org>
>
>> Best to find us on IRC when anthm is around and lets get into your box and
>> fix this.
>>
>>
>> /b
>>
>> On Jan 23, 2010, at 12:55 PM, Sergey Okhapkin wrote:
>>
>> > GOOD! You have a consistent way to reproduce the problem!
>>
>>
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
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From Russell.Mosemann at cune.org  Sun Jan 24 07:13:30 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sun, 24 Jan 2010 09:13:30 -0600
Subject: [Freeswitch-users] strange behavior of openzap/libpri (is that
	abug?)
In-Reply-To: <10128ef11001240425x5481679ahcebb94efce745933@mail.gmail.com>
References: <10128ef11001240425x5481679ahcebb94efce745933@mail.gmail.com>
Message-ID: <DA783286028641449A0B00D75CA40183@cune.pri>

Tzury Bar Yochay wrote:

> In my environment I have the following components:
> * FreeSWITCH Version 1.0.4

That is an old version. Upgrade to the latest version and try it, again.

--
Russell Mosemann




From a.afzali2003 at gmail.com  Sun Jan 24 07:39:16 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sun, 24 Jan 2010 19:09:16 +0330
Subject: [Freeswitch-users] How to get chat message via event
Message-ID: <debff5a81001240739q3dbe4680uf15fa1ce4ee79bf7@mail.gmail.com>

Hi,

It seems that the chat messages don't fire via events by default and just
exchange between parties.
Is it true? Is it possible to enable those via events?

appreciate all,
-- afshin
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From gmaruzz at celliax.org  Sun Jan 24 07:48:32 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 24 Jan 2010 16:48:32 +0100
Subject: [Freeswitch-users] How to get chat message via event
In-Reply-To: <debff5a81001240739q3dbe4680uf15fa1ce4ee79bf7@mail.gmail.com>
References: <debff5a81001240739q3dbe4680uf15fa1ce4ee79bf7@mail.gmail.com>
Message-ID: <7b197bef1001240748r248f53b8i5794f7cbc1fedfb1@mail.gmail.com>

you subscribe to them as MESSAGE events

eg, from a telnet session:

telnet localhost 8021
auth ClueCon
events plain message

then those events will show up in your telnet session.
-gm

On Sun, Jan 24, 2010 at 4:39 PM, afshin afzali <a.afzali2003 at gmail.com> wrote:
> Hi,
>
> It seems that the chat messages don't fire via events by default and just
> exchange between parties.
> Is it true? Is it possible to enable those via events?
>
> appreciate all,
> -- afshin
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From mailinglist at fribert.dk  Sun Jan 24 08:39:47 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Sun, 24 Jan 2010 17:39:47 +0100
Subject: [Freeswitch-users] Svar: Re: How do I invite group to join			
	existing call?
In-Reply-To: <4B5A1D99020000E100000418@mail.fribert.dk>
References: <4B59A44E020000E100000413@mail.fribert.dk>
	<4B5A1D99020000E100000418@mail.fribert.dk>
Message-ID: <4B5C85E3020000E100000425@mail.fribert.dk>

Somebody help me understand this features expression, it sounds like it sort of does what I need, I could just dial a group to have the group ring, and invite a third party to the conversation.
 
In the pfsense package there is this example:
 
<!-- In call Transfer for phones without a transfer button -->
<extension name="dx">
      <condition field="destination_number" expression="^dx$">
       <action application="answer"/>
       <action application="read" data="11 11 'tone_stream://%(10000,0,350,440)' digits 5000 #"/>
       <action application="execute_extension" data="is_transfer XML features"/>
      </condition>
</extension>
But what does it mean, how do I use it during a call?


>>> 22-01-2010 kl. 21:50 skrev "mailinglist" <mailinglist at fribert.dk> i meddelelsen <4B5A1D99020000E100000418 at mail.fribert.dk>:

Hmm, I don't get it, it might not do the right thing.
 
The situation is that I receive a call from the outside, answers it on a phone, and then wants to ask a third (local) party to join the conversation.
 
I thought from the example that I should just press *3, and then the extension I want to invite, but nothing happens.
 
I haven't the faintest how I accomplish this :-o


>>> 22-01-2010 kl. 13:12 skrev "mailinglist" <mailinglist at fribert.dk> i meddelelsen <4B59A44E020000E100000413 at mail.fribert.dk>:

Hi Michael et al.

I found the 'attended xfer' example in the documentation, and it looks like rxactly to what I want to do.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer

The example 2:
Do I add the first part to my dialplan/default.xml, and the second part to my dialplan/features.xml

How does it affect if I try to do a normal local dial if it just reacts on the fact that three digits have been dialed (which I should change to 2, as I use 2 digit local numbers).

Best regards
Fribse

>>> Michael Collins 18-01-10 21:19 >>>


On Sun, Jan 17, 2010 at 10:50 PM, mailinglist <mailinglist at fribert.dk> wrote:


Hi All

I would like to be able to invite a group / global to join an existing call, but how do I accomplish this, can it be done?


Malfunction! Need Input!

Could you give us a few more details on what you're trying to accomplish? What's the big picture? I'm curious what problem you're trying to solve. I'm sure the gang here will have thoughts to pass along. :)

-MC

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From camilin2212 at hotmail.com  Sun Jan 24 08:48:43 2010
From: camilin2212 at hotmail.com (juan camilo ospina quintero)
Date: Sun, 24 Jan 2010 11:48:43 -0500
Subject: [Freeswitch-users] FreeSwitch Integration With Sailfin
In-Reply-To: <84AB66EA-956C-4B42-87C8-42660CBEDCE3@jerris.com>
References: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>, ,
	<CFA0D82A-33B3-4B50-B6AE-BAA61E0862A4@freeswitch.org>, ,
	<BAY143-W13760770FF3FCA224AACEA1610@phx.gbl>, ,
	<9dc4a1671001231501t65b1281epeeb7e7fa1b0bf5bf@mail.gmail.com>,
	<BAY143-W760D1216FA26D7EFD8E9DA1600@phx.gbl>,
	<84AB66EA-956C-4B42-87C8-42660CBEDCE3@jerris.com>
Message-ID: <BAY143-W27CC94D97F584D192C5680A1600@phx.gbl>


Thanks mike,

i already read about dialplan, and it seems that the bridge application
is the one i need, but now i want to know, how to modify dialplan to use bridge
and where to put this in the dialplan:

<action application="bridge" data="sofia/sip/9998881111 at sip.yourprovider.com"/>

this is an example, but i think thats what i need, if someone has work with bridging,
please help with this.
thanks


From: mike at jerris.com
Date: Sun, 24 Jan 2010 05:20:23 -0500
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSwitch Integration With Sailfin



This is a good place to start reading on how to configure dialplan:
http://wiki.freeswitch.org/wiki/Dialplanhttp://wiki.freeswitch.org/wiki/Getting_Started_Guide#dialplan
Mike
On Jan 23, 2010, at 11:48 PM, juan camilo ospina quintero wrote:Hi 
thanks for fast answers
i already have sailfin installed and runnig, also freeswitch, both on the same machine.
now my quiestion is how to redirect o forward the SIP flow from freeswitch to salifin, the softphones must register in freeswitch, but to make a call the invite should go through sailfin and get back to freeswitch, this is for implementing some services of VoIP, for a project i'm on.

 		 	   		  
_________________________________________________________________
Windows Live: Make it easier for your friends to see what you?re up to on Facebook.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009
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From mike at jerris.com  Sun Jan 24 09:44:26 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 12:44:26 -0500
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001240642s680e770ao749dd9949cc14850@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<AEA6B751-211E-4B18-B195-102A7E31FBAE@freeswitch.org>
	<d361ca7b1001231038g1ba79602q9b6ca801c1d8a752@mail.gmail.com>
	<201001231355.31032.sos@sokhapkin.dyndns.org>
	<AC76DB08-C278-4F59-90D0-2C3724CDB39A@freeswitch.org>
	<d361ca7b1001240223s6d88d86ardc05b9c573987559@mail.gmail.com>
	<d361ca7b1001240642s680e770ao749dd9949cc14850@mail.gmail.com>
Message-ID: <D56D65EB-B150-4A59-8B66-68BF4C58108B@jerris.com>

Yes, ssh is best.  Are you able to meet up with me on irc so we can  
discuss this issue?

On Jan 24, 2010, at 9:42 AM, Yehavi Bourvine  
<yehavi.bourvine at gmail.com> wrote:

> I've tried printing the value of retval and it is 1 (and not 6...).   
> BTW, after adding the printf()'s I cannot examine anymore the value  
> of retval in gdb as it is optimized away. Maybe this is the reason  
> for a value of 6 which does not exist in the code that returns it.
>
> nua_base_server_report() returns one as "terminated" variable is  
> false.
> Hope this helps.
>
>                            Regards, __Yehavi:
> 2010/1/24 Yehavi Bourvine <yehavi.bourvine at gmail.com>
> About getting into my box: It is a production machine with quite  
> tight access control, so this is close to impossible...
> I am trying to reproduce it on a backup system. If it works, then I  
> might be able to give access to it. I am working on it now...
>
>                           __Yehavi:
>
> 2010/1/23 Brian West <brian at freeswitch.org>
> Best to find us on IRC when anthm is around and lets get into your  
> box and fix this.
>
>
> /b
>
> On Jan 23, 2010, at 12:55 PM, Sergey Okhapkin wrote:
>
> > GOOD! You have a consistent way to reproduce the problem!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From brian at freeswitch.org  Sun Jan 24 09:46:13 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 24 Jan 2010 11:46:13 -0600
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
Message-ID: <99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>

or sofia profile xxx flush_inbound_reg callid reboot

callid you can get from sofia status profile xxx

/b

On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:

> Actually I just found http://wiki.freeswitch.org/wiki/Mod_event_socket
> 
> If I telnet to FS as described
> http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do I
> just need to enter somthing like:
> 
> sendevent NOTIFY
> profile: internal
> event-string: resync
> user: 1000
> host: 192.168.1.121
> content-type: application/simple-message-summary
> 
> where 192.168.1.121 is the ip address of one of the Linksys devices?
> 
> I don't see any messages sent when I do this.  What am I doing wrong?
> 
> Thanks

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From mike at jerris.com  Sun Jan 24 09:46:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 12:46:20 -0500
Subject: [Freeswitch-users] strange behavior of openzap/libpri (is that
	a bug?)
In-Reply-To: <10128ef11001240425x5481679ahcebb94efce745933@mail.gmail.com>
References: <10128ef11001240425x5481679ahcebb94efce745933@mail.gmail.com>
Message-ID: <6A2CAE5A-90FE-441C-B383-DD8ED9B71F25@jerris.com>

Does this same behavior happen with svn trunk?

On Jan 24, 2010, at 7:25 AM, Tzury Bar Yochay  
<tzury.by at reguluslabs.com> wrote:

> Hi All,
>
> In my environment I have the following components:
> * FreeSWITCH Version 1.0.4
> * latest libpri
> * latest openZAP
>
> I have noticed that when originating calls to a specific number, the
> system sometimes reports this as UNALLOCATED_NUMBER and sometimes not.
> That is, I am calling a number which certainly exists, however, in
> most of the attempts to cal this number I get the UNALLOCATED_NUMBER
> error, while every 4th or 5th attempt the line get connected.
>
> I paste-bin the log at http://pastebin.freeswitch.org/11928.
> There is one with oz debug output at http://pastebin.freeswitch.org/11929
>
> please advise,
> Tzury
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From scott.torr.fs at letterboxes.org  Sun Jan 24 09:47:14 2010
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Mon, 25 Jan 2010 04:47:14 +1100
Subject: [Freeswitch-users] Application="record_session" records 16000Hz in
	mono now?
Message-ID: <1264355234.24126.1356300893@webmail.messagingengine.com>

Hi,

When recording a session in the past the caller audio would be in the
right channel and the callee audio in the left of a stereo recording.

<action application="record_session" data="/root/recordings/test.wav" />


A 8000Hz recording has remained the same and is in stereo,

but a 16000Hz recording is now in mono?


Noticed change since FreeSWITCH Version 1.0.trunk (16195).


What do I need to do to get stereo recording back?

Why the change?



Regards,
Scott Torr










From adam.falcone at gmail.com  Sun Jan 24 09:50:59 2010
From: adam.falcone at gmail.com (AFalcon)
Date: Sun, 24 Jan 2010 09:50:59 -0800 (PST)
Subject: [Freeswitch-users] Freeswitch process hangs, losses connection.
Message-ID: <1264355459996-4449929.post@n2.nabble.com>


Hi I am new here but have successfully configured Freeswitch to run on Snow
Leopard. I have configured Snow Leopard so that the computer will never go
to sleep. My issue though is that Freeswitch losses it connection at some
point while running. When I call in I get a busy signal or a message saying
the phone number is not in operation. 

To get around this I wrote a launchd process that restarts Freeswitch and my
softphone every 4 hours. 

I don't like this and am wondering how to go about trouble shooting the
issue of Freeswitch losing connectivity after a period of a few hours. 

Any help would be appreciated. Thanks.
-- 
View this message in context: http://n2.nabble.com/Freeswitch-process-hangs-losses-connection-tp4449929p4449929.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From brian at freeswitch.org  Sun Jan 24 09:52:29 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 24 Jan 2010 11:52:29 -0600
Subject: [Freeswitch-users] Application="record_session" records 16000Hz
	in mono now?
In-Reply-To: <1264355234.24126.1356300893@webmail.messagingengine.com>
References: <1264355234.24126.1356300893@webmail.messagingengine.com>
Message-ID: <7A25419A-110C-4390-A2B4-9AF8D9700B22@freeswitch.org>

Set the variable record_stereo=true

/b

On Jan 24, 2010, at 11:47 AM, Scott Torr wrote:

> Hi,
> 
> When recording a session in the past the caller audio would be in the
> right channel and the callee audio in the left of a stereo recording.
> 
> <action application="record_session" data="/root/recordings/test.wav" />
> 
> 
> A 8000Hz recording has remained the same and is in stereo,
> 
> but a 16000Hz recording is now in mono?
> 
> 
> Noticed change since FreeSWITCH Version 1.0.trunk (16195).
> 
> 
> What do I need to do to get stereo recording back?
> 
> Why the change?
> 
> 
> 
> Regards,
> Scott Torr
> 
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From scott.torr.fs at letterboxes.org  Sun Jan 24 10:27:05 2010
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Mon, 25 Jan 2010 05:27:05 +1100
Subject: [Freeswitch-users] Application="record_session" records 16000Hz
 in mono now?
In-Reply-To: <7A25419A-110C-4390-A2B4-9AF8D9700B22@freeswitch.org>
References: <1264355234.24126.1356300893@webmail.messagingengine.com>
	<7A25419A-110C-4390-A2B4-9AF8D9700B22@freeswitch.org>
Message-ID: <1264357625.29510.1356306415@webmail.messagingengine.com>

Yeap,
That did the trick :)

<action application="set" data="RECORD_STEREO=true"/>
<action application="record_session" data="/root/recordings/test.wav" />

D'oh!
And right there in the wiki:(
It was an unexpected change to the previous default, thanks.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session


regards,
Scott Torr


On Sun, 24 Jan 2010 11:52 -0600, "Brian West" <brian at freeswitch.org>
wrote:
> Set the variable record_stereo=true
> 
> /b
> 
> On Jan 24, 2010, at 11:47 AM, Scott Torr wrote:
> 
> > Hi,
> > 
> > When recording a session in the past the caller audio would be in the
> > right channel and the callee audio in the left of a stereo recording.
> > 
> > <action application="record_session" data="/root/recordings/test.wav" />
> > 
> > 
> > A 8000Hz recording has remained the same and is in stereo,
> > 
> > but a 16000Hz recording is now in mono?
> > 
> > 
> > Noticed change since FreeSWITCH Version 1.0.trunk (16195).
> > 
> > 
> > What do I need to do to get stereo recording back?
> > 
> > Why the change?
> > 
> > 
> > 
> > Regards,
> > Scott Torr
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From tayeb.meftah at gmail.com  Sun Jan 24 10:31:38 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 24 Jan 2010 19:31:38 +0100
Subject: [Freeswitch-users] strange behavior of openzap/libpri (is that
 a bug?)
In-Reply-To: <10128ef11001240425x5481679ahcebb94efce745933@mail.gmail.com>
References: <10128ef11001240425x5481679ahcebb94efce745933@mail.gmail.com>
Message-ID: <4B5C920A.2040602@gmail.com>

hi,
upgrade to the latest 1.0.5 release at:
http://latest.freeswitch.org
thanks
24/01/2010 13:25, Tzury Bar Yochay a ?crit :
> Hi All,
>
> In my environment I have the following components:
> * FreeSWITCH Version 1.0.4
> * latest libpri
> * latest openZAP
>
> I have noticed that when originating calls to a specific number, the
> system sometimes reports this as UNALLOCATED_NUMBER and sometimes not.
> That is, I am calling a number which certainly exists, however, in
> most of the attempts to cal this number I get the UNALLOCATED_NUMBER
> error, while every 4th or 5th attempt the line get connected.
>
> I paste-bin the log at http://pastebin.freeswitch.org/11928.
> There is one with oz debug output at http://pastebin.freeswitch.org/11929
>
> please advise,
> Tzury
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    




From brian at freeswitch.org  Sun Jan 24 10:39:45 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 24 Jan 2010 12:39:45 -0600
Subject: [Freeswitch-users] Application="record_session" records 16000Hz
	in mono now?
In-Reply-To: <1264357625.29510.1356306415@webmail.messagingengine.com>
References: <1264355234.24126.1356300893@webmail.messagingengine.com>
	<7A25419A-110C-4390-A2B4-9AF8D9700B22@freeswitch.org>
	<1264357625.29510.1356306415@webmail.messagingengine.com>
Message-ID: <DA0DDF32-DBA1-4753-8357-AB86DA516C98@freeswitch.org>

It didn't change.  I suspect you removed the directory entry or its an inbound call to a user... in the defaults its always been in conf/directory/default.xml as a variable.

/b

On Jan 24, 2010, at 12:27 PM, Scott Torr wrote:

> It was an unexpected change to the previous default, thanks.




From mastermind202 at gmail.com  Sun Jan 24 10:55:36 2010
From: mastermind202 at gmail.com (mm_202)
Date: Sun, 24 Jan 2010 13:55:36 -0500
Subject: [Freeswitch-users] reloadxml/rescan profile and killgw
In-Reply-To: <3F378D25-85F8-48B9-B814-7F7979EFE840@jerris.com>
References: <4B589A6E.8010205@kinetix.gr>
	<3F378D25-85F8-48B9-B814-7F7979EFE840@jerris.com>
Message-ID: <63de75711001241055p129739behbefda343b2435a08@mail.gmail.com>

I agree that it should be a different command (breaking backwards
compatibility is bad).

I would suggest something like 'sofia profile [profilename] update' that
would first run rescan, then any gateways that were not present can be
considered stale and deleted.

Apostolos:  How are you C skills?  ;)

-- mm_202.

On Sun, Jan 24, 2010 at 4:50 AM, Michael Jerris <mike at jerris.com> wrote:

> A lot of things would be nice.  Patches for example.  I find them quite
> nice.  This should probably be a different command or argument.
>
> Mike
>
> On Jan 21, 2010, at 1:18 PM, Apostolos Pantsiopoulos wrote:
>
> > Wouldn't it be nice a "reloadxml"/"rescan profile" to delete all
> > nonexistent gateways (when they are absent in the xml config) without
> > having to issue a "sofia profile xxxxx killgw yyyyyy" command?
> >
> > I always seem to find forgotten gateway's in the profile because of
> > this... Any thoughts?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From testa at voicetechnology.com.br  Sun Jan 24 11:21:51 2010
From: testa at voicetechnology.com.br (Fernando Gregianin Testa)
Date: Sun, 24 Jan 2010 17:21:51 -0200
Subject: [Freeswitch-users] Lua: io.popen/read blocking in
	other	incoming calls
In-Reply-To: <11387916.58.1263943057280.JavaMail.root@zimbra>
References: <11387916.58.1263943057280.JavaMail.root@zimbra>
Message-ID: <6BA36994-7FA2-44F2-9592-78597BB9E9A4@voicetechnology.com.br>

You may consider use lua socket.http package as an alternative to popen+wget. Check:
 https://web.tecgraf.puc-rio.br/luasocket/ 
http://www.tecgraf.puc-rio.br/~diego/professional/luasocket/http.html 

Maybe you can be interested also in http://github.com/fertesta/restinlua 

Em 19/01/2010, ?s 21:17, Dan escreveu:

> I would, but I need to post a a wav file that gets recorded, I didn't see a way to supply the location of a file to use as the post data.  It looks like you have to url encode the data in the script and pass it all in the call. 
> 
> Thanks
> Dan-
> ----- Original Message -----
> From: "Rupa Schomaker" <rupa at rupa.com>
> To: "freeswitch-users" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, January 19, 2010 4:06:36 PM
> Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other        incoming calls
> 
> 
> 
> On Tue, Jan 19, 2010 at 3:03 PM, Dan <freeswitch-users at digitaldan.com> wrote:
> My lua script is calling wget through lua's io.popen to send and receive data from a web service. While the f:read to wget  is running, other incoming calls will block on the same io.popen call until the first call closes the pipe (with f:close()).  
> 
> You might want to look at the api that mod_curl exposes to do what you want.  No need to do an expensive system call just to call a webservice.
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Fernando Gregianin Testa
testa at voicetechnology.com.br
+55 11 35882166



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From mrene_lists at avgs.ca  Sun Jan 24 11:42:40 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sun, 24 Jan 2010 14:42:40 -0500
Subject: [Freeswitch-users] Accessing Sangoma card inside a openVZ
	container
In-Reply-To: <59ad9ca11001240218w3accbb12r462e5c2fa7511024@mail.gmail.com>
References: <59ad9ca11001240218w3accbb12r462e5c2fa7511024@mail.gmail.com>
Message-ID: <E93E560E-3EF8-475C-B7C7-72155E600355@avgs.ca>

As far as openzap is concerned you only have to expose the proper  
device files in /dev, and allow then in your VE's config. The driver  
always runs on the host's kernel.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 24-Jan-10, at 5:18 AM, Henry Huang wrote:

> I know there are a couple of OpenVZ expert here on the mailing list.
>
> Has anyone of you tried to run freeswitch in a container with access  
> to physical Sangoma card?
> The reason for this is that I would like to create an "all in one  
> box" for small to medium size company that takes care of the most  
> essential services. Things like an IP PBX would be a huge plus. I  
> have been playing with OpenVZ for a while now, and I know how to let  
> containers access devices in the /dev directory from the physical  
> box. But to be able to use something like sangoma card, I think  
> there are some things in the /proc directory need to be made  
> available to the containers. But I don't know how to do this yet.  
> Maybe some of you know..
>
> Thanks,
>
> -- 
> Henry Huang
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From a.afzali2003 at gmail.com  Sun Jan 24 12:59:10 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 25 Jan 2010 00:29:10 +0330
Subject: [Freeswitch-users] How to get chat message via event
In-Reply-To: <7b197bef1001240748r248f53b8i5794f7cbc1fedfb1@mail.gmail.com>
References: <debff5a81001240739q3dbe4680uf15fa1ce4ee79bf7@mail.gmail.com>
	<7b197bef1001240748r248f53b8i5794f7cbc1fedfb1@mail.gmail.com>
Message-ID: <debff5a81001241259q3bc2c82dl40c73ac81e43b40b@mail.gmail.com>

Hi,

As you say, I've already done and unfortunately did not get the message
events although other events are fired as expected :(

-- afshin

On Sun, Jan 24, 2010 at 7:18 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> you subscribe to them as MESSAGE events
>
> eg, from a telnet session:
>
> telnet localhost 8021
> auth ClueCon
> events plain message
>
> then those events will show up in your telnet session.
> -gm
>
> On Sun, Jan 24, 2010 at 4:39 PM, afshin afzali <a.afzali2003 at gmail.com>
> wrote:
> > Hi,
> >
> > It seems that the chat messages don't fire via events by default and just
> > exchange between parties.
> > Is it true? Is it possible to enable those via events?
> >
> > appreciate all,
> > -- afshin
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at celliax.org  Sun Jan 24 13:05:54 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 24 Jan 2010 22:05:54 +0100
Subject: [Freeswitch-users] How to get chat message via event
In-Reply-To: <debff5a81001241259q3bc2c82dl40c73ac81e43b40b@mail.gmail.com>
References: <debff5a81001240739q3dbe4680uf15fa1ce4ee79bf7@mail.gmail.com> 
	<7b197bef1001240748r248f53b8i5794f7cbc1fedfb1@mail.gmail.com> 
	<debff5a81001241259q3bc2c82dl40c73ac81e43b40b@mail.gmail.com>
Message-ID: <7b197bef1001241305o3ee9bc9cq7decdc412f90575c@mail.gmail.com>

Which events you don't get? From which channel in which circumstances?
(I mean what you do and what do you expect?)

-giovanni

On Sun, Jan 24, 2010 at 9:59 PM, afshin afzali <a.afzali2003 at gmail.com> wrote:
> Hi,
>
> As you say, I've already done and unfortunately did not get the message
> events although other events are fired as expected :(
>
> -- afshin
>
> On Sun, Jan 24, 2010 at 7:18 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> you subscribe to them as MESSAGE events
>>
>> eg, from a telnet session:
>>
>> telnet localhost 8021
>> auth ClueCon
>> events plain message
>>
>> then those events will show up in your telnet session.
>> -gm
>>
>> On Sun, Jan 24, 2010 at 4:39 PM, afshin afzali <a.afzali2003 at gmail.com>
>> wrote:
>> > Hi,
>> >
>> > It seems that the chat messages don't fire via events by default and
>> > just
>> > exchange between parties.
>> > Is it true? Is it possible to enable those via events?
>> >
>> > appreciate all,
>> > -- afshin
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From tjardick at vanderkraan.net  Sun Jan 24 12:51:17 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Sun, 24 Jan 2010 21:51:17 +0100
Subject: [Freeswitch-users] Compile error sofia on Mac OS X
Message-ID: <F026A1E8-211C-444E-B6FA-AAFCDBE806C5@vanderkraan.net>

Hi,

I'm trying to compile freeswitch on my MacBook pro to have a local dev  
instance, but i run in to the following compile error during the make:

Compiling mod_sofia.c ...
cc1: warnings being treated as errors
mod_sofia.c: In function 'sofia_receive_message':
mod_sofia.c:1446: warning: 'from_host' may be used uninitialized in  
this function
mod_sofia.c:1446: warning: 'from_user' may be used uninitialized in  
this function
make[5]: *** [mod_sofia_la-mod_sofia.lo] Error 1
make[4]: *** [all] Error 2
make[3]: *** [mod_sofia-all] Error 1
make[2]: *** [all-recursive] Error 1


It's an MBP running Leopard version 10.5.8

Any help would be appreciated.

Kind regards,

Tjardick
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From brian at freeswitch.org  Sun Jan 24 13:15:15 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 24 Jan 2010 15:15:15 -0600
Subject: [Freeswitch-users] Compile error sofia on Mac OS X
In-Reply-To: <F026A1E8-211C-444E-B6FA-AAFCDBE806C5@vanderkraan.net>
References: <F026A1E8-211C-444E-B6FA-AAFCDBE806C5@vanderkraan.net>
Message-ID: <9AEDA6E3-E875-4E63-A4A5-9457317C1D1A@freeswitch.org>

Update this has been fixed now.

Thanks,
Brian

On Jan 24, 2010, at 2:51 PM, Tjardick van der Kraan wrote:

> Hi,
> 
> I'm trying to compile freeswitch on my MacBook pro to have a local dev instance, but i run in to the following compile error during the make:
> 
> Compiling mod_sofia.c ...
> cc1: warnings being treated as errors
> mod_sofia.c: In function 'sofia_receive_message':
> mod_sofia.c:1446: warning: 'from_host' may be used uninitialized in this function
> mod_sofia.c:1446: warning: 'from_user' may be used uninitialized in this function
> make[5]: *** [mod_sofia_la-mod_sofia.lo] Error 1
> make[4]: *** [all] Error 2
> make[3]: *** [mod_sofia-all] Error 1
> make[2]: *** [all-recursive] Error 1
> 
> 
> It's an MBP running Leopard version 10.5.8
> 
> Any help would be appreciated.
> 
> Kind regards,
> 
> Tjardick
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jmesquita at freeswitch.org  Sun Jan 24 15:47:16 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 24 Jan 2010 21:47:16 -0200
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <13529f9d1001222200p44afd96ema21729d0038e89ba@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
	<1264092620.14614.73.camel@luna.tc.commsmundi.com>
	<13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
	<ebbc2d551001211721n51cb5c54m996e0d494842c383@mail.gmail.com>
	<13529f9d1001211806o35fc3435j93b71bbb2ede3028@mail.gmail.com>
	<13529f9d1001222200p44afd96ema21729d0038e89ba@mail.gmail.com>
Message-ID: <ebbc2d551001241547j7e11d4e4se093ce5a60e008b5@mail.gmail.com>

What user_name? I don't understand that statement.

I think that you can always use
http://wiki.freeswitch.org/wiki/Channel_Variables#api_hangup_hook

Regards,
Jo?o Mesquita
FSComm Developer


On Sat, Jan 23, 2010 at 4:00 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:

> Hi Jo?o, do you know how to sign the agent off automatically when either
> party hangs up the call?
>
> Here's how I originate the call to the agent and sign him up in ACD1:
>
> originate skypiax/ANY/jingwei.yang 6*1
>
> However, I found the user_name property is empty. May I know how it is set?
>
>
> Thanks and best regards,
> -Jingwei
>
> On Fri, Jan 22, 2010 at 10:06 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>
>> Hi Jo?o, thanks for the reply. I'll try it out.
>>
>> Regards,
>> -Jingwei
>>
>> 2010/1/22 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> Jingwei, check my contrib dir. I think it may help you with one FIFO since
>>> we are able there to sign in and sign off dynamic agents as well as
>>> customize what we do when the FIFO makes a call to them.
>>>
>>> Regards,
>>> Jo?o Mesquita
>>> FSComm Developer
>>>
>>>
>>> On Thu, Jan 21, 2010 at 10:05 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>>>
>>>> Thanks for the reply. All the agents are dynamic and I can't predefine
>>>> them in the config file.
>>>>
>>>> Regards,
>>>> -Jingwei
>>>>
>>>>
>>>> On Fri, Jan 22, 2010 at 12:50 AM, Fran?ois Delawarde <
>>>> fdelawarde at wirelessmundi.com> wrote:
>>>>
>>>>> Why do you need 2 fifos? You could have callback agents connected to
>>>>> the
>>>>> fifo and send incoming calls there, mod_fifo will do the rest.
>>>>>
>>>>> To configure agents for callback:
>>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>>>>>
>>>>> To place a call into a fifo:
>>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue
>>>>>
>>>>> Fran?ois.
>>>>>
>>>>> On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
>>>>> > Sorry about the confusion, I'm just trying to think over all the
>>>>> > abnormal situations before the implementation and hope the flow has
>>>>> no
>>>>> > design flaws.
>>>>> >
>>>>> > Client A is parked in Queue 1, multiple consumers will be ringed to
>>>>> > answer him. And once the first one is connected, all the rest will
>>>>> > hang up. This is the targeted function.
>>>>> >
>>>>> > To achieve this, I'm considering to originate a call to each consumer
>>>>> > and put the calls in Queue 2. Then intercept client A to the first
>>>>> > element of Queue 2.
>>>>> >
>>>>> > I'm not sure if it's the usual or the best way. If you feel not,
>>>>> > please don't hesitate to correct me. Any thoughts are warmly
>>>>> > appreciated.
>>>>> >
>>>>> > On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com>
>>>>> wrote:
>>>>> >         what no errors mean? how do you originate calls to consumers?
>>>>> >         I don't
>>>>> >         understand your scenario.
>>>>> >
>>>>> >         2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
>>>>> >
>>>>> >         > Hi All,
>>>>> >         >
>>>>> >         > Please advise whether the following flow makes sense.
>>>>> >         >
>>>>> >         > 1. Client A calls in and parked in Queue 1
>>>>> >         > 2. Originate calls to several consumers simultaneously and
>>>>> >         park them in
>>>>> >         > Queue 2
>>>>> >         > 3. Intercept A's call to the first consumer of Queue 2
>>>>> >         >
>>>>> >         > Basically I want A's call picked up by the first among many
>>>>> >         consumers with
>>>>> >         > no errors. Please let me know whether I'm on the right
>>>>> >         track.
>>>>> >         >
>>>>> >         > Thanks and best regards,
>>>>> >         > -Jingwei
>>>>> >         >
>>>>> >
>>>>> >
>>>>> >         _______________________________________________
>>>>> >         FreeSWITCH-users mailing list
>>>>> >         FreeSWITCH-users at lists.freeswitch.org
>>>>> >
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >         UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >         http://www.freeswitch.org
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Sun Jan 24 15:58:35 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 25 Jan 2010 10:58:35 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
Message-ID: <33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>

Thanks Brian.. this still does not work.  Maybe I need to open a Jira?
 Notice the username in the authorization field.  It should be 1000.

Cheers
Mark

freeswitch at internal> sofia status profile internal user 1000 at 192.168.1.120

Registrations:
=================================================================================================
Call-ID:        bd783b73-66877627 at 192.168.1.121
User:           1000 at 192.168.1.120
Contact:        1000 <sip:1000 at 192.168.1.121:5060>
Agent:          Linksys/PAP2T-5.1.6(LS)
Status:         Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
Host:           freeswitch
IP:             192.168.1.121
Port:           5060
Auth-User:      1000
Auth-Realm:     192.168.1.120
MWI-Account:    1000 at 192.168.1.120

=================================================================================================

freeswitch at internal> sofia profile internal flush_inbound_reg
bd783b73-66877627 at 192.168.1.121 reboot
+OK rebooting all registrations matching specified call_id

freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060 at
23:55:49.012627:
   ------------------------------------------------------------------------
   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
   Max-Forwards: 70
   From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
   To: <sip:1000 at 192.168.1.120>
   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
   CSeq: 126070338 NOTIFY
   Contact: <sip:mod_sofia at 192.168.1.120:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Event: reboot_now
   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Subscription-State: terminated;reason=timeout
   Content-Type: application/simple-message-summary
   Content-Length: 0

   ------------------------------------------------------------------------
recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
   From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
   CSeq: 126070338 NOTIFY
   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
   Server: Linksys/PAP2T-5.1.6(LS)
   WWW-Authenticate: Digest realm="192.168.1.120", nonce="8e54805b",
qop="auth", algorithm=md5
   Content-Length: 0

   ------------------------------------------------------------------------
send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
   ------------------------------------------------------------------------
   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
   Max-Forwards: 70
   From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
   To: <sip:1000 at 192.168.1.120>
   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
   CSeq: 126070339 NOTIFY
   Contact: <sip:mod_sofia at 192.168.1.120:5060>
   Expires: 3590
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Event: reboot_now
   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Subscription-State: terminated;reason=timeout
   Authorization: Digest username="1115633124", realm="192.168.1.120",
nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
uri="sip:1000 at 192.168.1.121:5060",
response="747b4d04544c84535dbbd987f2999ca7", qop=auth, nc=00000001
   Content-Type: application/simple-message-summary
   Content-Length: 0

   ------------------------------------------------------------------------
recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
   From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
   CSeq: 126070339 NOTIFY
   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
   Server: Linksys/PAP2T-5.1.6(LS)
   WWW-Authenticate: Digest realm="192.168.1.120", nonce="5339c7ba",
qop="auth", algorithm=md5
   Content-Length: 0

   ------------------------------------------------------------------------

On Mon, Jan 25, 2010 at 4:46 AM, Brian West <brian at freeswitch.org> wrote:
> or sofia profile xxx flush_inbound_reg callid reboot
> callid you can get from sofia status profile xxx
> /b
> On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
>
> Actually I just found?http://wiki.freeswitch.org/wiki/Mod_event_socket
>
> If I telnet to FS as described
> http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do I
> just need to enter somthing like:
>
> sendevent NOTIFY
> profile: internal
> event-string: resync
> user: 1000
> host: 192.168.1.121
> content-type: application/simple-message-summary
>
> where 192.168.1.121 is the ip address of one of the Linksys devices?
>
> I don't see any messages sent when I do this. ?What am I doing wrong?
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From oseslija at gmail.com  Sun Jan 24 16:15:13 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 25 Jan 2010 01:15:13 +0100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
Message-ID: <4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>

You  should not authenticate those NOTIFYs (this will work only with SPA9000
afaik). The option to change for this is in EXT tabs:

Auth Resync-Reboot: No

Also, FSs code will do a reboot of a phone, not resync (it sends reboot_now
event). For that to work a patch is required.

I've just tried to reboot my 942 (rev 16506) and it definitely works.

Regards,
Ognjen


On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Thanks Brian.. this still does not work.  Maybe I need to open a Jira?
>  Notice the username in the authorization field.  It should be 1000.
>
> Cheers
> Mark
>
> freeswitch at internal> sofia status profile internal user 1000 at 192.168.1.120
>
> Registrations:
>
> =================================================================================================
> Call-ID:        bd783b73-66877627 at 192.168.1.121
> User:           1000 at 192.168.1.120
> Contact:        1000 <sip:1000 at 192.168.1.121:5060>
> Agent:          Linksys/PAP2T-5.1.6(LS)
> Status:         Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
> Host:           freeswitch
> IP:             192.168.1.121
> Port:           5060
> Auth-User:      1000
> Auth-Realm:     192.168.1.120
> MWI-Account:    1000 at 192.168.1.120
>
>
> =================================================================================================
>
> freeswitch at internal> sofia profile internal flush_inbound_reg
> bd783b73-66877627 at 192.168.1.121 reboot
> +OK rebooting all registrations matching specified call_id
>
> freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060 at
> 23:55:49.012627:
>   ------------------------------------------------------------------------
>    NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
>   Max-Forwards: 70
>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
>    To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
>    Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>   CSeq: 126070338 NOTIFY
>    Contact: <sip:mod_sofia at 192.168.1.120:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>    Event: reboot_now
>    Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>   Subscription-State: terminated;reason=timeout
>    Content-Type: application/simple-message-summary
>   Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
>   ------------------------------------------------------------------------
>   SIP/2.0 401 Unauthorized
>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>   CSeq: 126070338 NOTIFY
>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
>   Server: Linksys/PAP2T-5.1.6(LS)
>   WWW-Authenticate: Digest realm="192.168.1.120", nonce="8e54805b",
> qop="auth", algorithm=md5
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
>   ------------------------------------------------------------------------
>    NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
>   Max-Forwards: 70
>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
>    To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
>    Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>   CSeq: 126070339 NOTIFY
>    Contact: <sip:mod_sofia at 192.168.1.120:5060>
>   Expires: 3590
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>    Event: reboot_now
>    Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>   Subscription-State: terminated;reason=timeout
>    Authorization: Digest username="1115633124", realm="192.168.1.120",
> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
> uri="sip:1000 at 192.168.1.121:5060",
> response="747b4d04544c84535dbbd987f2999ca7", qop=auth, nc=00000001
>    Content-Type: application/simple-message-summary
>   Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
>   ------------------------------------------------------------------------
>   SIP/2.0 401 Unauthorized
>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>   CSeq: 126070339 NOTIFY
>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
>   Server: Linksys/PAP2T-5.1.6(LS)
>   WWW-Authenticate: Digest realm="192.168.1.120", nonce="5339c7ba",
> qop="auth", algorithm=md5
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
>
> On Mon, Jan 25, 2010 at 4:46 AM, Brian West <brian at freeswitch.org> wrote:
> > or sofia profile xxx flush_inbound_reg callid reboot
> > callid you can get from sofia status profile xxx
> > /b
> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
> >
> > Actually I just found http://wiki.freeswitch.org/wiki/Mod_event_socket
> >
> > If I telnet to FS as described
> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do I
> > just need to enter somthing like:
> >
> > sendevent NOTIFY
> > profile: internal
> > event-string: resync
> > user: 1000
> > host: 192.168.1.121
> > content-type: application/simple-message-summary
> >
> > where 192.168.1.121 is the ip address of one of the Linksys devices?
> >
> > I don't see any messages sent when I do this.  What am I doing wrong?
> >
> > Thanks
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mcampbellsmith at gmail.com  Sun Jan 24 16:29:46 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 25 Jan 2010 11:29:46 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
Message-ID: <33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>

Hi Ognjen,

Thanks for the tip on the resync under the EXT tab.  It now works
using mod_event_socket and the following:

sendevent NOTIFY
profile: internal
event-string: resync
user: 1000
host: 192.168.1.121
content-type: application/simple-message-summary

However, if AUTH is required, why does FS send the wrong information to the SPA?

On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija <oseslija at gmail.com> wrote:
> You? should not authenticate those NOTIFYs (this will work only with SPA9000
> afaik). The option to change for this is in EXT tabs:
>
> Auth Resync-Reboot: No
>
> Also, FSs code will do a reboot of a phone, not resync (it sends reboot_now
> event). For that to work a patch is required.
>
> I've just tried to reboot my 942 (rev 16506) and it definitely works.
>
> Regards,
> Ognjen
>
>
> On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>>
>> Thanks Brian.. this still does not work. ?Maybe I need to open a Jira?
>> ?Notice the username in the authorization field. ?It should be 1000.
>>
>> Cheers
>> Mark
>>
>> freeswitch at internal> sofia status profile internal user 1000 at 192.168.1.120
>>
>> Registrations:
>>
>> =================================================================================================
>> Call-ID: ? ? ? ?bd783b73-66877627 at 192.168.1.121
>> User: ? ? ? ? ? 1000 at 192.168.1.120
>> Contact: ? ? ? ?1000 <sip:1000 at 192.168.1.121:5060>
>> Agent: ? ? ? ? ?Linksys/PAP2T-5.1.6(LS)
>> Status: ? ? ? ? Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
>> Host: ? ? ? ? ? freeswitch
>> IP: ? ? ? ? ? ? 192.168.1.121
>> Port: ? ? ? ? ? 5060
>> Auth-User: ? ? ?1000
>> Auth-Realm: ? ? 192.168.1.120
>> MWI-Account: ? ?1000 at 192.168.1.120
>>
>>
>> =================================================================================================
>>
>> freeswitch at internal> sofia profile internal flush_inbound_reg
>> bd783b73-66877627 at 192.168.1.121 reboot
>> +OK rebooting all registrations matching specified call_id
>>
>> freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060 at
>> 23:55:49.012627:
>> ? ------------------------------------------------------------------------
>> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> ? Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
>> ? Max-Forwards: 70
>> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> ? To: <sip:1000 at 192.168.1.120>
>> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> ? CSeq: 126070338 NOTIFY
>> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> ? Supported: timer, precondition, path, replaces
>> ? Event: reboot_now
>> ? Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer
>> ? Subscription-State: terminated;reason=timeout
>> ? Content-Type: application/simple-message-summary
>> ? Content-Length: 0
>>
>> ? ------------------------------------------------------------------------
>> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
>> ? ------------------------------------------------------------------------
>> ? SIP/2.0 401 Unauthorized
>> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> ? CSeq: 126070338 NOTIFY
>> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
>> ? Server: Linksys/PAP2T-5.1.6(LS)
>> ? WWW-Authenticate: Digest realm="192.168.1.120", nonce="8e54805b",
>> qop="auth", algorithm=md5
>> ? Content-Length: 0
>>
>> ? ------------------------------------------------------------------------
>> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
>> ? ------------------------------------------------------------------------
>> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> ? Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
>> ? Max-Forwards: 70
>> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> ? To: <sip:1000 at 192.168.1.120>
>> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> ? CSeq: 126070339 NOTIFY
>> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> ? Expires: 3590
>> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> ? Supported: timer, precondition, path, replaces
>> ? Event: reboot_now
>> ? Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer
>> ? Subscription-State: terminated;reason=timeout
>> ? Authorization: Digest username="1115633124", realm="192.168.1.120",
>> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
>> uri="sip:1000 at 192.168.1.121:5060",
>> response="747b4d04544c84535dbbd987f2999ca7", qop=auth, nc=00000001
>> ? Content-Type: application/simple-message-summary
>> ? Content-Length: 0
>>
>> ? ------------------------------------------------------------------------
>> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
>> ? ------------------------------------------------------------------------
>> ? SIP/2.0 401 Unauthorized
>> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> ? CSeq: 126070339 NOTIFY
>> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
>> ? Server: Linksys/PAP2T-5.1.6(LS)
>> ? WWW-Authenticate: Digest realm="192.168.1.120", nonce="5339c7ba",
>> qop="auth", algorithm=md5
>> ? Content-Length: 0
>>
>> ? ------------------------------------------------------------------------
>>
>> On Mon, Jan 25, 2010 at 4:46 AM, Brian West <brian at freeswitch.org> wrote:
>> > or sofia profile xxx flush_inbound_reg callid reboot
>> > callid you can get from sofia status profile xxx
>> > /b
>> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
>> >
>> > Actually I just found?http://wiki.freeswitch.org/wiki/Mod_event_socket
>> >
>> > If I telnet to FS as described
>> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do I
>> > just need to enter somthing like:
>> >
>> > sendevent NOTIFY
>> > profile: internal
>> > event-string: resync
>> > user: 1000
>> > host: 192.168.1.121
>> > content-type: application/simple-message-summary
>> >
>> > where 192.168.1.121 is the ip address of one of the Linksys devices?
>> >
>> > I don't see any messages sent when I do this. ?What am I doing wrong?
>> >
>> > Thanks
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Sun Jan 24 16:39:49 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 24 Jan 2010 18:39:49 -0600
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
Message-ID: <A5B60A4C-C827-4025-823B-1007E178DB2E@freeswitch.org>

Its not WRONG its just we don't know how to answer the challenge.

/b

On Jan 24, 2010, at 6:29 PM, Mark Campbell-Smith wrote:

> However, if AUTH is required, why does FS send the wrong information to the SPA?




From oseslija at gmail.com  Sun Jan 24 16:50:06 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 25 Jan 2010 01:50:06 +0100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
Message-ID: <4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>

The phone is asking FS to authenticate prior then accepting a NOTIFY from
it.
The authentication of notify's from spa endpoints work (afaik) only with
Linksys SPA9000 PBX , and FS doesn't have the code for that for obvious
reasons.
If you have SPA9000 maybe you can collect SIP traces.

Ognjen





On Mon, Jan 25, 2010 at 1:29 AM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Hi Ognjen,
>
> Thanks for the tip on the resync under the EXT tab.  It now works
> using mod_event_socket and the following:
>
> sendevent NOTIFY
> profile: internal
> event-string: resync
> user: 1000
> host: 192.168.1.121
> content-type: application/simple-message-summary
>
> However, if AUTH is required, why does FS send the wrong information to the
> SPA?
>
> On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija <oseslija at gmail.com>
> wrote:
> > You  should not authenticate those NOTIFYs (this will work only with
> SPA9000
> > afaik). The option to change for this is in EXT tabs:
> >
> > Auth Resync-Reboot: No
> >
> > Also, FSs code will do a reboot of a phone, not resync (it sends
> reboot_now
> > event). For that to work a patch is required.
> >
> > I've just tried to reboot my 942 (rev 16506) and it definitely works.
> >
> > Regards,
> > Ognjen
> >
> >
> > On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
> > <mcampbellsmith at gmail.com> wrote:
> >>
> >> Thanks Brian.. this still does not work.  Maybe I need to open a Jira?
> >>  Notice the username in the authorization field.  It should be 1000.
> >>
> >> Cheers
> >> Mark
> >>
> >> freeswitch at internal> sofia status profile internal user
> 1000 at 192.168.1.120
> >>
> >> Registrations:
> >>
> >>
> =================================================================================================
> >> Call-ID:        bd783b73-66877627 at 192.168.1.121
> >> User:           1000 at 192.168.1.120
> >> Contact:        1000 <sip:1000 at 192.168.1.121:5060>
> >> Agent:          Linksys/PAP2T-5.1.6(LS)
> >> Status:         Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
> >> Host:           freeswitch
> >> IP:             192.168.1.121
> >> Port:           5060
> >> Auth-User:      1000
> >> Auth-Realm:     192.168.1.120
> >> MWI-Account:    1000 at 192.168.1.120
> >>
> >>
> >>
> =================================================================================================
> >>
> >> freeswitch at internal> sofia profile internal flush_inbound_reg
> >> bd783b73-66877627 at 192.168.1.121 reboot
> >> +OK rebooting all registrations matching specified call_id
> >>
> >> freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060 at
> >> 23:55:49.012627:
> >>
> ------------------------------------------------------------------------
> >>   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
> >>   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
> >>   Max-Forwards: 70
> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >>   CSeq: 126070338 NOTIFY
> >>   Contact: <sip:mod_sofia at 192.168.1.120:5060>
> >>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
> >>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >>   Supported: timer, precondition, path, replaces
> >>   Event: reboot_now
> >>   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >> include-session-description, presence.winfo, message-summary, refer
> >>   Subscription-State: terminated;reason=timeout
> >>   Content-Type: application/simple-message-summary
> >>   Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
> >>
> ------------------------------------------------------------------------
> >>   SIP/2.0 401 Unauthorized
> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >>   CSeq: 126070338 NOTIFY
> >>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
> >>   Server: Linksys/PAP2T-5.1.6(LS)
> >>   WWW-Authenticate: Digest realm="192.168.1.120", nonce="8e54805b",
> >> qop="auth", algorithm=md5
> >>   Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
> >>
> ------------------------------------------------------------------------
> >>   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
> >>   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
> >>   Max-Forwards: 70
> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >>   CSeq: 126070339 NOTIFY
> >>   Contact: <sip:mod_sofia at 192.168.1.120:5060>
> >>   Expires: 3590
> >>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
> >>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >>   Supported: timer, precondition, path, replaces
> >>   Event: reboot_now
> >>   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >> include-session-description, presence.winfo, message-summary, refer
> >>   Subscription-State: terminated;reason=timeout
> >>   Authorization: Digest username="1115633124", realm="192.168.1.120",
> >> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
> >> uri="sip:1000 at 192.168.1.121:5060",
> >> response="747b4d04544c84535dbbd987f2999ca7", qop=auth, nc=00000001
> >>   Content-Type: application/simple-message-summary
> >>   Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
> >>
> ------------------------------------------------------------------------
> >>   SIP/2.0 401 Unauthorized
> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >>   CSeq: 126070339 NOTIFY
> >>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
> >>   Server: Linksys/PAP2T-5.1.6(LS)
> >>   WWW-Authenticate: Digest realm="192.168.1.120", nonce="5339c7ba",
> >> qop="auth", algorithm=md5
> >>   Content-Length: 0
> >>
> >>
> ------------------------------------------------------------------------
> >>
> >> On Mon, Jan 25, 2010 at 4:46 AM, Brian West <brian at freeswitch.org>
> wrote:
> >> > or sofia profile xxx flush_inbound_reg callid reboot
> >> > callid you can get from sofia status profile xxx
> >> > /b
> >> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
> >> >
> >> > Actually I just found
> http://wiki.freeswitch.org/wiki/Mod_event_socket
> >> >
> >> > If I telnet to FS as described
> >> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do I
> >> > just need to enter somthing like:
> >> >
> >> > sendevent NOTIFY
> >> > profile: internal
> >> > event-string: resync
> >> > user: 1000
> >> > host: 192.168.1.121
> >> > content-type: application/simple-message-summary
> >> >
> >> > where 192.168.1.121 is the ip address of one of the Linksys devices?
> >> >
> >> > I don't see any messages sent when I do this.  What am I doing wrong?
> >> >
> >> > Thanks
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Sun Jan 24 20:04:41 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 23:04:41 -0500
Subject: [Freeswitch-users] E1 hangups
In-Reply-To: <4B59EC64.3080907@comgate.cz>
References: <4B59EC64.3080907@comgate.cz>
Message-ID: <A13D010D-25B0-4D05-BC14-5388F4AB4234@jerris.com>

We just fixed this issue friday and have been testing it more over the weekend.  The new drivers that should go with the latest openzap code have not been released yet but should be early next week.

Mike

On Jan 22, 2010, at 1:20 PM, Michal Zub?? wrote:

> Hi.
> 
> I'm trying to correct this behaviour, but can't figure out, where is the
> problem. Here's the scenario:
> 
> - we're trying to execute simple dialplan
>    * answer
>    * play sound
>    * wait for 3 seconds
>    * hangup
> 
> - for SIP caller it works as expected
> - problems are when, we try to call into it from E1 line
> - for E1 we're using sangoma winpipe & openzap
> - dialplan in freeswitch console is done in a moment ending with hangup
> - on the E1 line I hear nothing and after 2 seconds it disconnects
> - similar problem when there's only bridge to another number (E1)
> - it rings (on the destination phone) for a short moment (0.5-1s), but
> then hangs up spontaneously
> 
> Thanks for any clues.
> 
> mZubac
> <problem_background.zip>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Sun Jan 24 20:17:27 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 24 Jan 2010 23:17:27 -0500
Subject: [Freeswitch-users] FreeSwitch Integration With Sailfin
In-Reply-To: <BAY143-W27CC94D97F584D192C5680A1600@phx.gbl>
References: <BAY143-W13FCC92F6FD8F62B8993B8A1610@phx.gbl>, ,
	<CFA0D82A-33B3-4B50-B6AE-BAA61E0862A4@freeswitch.org>, ,
	<BAY143-W13760770FF3FCA224AACEA1610@phx.gbl>, ,
	<9dc4a1671001231501t65b1281epeeb7e7fa1b0bf5bf@mail.gmail.com>,
	<BAY143-W760D1216FA26D7EFD8E9DA1600@phx.gbl>,
	<84AB66EA-956C-4B42-87C8-42660CBEDCE3@jerris.com>
	<BAY143-W27CC94D97F584D192C5680A1600@phx.gbl>
Message-ID: <F8985501-2A06-4596-913C-081E899B396A@jerris.com>

There is more information on how to create a proper bridge string for sofia here:

http://wiki.freeswitch.org/wiki/Sofia#Syntax

You should put it in the dialplan in the context that you need to call it from.

Mie


On Jan 24, 2010, at 11:48 AM, juan camilo ospina quintero wrote:

> Thanks mike,
> 
> i already read about dialplan, and it seems that the bridge application
> is the one i need, but now i want to know, how to modify dialplan to use bridge
> and where to put this in the dialplan:
> 
> <action application="bridge" data="sofia/sip/9998881111 at sip.yourprovider.com"/>
> 
> this is an example, but i think thats what i need, if someone has work with bridging,
> please help with this.
> thanks
> 
> 
> From: mike at jerris.com
> Date: Sun, 24 Jan 2010 05:20:23 -0500
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FreeSwitch Integration With Sailfin
> 
> This is a good place to start reading on how to configure dialplan:
> 
> http://wiki.freeswitch.org/wiki/Dialplan
> http://wiki.freeswitch.org/wiki/Getting_Started_Guide#dialplan
> 

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From thangappan143 at gmail.com  Sun Jan 24 20:25:59 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Mon, 25 Jan 2010 09:55:59 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
	<7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>
	<7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>
Message-ID: <7aa29e791001242025y2adc91aes622bd22ea8d5ae1c@mail.gmail.com>

Here I mentioned the link which has the details of /etc/wanpipe/smg_pri.conf

http://www.pastebin.org/81895

On Sat, Jan 23, 2010 at 10:02 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Updated the latest version of freeswitch ( 1.0.5-20100121-0400) and run the
> wanrouter then freeswitch. While executing the freeswtich it said the
> following error.
>
> [ERR] zap_io.c:2562 Error loading/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
>
> [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
> file: No such file or directory]
> [ERR] zap_io.c:2722 can't find 'sangoma_boost
>
>
>
>
> Searched about this in freeswitch mailing list and found one post was there
> regarding the same problem. Finally found the problem. I missed to install
> the SCTP packages. Installed it and compiled the freeswitch again now the
> inbound call was landed on freeswitch.
>
> But I am unable to make a outbound call. When I was trying the following
> was get.
>
> freeswitch at internal> originate openzap/smg_prid/a/9940464753 at g1openzap/smg_prid/a/9940464753 at g1
> -ERR NORMAL_CIRCUIT_CONGESTION
>
> 2010-01-23 10:00:45.688854 [WARNING] ozmod_sangoma_boost.c:348 TX EVENT:
> CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[9940464753]
> Ci=[0000000000] Rdnis=[]
> freeswitch at internal> 2010-01-23 10:00:46.709355 [WARNING]
> ozmod_sangoma_boost.c:1373 RX EVENT (N): CALL_START_NACK:(82) [w256g256]
> Rc=0 CSid=2 Seq=2
> 2010-01-23 10:00:46.709355 [WARNING] sangoma_boost_client.c:220 TX EVENT
> (N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=2 Seq=3
> 2010-01-23 10:00:46.709355 [ERR] mod_openzap.c:1162 No channels available
> 2010-01-23 10:00:46.709355 [ERR] switch_ivr_originate.c:2411 Cannot create
> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
> 2010-01-23 10:00:46.709355 [DEBUG] switch_ivr_originate.c:3211 Originate
> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>
> Please help me...........
>
>
>
> On Fri, Jan 22, 2010 at 10:15 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>> The following link have the openzap.conf,openzap.conf.xml ,smg_prid.conf ,
>> debug log of mod_openzap , oz list and oz dump 1 output.
>>
>> http://pastebin.org/80095
>>
>>
>>
>> On Thu, Jan 21, 2010 at 10:34 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>
>>> OpenZap is loading the ss7 signalling type. As per your concern openzap
>>> does not know the details of the signalling then how it is loading the
>>> ss7_boost libraries?
>>>
>>> FreeSWITCH log:
>>> -----------------------------
>>> 2010-01-21 10:10:46.707844 [INFO] zap_io.c:2374 Configured 30 channel(s)
>>> 2010-01-21 10:10:46.708600 [INFO] zap_io.c:2468 Loading SIG from
>>> /usr/local/freeswitch/mod/ozmod_ss7_boost.so
>>> 2010-01-21 10:10:46.709031 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost'
>>> 2010-01-21 10:10:46.709466 [DEBUG] ss7_boost_client.c:124 Creating L=
>>> 127.0.0.65:53000 R=127.0.0.66:53000
>>> 2010-01-21 10:10:46.709834 [DEBUG] ss7_boost_client.c:124 Creating L=
>>> 127.0.0.65:53001 R=127.0.0.66:53001
>>> 2010-01-21 10:10:46.710424 [WARNING] ss7_boost_client.c:244 TX EVENT (P):
>>> SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>>>
>>> The signalling type might be anything but when I used the oz list command
>>> it showed the span details. But I am unable to make a inbound and outbound
>>> call.
>>>
>>> Outbound call result:
>>> ============
>>> >  originate openzap/smg_prid/a/9940464753 openzap/smg_prid/a/9843171457
>>> -ERR NORMAL_CIRCUIT_CONGESTION
>>>
>>> 2010-01-21 10:26:14.304816 [CRIT] ozmod_ss7_boost.c:244 SPAN is not
>>> online.
>>> freeswitch at internal> 2010-01-21 10:26:14.304816 [ERR] mod_openzap.c:1043
>>> No channels available
>>> 2010-01-21 10:26:14.304816 [ERR] switch_ivr_originate.c:1510 Cannot
>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>> 2010-01-21 10:26:14.304816 [DEBUG] switch_ivr_originate.c:2138 Originate
>>> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>>>
>>> Inbound call result:
>>> -----------------------------
>>>
>>>      I made incoming call for the dial plan which is specified in the
>>> earlier post at that time it said the number is busy. We did the packet
>>> capture using the following command.
>>>
>>> wanpipemon -i w1g1 -pcap -pcap file isdn.pcap -port ISDN -full -systime
>>> -c trd
>>>
>>> Here I attached the pcap file for that.
>>>
>>>
>>> Where I did mistake or Did I miss any thing to do?
>>> Please help me.......
>>>
>>>
>>>
>>> On Wed, Jan 20, 2010 at 7:40 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>>
>>>>
>>>>       I noticed the 'oz list' output  in that span type is 'ss7
>>>> (boost)'. How can I change this to isdn?
>>>>
>>>>
>>>>
>>>> On Wed, Jan 20, 2010 at 12:43 PM, Thangappan.M <thangappan143 at gmail.com
>>>> > wrote:
>>>>
>>>>> I found the error in it. The file name is used as openzap.conf.xml (
>>>>> smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
>>>>> PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
>>>>> openzap.conf.wiki.xml  file.
>>>>>
>>>>> Now the another problem is raised here.
>>>>> When I was using oz list command , the details of the smg_prid shown.
>>>>> When I was using 'oz dump smg_prid' command it shows all the channels'
>>>>> details. But all the channels' states are DOWN. why? How can I make it the
>>>>> states to UP?
>>>>>
>>>>> When I was making the call , the number is busy message was get. The
>>>>> call was not  at all landed to the freeswitch.
>>>>>
>>>>> Dial plan Example:
>>>>> -------------------------------
>>>>>  <extension name="PRI_Testing">
>>>>>         <condition field="destination_number" expression="^39114600$">
>>>>>                 <action application="playback"
>>>>> data="ivr-welcome_to_freeswitch"/>
>>>>>         </condition>
>>>>>   </extension>
>>>>>
>>>>> Please help me...........
>>>>>
>>>>> *Output Reference:*
>>>>> http://pastebin.org/79074
>>>>>
>>>>>
>>>>> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <
>>>>> thangappan143 at gmail.com> wrote:
>>>>>
>>>>>> Dear all,
>>>>>>
>>>>>>          I have successfully configured wanpipe with freeswitch. When
>>>>>> I was the running wancfg_fs script the following files openzap.conf ,
>>>>>> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
>>>>>> are created.
>>>>>>
>>>>>>          I started the wanrouter command then executed the freeswitch.
>>>>>>          When I was executing freeswitch mod_openzap.c said the error
>>>>>> as  "Error  for finding the span id. name:PRI_1".
>>>>>>          But in the openzap.conf and openzap.conf.xml files the span
>>>>>> name is smg_prid.
>>>>>>
>>>>>>          Why the freeswitch is referring the span name as PRI_1 ?
>>>>>>          Whether this has to configured in anywhere?
>>>>>>
>>>>>>          In the freeswitch CLI using oz command I tried to dump the
>>>>>> PRI_1 span id but it said te error  as "PRI_1 is not found".  When I was
>>>>>> trying the command 'oz dump smg_prid'  all the channel states and details
>>>>>> shown.
>>>>>>
>>>>>>         It seems that smg_prid span configured in openzap perfectly
>>>>>> (Its my assumption). Then Why freeswitch is referring the span name as
>>>>>> PRI_1.
>>>>>>
>>>>>> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>>>>>>
>>>>>> Could anyone please help me?
>>>>>>
>>>>>> REFERENCE:
>>>>>>
>>>>>> openzap.conf
>>>>>> [span wanpipe smg_prid]
>>>>>> name => smg_prid
>>>>>> trunk_type =>e1
>>>>>> b-channel => 1:1-15
>>>>>> b-channel => 1:17-31
>>>>>>
>>>>>>
>>>>>> openzap.conf.xml
>>>>>> <configuration name="openzap.conf" description="OpenZAP
>>>>>> Configuration">
>>>>>>   <settings>
>>>>>>     <param name="debug" value="0"/>
>>>>>>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>>>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>>>>     <!--<param name="enable-analog-option" value="3-way"/>-->
>>>>>>   </settings>
>>>>>>   <boost_spans>
>>>>>>         <span id="smg_prid">
>>>>>>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>>>         <param name="dialplan" value="XML"/>
>>>>>>         <param name="context" value="default"/>
>>>>>>          <!-- regex to stop dialing when it matches -->
>>>>>>         <!--<param name="dial-regex" value="5555"/>-->
>>>>>>         <!-- regex to stop dialing when it does not match -->
>>>>>>         <!--<param name="fail-dial-regex" value="^5"/>-->
>>>>>>         </span>
>>>>>>  </boost_spans>
>>>>>>  <analog_spans>
>>>>>>  </analog_spans>
>>>>>> </configuration>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Regards,
>>>>>> Thangappan.M
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Regards,
>>>>> Thangappan.M
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Thangappan.M
>>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Thangappan.M
>>>
>>
>>
>>
>> --
>> Regards,
>> Thangappan.M
>>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From thangappan143 at gmail.com  Sun Jan 24 21:50:02 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Mon, 25 Jan 2010 11:20:02 +0530
Subject: [Freeswitch-users] Need Help to setup freeswitch with sangoma
	card
In-Reply-To: <7aa29e791001242025y2adc91aes622bd22ea8d5ae1c@mail.gmail.com>
References: <7aa29e791001192155l3c1f06e6w69769a69c9b761a1@mail.gmail.com>
	<7aa29e791001192313p73b20c8fk7e9b78e577c305a9@mail.gmail.com>
	<7aa29e791001200610m5a7b64cdx356e527ea7285b23@mail.gmail.com>
	<7aa29e791001202104n276ca48bo56d55ca82f3aac0f@mail.gmail.com>
	<7aa29e791001212045u20d85213wa0c8523761fee826@mail.gmail.com>
	<7aa29e791001222032t6f381e34x9f3c5eb099af9ba8@mail.gmail.com>
	<7aa29e791001242025y2adc91aes622bd22ea8d5ae1c@mail.gmail.com>
Message-ID: <7aa29e791001242150o3a582e2end69f1f5e9f1aef1c@mail.gmail.com>

The following link have the openzap.conf,openzap.conf.xml
,smg_pri.conf, output of oz list
and oz dump.

http://www.pastebin.org/81929



On Mon, Jan 25, 2010 at 9:55 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Here I mentioned the link which has the details of
> /etc/wanpipe/smg_pri.conf
> http://www.pastebin.org/81895
>
>
> On Sat, Jan 23, 2010 at 10:02 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>> Updated the latest version of freeswitch ( 1.0.5-20100121-0400) and run
>> the wanrouter then freeswitch. While executing the freeswtich it said the
>> following error.
>>
>> [ERR] zap_io.c:2562 Error loading/usr/local/freeswitch/mod/ozmod_sangoma_boost.so
>>
>> [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object
>> file: No such file or directory]
>> [ERR] zap_io.c:2722 can't find 'sangoma_boost
>>
>>
>>
>>
>> Searched about this in freeswitch mailing list and found one post was
>> there regarding the same problem. Finally found the problem. I missed to
>> install the SCTP packages. Installed it and compiled the freeswitch again
>> now the inbound call was landed on freeswitch.
>>
>> But I am unable to make a outbound call. When I was trying the following
>> was get.
>>
>> freeswitch at internal> originate openzap/smg_prid/a/9940464753 at g1openzap/smg_prid/a/9940464753 at g1
>> -ERR NORMAL_CIRCUIT_CONGESTION
>>
>> 2010-01-23 10:00:45.688854 [WARNING] ozmod_sangoma_boost.c:348 TX EVENT:
>> CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[9940464753]
>> Ci=[0000000000] Rdnis=[]
>> freeswitch at internal> 2010-01-23 10:00:46.709355 [WARNING]
>> ozmod_sangoma_boost.c:1373 RX EVENT (N): CALL_START_NACK:(82) [w256g256]
>> Rc=0 CSid=2 Seq=2
>> 2010-01-23 10:00:46.709355 [WARNING] sangoma_boost_client.c:220 TX EVENT
>> (N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=2 Seq=3
>> 2010-01-23 10:00:46.709355 [ERR] mod_openzap.c:1162 No channels available
>> 2010-01-23 10:00:46.709355 [ERR] switch_ivr_originate.c:2411 Cannot create
>> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>> 2010-01-23 10:00:46.709355 [DEBUG] switch_ivr_originate.c:3211 Originate
>> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>>
>> Please help me...........
>>
>>
>>
>> On Fri, Jan 22, 2010 at 10:15 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>
>>> The following link have the openzap.conf,openzap.conf.xml ,smg_prid.conf
>>> , debug log of mod_openzap , oz list and oz dump 1 output.
>>>
>>> http://pastebin.org/80095
>>>
>>>
>>>
>>> On Thu, Jan 21, 2010 at 10:34 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>>
>>>> OpenZap is loading the ss7 signalling type. As per your concern openzap
>>>> does not know the details of the signalling then how it is loading the
>>>> ss7_boost libraries?
>>>>
>>>> FreeSWITCH log:
>>>> -----------------------------
>>>> 2010-01-21 10:10:46.707844 [INFO] zap_io.c:2374 Configured 30 channel(s)
>>>> 2010-01-21 10:10:46.708600 [INFO] zap_io.c:2468 Loading SIG from
>>>> /usr/local/freeswitch/mod/ozmod_ss7_boost.so
>>>> 2010-01-21 10:10:46.709031 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost'
>>>> 2010-01-21 10:10:46.709466 [DEBUG] ss7_boost_client.c:124 Creating L=
>>>> 127.0.0.65:53000 R=127.0.0.66:53000
>>>> 2010-01-21 10:10:46.709834 [DEBUG] ss7_boost_client.c:124 Creating L=
>>>> 127.0.0.65:53001 R=127.0.0.66:53001
>>>> 2010-01-21 10:10:46.710424 [WARNING] ss7_boost_client.c:244 TX EVENT
>>>> (P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>>>>
>>>> The signalling type might be anything but when I used the oz list
>>>> command it showed the span details. But I am unable to make a inbound and
>>>> outbound call.
>>>>
>>>> Outbound call result:
>>>> ============
>>>> >  originate openzap/smg_prid/a/9940464753 openzap/smg_prid/a/9843171457
>>>> -ERR NORMAL_CIRCUIT_CONGESTION
>>>>
>>>> 2010-01-21 10:26:14.304816 [CRIT] ozmod_ss7_boost.c:244 SPAN is not
>>>> online.
>>>> freeswitch at internal> 2010-01-21 10:26:14.304816 [ERR]
>>>> mod_openzap.c:1043 No channels available
>>>> 2010-01-21 10:26:14.304816 [ERR] switch_ivr_originate.c:1510 Cannot
>>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>>> 2010-01-21 10:26:14.304816 [DEBUG] switch_ivr_originate.c:2138 Originate
>>>> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>>>>
>>>> Inbound call result:
>>>> -----------------------------
>>>>
>>>>      I made incoming call for the dial plan which is specified in the
>>>> earlier post at that time it said the number is busy. We did the packet
>>>> capture using the following command.
>>>>
>>>> wanpipemon -i w1g1 -pcap -pcap file isdn.pcap -port ISDN -full -systime
>>>> -c trd
>>>>
>>>> Here I attached the pcap file for that.
>>>>
>>>>
>>>> Where I did mistake or Did I miss any thing to do?
>>>> Please help me.......
>>>>
>>>>
>>>>
>>>> On Wed, Jan 20, 2010 at 7:40 PM, Thangappan.M <thangappan143 at gmail.com>wrote:
>>>>
>>>>>
>>>>>       I noticed the 'oz list' output  in that span type is 'ss7
>>>>> (boost)'. How can I change this to isdn?
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Jan 20, 2010 at 12:43 PM, Thangappan.M <
>>>>> thangappan143 at gmail.com> wrote:
>>>>>
>>>>>> I found the error in it. The file name is used as openzap.conf.xml (
>>>>>> smg_prid is specified here) and another file name as openzap.conf.wiki.xml (
>>>>>> PRI_1 span is specified here ). FreeSWITCH referred the PRI_1 span from
>>>>>> openzap.conf.wiki.xml  file.
>>>>>>
>>>>>> Now the another problem is raised here.
>>>>>> When I was using oz list command , the details of the smg_prid shown.
>>>>>> When I was using 'oz dump smg_prid' command it shows all the channels'
>>>>>> details. But all the channels' states are DOWN. why? How can I make it the
>>>>>> states to UP?
>>>>>>
>>>>>> When I was making the call , the number is busy message was get. The
>>>>>> call was not  at all landed to the freeswitch.
>>>>>>
>>>>>> Dial plan Example:
>>>>>> -------------------------------
>>>>>>  <extension name="PRI_Testing">
>>>>>>         <condition field="destination_number" expression="^39114600$">
>>>>>>                 <action application="playback"
>>>>>> data="ivr-welcome_to_freeswitch"/>
>>>>>>         </condition>
>>>>>>   </extension>
>>>>>>
>>>>>> Please help me...........
>>>>>>
>>>>>> *Output Reference:*
>>>>>> http://pastebin.org/79074
>>>>>>
>>>>>>
>>>>>> On Wed, Jan 20, 2010 at 11:25 AM, Thangappan.M <
>>>>>> thangappan143 at gmail.com> wrote:
>>>>>>
>>>>>>> Dear all,
>>>>>>>
>>>>>>>          I have successfully configured wanpipe with freeswitch. When
>>>>>>> I was the running wancfg_fs script the following files openzap.conf ,
>>>>>>> autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf
>>>>>>> are created.
>>>>>>>
>>>>>>>          I started the wanrouter command then executed the
>>>>>>> freeswitch.
>>>>>>>          When I was executing freeswitch mod_openzap.c said the error
>>>>>>> as  "Error  for finding the span id. name:PRI_1".
>>>>>>>          But in the openzap.conf and openzap.conf.xml files the span
>>>>>>> name is smg_prid.
>>>>>>>
>>>>>>>          Why the freeswitch is referring the span name as PRI_1 ?
>>>>>>>          Whether this has to configured in anywhere?
>>>>>>>
>>>>>>>          In the freeswitch CLI using oz command I tried to dump the
>>>>>>> PRI_1 span id but it said te error  as "PRI_1 is not found".  When I was
>>>>>>> trying the command 'oz dump smg_prid'  all the channel states and details
>>>>>>> shown.
>>>>>>>
>>>>>>>         It seems that smg_prid span configured in openzap perfectly
>>>>>>> (Its my assumption). Then Why freeswitch is referring the span name as
>>>>>>> PRI_1.
>>>>>>>
>>>>>>> DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO?
>>>>>>>
>>>>>>> Could anyone please help me?
>>>>>>>
>>>>>>> REFERENCE:
>>>>>>>
>>>>>>> openzap.conf
>>>>>>> [span wanpipe smg_prid]
>>>>>>> name => smg_prid
>>>>>>> trunk_type =>e1
>>>>>>> b-channel => 1:1-15
>>>>>>> b-channel => 1:17-31
>>>>>>>
>>>>>>>
>>>>>>> openzap.conf.xml
>>>>>>> <configuration name="openzap.conf" description="OpenZAP
>>>>>>> Configuration">
>>>>>>>   <settings>
>>>>>>>     <param name="debug" value="0"/>
>>>>>>>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>>>>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>>>>>     <!--<param name="enable-analog-option" value="3-way"/>-->
>>>>>>>   </settings>
>>>>>>>   <boost_spans>
>>>>>>>         <span id="smg_prid">
>>>>>>>         <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>>>>>         <param name="dialplan" value="XML"/>
>>>>>>>         <param name="context" value="default"/>
>>>>>>>          <!-- regex to stop dialing when it matches -->
>>>>>>>         <!--<param name="dial-regex" value="5555"/>-->
>>>>>>>         <!-- regex to stop dialing when it does not match -->
>>>>>>>         <!--<param name="fail-dial-regex" value="^5"/>-->
>>>>>>>         </span>
>>>>>>>  </boost_spans>
>>>>>>>  <analog_spans>
>>>>>>>  </analog_spans>
>>>>>>> </configuration>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Regards,
>>>>>>> Thangappan.M
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Regards,
>>>>>> Thangappan.M
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Regards,
>>>>> Thangappan.M
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Thangappan.M
>>>>
>>>
>>>
>>>
>>> --
>>> Regards,
>>> Thangappan.M
>>>
>>
>>
>>
>> --
>> Regards,
>> Thangappan.M
>>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From elihayun at gmail.com  Sun Jan 24 22:17:33 2010
From: elihayun at gmail.com (Eli Hayun)
Date: Mon, 25 Jan 2010 08:17:33 +0200
Subject: [Freeswitch-users] Module multicast fail
In-Reply-To: <2A8BE7A9-EC50-47CD-9FE8-172BF0F97DB6@freeswitch.org>
References: <4B56F0F9.9090808@savion.huji.ac.il>
	<2A8BE7A9-EC50-47CD-9FE8-172BF0F97DB6@freeswitch.org>
Message-ID: <4B5D377D.8050408@savion.huji.ac.il>

On 01/20/2010 05:18 PM, Brian West wrote:
> Please visit http://latest.freeswitch.org  and update to the latest ;)  Its the best you can get to date!  All the preX releases are gone from the download site.
>
> /b
>
> On Jan 20, 2010, at 6:03 AM, Eli Hayun wrote:
>
>   
>> (ver 1.0.5pre9)
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
Hi
I run the latest version and still getting an error

2010-01-25 07:42:53.449509 [ERR] mod_event_multicast.c:410 Multicast Error^M
2010-01-25 07:42:53.449761 [CRIT] switch_loadable_module.c:872 Error
Loading module /freeswitch-1.0.5_20100120-0400/mod/mod_event_multicast.so^M
**Module load routine returned an error**^M
2010-01-25 07:42:53.450424 [CONSOLE] switch_loadable_module.c:890
Successfully Loaded [mod_event_socket]^M

Eli





From brian at freeswitch.org  Sun Jan 24 22:34:55 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 25 Jan 2010 00:34:55 -0600
Subject: [Freeswitch-users] Module multicast fail
In-Reply-To: <4B5D377D.8050408@savion.huji.ac.il>
References: <4B56F0F9.9090808@savion.huji.ac.il>
	<2A8BE7A9-EC50-47CD-9FE8-172BF0F97DB6@freeswitch.org>
	<4B5D377D.8050408@savion.huji.ac.il>
Message-ID: <6F3B0496-C94E-4515-ADE3-89D12EAD6379@freeswitch.org>

+OK log level  [7]
freeswitch at internal> load mod_event_multicast
+OK

2010-01-25 00:34:37.581408 [CONSOLE] switch_loadable_module.c:890 Successfully Loaded [mod_event_multicast]
freeswitch at internal> 2010-01-25 00:34:37.581408 [NOTICE] switch_loadable_module.c:271 Adding API Function 'multicast_peers'

What distro are you on?

/b

On Jan 25, 2010, at 12:17 AM, Eli Hayun wrote:

> mod_event_multicast




From a.afzali2003 at gmail.com  Sun Jan 24 22:36:21 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 25 Jan 2010 10:06:21 +0330
Subject: [Freeswitch-users] How to get chat message via event
In-Reply-To: <7b197bef1001241305o3ee9bc9cq7decdc412f90575c@mail.gmail.com>
References: <debff5a81001240739q3dbe4680uf15fa1ce4ee79bf7@mail.gmail.com>
	<7b197bef1001240748r248f53b8i5794f7cbc1fedfb1@mail.gmail.com>
	<debff5a81001241259q3bc2c82dl40c73ac81e43b40b@mail.gmail.com>
	<7b197bef1001241305o3ee9bc9cq7decdc412f90575c@mail.gmail.com>
Message-ID: <debff5a81001242236u4fb153f3j6cfee6550429573f@mail.gmail.com>

I have two X-Lite logged in FreeSWITCH that could call , monitor presence
data and also send text messages to each others. So as I'm able to have
every step of those operations via events mechanism, I expect to see MESSAGE
event in case of sending text messages between them.

appreciate your help
-- afshin

On Mon, Jan 25, 2010 at 12:35 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org>wrote:

> Which events you don't get? From which channel in which circumstances?
> (I mean what you do and what do you expect?)
>
> -giovanni
>
> On Sun, Jan 24, 2010 at 9:59 PM, afshin afzali <a.afzali2003 at gmail.com>
> wrote:
> > Hi,
> >
> > As you say, I've already done and unfortunately did not get the message
> > events although other events are fired as expected :(
> >
> > -- afshin
> >
> > On Sun, Jan 24, 2010 at 7:18 PM, Giovanni Maruzzelli <
> gmaruzz at celliax.org>
> > wrote:
> >>
> >> you subscribe to them as MESSAGE events
> >>
> >> eg, from a telnet session:
> >>
> >> telnet localhost 8021
> >> auth ClueCon
> >> events plain message
> >>
> >> then those events will show up in your telnet session.
> >> -gm
> >>
> >> On Sun, Jan 24, 2010 at 4:39 PM, afshin afzali <a.afzali2003 at gmail.com>
> >> wrote:
> >> > Hi,
> >> >
> >> > It seems that the chat messages don't fire via events by default and
> >> > just
> >> > exchange between parties.
> >> > Is it true? Is it possible to enable those via events?
> >> >
> >> > appreciate all,
> >> > -- afshin
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From abhishek.dixit at nagarro.com  Sun Jan 24 20:47:29 2010
From: abhishek.dixit at nagarro.com (Abhishek Dixit)
Date: Mon, 25 Jan 2010 10:17:29 +0530
Subject: [Freeswitch-users] Configuring duration of conference
Message-ID: <000001ca9d79$81aa0f50$84fe2df0$@dixit@nagarro.com>

Hi,

 

I am working on FreeSWITCH Version 1.0.4 (14460).

I am trying to set up a conference room which will allow conferencing for
limited duration. This means that when parties connect to this conference
room they can remain connected only till a specified duration. After this
duration the conference should end with announcement.

I have read wiki for mod_conference and for dialplan and studied IVR
conferencing example using javascript as well.

But I am unable to find a way to specify duration for the conference using
dialplan and javascript application.

 

I am planning now to use mod_event_socket interface to end conference after
a specified duration.

 

Please tell me if there can be a way to configure duration of conference
using dialplan configurations or javascript application.

 

Thanks

 

Abhishek Dixit

 

 

 

 

 

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From jingwei.yang at gmail.com  Sun Jan 24 23:54:28 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 25 Jan 2010 15:54:28 +0800
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <ebbc2d551001241547j7e11d4e4se093ce5a60e008b5@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
	<1264092620.14614.73.camel@luna.tc.commsmundi.com>
	<13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
	<ebbc2d551001211721n51cb5c54m996e0d494842c383@mail.gmail.com>
	<13529f9d1001211806o35fc3435j93b71bbb2ede3028@mail.gmail.com>
	<13529f9d1001222200p44afd96ema21729d0038e89ba@mail.gmail.com>
	<ebbc2d551001241547j7e11d4e4se093ce5a60e008b5@mail.gmail.com>
Message-ID: <13529f9d1001242354q18e8b05fla3cc37de466f0767@mail.gmail.com>

Thanks Jo?o. In 01_fifo.xml, the agent uses the action below to sign up.

<action application="set" data="result=${fifo_member(add ACD$1
{fifo_member_wait=nowait}loopback/1${user_name}/default/XML )"/>

The variable ${user_name} was empty when I tried both skypiax and dingaling.


2010/1/25 Jo?o Mesquita <jmesquita at freeswitch.org>

> What user_name? I don't understand that statement.
>
> I think that you can always use
> http://wiki.freeswitch.org/wiki/Channel_Variables#api_hangup_hook
>
> Regards,
> Jo?o Mesquita
> FSComm Developer
>
>
> On Sat, Jan 23, 2010 at 4:00 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>
>> Hi Jo?o, do you know how to sign the agent off automatically when either
>> party hangs up the call?
>>
>> Here's how I originate the call to the agent and sign him up in ACD1:
>>
>> originate skypiax/ANY/jingwei.yang 6*1
>>
>> However, I found the user_name property is empty. May I know how it is
>> set?
>>
>>
>> Thanks and best regards,
>> -Jingwei
>>
>> On Fri, Jan 22, 2010 at 10:06 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>>
>>> Hi Jo?o, thanks for the reply. I'll try it out.
>>>
>>> Regards,
>>> -Jingwei
>>>
>>> 2010/1/22 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>
>>> Jingwei, check my contrib dir. I think it may help you with one FIFO
>>>> since we are able there to sign in and sign off dynamic agents as well as
>>>> customize what we do when the FIFO makes a call to them.
>>>>
>>>> Regards,
>>>> Jo?o Mesquita
>>>> FSComm Developer
>>>>
>>>>
>>>> On Thu, Jan 21, 2010 at 10:05 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>>>>
>>>>> Thanks for the reply. All the agents are dynamic and I can't predefine
>>>>> them in the config file.
>>>>>
>>>>> Regards,
>>>>> -Jingwei
>>>>>
>>>>>
>>>>> On Fri, Jan 22, 2010 at 12:50 AM, Fran?ois Delawarde <
>>>>> fdelawarde at wirelessmundi.com> wrote:
>>>>>
>>>>>> Why do you need 2 fifos? You could have callback agents connected to
>>>>>> the
>>>>>> fifo and send incoming calls there, mod_fifo will do the rest.
>>>>>>
>>>>>> To configure agents for callback:
>>>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>>>>>>
>>>>>> To place a call into a fifo:
>>>>>>
>>>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue
>>>>>>
>>>>>> Fran?ois.
>>>>>>
>>>>>> On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
>>>>>> > Sorry about the confusion, I'm just trying to think over all the
>>>>>> > abnormal situations before the implementation and hope the flow has
>>>>>> no
>>>>>> > design flaws.
>>>>>> >
>>>>>> > Client A is parked in Queue 1, multiple consumers will be ringed to
>>>>>> > answer him. And once the first one is connected, all the rest will
>>>>>> > hang up. This is the targeted function.
>>>>>> >
>>>>>> > To achieve this, I'm considering to originate a call to each
>>>>>> consumer
>>>>>> > and put the calls in Queue 2. Then intercept client A to the first
>>>>>> > element of Queue 2.
>>>>>> >
>>>>>> > I'm not sure if it's the usual or the best way. If you feel not,
>>>>>> > please don't hesitate to correct me. Any thoughts are warmly
>>>>>> > appreciated.
>>>>>> >
>>>>>> > On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com>
>>>>>> wrote:
>>>>>> >         what no errors mean? how do you originate calls to
>>>>>> consumers?
>>>>>> >         I don't
>>>>>> >         understand your scenario.
>>>>>> >
>>>>>> >         2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
>>>>>> >
>>>>>> >         > Hi All,
>>>>>> >         >
>>>>>> >         > Please advise whether the following flow makes sense.
>>>>>> >         >
>>>>>> >         > 1. Client A calls in and parked in Queue 1
>>>>>> >         > 2. Originate calls to several consumers simultaneously and
>>>>>> >         park them in
>>>>>> >         > Queue 2
>>>>>> >         > 3. Intercept A's call to the first consumer of Queue 2
>>>>>> >         >
>>>>>> >         > Basically I want A's call picked up by the first among
>>>>>> many
>>>>>> >         consumers with
>>>>>> >         > no errors. Please let me know whether I'm on the right
>>>>>> >         track.
>>>>>> >         >
>>>>>> >         > Thanks and best regards,
>>>>>> >         > -Jingwei
>>>>>> >         >
>>>>>> >
>>>>>> >
>>>>>> >         _______________________________________________
>>>>>> >         FreeSWITCH-users mailing list
>>>>>> >         FreeSWITCH-users at lists.freeswitch.org
>>>>>> >
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >         UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >         http://www.freeswitch.org
>>>>>> >
>>>>>> > _______________________________________________
>>>>>> > FreeSWITCH-users mailing list
>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> > http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lei.tlfly at gmail.com  Mon Jan 25 00:16:10 2010
From: lei.tlfly at gmail.com (Lei Tang)
Date: Mon, 25 Jan 2010 16:16:10 +0800
Subject: [Freeswitch-users] Question about bridge_answer_timeout variable
Message-ID: <50c41b4e1001250016v6633eaf1r7e4c10a656d649f6@mail.gmail.com>

Hi all, I'm using freeswitch-1.0.5pre9
When I set bridge_answer_timeout variable, the call is hangup by fs even the
callee has answered the call. I  try to read the source code, found the
cause is in
file switch_ivr_bridge.c  audio_bridge_thread function, It seems  in the
thread loop, ans_a flag is not updated, So when   bridge_answer_timeout is
set, FS will think the channel is still not answered and hangup the call
when timeout.  I tried add "ans_a = switch_channel_test_flag(chan_a,
CF_ANSWERED);  "  in  "for(;;)"  loop, it seem ok now.
Does someone known something about this problem?  Or it's a known bug of
freeswitch?

BTW, my scenario is as follow:

1. A call B in FS
2.FS set bridge_answer_timeout to 30 and bridge the call to B
3.B answer the call
4.after 30 secs, FS hangup call, the cause is "ALLOTTED_TIMEOUT"


-- 
Lei.Tang
lei.tlfly at gmail.com
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From dujinfang at gmail.com  Mon Jan 25 00:28:13 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 25 Jan 2010 16:28:13 +0800
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <13529f9d1001242354q18e8b05fla3cc37de466f0767@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<23f91031001210341x78eb8e61h8938ca525950eda7@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
	<1264092620.14614.73.camel@luna.tc.commsmundi.com>
	<13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
	<ebbc2d551001211721n51cb5c54m996e0d494842c383@mail.gmail.com>
	<13529f9d1001211806o35fc3435j93b71bbb2ede3028@mail.gmail.com>
	<13529f9d1001222200p44afd96ema21729d0038e89ba@mail.gmail.com>
	<ebbc2d551001241547j7e11d4e4se093ce5a60e008b5@mail.gmail.com>
	<13529f9d1001242354q18e8b05fla3cc37de466f0767@mail.gmail.com>
Message-ID: <23f91031001250028t46e5ee06w264aeb2a6c69fdd4@mail.gmail.com>

try to use <action application="info" ...  to see all the available params

2010/1/25 Jingwei Yang <jingwei.yang at gmail.com>:
> Thanks Jo?o. In 01_fifo.xml, the agent uses the action below to sign up.
>
> <action application="set" data="result=${fifo_member(add ACD$1
> {fifo_member_wait=nowait}loopback/1${user_name}/default/XML )"/>
>
> The variable ${user_name} was empty when I tried both skypiax and dingaling.
>
> 2010/1/25 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> What user_name? I don't understand that statement.
>> I think that you can always use
>> http://wiki.freeswitch.org/wiki/Channel_Variables#api_hangup_hook
>> Regards,
>> Jo?o Mesquita
>> FSComm Developer
>>
>>
>> On Sat, Jan 23, 2010 at 4:00 AM, Jingwei Yang <jingwei.yang at gmail.com>
>> wrote:
>>>
>>> Hi Jo?o, do you know how to sign the agent off automatically when either
>>> party hangs up the call?
>>>
>>> Here's how I originate the call to the agent and sign him up in ACD1:
>>>
>>> originate skypiax/ANY/jingwei.yang 6*1
>>>
>>> However, I found the user_name property is empty. May I know how it is
>>> set?
>>>
>>> Thanks and best regards,
>>> -Jingwei
>>>
>>> On Fri, Jan 22, 2010 at 10:06 AM, Jingwei Yang <jingwei.yang at gmail.com>
>>> wrote:
>>>>
>>>> Hi Jo?o, thanks for the reply. I'll try it out.
>>>>
>>>> Regards,
>>>> -Jingwei
>>>>
>>>> 2010/1/22 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>>>
>>>>> Jingwei, check my contrib dir. I think it may help you with one FIFO
>>>>> since we are able there to sign in and sign off dynamic agents as well as
>>>>> customize what we do when the FIFO makes a call to them.
>>>>>
>>>>> Regards,
>>>>> Jo?o Mesquita
>>>>> FSComm Developer
>>>>>
>>>>> On Thu, Jan 21, 2010 at 10:05 PM, Jingwei Yang <jingwei.yang at gmail.com>
>>>>> wrote:
>>>>>>
>>>>>> Thanks for the reply. All the agents are dynamic and I can't predefine
>>>>>> them in the config file.
>>>>>>
>>>>>> Regards,
>>>>>> -Jingwei
>>>>>>
>>>>>> On Fri, Jan 22, 2010 at 12:50 AM, Fran?ois Delawarde
>>>>>> <fdelawarde at wirelessmundi.com> wrote:
>>>>>>>
>>>>>>> Why do you need 2 fifos? You could have callback agents connected to
>>>>>>> the
>>>>>>> fifo and send incoming calls there, mod_fifo will do the rest.
>>>>>>>
>>>>>>> To configure agents for callback:
>>>>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
>>>>>>>
>>>>>>> To place a call into a fifo:
>>>>>>>
>>>>>>> http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue
>>>>>>>
>>>>>>> Fran?ois.
>>>>>>>
>>>>>>> On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
>>>>>>> > Sorry about the confusion, I'm just trying to think over all the
>>>>>>> > abnormal situations before the implementation and hope the flow has
>>>>>>> > no
>>>>>>> > design flaws.
>>>>>>> >
>>>>>>> > Client A is parked in Queue 1, multiple consumers will be ringed to
>>>>>>> > answer him. And once the first one is connected, all the rest will
>>>>>>> > hang up. This is the targeted function.
>>>>>>> >
>>>>>>> > To achieve this, I'm considering to originate a call to each
>>>>>>> > consumer
>>>>>>> > and put the calls in Queue 2. Then intercept client A to the first
>>>>>>> > element of Queue 2.
>>>>>>> >
>>>>>>> > I'm not sure if it's the usual or the best way. If you feel not,
>>>>>>> > please don't hesitate to correct me. Any thoughts are warmly
>>>>>>> > appreciated.
>>>>>>> >
>>>>>>> > On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com>
>>>>>>> > wrote:
>>>>>>> > ? ? ? ? what no errors mean? how do you originate calls to
>>>>>>> > consumers?
>>>>>>> > ? ? ? ? I don't
>>>>>>> > ? ? ? ? understand your scenario.
>>>>>>> >
>>>>>>> > ? ? ? ? 2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
>>>>>>> >
>>>>>>> > ? ? ? ? > Hi All,
>>>>>>> > ? ? ? ? >
>>>>>>> > ? ? ? ? > Please advise whether the following flow makes sense.
>>>>>>> > ? ? ? ? >
>>>>>>> > ? ? ? ? > 1. Client A calls in and parked in Queue 1
>>>>>>> > ? ? ? ? > 2. Originate calls to several consumers simultaneously
>>>>>>> > and
>>>>>>> > ? ? ? ? park them in
>>>>>>> > ? ? ? ? > Queue 2
>>>>>>> > ? ? ? ? > 3. Intercept A's call to the first consumer of Queue 2
>>>>>>> > ? ? ? ? >
>>>>>>> > ? ? ? ? > Basically I want A's call picked up by the first among
>>>>>>> > many
>>>>>>> > ? ? ? ? consumers with
>>>>>>> > ? ? ? ? > no errors. Please let me know whether I'm on the right
>>>>>>> > ? ? ? ? track.
>>>>>>> > ? ? ? ? >
>>>>>>> > ? ? ? ? > Thanks and best regards,
>>>>>>> > ? ? ? ? > -Jingwei
>>>>>>> > ? ? ? ? >
>>>>>>> >
>>>>>>> >
>>>>>>> > ? ? ? ? _______________________________________________
>>>>>>> > ? ? ? ? FreeSWITCH-users mailing list
>>>>>>> > ? ? ? ? FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >
>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >
>>>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> > ? ? ? ? http://www.freeswitch.org
>>>>>>> >
>>>>>>> > _______________________________________________
>>>>>>> > FreeSWITCH-users mailing list
>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >
>>>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> > http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From mike at jerris.com  Mon Jan 25 00:59:43 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 25 Jan 2010 03:59:43 -0500
Subject: [Freeswitch-users] Module multicast fail
In-Reply-To: <4B5D377D.8050408@savion.huji.ac.il>
References: <4B56F0F9.9090808@savion.huji.ac.il>
	<2A8BE7A9-EC50-47CD-9FE8-172BF0F97DB6@freeswitch.org>
	<4B5D377D.8050408@savion.huji.ac.il>
Message-ID: <85B99B6D-E4E2-457D-B643-4168ADA364A3@jerris.com>

This means it was unable to joint he multicast group you specified in the "address" in your conf file for the module.

Mie

On Jan 25, 2010, at 1:17 AM, Eli Hayun wrote:
> Hi
> I run the latest version and still getting an error
> 
> 2010-01-25 07:42:53.449509 [ERR] mod_event_multicast.c:410 Multicast Error^M
> 2010-01-25 07:42:53.449761 [CRIT] switch_loadable_module.c:872 Error
> Loading module /freeswitch-1.0.5_20100120-0400/mod/mod_event_multicast.so^M
> **Module load routine returned an error**^M
> 2010-01-25 07:42:53.450424 [CONSOLE] switch_loadable_module.c:890
> Successfully Loaded [mod_event_socket]^M




From mike at jerris.com  Mon Jan 25 01:03:28 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 25 Jan 2010 04:03:28 -0500
Subject: [Freeswitch-users] mod_radius_cdr module load error
In-Reply-To: <BAY141-W1AF84CAAD730644C7A09790610@phx.gbl>
References: <BAY141-W1AF84CAAD730644C7A09790610@phx.gbl>
Message-ID: <24760B4B-040C-46B7-8AD3-A12F6C0AE140@jerris.com>

the issue is 

"Open of mod_radius_cdr.conf failed"

You need a configuration file for that module that is not there.

Mie

On Jan 23, 2010, at 2:08 AM, satish patel wrote:

> Hi All,
> 
> I am following this wiki http://wiki.freeswitch.org/wiki/Mod_radius_cdr to hook up freeradius with freeswitch but i am getting following error in log
> 
> 2010-01-23 01:56:25.717201 [ERR] mod_radius_cdr.c:662 Open of mod_radius_cdr.conf failed
> 2010-01-23 01:56:25.717225 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_radius_cdr.so
> **Module load routine returned an error**

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From mike at jerris.com  Mon Jan 25 01:04:56 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 25 Jan 2010 04:04:56 -0500
Subject: [Freeswitch-users] Freeswitch process hangs, losses connection.
In-Reply-To: <1264355459996-4449929.post@n2.nabble.com>
References: <1264355459996-4449929.post@n2.nabble.com>
Message-ID: <FFC27F5A-E1AF-427C-954D-C37EF6D8862B@jerris.com>

Have you looked at all a the freeswitch logs?  Try tuning them up to debug, maybe adding some sort of monitoring to see when it is down to see what happened in the logs about that time.
Mike

On Jan 24, 2010, at 12:50 PM, AFalcon wrote:

> 
> Hi I am new here but have successfully configured Freeswitch to run on Snow
> Leopard. I have configured Snow Leopard so that the computer will never go
> to sleep. My issue though is that Freeswitch losses it connection at some
> point while running. When I call in I get a busy signal or a message saying
> the phone number is not in operation. 
> 
> To get around this I wrote a launchd process that restarts Freeswitch and my
> softphone every 4 hours. 
> 
> I don't like this and am wondering how to go about trouble shooting the
> issue of Freeswitch losing connectivity after a period of a few hours. 




From mike at jerris.com  Mon Jan 25 01:05:54 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 25 Jan 2010 04:05:54 -0500
Subject: [Freeswitch-users] Configuring duration of conference
In-Reply-To: <000001ca9d79$81aa0f50$84fe2df0$@dixit@nagarro.com>
References: <000001ca9d79$81aa0f50$84fe2df0$@dixit@nagarro.com>
Message-ID: <438CA6A7-1F44-499C-98D0-BE739B3A71AF@jerris.com>

sched_hangup
sched_broadcast


On Jan 24, 2010, at 11:47 PM, Abhishek Dixit wrote:

> Hi,
>  
> I am working on FreeSWITCH Version 1.0.4 (14460).
> I am trying to set up a conference room which will allow conferencing for limited duration. This means that when parties connect to this conference room they can remain connected only till a specified duration. After this duration the conference should end with announcement.
> I have read wiki for mod_conference and for dialplan and studied IVR conferencing example using javascript as well.
> But I am unable to find a way to specify duration for the conference using dialplan and javascript application.
>  
> I am planning now to use mod_event_socket interface to end conference after a specified duration.
>  
> Please tell me if there can be a way to configure duration of conference using dialplan configurations or javascript application.

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From mike at jerris.com  Mon Jan 25 01:07:04 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 25 Jan 2010 04:07:04 -0500
Subject: [Freeswitch-users] Question about bridge_answer_timeout variable
In-Reply-To: <50c41b4e1001250016v6633eaf1r7e4c10a656d649f6@mail.gmail.com>
References: <50c41b4e1001250016v6633eaf1r7e4c10a656d649f6@mail.gmail.com>
Message-ID: <504AA55B-102A-4755-BFB8-205B40DB3172@jerris.com>

Check if this is an issue in latest svn trunk.  I suspect it is already fixed.

Mike

On Jan 25, 2010, at 3:16 AM, Lei Tang wrote:

> Hi all, I'm using freeswitch-1.0.5pre9
> When I set bridge_answer_timeout variable, the call is hangup by fs even the callee has answered the call. I  try to read the source code, found the cause is in 
> file switch_ivr_bridge.c  audio_bridge_thread function, It seems  in the thread loop, ans_a flag is not updated, So when   bridge_answer_timeout is set, FS will think the channel is still not answered and hangup the call when timeout.  I tried add "ans_a = switch_channel_test_flag(chan_a, CF_ANSWERED);  "  in  "for(;;)"  loop, it seem ok now. 
> Does someone known something about this problem?  Or it's a known bug of freeswitch?
> 
> BTW, my scenario is as follow:
> 
> 1. A call B in FS
> 2.FS set bridge_answer_timeout to 30 and bridge the call to B
> 3.B answer the call
> 4.after 30 secs, FS hangup call, the cause is "ALLOTTED_TIMEOUT"




From irmatov at gmail.com  Mon Jan 25 01:13:21 2010
From: irmatov at gmail.com (Timur Irmatov)
Date: Mon, 25 Jan 2010 14:13:21 +0500
Subject: [Freeswitch-users] mod_erlang_event: disconnects
In-Reply-To: <20100122154658.GC25693@hijacked.us>
References: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>
	<20100121134241.GD1036@hijacked.us>
	<241d382f1001220522g27f67ef5p5ba86ebb3afe63c@mail.gmail.com>
	<20100122154658.GC25693@hijacked.us>
Message-ID: <241d382f1001250113q2885a27dxebdf90ee836f337@mail.gmail.com>

Hi, Andrew!

On Fri, Jan 22, 2010 at 8:46 PM, Andrew Thompson <andrew at hijacked.us> wrote:
> Give this patch a shot:
>
> http://eagle.bsd.st/~andrew/erlang_session_fix.diff
>
> And see if it makes a difference.

I have just installed this patch. Thank you. I will let you know the results.


-- 
Timur Irmatov, xmpp:irmatov at jabber.ru



From jingwei.yang at gmail.com  Mon Jan 25 01:13:21 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 25 Jan 2010 17:13:21 +0800
Subject: [Freeswitch-users] Is this queue flow correct?
In-Reply-To: <23f91031001250028t46e5ee06w264aeb2a6c69fdd4@mail.gmail.com>
References: <13529f9d1001210122m38431b3bq9bcb5de4e8ccd16@mail.gmail.com>
	<13529f9d1001210739v8585a54h93b20d12494a58cf@mail.gmail.com>
	<1264092620.14614.73.camel@luna.tc.commsmundi.com>
	<13529f9d1001211705s39cab157w2632fe371225f89e@mail.gmail.com>
	<ebbc2d551001211721n51cb5c54m996e0d494842c383@mail.gmail.com>
	<13529f9d1001211806o35fc3435j93b71bbb2ede3028@mail.gmail.com>
	<13529f9d1001222200p44afd96ema21729d0038e89ba@mail.gmail.com>
	<ebbc2d551001241547j7e11d4e4se093ce5a60e008b5@mail.gmail.com>
	<13529f9d1001242354q18e8b05fla3cc37de466f0767@mail.gmail.com>
	<23f91031001250028t46e5ee06w264aeb2a6c69fdd4@mail.gmail.com>
Message-ID: <13529f9d1001250113p63dacecbx946f0457875eaaa3@mail.gmail.com>

I see.. thanks!

On Mon, Jan 25, 2010 at 4:28 PM, Seven Du <dujinfang at gmail.com> wrote:

> try to use <action application="info" ...  to see all the available params
>
> 2010/1/25 Jingwei Yang <jingwei.yang at gmail.com>:
> > Thanks Jo?o. In 01_fifo.xml, the agent uses the action below to sign up.
> >
> > <action application="set" data="result=${fifo_member(add ACD$1
> > {fifo_member_wait=nowait}loopback/1${user_name}/default/XML )"/>
> >
> > The variable ${user_name} was empty when I tried both skypiax and
> dingaling.
> >
> > 2010/1/25 Jo?o Mesquita <jmesquita at freeswitch.org>
> >>
> >> What user_name? I don't understand that statement.
> >> I think that you can always use
> >> http://wiki.freeswitch.org/wiki/Channel_Variables#api_hangup_hook
> >> Regards,
> >> Jo?o Mesquita
> >> FSComm Developer
> >>
> >>
> >> On Sat, Jan 23, 2010 at 4:00 AM, Jingwei Yang <jingwei.yang at gmail.com>
> >> wrote:
> >>>
> >>> Hi Jo?o, do you know how to sign the agent off automatically when
> either
> >>> party hangs up the call?
> >>>
> >>> Here's how I originate the call to the agent and sign him up in ACD1:
> >>>
> >>> originate skypiax/ANY/jingwei.yang 6*1
> >>>
> >>> However, I found the user_name property is empty. May I know how it is
> >>> set?
> >>>
> >>> Thanks and best regards,
> >>> -Jingwei
> >>>
> >>> On Fri, Jan 22, 2010 at 10:06 AM, Jingwei Yang <jingwei.yang at gmail.com
> >
> >>> wrote:
> >>>>
> >>>> Hi Jo?o, thanks for the reply. I'll try it out.
> >>>>
> >>>> Regards,
> >>>> -Jingwei
> >>>>
> >>>> 2010/1/22 Jo?o Mesquita <jmesquita at freeswitch.org>
> >>>>>
> >>>>> Jingwei, check my contrib dir. I think it may help you with one FIFO
> >>>>> since we are able there to sign in and sign off dynamic agents as
> well as
> >>>>> customize what we do when the FIFO makes a call to them.
> >>>>>
> >>>>> Regards,
> >>>>> Jo?o Mesquita
> >>>>> FSComm Developer
> >>>>>
> >>>>> On Thu, Jan 21, 2010 at 10:05 PM, Jingwei Yang <
> jingwei.yang at gmail.com>
> >>>>> wrote:
> >>>>>>
> >>>>>> Thanks for the reply. All the agents are dynamic and I can't
> predefine
> >>>>>> them in the config file.
> >>>>>>
> >>>>>> Regards,
> >>>>>> -Jingwei
> >>>>>>
> >>>>>> On Fri, Jan 22, 2010 at 12:50 AM, Fran?ois Delawarde
> >>>>>> <fdelawarde at wirelessmundi.com> wrote:
> >>>>>>>
> >>>>>>> Why do you need 2 fifos? You could have callback agents connected
> to
> >>>>>>> the
> >>>>>>> fifo and send incoming calls there, mod_fifo will do the rest.
> >>>>>>>
> >>>>>>> To configure agents for callback:
> >>>>>>>
> http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback
> >>>>>>>
> >>>>>>> To place a call into a fifo:
> >>>>>>>
> >>>>>>>
> http://wiki.freeswitch.org/wiki/Mod_fifo#Put_a_caller_into_a_FIFO_queue
> >>>>>>>
> >>>>>>> Fran?ois.
> >>>>>>>
> >>>>>>> On Thu, 2010-01-21 at 23:39 +0800, Jingwei Yang wrote:
> >>>>>>> > Sorry about the confusion, I'm just trying to think over all the
> >>>>>>> > abnormal situations before the implementation and hope the flow
> has
> >>>>>>> > no
> >>>>>>> > design flaws.
> >>>>>>> >
> >>>>>>> > Client A is parked in Queue 1, multiple consumers will be ringed
> to
> >>>>>>> > answer him. And once the first one is connected, all the rest
> will
> >>>>>>> > hang up. This is the targeted function.
> >>>>>>> >
> >>>>>>> > To achieve this, I'm considering to originate a call to each
> >>>>>>> > consumer
> >>>>>>> > and put the calls in Queue 2. Then intercept client A to the
> first
> >>>>>>> > element of Queue 2.
> >>>>>>> >
> >>>>>>> > I'm not sure if it's the usual or the best way. If you feel not,
> >>>>>>> > please don't hesitate to correct me. Any thoughts are warmly
> >>>>>>> > appreciated.
> >>>>>>> >
> >>>>>>> > On Thu, Jan 21, 2010 at 7:41 PM, Seven Du <dujinfang at gmail.com>
> >>>>>>> > wrote:
> >>>>>>> >         what no errors mean? how do you originate calls to
> >>>>>>> > consumers?
> >>>>>>> >         I don't
> >>>>>>> >         understand your scenario.
> >>>>>>> >
> >>>>>>> >         2010/1/21, Jingwei Yang <jingwei.yang at gmail.com>:
> >>>>>>> >
> >>>>>>> >         > Hi All,
> >>>>>>> >         >
> >>>>>>> >         > Please advise whether the following flow makes sense.
> >>>>>>> >         >
> >>>>>>> >         > 1. Client A calls in and parked in Queue 1
> >>>>>>> >         > 2. Originate calls to several consumers simultaneously
> >>>>>>> > and
> >>>>>>> >         park them in
> >>>>>>> >         > Queue 2
> >>>>>>> >         > 3. Intercept A's call to the first consumer of Queue 2
> >>>>>>> >         >
> >>>>>>> >         > Basically I want A's call picked up by the first among
> >>>>>>> > many
> >>>>>>> >         consumers with
> >>>>>>> >         > no errors. Please let me know whether I'm on the right
> >>>>>>> >         track.
> >>>>>>> >         >
> >>>>>>> >         > Thanks and best regards,
> >>>>>>> >         > -Jingwei
> >>>>>>> >         >
> >>>>>>> >
> >>>>>>> >
> >>>>>>> >         _______________________________________________
> >>>>>>> >         FreeSWITCH-users mailing list
> >>>>>>> >         FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> >
> >>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> >
> >>>>>>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> >         http://www.freeswitch.org
> >>>>>>> >
> >>>>>>> > _______________________________________________
> >>>>>>> > FreeSWITCH-users mailing list
> >>>>>>> > FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> >
> >>>>>>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> > http://www.freeswitch.org
> >>>>>>>
> >>>>>>>
> >>>>>>> _______________________________________________
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>>
> >>>>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> http://www.freeswitch.org
> >>>>>>
> >>>>>>
> >>>>>> _______________________________________________
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>
> >>>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>> http://www.freeswitch.org
> >>>>>>
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>>
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From michal.zubac at comgate.cz  Mon Jan 25 04:26:38 2010
From: michal.zubac at comgate.cz (=?ISO-8859-2?Q?Michal_Zub=E1=E8?=)
Date: Mon, 25 Jan 2010 13:26:38 +0100
Subject: [Freeswitch-users] sangoma_prid & freeswitch openzap - conflict
Message-ID: <4B5D8DFE.30904@comgate.cz>

Hi.

I'm just curious. Is sangoma_prid neccessary for Freeswitch to work with
E1 (PRI) line? (wanpipe & openzap mode)
I stopped sangoma_prid because, when I try to start Freeswitch, openzap
yells that it cannot open D-channel (/dev/wanpipe1_if16). It is already
used by sangoma_prid.

But PRI calls are behaving strangely for me. Maybe this is the cause.
How can I resolve this conflict?

Thanks for advice. It's possible, that I am doing some newbie mistake.

Michal Zubac



From elihayun at gmail.com  Mon Jan 25 04:42:42 2010
From: elihayun at gmail.com (Eli Hayun)
Date: Mon, 25 Jan 2010 14:42:42 +0200
Subject: [Freeswitch-users] Module multicast fail
In-Reply-To: <85B99B6D-E4E2-457D-B643-4168ADA364A3@jerris.com>
References: <4B56F0F9.9090808@savion.huji.ac.il>	<2A8BE7A9-EC50-47CD-9FE8-172BF0F97DB6@freeswitch.org>	<4B5D377D.8050408@savion.huji.ac.il>
	<85B99B6D-E4E2-457D-B643-4168ADA364A3@jerris.com>
Message-ID: <4B5D91C2.1030707@savion.huji.ac.il>

On 01/25/2010 10:59 AM, Michael Jerris wrote:
> This means it was unable to joint he multicast group you specified in the "address" in your conf file for the module.
>
> Mie
>
> On Jan 25, 2010, at 1:17 AM, Eli Hayun wrote:
>   
>> Hi
>> I run the latest version and still getting an error
>>
>> 2010-01-25 07:42:53.449509 [ERR] mod_event_multicast.c:410 Multicast Error^M
>> 2010-01-25 07:42:53.449761 [CRIT] switch_loadable_module.c:872 Error
>> Loading module /freeswitch-1.0.5_20100120-0400/mod/mod_event_multicast.so^M
>> **Module load routine returned an error**^M
>> 2010-01-25 07:42:53.450424 [CONSOLE] switch_loadable_module.c:890
>> Successfully Loaded [mod_event_socket]^M
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
Hi
And How do I solve this problem? I put the address of my server and it
gave me this error. I tried to stay with the default, same problem.
I want to be able to notify one FS server of all the activities of
onother FS server

Thanks
Eli




From satish_lx at hotmail.com  Mon Jan 25 05:12:08 2010
From: satish_lx at hotmail.com (Satish Patel)
Date: Mon, 25 Jan 2010 08:12:08 -0500
Subject: [Freeswitch-users] Freeswitch billing
Message-ID: <BLU0-SMTP61C041045C3AA7CC15F147905F0@phx.gbl>

I'm planing to intigrate billing fuctionality with freeswitch us there  
any thing available which I can use ?

Mod_nibble is available but is there any GUI for billing ?

Thanks,
Satish



From wasim at convergence.pk  Mon Jan 25 05:51:11 2010
From: wasim at convergence.pk (Wasim Baig)
Date: Mon, 25 Jan 2010 18:51:11 +0500
Subject: [Freeswitch-users] Freeswitch billing
In-Reply-To: <BLU0-SMTP61C041045C3AA7CC15F147905F0@phx.gbl>
References: <BLU0-SMTP61C041045C3AA7CC15F147905F0@phx.gbl>
Message-ID: <b8ad2a5b1001250551t641c53b9s182627f967e904af@mail.gmail.com>

On Mon, Jan 25, 2010 at 6:12 PM, Satish Patel <satish_lx at hotmail.com> wrote:

> I'm planing to intigrate billing fuctionality with freeswitch us there
> any thing available which I can use ?
>
> Mod_nibble is available but is there any GUI for billing ?
>

try astpp, it has a working calling card setup

-- 
wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
peace be upon you ...
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From brian at freeswitch.org  Mon Jan 25 06:37:13 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 25 Jan 2010 08:37:13 -0600
Subject: [Freeswitch-users] Module multicast fail
In-Reply-To: <4B5D91C2.1030707@savion.huji.ac.il>
References: <4B56F0F9.9090808@savion.huji.ac.il>	<2A8BE7A9-EC50-47CD-9FE8-172BF0F97DB6@freeswitch.org>	<4B5D377D.8050408@savion.huji.ac.il>
	<85B99B6D-E4E2-457D-B643-4168ADA364A3@jerris.com>
	<4B5D91C2.1030707@savion.huji.ac.il>
Message-ID: <0075CC82-4653-446C-8F96-89A0AF08B84E@freeswitch.org>

The address of your server isn't a mulitcast address.  Restore the defaults.

/b

On Jan 25, 2010, at 6:42 AM, Eli Hayun wrote:

> Hi
> And How do I solve this problem? I put the address of my server and it
> gave me this error. I tried to stay with the default, same problem.
> I want to be able to notify one FS server of all the activities of
> onother FS server
> 
> Thanks
> Eli




From brian at freeswitch.org  Mon Jan 25 06:41:24 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 25 Jan 2010 08:41:24 -0600
Subject: [Freeswitch-users] sangoma_prid & freeswitch openzap - conflict
In-Reply-To: <4B5D8DFE.30904@comgate.cz>
References: <4B5D8DFE.30904@comgate.cz>
Message-ID: <6B96C690-1E6C-4C9E-9B5F-B23DA1354B48@freeswitch.org>

If you are using PRID you do not configure D channels at all.  Sangoma PRID will use those already.

/b

On Jan 25, 2010, at 6:26 AM, Michal Zub?? wrote:

> Hi.
> 
> I'm just curious. Is sangoma_prid neccessary for Freeswitch to work with
> E1 (PRI) line? (wanpipe & openzap mode)
> I stopped sangoma_prid because, when I try to start Freeswitch, openzap
> yells that it cannot open D-channel (/dev/wanpipe1_if16). It is already
> used by sangoma_prid.
> 
> But PRI calls are behaving strangely for me. Maybe this is the cause.
> How can I resolve this conflict?
> 
> Thanks for advice. It's possible, that I am doing some newbie mistake.
> 
> Michal Zubac




From tjardick at vanderkraan.net  Mon Jan 25 07:21:22 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Mon, 25 Jan 2010 16:21:22 +0100
Subject: [Freeswitch-users] Compile error sofia on Mac OS X
In-Reply-To: <9AEDA6E3-E875-4E63-A4A5-9457317C1D1A@freeswitch.org>
References: <F026A1E8-211C-444E-B6FA-AAFCDBE806C5@vanderkraan.net>
	<9AEDA6E3-E875-4E63-A4A5-9457317C1D1A@freeswitch.org>
Message-ID: <D081C5BA-C32F-463C-A5F2-BC38AF438E05@vanderkraan.net>

Hi Brian,

Just little message to confirm it worked now.

Thanks for the quick fix!

Regards,

Tjardick


On 24 Jan 2010, at 22:15, Brian West wrote:

> Update this has been fixed now.
>
> Thanks,
> Brian
>
> On Jan 24, 2010, at 2:51 PM, Tjardick van der Kraan wrote:
>
>> Hi,
>>
>> I'm trying to compile freeswitch on my MacBook pro to have a local  
>> dev instance, but i run in to the following compile error during  
>> the make:
>>
>> Compiling mod_sofia.c ...
>> cc1: warnings being treated as errors
>> mod_sofia.c: In function 'sofia_receive_message':
>> mod_sofia.c:1446: warning: 'from_host' may be used uninitialized in  
>> this function
>> mod_sofia.c:1446: warning: 'from_user' may be used uninitialized in  
>> this function
>> make[5]: *** [mod_sofia_la-mod_sofia.lo] Error 1
>> make[4]: *** [all] Error 2
>> make[3]: *** [mod_sofia-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>>
>>
>> It's an MBP running Leopard version 10.5.8
>>
>> Any help would be appreciated.
>>
>> Kind regards,
>>
>> Tjardick
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From yehavi.bourvine at gmail.com  Mon Jan 25 07:38:25 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 25 Jan 2010 17:38:25 +0200
Subject: [Freeswitch-users] Sending CDRs to both file and SQL?
Message-ID: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com>

Hello,

  At present we send all our CDRs to a flat file using Asterisk's format
(template "asterisk" in cdr_csv.conf.xml). This file is used by our billing
software.

  For an interim period I would like to send the CDRs to both file and MySQL
database (until I finish writing script to retreive the CDRs from the
database to a file). Is it possible to send the CDRs to both?

                       Thanks, __Yehavi:
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From christian.loeschenkohl at xpirio.com  Mon Jan 25 07:52:25 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 25 Jan 2010 16:52:25 +0100
Subject: [Freeswitch-users] call drops on unanswered UPDATE messages after
	about 30s
Message-ID: <4B5DBE39.7020101@xpirio.com>

hello

we do see some new (interfering) behavior after updating to trunk (revision 16456)
the call gets up normal

- normal invite to sip endpoint (OpenCom 130, OpenCom X320)
- we get 100 trying back
- we get 180 ringing back + rtp
- we get 200 ok
- our freeswitch sends an ack message
- after that freeswitch starts sending UPDATE messages
   none of these UPDATE message is answered by the sip endpoint, so the call gets dropped
   after about 30s.

how can this UPDATE messages be disabled.
i didn't find any option for that.

this is also not nat problem (official ip on the sip device)

please advise

br

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From anthony.minessale at gmail.com  Mon Jan 25 07:56:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 25 Jan 2010 09:56:14 -0600
Subject: [Freeswitch-users] Sending CDRs to both file and SQL?
In-Reply-To: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com>
References: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com>
Message-ID: <191c3a031001250756y9ea5211hf01d2839cbf9bd74@mail.gmail.com>

we leave that exercise to the user, there is no module to write cdr's direct
to a db.


On Mon, Jan 25, 2010 at 9:38 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Hello,
>
>   At present we send all our CDRs to a flat file using Asterisk's format
> (template "asterisk" in cdr_csv.conf.xml). This file is used by our billing
> software.
>
>   For an interim period I would like to send the CDRs to both file and
> MySQL database (until I finish writing script to retreive the CDRs from the
> database to a file). Is it possible to send the CDRs to both?
>
>                        Thanks, __Yehavi:
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From yehavi.bourvine at gmail.com  Mon Jan 25 07:56:43 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 25 Jan 2010 17:56:43 +0200
Subject: [Freeswitch-users] Guide of creating national fonts for Polycom
	phones
Message-ID: <d361ca7b1001250756s490d1924iaba80921a13eb3ce@mail.gmail.com>

Hello,

  We had to add Hebrew labels support for Polycom phones. The way we did it
is described now on the wiki at "polycom configuration" page. The process
for other languages should be quite identical.

  We did not bother with right-to-lef issues, as these cannot be dealt with
without Polycom's help.

                               Regards, __Yehavi:
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From fvillarroel at yahoo.com  Mon Jan 25 08:01:53 2010
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Mon, 25 Jan 2010 08:01:53 -0800 (PST)
Subject: [Freeswitch-users] CDR Gateways
Message-ID: <961072.63349.qm@web34305.mail.mud.yahoo.com>

Dear All.

I have defined various gateways in ~/sip-profiles/external

My questions is if a gateway named foo that send calls from diferents IP address like x.x.x.x and x.x.x.y to my FS. How i can doing a group for both ip address where i can doing later a sql like accountcode=foo?

In this moment if i need know the traffic of customer foo, i should doing two differents cdr, one for every ip address.

It's fine or i can doing of different way?

I hope anyone could me explain how i can doing. 

my gateway foo.xml

<include>
    <gateway name="foo">
      <param name="username" value="foo"/>
      <param name="password" value="foo"/>
      <param name="realm" value="x.x.x.x"/>
      <param name="register" value="false"/>
      <param name="context" value="public"/>
    </gateway>
</include>

foo1.xml

<include>
    <gateway name="foo1">
      <param name="username" value="foo1"/>
      <param name="password" value="foo1"/>
      <param name="realm" value="x.x.x.y"/>
      <param name="register" value="false"/>
      <param name="context" value="public"/>
    </gateway>
</include>

Both gateways foo and foo1 are the same customer

my cdr_csv.conf.xml

<configuration name="cdr_csv.conf" description="CDR CSV Format">
  <settings>
    <!-- 'cdr-csv' will always be appended to log-base -->
    <!--<param name="log-base" value="/var/log"/>-->
    <param name="default-template" value="sql"/>
    <!-- This is like the info app but after the call is hung up -->
    <!--<param name="debug" value="true"/>-->
    <param name="rotate-on-hup" value="false"/>
    <!-- may be a b or ab -->
    <param name="legs" value="ab"/>
  </settings>
  <templates>
        <template name="sql">INSERT INTO v_cdr (v_id, caller_id_name, caller_id_number, destination_number, context, start_stamp, answer_stamp, end_stamp,
duration, billsec, hangup_cause, uuid, bleg_uuid, accountcode, read_codec, write_codec, remote_media_ip, network_addr) VALUES ("{v_id}", "${caller_id_name}"
,"${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}"
,"${uuid}","${bleg_uuid}", "${accountcode}", "${read_codec}", "${write_codec}", "${remote_media_ip}", "${network_addr}");</template>

The argument accountcode on my database is Blank or None for all records of gateways.

Regards.



      



From anthony.minessale at gmail.com  Mon Jan 25 08:03:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 25 Jan 2010 10:03:45 -0600
Subject: [Freeswitch-users] call drops on unanswered UPDATE messages
	after about 30s
In-Reply-To: <4B5DBE39.7020101@xpirio.com>
References: <4B5DBE39.7020101@xpirio.com>
Message-ID: <191c3a031001250803m1177a0f2w853346a17e9be0fd@mail.gmail.com>

you must have polycoms,
if you are running a stable firmware on them the updates should be accepted
and replied to by your phone.
otherwise add the global variable ignore_display_updates=true


2010/1/25 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> hello
>
> we do see some new (interfering) behavior after updating to trunk (revision
> 16456)
> the call gets up normal
>
> - normal invite to sip endpoint (OpenCom 130, OpenCom X320)
> - we get 100 trying back
> - we get 180 ringing back + rtp
> - we get 200 ok
> - our freeswitch sends an ack message
> - after that freeswitch starts sending UPDATE messages
>   none of these UPDATE message is answered by the sip endpoint, so the call
> gets dropped
>   after about 30s.
>
> how can this UPDATE messages be disabled.
> i didn't find any option for that.
>
> this is also not nat problem (official ip on the sip device)
>
> please advise
>
> br
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From christian.loeschenkohl at xpirio.com  Mon Jan 25 08:21:44 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 25 Jan 2010 17:21:44 +0100
Subject: [Freeswitch-users] call drops on unanswered UPDATE
 messages	after about 30s
In-Reply-To: <191c3a031001250803m1177a0f2w853346a17e9be0fd@mail.gmail.com>
References: <4B5DBE39.7020101@xpirio.com>
	<191c3a031001250803m1177a0f2w853346a17e9be0fd@mail.gmail.com>
Message-ID: <4B5DC518.5080309@xpirio.com>

thank you for you quick reply

these devices are sip pbx kind-of-do-it-all (sip, dect, isdn, analog ...)
see also http://www.aastra.com/cps/rde/xchg/04/hs.xsl/15668.htm

i also posted a call setup done right now with ignore_display_updates=true and i
see no difference - please note the many update packages send from our freeswitch (92.63.208.24)
after 200 ok.

http://pastebin.freeswitch.org/11934

br

On 2010-01-25 17:03, Anthony Minessale wrote:

> you must have polycoms,
> if you are running a stable firmware on them the updates should be
> accepted and replied to by your phone.
> otherwise add the global variable ignore_display_updates=true
>
>
> 2010/1/25 Christian L?schenkohl <christian.loeschenkohl at xpirio.com
> <mailto:christian.loeschenkohl at xpirio.com>>
>
>     hello
>
>     we do see some new (interfering) behavior after updating to trunk
>     (revision 16456)
>     the call gets up normal
>
>     - normal invite to sip endpoint (OpenCom 130, OpenCom X320)
>     - we get 100 trying back
>     - we get 180 ringing back + rtp
>     - we get 200 ok
>     - our freeswitch sends an ack message
>     - after that freeswitch starts sending UPDATE messages
>        none of these UPDATE message is answered by the sip endpoint, so
>     the call gets dropped
>        after about 30s.
>
>     how can this UPDATE messages be disabled.
>     i didn't find any option for that.
>
>     this is also not nat problem (official ip on the sip device)
>
>     please advise
>
>     br
>
>     --
>     Ing. Christian L?schenkohl
>     Technische Leitung, Forschung & Entwicklung VoIP
>
>     xpirio
>     Telekommunikation & Service GmbH
>     Lakeside B04
>     9020 Klagenfurt
>     Austria
>
>     T  +43 (0) 5 77 11 - 1000
>     F  +43 (0) 5 77 11 - 1002
>     E christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From brian at freeswitch.org  Mon Jan 25 08:27:34 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 25 Jan 2010 10:27:34 -0600
Subject: [Freeswitch-users] call drops on unanswered UPDATE
	messages	after about 30s
In-Reply-To: <4B5DC518.5080309@xpirio.com>
References: <4B5DBE39.7020101@xpirio.com>
	<191c3a031001250803m1177a0f2w853346a17e9be0fd@mail.gmail.com>
	<4B5DC518.5080309@xpirio.com>
Message-ID: <C737B7E7-1BBA-4B5C-868C-DCED4E667E22@freeswitch.org>

global_setvar ignore_display_updates=true

/b

On Jan 25, 2010, at 10:21 AM, Christian L?schenkohl wrote:

> thank you for you quick reply
> 
> these devices are sip pbx kind-of-do-it-all (sip, dect, isdn, analog ...)
> see also http://www.aastra.com/cps/rde/xchg/04/hs.xsl/15668.htm
> 
> i also posted a call setup done right now with ignore_display_updates=true and i
> see no difference - please note the many update packages send from our freeswitch (92.63.208.24)
> after 200 ok.
> 
> http://pastebin.freeswitch.org/11934
> 
> br

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From christian.loeschenkohl at xpirio.com  Mon Jan 25 08:59:11 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 25 Jan 2010 17:59:11 +0100
Subject: [Freeswitch-users] call drops on unanswered
 UPDATE	messages	after about 30s
In-Reply-To: <C737B7E7-1BBA-4B5C-868C-DCED4E667E22@freeswitch.org>
References: <4B5DBE39.7020101@xpirio.com>	<191c3a031001250803m1177a0f2w853346a17e9be0fd@mail.gmail.com>	<4B5DC518.5080309@xpirio.com>
	<C737B7E7-1BBA-4B5C-868C-DCED4E667E22@freeswitch.org>
Message-ID: <4B5DCDDF.6050305@xpirio.com>

thank you, i works now

---
for documentation

would conf/vars.xml or conf/autoload_configs/switch.conf.xml be the right place for this setting

like
<X-PRE-PROCESS cmd="set" data="ignore_display_updates=true"/>
or
<param name="ignore_display_updates" value="true"/>

so i can put it in the wiki

br

On 2010-01-25 17:27, Brian West wrote:

> global_setvar ignore_display_updates=true
>
> /b
>
> On Jan 25, 2010, at 10:21 AM, Christian L?schenkohl wrote:
>
>> thank you for you quick reply
>>
>> these devices are sip pbx kind-of-do-it-all (sip, dect, isdn, analog ...)
>> see also http://www.aastra.com/cps/rde/xchg/04/hs.xsl/15668.htm
>>
>> i also posted a call setup done right now with
>> ignore_display_updates=true and i
>> see no difference - please note the many update packages send from our
>> freeswitch (92.63.208.24)
>> after 200 ok.
>>
>> http://pastebin.freeswitch.org/11934
>>
>> br
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From brian at freeswitch.org  Mon Jan 25 09:10:38 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 25 Jan 2010 11:10:38 -0600
Subject: [Freeswitch-users] call drops on unanswered
	UPDATE	messages	after about 30s
In-Reply-To: <4B5DCDDF.6050305@xpirio.com>
References: <4B5DBE39.7020101@xpirio.com>	<191c3a031001250803m1177a0f2w853346a17e9be0fd@mail.gmail.com>	<4B5DC518.5080309@xpirio.com>
	<C737B7E7-1BBA-4B5C-868C-DCED4E667E22@freeswitch.org>
	<4B5DCDDF.6050305@xpirio.com>
Message-ID: <0B75CBBB-4D09-4988-A1CD-3DCF04066C8E@freeswitch.org>

see vars.xml you'll see others like this... also you're in bypass media mode.  Didn't mention that eh?

/b

On Jan 25, 2010, at 10:59 AM, Christian L?schenkohl wrote:

> <X-PRE-PROCESS cmd="set" data="ignore_display_updates=true"/>




From testeador01 at gmail.com  Mon Jan 25 09:11:06 2010
From: testeador01 at gmail.com (Milena)
Date: Mon, 25 Jan 2010 12:11:06 -0500
Subject: [Freeswitch-users] making calls through an FXO gw without knowing
	the IP
Message-ID: <c29b930b1001250911vfdc4cb7t30f0fbc2e36d0a92@mail.gmail.com>

Hello everyone,

I need some help in my dialplan, I have an FXO gateway and i can
receive calls from it,

now, if I assign a static IP on it, i can make calls through the fxo
gateway using this:
                 <action application="bridge"
data="sofia/<profile>/<destination>@<gw_IP>"/>

I want to know if it is possible to make calls through this gateway if
it has a variable IP, and how to do it if so;
the gateway uses SIP accounts to register to FS so i think it might be
possible to route the call using one of the registered extension
numbers but I am not sure how so I need some pointers,

Anyone has ideas or knows how to do this?, help is greatly appreciated!



From christian.loeschenkohl at xpirio.com  Mon Jan 25 09:18:29 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 25 Jan 2010 18:18:29 +0100
Subject: [Freeswitch-users] call drops on unanswered
 UPDATE	messages	after about 30s
In-Reply-To: <0B75CBBB-4D09-4988-A1CD-3DCF04066C8E@freeswitch.org>
References: <4B5DBE39.7020101@xpirio.com>	<191c3a031001250803m1177a0f2w853346a17e9be0fd@mail.gmail.com>	<4B5DC518.5080309@xpirio.com>
	<C737B7E7-1BBA-4B5C-868C-DCED4E667E22@freeswitch.org>
	<4B5DCDDF.6050305@xpirio.com>
	<0B75CBBB-4D09-4988-A1CD-3DCF04066C8E@freeswitch.org>
Message-ID: <4B5DD265.7030500@xpirio.com>

yes, caught :-)
we do use bypass media mode (because of t.38)

yes, i also think vars.xml is a good place for this - will put it in the wiki there

br

On 2010-01-25 18:10, Brian West wrote:

> see vars.xml you'll see others like this... also you're in bypass media mode.  Didn't mention that eh?
>
> /b
>
> On Jan 25, 2010, at 10:59 AM, Christian L?schenkohl wrote:
>
>> <X-PRE-PROCESS cmd="set" data="ignore_display_updates=true"/>
>


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From testeador01 at gmail.com  Mon Jan 25 09:20:21 2010
From: testeador01 at gmail.com (Milena)
Date: Mon, 25 Jan 2010 12:20:21 -0500
Subject: [Freeswitch-users] Sending CDRs to both file and SQL?
In-Reply-To: <191c3a031001250756y9ea5211hf01d2839cbf9bd74@mail.gmail.com>
References: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com>
	<191c3a031001250756y9ea5211hf01d2839cbf9bd74@mail.gmail.com>
Message-ID: <c29b930b1001250920g5999d707i1caa38298795cbae@mail.gmail.com>

hello Yehavi,

This is what the wiki suggests anyways, in case you didn't read this yet:
http://wiki.freeswitch.org/wiki/Mod_cdr_csv#Example_Perl_Script_for_CDR_into_Mysql


2010/1/25 Anthony Minessale <anthony.minessale at gmail.com>:
> we leave that exercise to the user, there is no module to write cdr's
direct
> to a db.
>
>
> On Mon, Jan 25, 2010 at 9:38 AM, Yehavi Bourvine <
yehavi.bourvine at gmail.com>
> wrote:
>>
>> Hello,
>>
>>   At present we send all our CDRs to a flat file using Asterisk's format
>> (template "asterisk" in cdr_csv.conf.xml). This file is used by our
billing
>> software.
>>
>>   For an interim period I would like to send the CDRs to both file and
>> MySQL database (until I finish writing script to retreive the CDRs from
the
>> database to a file). Is it possible to send the CDRs to both?
>>
>>                        Thanks, __Yehavi:
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Jan 25 09:20:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 25 Jan 2010 11:20:30 -0600
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <9881D312-67D1-40D3-B169-A178202F4E6C@jerris.com>
References: <984278.36075.qm@web33504.mail.mud.yahoo.com>
	<9881D312-67D1-40D3-B169-A178202F4E6C@jerris.com>
Message-ID: <191c3a031001250920y5e622c0ewa7a80b9fe5799388@mail.gmail.com>

its possible your string hits the parser more than once.
try using 4 \

\\\\sound


On Sun, Jan 24, 2010 at 4:03 AM, Michael Jerris <mike at jerris.com> wrote:

> As noted on that bug, you should be able to either use \\ or / for the path
> separator there and it should work.
>
> Mike
>
> On Jan 22, 2010, at 9:18 AM, Diego Toro wrote:
>
> > Hi, with svn r16440 the problem persists, I creted a jira report
> http://jira.freeswitch.org/browse/LBSNDF-8 this is a minor issue, but
> activing playback delimiter no audio file can be played. On FS the audio
> files are placed in the \sound\ directory, building the path on Windows
> would be \sound '\s' which is replaced by 'ound'.
> >
> > Thank you
> >
> > Diego Toro
> > http://lacarretade.blogspot.com/
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From vfclists at gmail.com  Mon Jan 25 09:36:42 2010
From: vfclists at gmail.com (vfclists)
Date: Mon, 25 Jan 2010 09:36:42 -0800 (PST)
Subject: [Freeswitch-users] Where can I set the IP address for Freeswitch to
 bind to?
Message-ID: <27308006.post@talk.nabble.com>


I have a rather complicated but necessary IP setup. 

I have a VPN running on the system, and some of the IP address a bridged
with other network devices, and Freeswitch is not quite sure which one to
bind to.

It has an IP for the local network, one for the VPN and one for the gateway,
but it appears to bind to the gateway IP or the VPN when the VPN comes up. I
want it bind to the local IP, and let the VPN do its normal stuff on the
outbound,

I edited vars.xml and set local ip but it doesn't seem to be working.

  <X-PRE-PROCESS cmd="set" data="domain=192.168.1.41"/>
  <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>

Is the vars.xml the right place to do it?

I take it that vars.xml is where you change stuff in the default settings?

Thanks

/vfclists

-- 
View this message in context: http://old.nabble.com/Where-can-I-set-the-IP-address-for-Freeswitch-to-bind-to--tp27308006p27308006.html
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From brian at freeswitch.org  Mon Jan 25 10:07:55 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 25 Jan 2010 12:07:55 -0600
Subject: [Freeswitch-users] Where can I set the IP address for
	Freeswitch to bind to?
In-Reply-To: <27308006.post@talk.nabble.com>
References: <27308006.post@talk.nabble.com>
Message-ID: <4F1F4A28-19BC-4EC2-BCA2-C656354B3071@freeswitch.org>

open up sip_profiles/*.xml and put the IP in there for rtp-ip and sip-ip

/b

On Jan 25, 2010, at 11:36 AM, vfclists wrote:

> 
> I have a rather complicated but necessary IP setup. 
> 
> I have a VPN running on the system, and some of the IP address a bridged
> with other network devices, and Freeswitch is not quite sure which one to
> bind to.
> 
> It has an IP for the local network, one for the VPN and one for the gateway,
> but it appears to bind to the gateway IP or the VPN when the VPN comes up. I
> want it bind to the local IP, and let the VPN do its normal stuff on the
> outbound,
> 
> I edited vars.xml and set local ip but it doesn't seem to be working.
> 
>  <X-PRE-PROCESS cmd="set" data="domain=192.168.1.41"/>
>  <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
> 
> Is the vars.xml the right place to do it?
> 
> I take it that vars.xml is where you change stuff in the default settings?
> 
> Thanks
> 
> /vfclists




From mgg at giagnocavo.net  Mon Jan 25 10:08:50 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Mon, 25 Jan 2010 13:08:50 -0500
Subject: [Freeswitch-users] Where can I set the IP address for
 Freeswitch to bind to?
In-Reply-To: <27308006.post@talk.nabble.com>
References: <27308006.post@talk.nabble.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C67032C749445@mse17be1.mse17.exchange.ms>

That'll be in the per-profile settings (sofia).
-Michael

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of vfclists
Sent: Monday, January 25, 2010 10:37 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Where can I set the IP address for Freeswitch to bind to?


I have a rather complicated but necessary IP setup. 

I have a VPN running on the system, and some of the IP address a bridged with other network devices, and Freeswitch is not quite sure which one to bind to.

It has an IP for the local network, one for the VPN and one for the gateway, but it appears to bind to the gateway IP or the VPN when the VPN comes up. I want it bind to the local IP, and let the VPN do its normal stuff on the outbound,

I edited vars.xml and set local ip but it doesn't seem to be working.

  <X-PRE-PROCESS cmd="set" data="domain=192.168.1.41"/>
  <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>

Is the vars.xml the right place to do it?

I take it that vars.xml is where you change stuff in the default settings?

Thanks

/vfclists

--
View this message in context: http://old.nabble.com/Where-can-I-set-the-IP-address-for-Freeswitch-to-bind-to--tp27308006p27308006.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From sos at sokhapkin.dyndns.org  Mon Jan 25 10:09:22 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 25 Jan 2010 13:09:22 -0500
Subject: [Freeswitch-users] Where can I set the IP address for
	Freeswitch to bind to?
In-Reply-To: <27308006.post@talk.nabble.com>
References: <27308006.post@talk.nabble.com>
Message-ID: <201001251309.22400.sos@sokhapkin.dyndns.org>

You need to set IP in the  profile settings, rtp-ip and sip-ip parameters.

On Monday 25 January 2010, vfclists wrote:
> I have a rather complicated but necessary IP setup.
>
> I have a VPN running on the system, and some of the IP address a bridged
> with other network devices, and Freeswitch is not quite sure which one to
> bind to.
>
> It has an IP for the local network, one for the VPN and one for the
> gateway, but it appears to bind to the gateway IP or the VPN when the VPN
> comes up. I want it bind to the local IP, and let the VPN do its normal
> stuff on the outbound,
>
> I edited vars.xml and set local ip but it doesn't seem to be working.
>
>   <X-PRE-PROCESS cmd="set" data="domain=192.168.1.41"/>
>   <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
>
> Is the vars.xml the right place to do it?
>
> I take it that vars.xml is where you change stuff in the default settings?
>
> Thanks
>
> /vfclists





From yehavi.bourvine at gmail.com  Mon Jan 25 10:11:46 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 25 Jan 2010 20:11:46 +0200
Subject: [Freeswitch-users] Sending CDRs to both file and SQL?
In-Reply-To: <c29b930b1001250920g5999d707i1caa38298795cbae@mail.gmail.com>
References: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com>
	<191c3a031001250756y9ea5211hf01d2839cbf9bd74@mail.gmail.com>
	<c29b930b1001250920g5999d707i1caa38298795cbae@mail.gmail.com>
Message-ID: <d361ca7b1001251011q32120d50l5f14a98f33fcee29@mail.gmail.com>

Thanks. I've seen that. Will try to implement this.

                          Thanks, __Yehavi:

2010/1/25 Milena <testeador01 at gmail.com>

> hello Yehavi,
>
> This is what the wiki suggests anyways, in case you didn't read this yet:
>
> http://wiki.freeswitch.org/wiki/Mod_cdr_csv#Example_Perl_Script_for_CDR_into_Mysql
>
>
> 2010/1/25 Anthony Minessale <anthony.minessale at gmail.com>:
>
> > we leave that exercise to the user, there is no module to write cdr's
> direct
> > to a db.
> >
> >
> > On Mon, Jan 25, 2010 at 9:38 AM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com>
> > wrote:
> >>
> >> Hello,
> >>
> >>   At present we send all our CDRs to a flat file using Asterisk's format
> >> (template "asterisk" in cdr_csv.conf.xml). This file is used by our
> billing
> >> software.
> >>
> >>   For an interim period I would like to send the CDRs to both file and
> >> MySQL database (until I finish writing script to retreive the CDRs from
> the
> >> database to a file). Is it possible to send the CDRs to both?
> >>
> >>                        Thanks, __Yehavi:
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From info at daccii.it  Mon Jan 25 10:15:33 2010
From: info at daccii.it (Daniele Salvatore Albano)
Date: Mon, 25 Jan 2010 19:15:33 +0100
Subject: [Freeswitch-users] making calls through an FXO gw without
 knowing the IP
In-Reply-To: <c29b930b1001250911vfdc4cb7t30f0fbc2e36d0a92@mail.gmail.com>
References: <c29b930b1001250911vfdc4cb7t30f0fbc2e36d0a92@mail.gmail.com>
Message-ID: <4B5DDFC5.7000402@daccii.it>

I don't know if it's possible to do using plain xml, but you can try 
writing some javascript/lua/what-you-want code to acquire parameters 
from the registration and use them to do the call

Milena ha scritto:
> Hello everyone,
>
> I need some help in my dialplan, I have an FXO gateway and i can
> receive calls from it,
>
> now, if I assign a static IP on it, i can make calls through the fxo
> gateway using this:
>                  <action application="bridge"
> data="sofia/<profile>/<destination>@<gw_IP>"/>
>
> I want to know if it is possible to make calls through this gateway if
> it has a variable IP, and how to do it if so;
> the gateway uses SIP accounts to register to FS so i think it might be
> possible to route the call using one of the registered extension
> numbers but I am not sure how so I need some pointers,
>
> Anyone has ideas or knows how to do this?, help is greatly appreciated!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From lon at kickasspixels.com  Mon Jan 25 10:22:35 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Mon, 25 Jan 2010 10:22:35 -0800
Subject: [Freeswitch-users] Sending CDRs to both file and SQL?
In-Reply-To: <d361ca7b1001251011q32120d50l5f14a98f33fcee29@mail.gmail.com>
References: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com> 
	<191c3a031001250756y9ea5211hf01d2839cbf9bd74@mail.gmail.com> 
	<c29b930b1001250920g5999d707i1caa38298795cbae@mail.gmail.com> 
	<d361ca7b1001251011q32120d50l5f14a98f33fcee29@mail.gmail.com>
Message-ID: <5d3e0dc61001251022h6c98722ag8b59fb12ce2d8ee8@mail.gmail.com>

For what its worth, I wrote a simple Sinatra-based web service to
accept the CDRs over HTTP. It then queues them for processing into a
database, syslog and/or file.

Its easy to load balance and lets me throttle CDR processing so I
don't put excessive load on the database server at peak times.

Lon



From jerry.richards at teotech.com  Mon Jan 25 10:30:47 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 25 Jan 2010 10:30:47 -0800
Subject: [Freeswitch-users] Presence Change Distribution
In-Reply-To: <2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com>
References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com>
	<2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com>
Message-ID: <26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com>

Okay, I notice that if I reduce the Presence SUBSCRIBE Expires duration
(from 3600 seconds to 60 seconds), then the delay between PUBLISH's and
NOTIFY's is reduced, but FS still waits nearly 45 seconds to send the
NOTIFY's after it receives a PUBLISH.
 
Can a change be made in FS so that NOTIFYs are sent as a direct result of
receipt of the PUBLISH message?  And not tied to the SUBSCRIBE expiration?
I really don't want to configure all my phones to re-subscribe every 30 or
15 seconds.
 
Thanks and Best Regards,
Jerry
 


  _____  

From: RobertT [mailto:siniypin at gmail.com] 
Sent: Tuesday, December 29, 2009 12:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


You can try to reduce your registration time. 
I for one made my client apps send PUBLISH message every minute in addition
to reduced registration time.

Regards, Robert.


2009/12/28 Jerry Richards <jerry.richards at teotech.com>


Is there a setting to control how fast FS distributes presence changes to
subscribers?  Currently, it appears to take several minutes before I see
presence changes.  I would like to see them almost instantaneously, if
possible.

Thanks and Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



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From anthony.minessale at gmail.com  Mon Jan 25 11:04:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 25 Jan 2010 13:04:57 -0600
Subject: [Freeswitch-users] Presence Change Distribution
In-Reply-To: <26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com>
References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com>
	<2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com>
	<26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com>
Message-ID: <191c3a031001251104p55ba7009g9381841f7de56d65@mail.gmail.com>

the notify will be instant after the publish
the notify you see are not triggered by the publish or they would be
instant.

Same drill, turn on presence debugging in sofia.conf.xml
and look at the sql stmts and see why


On Mon, Jan 25, 2010 at 12:30 PM, Jerry Richards <jerry.richards at teotech.com
> wrote:

>  Okay, I notice that if I reduce the Presence SUBSCRIBE Expires duration
> (from 3600 seconds to 60 seconds), then the delay between PUBLISH's and
> NOTIFY's is reduced, but FS still waits nearly 45 seconds to send the
> NOTIFY's after it receives a PUBLISH.
>
> Can a change be made in FS so that NOTIFYs are sent as a direct result of
> receipt of the PUBLISH message?  And not tied to the SUBSCRIBE expiration?
> I really don't want to configure all my phones to re-subscribe every 30 or
> 15 seconds.
>
> Thanks and Best Regards,
> Jerry
>
>
>  ------------------------------
> *From:* RobertT [mailto:siniypin at gmail.com]
> *Sent:* Tuesday, December 29, 2009 12:02 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>
> You can try to reduce your registration time.
> I for one made my client apps send PUBLISH message every minute in addition
> to reduced registration time.
>
> Regards, Robert.
>
> 2009/12/28 Jerry Richards <jerry.richards at teotech.com>
>
>> Is there a setting to control how fast FS distributes presence changes to
>> subscribers?  Currently, it appears to take several minutes before I see
>> presence changes.  I would like to see them almost instantaneously, if
>> possible.
>>
>> Thanks and Best Regards,
>> Jerry
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From dftoro at yahoo.com  Mon Jan 25 11:53:07 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Mon, 25 Jan 2010 11:53:07 -0800 (PST)
Subject: [Freeswitch-users] How to get chat message via event
In-Reply-To: <7b197bef1001241305o3ee9bc9cq7decdc412f90575c@mail.gmail.com>
Message-ID: <561160.39572.qm@web33504.mail.mud.yahoo.com>

Hi, try with ESL  (libs/esl).


Diego Toro
http://lacarretade.blogspot.com/


--- On Sun, 1/24/10, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:

> From: Giovanni Maruzzelli <gmaruzz at celliax.org>
> Subject: Re: [Freeswitch-users] How to get chat message via event
> To: freeswitch-users at lists.freeswitch.org
> Date: Sunday, January 24, 2010, 4:05 PM
> Which events you don't get? From
> which channel in which circumstances?
> (I mean what you do and what do you expect?)
> 
> -giovanni
> 
> On Sun, Jan 24, 2010 at 9:59 PM, afshin afzali <a.afzali2003 at gmail.com>
> wrote:
> > Hi,
> >
> > As you say, I've already done and unfortunately did
> not get the message
> > events although other events are fired as expected :(
> >
> > -- afshin
> >
> > On Sun, Jan 24, 2010 at 7:18 PM, Giovanni Maruzzelli
> <gmaruzz at celliax.org>
> > wrote:
> >>
> >> you subscribe to them as MESSAGE events
> >>
> >> eg, from a telnet session:
> >>
> >> telnet localhost 8021
> >> auth ClueCon
> >> events plain message
> >>
> >> then those events will show up in your telnet
> session.
> >> -gm
> >>
> >> On Sun, Jan 24, 2010 at 4:39 PM, afshin afzali
> <a.afzali2003 at gmail.com>
> >> wrote:
> >> > Hi,
> >> >
> >> > It seems that the chat messages don't fire
> via events by default and
> >> > just
> >> > exchange between parties.
> >> > Is it true? Is it possible to enable those
> via events?
> >> >
> >> > appreciate all,
> >> > -- afshin
> >> >
> >> >
> _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From christian.loeschenkohl at xpirio.com  Mon Jan 25 12:58:48 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 25 Jan 2010 21:58:48 +0100
Subject: [Freeswitch-users] little hangup problem - prepaid application
Message-ID: <4B5E0608.3070001@xpirio.com>

hello

i try to implement a prepaid application and mod_nibblebill isn't my first choice.
the implementation is quite ok so far. now my problem:

- the call setup is normal (a little scripting)
- the calls get ended by sched_hangup (max duration of calls)
- then i would like to execute a script with api_hangup_hook
   in this last script i would also like to use some channel vars (a-number, call duration
   after answer) but i can't do that

i try to use an outbound event socket script but i couldn't get the variables/information
i need to calculate the amount charge

any hint here

br

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From sos at sokhapkin.dyndns.org  Mon Jan 25 13:20:35 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 25 Jan 2010 16:20:35 -0500
Subject: [Freeswitch-users] little hangup problem - prepaid application
In-Reply-To: <4B5E0608.3070001@xpirio.com>
References: <4B5E0608.3070001@xpirio.com>
Message-ID: <201001251620.35308.sos@sokhapkin.dyndns.org>

To my experience most of "interesting" billing-related variables are created 
not when channel is hung up, but later, when channel enters REPORTING state. 
I use mod_cdr_csv to access these variables:

cdr_csv.conf.xml:

<param name="default-template" value="lua"/>

<template name="lua">${lua(onhangup.lua)}</template>

onhangup.lua script accesses required variables like "billsec", does final 
calculations and writes CDR to DB.


On Monday 25 January 2010, Christian L?schenkohl wrote:
> hello
>
> i try to implement a prepaid application and mod_nibblebill isn't my first
> choice. the implementation is quite ok so far. now my problem:
>
> - the call setup is normal (a little scripting)
> - the calls get ended by sched_hangup (max duration of calls)
> - then i would like to execute a script with api_hangup_hook
>    in this last script i would also like to use some channel vars
> (a-number, call duration after answer) but i can't do that
>
> i try to use an outbound event socket script but i couldn't get the
> variables/information i need to calculate the amount charge
>
> any hint here
>
> br





From mike at van.lammeren.net  Mon Jan 25 14:02:27 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Mon, 25 Jan 2010 17:02:27 -0500
Subject: [Freeswitch-users] making calls through an FXO gw without
	knowing the IP
In-Reply-To: <4B5DDFC5.7000402@daccii.it>
References: <c29b930b1001250911vfdc4cb7t30f0fbc2e36d0a92@mail.gmail.com>
	<4B5DDFC5.7000402@daccii.it>
Message-ID: <5d2828f1001251402w28a5e962td9970fab6e1ff3e8@mail.gmail.com>

Hi Milena!

A device with an IP address has either been assigned a static IP by a
person, or assigned a dynamic IP by, probably, a DHCP server. If your FXO
gateway has been assigned a dynamic IP, then before you can figure out how
to get that value in the configuration, you need to determine what that IP
is.

How do you know the IP assigned to your FXO gateway? Does it implement the
UPnP standard?

Just out of curiosity, why not let it have a static IP address?


Mike van Lammeren


On Mon, Jan 25, 2010 at 1:15 PM, Daniele Salvatore Albano <info at daccii.it>wrote:

> I don't know if it's possible to do using plain xml, but you can try
> writing some javascript/lua/what-you-want code to acquire parameters
> from the registration and use them to do the call
>
> Milena ha scritto:
> > Hello everyone,
> >
> > I need some help in my dialplan, I have an FXO gateway and i can
> > receive calls from it,
> >
> > now, if I assign a static IP on it, i can make calls through the fxo
> > gateway using this:
> >                  <action application="bridge"
> > data="sofia/<profile>/<destination>@<gw_IP>"/>
> >
> > I want to know if it is possible to make calls through this gateway if
> > it has a variable IP, and how to do it if so;
> > the gateway uses SIP accounts to register to FS so i think it might be
> > possible to route the call using one of the registered extension
> > numbers but I am not sure how so I need some pointers,
> >
> > Anyone has ideas or knows how to do this?, help is greatly appreciated!
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Mon Jan 25 14:09:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 25 Jan 2010 16:09:06 -0600
Subject: [Freeswitch-users] making calls through an FXO gw without
	knowing the IP
In-Reply-To: <4B5DDFC5.7000402@daccii.it>
References: <c29b930b1001250911vfdc4cb7t30f0fbc2e36d0a92@mail.gmail.com>
	<4B5DDFC5.7000402@daccii.it>
Message-ID: <191c3a031001251409u50976653ib88f586f7aab9e1b@mail.gmail.com>

Either:
sofa/<profile>/<ext gateway regged as>

of if domain was different:

sofa/<profile>/<ext gateway regged as>%<domain gateway regged as>


On Mon, Jan 25, 2010 at 12:15 PM, Daniele Salvatore Albano
<info at daccii.it>wrote:

> I don't know if it's possible to do using plain xml, but you can try
> writing some javascript/lua/what-you-want code to acquire parameters
> from the registration and use them to do the call
>
> Milena ha scritto:
> > Hello everyone,
> >
> > I need some help in my dialplan, I have an FXO gateway and i can
> > receive calls from it,
> >
> > now, if I assign a static IP on it, i can make calls through the fxo
> > gateway using this:
> >                  <action application="bridge"
> > data="sofia/<profile>/<destination>@<gw_IP>"/>
> >
> > I want to know if it is possible to make calls through this gateway if
> > it has a variable IP, and how to do it if so;
> > the gateway uses SIP accounts to register to FS so i think it might be
> > possible to route the call using one of the registered extension
> > numbers but I am not sure how so I need some pointers,
> >
> > Anyone has ideas or knows how to do this?, help is greatly appreciated!
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From nazim.agabekov at gmail.com  Mon Jan 25 14:31:12 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Tue, 26 Jan 2010 02:31:12 +0400
Subject: [Freeswitch-users] Sending CDRs to both file and SQL?
In-Reply-To: <5d3e0dc61001251022h6c98722ag8b59fb12ce2d8ee8@mail.gmail.com>
References: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com>
	<191c3a031001250756y9ea5211hf01d2839cbf9bd74@mail.gmail.com>
	<c29b930b1001250920g5999d707i1caa38298795cbae@mail.gmail.com>
	<d361ca7b1001251011q32120d50l5f14a98f33fcee29@mail.gmail.com>
	<5d3e0dc61001251022h6c98722ag8b59fb12ce2d8ee8@mail.gmail.com>
Message-ID: <4B5E1BB0.2020104@gmail.com>

I have another implementation based on fastcgi and libxml2. Source could
be easily modified to log into the file as well.

http://blog.buta-tech.com/?p=1

On 01/25/2010 10:22 PM, Lon Baker wrote:
> For what its worth, I wrote a simple Sinatra-based web service to
> accept the CDRs over HTTP. It then queues them for processing into a
> database, syslog and/or file.
>
> Its easy to load balance and lets me throttle CDR processing so I
> don't put excessive load on the database server at peak times.
>
> Lon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From m.sobkow at marketelsystems.com  Mon Jan 25 17:49:41 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 25 Jan 2010 19:49:41 -0600
Subject: [Freeswitch-users] Has anyone had Freeswitch crash on loading
	mod_voicemail?
Message-ID: <4B5E4A35.3060803@marketelsystems.com>

I tried a "load mod_voicemail" in fs_cli, hoping to see what 
configuration section it requested from Erlang, but instead of loading 
the module, Freeswitch crashed without any error messages.  SVN 15188 
built on Ubuntu Hardy 32-bit.



From vfclists at googlemail.com  Mon Jan 25 17:13:50 2010
From: vfclists at googlemail.com (Frank Church)
Date: Tue, 26 Jan 2010 01:13:50 +0000
Subject: [Freeswitch-users] little hangup problem - prepaid application
In-Reply-To: <201001251620.35308.sos@sokhapkin.dyndns.org>
References: <4B5E0608.3070001@xpirio.com>
	<201001251620.35308.sos@sokhapkin.dyndns.org>
Message-ID: <7abab2411001251713p31e542b2xf6204ff13c556d03@mail.gmail.com>

I am new to Freeswitch and I am interested in how it works. When the
record is sent to the lua program what format is it sent in?

Are the details sent like parameters on the command line like

script_lua -calleridname=XXXXXX --caller_id_number=XXXXXXX
--start_stamp=XXXXXX ...?

On looking at the template as well, how is the SQL configured?

<template name="sql">INSERT INTO cdr VALUES
("${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}",
"${accountcode}");</template>

Is it a matter of configuring with the

2010/1/25 Sergey Okhapkin <sos at sokhapkin.dyndns.org>:
> To my experience most of "interesting" billing-related variables are created
> not when channel is hung up, but later, when channel enters REPORTING state.
> I use mod_cdr_csv to access these variables:
>
> cdr_csv.conf.xml:
>
> <param name="default-template" value="lua"/>
>
> <template name="lua">${lua(onhangup.lua)}</template>
>
> onhangup.lua script accesses required variables like "billsec", does final
> calculations and writes CDR to DB.
>
>
> On Monday 25 January 2010, Christian L?schenkohl wrote:
>> hello
>>
>> i try to implement a prepaid application and mod_nibblebill isn't my first
>> choice. the implementation is quite ok so far. now my problem:
>>
>> - the call setup is normal (a little scripting)
>> - the calls get ended by sched_hangup (max duration of calls)
>> - then i would like to execute a script with api_hangup_hook
>> ? ?in this last script i would also like to use some channel vars
>> (a-number, call duration after answer) but i can't do that
>>
>> i try to use an outbound event socket script but i couldn't get the
>> variables/information i need to calculate the amount charge
>>
>> any hint here
>>
>> br
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Frank Church

=======================
<a href="http://devblog.brahmancreations.com">devblog.brahmancreations.com</a>



From mike at jerris.com  Mon Jan 25 19:26:25 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 25 Jan 2010 22:26:25 -0500
Subject: [Freeswitch-users] Has anyone had Freeswitch crash on loading
	mod_voicemail?
In-Reply-To: <4B5E4A35.3060803@marketelsystems.com>
References: <4B5E4A35.3060803@marketelsystems.com>
Message-ID: <189DAD7E-77B3-4A09-AA32-1375554A2C0C@jerris.com>

Please confirm this is still the case in svn trunk and if so, report a bug with a backtrace to http://jira.freeswitch.org.

Mike

On Jan 25, 2010, at 8:49 PM, Mark Sobkow wrote:

> I tried a "load mod_voicemail" in fs_cli, hoping to see what 
> configuration section it requested from Erlang, but instead of loading 
> the module, Freeswitch crashed without any error messages.  SVN 15188 
> built on Ubuntu Hardy 32-bit.




From sos at sokhapkin.dyndns.org  Mon Jan 25 19:34:07 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 25 Jan 2010 22:34:07 -0500
Subject: [Freeswitch-users] little hangup problem - prepaid application
In-Reply-To: <7abab2411001251713p31e542b2xf6204ff13c556d03@mail.gmail.com>
References: <4B5E0608.3070001@xpirio.com>
	<201001251620.35308.sos@sokhapkin.dyndns.org>
	<7abab2411001251713p31e542b2xf6204ff13c556d03@mail.gmail.com>
Message-ID: <201001252234.08162.sos@sokhapkin.dyndns.org>

No record is sent to the script, but the script has access to all channel 
variables.

On Monday 25 January 2010, Frank Church wrote:
> I am new to Freeswitch and I am interested in how it works. When the
> record is sent to the lua program what format is it sent in?
>
> Are the details sent like parameters on the command line like
>
> script_lua -calleridname=XXXXXX --caller_id_number=XXXXXXX
> --start_stamp=XXXXXX ...?
>
> On looking at the template as well, how is the SQL configured?
>
> <template name="sql">INSERT INTO cdr VALUES
> ("${caller_id_name}","${caller_id_number}","${destination_number}","${conte
>xt}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${bill
>sec}","${hangup_cause}","${uuid}","${bleg_uuid}",
> "${accountcode}");</template>
>
> Is it a matter of configuring with the
>
> 2010/1/25 Sergey Okhapkin <sos at sokhapkin.dyndns.org>:
> > To my experience most of "interesting" billing-related variables are
> > created not when channel is hung up, but later, when channel enters
> > REPORTING state. I use mod_cdr_csv to access these variables:
> >
> > cdr_csv.conf.xml:
> >
> > <param name="default-template" value="lua"/>
> >
> > <template name="lua">${lua(onhangup.lua)}</template>
> >
> > onhangup.lua script accesses required variables like "billsec", does
> > final calculations and writes CDR to DB.
> >
> > On Monday 25 January 2010, Christian L?schenkohl wrote:
> >> hello
> >>
> >> i try to implement a prepaid application and mod_nibblebill isn't my
> >> first choice. the implementation is quite ok so far. now my problem:
> >>
> >> - the call setup is normal (a little scripting)
> >> - the calls get ended by sched_hangup (max duration of calls)
> >> - then i would like to execute a script with api_hangup_hook
> >> ? ?in this last script i would also like to use some channel vars
> >> (a-number, call duration after answer) but i can't do that
> >>
> >> i try to use an outbound event socket script but i couldn't get the
> >> variables/information i need to calculate the amount charge
> >>
> >> any hint here
> >>
> >> br
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From sos at sokhapkin.dyndns.org  Mon Jan 25 19:41:22 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 25 Jan 2010 22:41:22 -0500
Subject: [Freeswitch-users] IAX2 Support Removed.
In-Reply-To: <0C2714A2-355A-42C7-B589-F0704D436607@freeswitch.org>
References: <0C2714A2-355A-42C7-B589-F0704D436607@freeswitch.org>
Message-ID: <201001252241.22488.sos@sokhapkin.dyndns.org>

Where is "unsupported" directory in SVN if I need to build mod_iax?

On Friday 22 January 2010, Brian West wrote:
> Due to lack of support for the libiax2 being updated to support the newer
> protocol changes and the lack of interest from anyone willing to actually
> work on it.  I have moved mod_iax to unsupported where it will stay until
> someone steps up to rewrite a new IAX2 lib.
>
> Thanks,
> Brian
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From brian at freeswitch.org  Mon Jan 25 19:58:39 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 25 Jan 2010 21:58:39 -0600
Subject: [Freeswitch-users] IAX2 Support Removed.
In-Reply-To: <201001252241.22488.sos@sokhapkin.dyndns.org>
References: <0C2714A2-355A-42C7-B589-F0704D436607@freeswitch.org>
	<201001252241.22488.sos@sokhapkin.dyndns.org>
Message-ID: <C67C1F33-9462-489A-9904-4B4B1BBCB1A6@freeswitch.org>

Not recommend it will crash if someone calls it with a newer iax lib...

http://svn.freeswitch.org/svn/unsupported/

/b

On Jan 25, 2010, at 9:41 PM, Sergey Okhapkin wrote:

> Where is "unsupported" directory in SVN if I need to build mod_iax?




From vfclists at googlemail.com  Mon Jan 25 20:34:07 2010
From: vfclists at googlemail.com (Frank Church)
Date: Tue, 26 Jan 2010 04:34:07 +0000
Subject: [Freeswitch-users] little hangup problem - prepaid application
In-Reply-To: <201001252234.08162.sos@sokhapkin.dyndns.org>
References: <4B5E0608.3070001@xpirio.com>
	<201001251620.35308.sos@sokhapkin.dyndns.org>
	<7abab2411001251713p31e542b2xf6204ff13c556d03@mail.gmail.com>
	<201001252234.08162.sos@sokhapkin.dyndns.org>
Message-ID: <7abab2411001252034kaf90e64g97b5d5c8f7d65e20@mail.gmail.com>

That is new to me, does that mean that all the languages linked in
with Freeswitch have access to the events and variables in FS at all
times?

Can you link me to the documenation that describes this part in more
detail and some examples?

2010/1/26 Sergey Okhapkin <sos at sokhapkin.dyndns.org>:
> No record is sent to the script, but the script has access to all channel
> variables.
>
> On Monday 25 January 2010, Frank Church wrote:
>> I am new to Freeswitch and I am interested in how it works. When the
>> record is sent to the lua program what format is it sent in?
>>
>> Are the details sent like parameters on the command line like
>>
>> script_lua -calleridname=XXXXXX --caller_id_number=XXXXXXX
>> --start_stamp=XXXXXX ...?
>>
>> On looking at the template as well, how is the SQL configured?
>>
>> <template name="sql">INSERT INTO cdr VALUES
>> ("${caller_id_name}","${caller_id_number}","${destination_number}","${conte
>>xt}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${bill
>>sec}","${hangup_cause}","${uuid}","${bleg_uuid}",
>> "${accountcode}");</template>
>>
>> Is it a matter of configuring with the
>>
>> 2010/1/25 Sergey Okhapkin <sos at sokhapkin.dyndns.org>:
>> > To my experience most of "interesting" billing-related variables are
>> > created not when channel is hung up, but later, when channel enters
>> > REPORTING state. I use mod_cdr_csv to access these variables:
>> >
>> > cdr_csv.conf.xml:
>> >
>> > <param name="default-template" value="lua"/>
>> >
>> > <template name="lua">${lua(onhangup.lua)}</template>
>> >
>> > onhangup.lua script accesses required variables like "billsec", does
>> > final calculations and writes CDR to DB.
>> >
>> > On Monday 25 January 2010, Christian L?schenkohl wrote:
>> >> hello
>> >>
>> >> i try to implement a prepaid application and mod_nibblebill isn't my
>> >> first choice. the implementation is quite ok so far. now my problem:
>> >>
>> >> - the call setup is normal (a little scripting)
>> >> - the calls get ended by sched_hangup (max duration of calls)
>> >> - then i would like to execute a script with api_hangup_hook
>> >> ? ?in this last script i would also like to use some channel vars
>> >> (a-number, call duration after answer) but i can't do that
>> >>
>> >> i try to use an outbound event socket script but i couldn't get the
>> >> variables/information i need to calculate the amount charge
>> >>
>> >> any hint here
>> >>
>> >> br
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Frank Church

=======================
<a href="http://devblog.brahmancreations.com">devblog.brahmancreations.com</a>



From a.afzali2003 at gmail.com  Mon Jan 25 23:51:02 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Tue, 26 Jan 2010 11:21:02 +0330
Subject: [Freeswitch-users] How to get chat message via event
In-Reply-To: <561160.39572.qm@web33504.mail.mud.yahoo.com>
References: <7b197bef1001241305o3ee9bc9cq7decdc412f90575c@mail.gmail.com>
	<561160.39572.qm@web33504.mail.mud.yahoo.com>
Message-ID: <debff5a81001252351g1befbe63m8d6c9b97dfa31cf5@mail.gmail.com>

As I see in the code, exchanging chat messages don't fire by events at least
if there is not any sip session between parties.

--afshin

On Mon, Jan 25, 2010 at 11:23 PM, Diego Toro <dftoro at yahoo.com> wrote:

> Hi, try with ESL  (libs/esl).
>
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
> --- On Sun, 1/24/10, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>
> > From: Giovanni Maruzzelli <gmaruzz at celliax.org>
> > Subject: Re: [Freeswitch-users] How to get chat message via event
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Sunday, January 24, 2010, 4:05 PM
> > Which events you don't get? From
> > which channel in which circumstances?
> > (I mean what you do and what do you expect?)
> >
> > -giovanni
> >
> > On Sun, Jan 24, 2010 at 9:59 PM, afshin afzali <a.afzali2003 at gmail.com>
> > wrote:
> > > Hi,
> > >
> > > As you say, I've already done and unfortunately did
> > not get the message
> > > events although other events are fired as expected :(
> > >
> > > -- afshin
> > >
> > > On Sun, Jan 24, 2010 at 7:18 PM, Giovanni Maruzzelli
> > <gmaruzz at celliax.org>
> > > wrote:
> > >>
> > >> you subscribe to them as MESSAGE events
> > >>
> > >> eg, from a telnet session:
> > >>
> > >> telnet localhost 8021
> > >> auth ClueCon
> > >> events plain message
> > >>
> > >> then those events will show up in your telnet
> > session.
> > >> -gm
> > >>
> > >> On Sun, Jan 24, 2010 at 4:39 PM, afshin afzali
> > <a.afzali2003 at gmail.com>
> > >> wrote:
> > >> > Hi,
> > >> >
> > >> > It seems that the chat messages don't fire
> > via events by default and
> > >> > just
> > >> > exchange between parties.
> > >> > Is it true? Is it possible to enable those
> > via events?
> > >> >
> > >> > appreciate all,
> > >> > -- afshin
> > >> >
> > >> >
> > _______________________________________________
> > >> > FreeSWITCH-users mailing list
> > >> > FreeSWITCH-users at lists.freeswitch.org
> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> > http://www.freeswitch.org
> > >> >
> > >> >
> > >>
> > >>
> > >>
> > >> --
> > >> Sincerely,
> > >>
> > >> Giovanni Maruzzelli
> > >> Cell : +39-347-2665618
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> >
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.villasmil.work at gmail.com  Tue Jan 26 00:14:10 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 26 Jan 2010 09:14:10 +0100
Subject: [Freeswitch-users] Sending CDRs to both file and SQL?
In-Reply-To: <4B5E1BB0.2020104@gmail.com>
References: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com>
	<191c3a031001250756y9ea5211hf01d2839cbf9bd74@mail.gmail.com>
	<c29b930b1001250920g5999d707i1caa38298795cbae@mail.gmail.com>
	<d361ca7b1001251011q32120d50l5f14a98f33fcee29@mail.gmail.com>
	<5d3e0dc61001251022h6c98722ag8b59fb12ce2d8ee8@mail.gmail.com>
	<4B5E1BB0.2020104@gmail.com>
Message-ID: <9853f4ff1001260014q7f68206ek2783b509b1e9dc5a@mail.gmail.com>

xml_cdr is perfect for that, why not use it?

On Mon, Jan 25, 2010 at 11:31 PM, Nazim Agabekov
<nazim.agabekov at gmail.com> wrote:
> I have another implementation based on fastcgi and libxml2. Source could
> be easily modified to log into the file as well.
>
> http://blog.buta-tech.com/?p=1
>
> On 01/25/2010 10:22 PM, Lon Baker wrote:
>> For what its worth, I wrote a simple Sinatra-based web service to
>> accept the CDRs over HTTP. It then queues them for processing into a
>> database, syslog and/or file.
>>
>> Its easy to load balance and lets me throttle CDR processing so I
>> don't put excessive load on the database server at peak times.
>>
>> Lon
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From irmatov at gmail.com  Tue Jan 26 01:37:18 2010
From: irmatov at gmail.com (Timur Irmatov)
Date: Tue, 26 Jan 2010 14:37:18 +0500
Subject: [Freeswitch-users] mod_erlang_event: disconnects
In-Reply-To: <20100122154658.GC25693@hijacked.us>
References: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>
	<20100121134241.GD1036@hijacked.us>
	<241d382f1001220522g27f67ef5p5ba86ebb3afe63c@mail.gmail.com>
	<20100122154658.GC25693@hijacked.us>
Message-ID: <241d382f1001260137o535b9e3boe479160378a4a747@mail.gmail.com>

Hi, Andrew!


On Fri, Jan 22, 2010 at 8:46 PM, Andrew Thompson <andrew at hijacked.us> wrote:
> Give this patch a shot:
>
> http://eagle.bsd.st/~andrew/erlang_session_fix.diff
>
> And see if it makes a difference.

24 hours passed since I have installed this patch. Seems to be working
fine - I haven't seen a single disconnect between FreeSWITCH and my
application.

Thank you very much! I owe you a beer, if you ever to visit Tashkent,
Uzbekistan.. :-)


-- 
Timur Irmatov, xmpp:irmatov at jabber.ru



From michal.zubac at comgate.cz  Tue Jan 26 02:59:12 2010
From: michal.zubac at comgate.cz (=?UTF-8?B?TWljaGFsIFp1YsOhxI0=?=)
Date: Tue, 26 Jan 2010 11:59:12 +0100
Subject: [Freeswitch-users] sangoma_prid & freeswitch openzap - conflict
In-Reply-To: <6B96C690-1E6C-4C9E-9B5F-B23DA1354B48@freeswitch.org>
References: <4B5D8DFE.30904@comgate.cz>
	<6B96C690-1E6C-4C9E-9B5F-B23DA1354B48@freeswitch.org>
Message-ID: <4B5ECB00.2070800@comgate.cz>

I've finally figured it out.
I have to use sangoma_prid and (ozmod_sangoma_boost) <boost_spans>
configuration in openzap.conf.xml.

Now  we have problems with sangoma_prid version (I assume), because
Freeswitch logs following messages:
[CRIT] sangoma_boost_client.c:356 Invalid Boost Version 100  Expecting 101
[CRIT] sangoma_boost_client.c:370 Critical Error: PQ Invalid Event
lenght from boost rxlen=23 evsz=1031

Our versions:
sangoma_prid: = Sangoma PRI Protocol Stack Daemon       =
virtual sangoma_prid: = Version: 1.25                           =
virtual sangoma_prid: = Date: Dec 22 2009                       =
virtual sangoma_prid: = Wanpipe Release: wanpipe-3.5.8.11          =
FreeSwitch from SVN trunk (r16509)

I assume, we have to wait for new drivers from Sangoma according to post
<mike at jerris.com> Re: [Freeswitch-users] Need Help to setup freeswitch
with sangoma card [Sun, 24 Jan 2010 05:13:32 -0500]

Mighq

Brian West napsal(a):
> If you are using PRID you do not configure D channels at all.  Sangoma PRID will use those already.
>
> /b
>
> On Jan 25, 2010, at 6:26 AM, Michal Zub?? wrote:
>
>   
>> Hi.
>>
>> I'm just curious. Is sangoma_prid neccessary for Freeswitch to work with
>> E1 (PRI) line? (wanpipe & openzap mode)
>> I stopped sangoma_prid because, when I try to start Freeswitch, openzap
>> yells that it cannot open D-channel (/dev/wanpipe1_if16). It is already
>> used by sangoma_prid.
>>
>> But PRI calls are behaving strangely for me. Maybe this is the cause.
>> How can I resolve this conflict?
>>
>> Thanks for advice. It's possible, that I am doing some newbie mistake.
>>
>> Michal Zubac
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From nazim.agabekov at gmail.com  Tue Jan 26 03:12:39 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Tue, 26 Jan 2010 15:12:39 +0400
Subject: [Freeswitch-users] Sending CDRs to both file and SQL?
In-Reply-To: <9853f4ff1001260014q7f68206ek2783b509b1e9dc5a@mail.gmail.com>
References: <d361ca7b1001250738s6e420f96ica3c437bb568cb79@mail.gmail.com>	<191c3a031001250756y9ea5211hf01d2839cbf9bd74@mail.gmail.com>	<c29b930b1001250920g5999d707i1caa38298795cbae@mail.gmail.com>	<d361ca7b1001251011q32120d50l5f14a98f33fcee29@mail.gmail.com>	<5d3e0dc61001251022h6c98722ag8b59fb12ce2d8ee8@mail.gmail.com>	<4B5E1BB0.2020104@gmail.com>
	<9853f4ff1001260014q7f68206ek2783b509b1e9dc5a@mail.gmail.com>
Message-ID: <4B5ECE27.5090603@gmail.com>

We do use it for cdr generation, for parsing every one has it's own recipe.
In my opinion php is an easiest way to parse:
http://www.0xdecafbad.com/?p=28
Personally, I prefer fastcgi and C.

On 01/26/2010 12:14 PM, David Villasmil wrote:
> xml_cdr is perfect for that, why not use it?
>
> On Mon, Jan 25, 2010 at 11:31 PM, Nazim Agabekov
> <nazim.agabekov at gmail.com> wrote:
>   
>> I have another implementation based on fastcgi and libxml2. Source could
>> be easily modified to log into the file as well.
>>
>> http://blog.buta-tech.com/?p=1
>>
>> On 01/25/2010 10:22 PM, Lon Baker wrote:
>>     
>>> For what its worth, I wrote a simple Sinatra-based web service to
>>> accept the CDRs over HTTP. It then queues them for processing into a
>>> database, syslog and/or file.
>>>
>>> Its easy to load balance and lets me throttle CDR processing so I
>>> don't put excessive load on the database server at peak times.
>>>
>>> Lon
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From null at invalid.name  Tue Jan 26 06:14:25 2010
From: null at invalid.name (Dan Lane)
Date: Tue, 26 Jan 2010 14:14:25 +0000
Subject: [Freeswitch-users] Conference talk detection
Message-ID: <b042b23b1001260614ie185801h6fe871607afe6615@mail.gmail.com>

Hi,

I have a conference bridge with a simple web interface that shows who
is talking using the event api. Unfortunately the users don't like the
way the first few ms of speech after silence is cut off. Presumably
this is an unavoidable side-effect of cutting out audio when someone
isn't speaking.

Is there any way to have a conference where all audio is sent to the
conference but start/stop talking events are still generated?
Currently setting energy-level to 0 disables start/stop talking events
:(

Cheers,
Dan



From moises.silva at gmail.com  Tue Jan 26 06:18:17 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 26 Jan 2010 09:18:17 -0500
Subject: [Freeswitch-users] sangoma_prid & freeswitch openzap - conflict
In-Reply-To: <4B5ECB00.2070800@comgate.cz>
References: <4B5D8DFE.30904@comgate.cz>
	<6B96C690-1E6C-4C9E-9B5F-B23DA1354B48@freeswitch.org>
	<4B5ECB00.2070800@comgate.cz>
Message-ID: <c4d05cbe1001260618o11516ecax1642fd8510184b9@mail.gmail.com>

On Tue, Jan 26, 2010 at 5:59 AM, Michal Zub?? <michal.zubac at comgate.cz>wrote:

> Now  we have problems with sangoma_prid version (I assume), because
> Freeswitch logs following messages:
> [CRIT] sangoma_boost_client.c:356 Invalid Boost Version 100  Expecting 101
> [CRIT] sangoma_boost_client.c:370 Critical Error: PQ Invalid Event
> lenght from boost rxlen=23 evsz=1031
>
> I assume, we have to wait for new drivers from Sangoma according to post
> <mike at jerris.com> Re: [Freeswitch-users] Need Help to setup freeswitch
> with sangoma card [Sun, 24 Jan 2010 05:13:32 -0500]
>
> Mighq
>

That is correct. We apologize for the inconvenience. The waiting shouldn't
be long.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From rob4manhere at gmail.com  Tue Jan 26 06:30:18 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 26 Jan 2010 08:30:18 -0600
Subject: [Freeswitch-users] Conference talk detection
In-Reply-To: <b042b23b1001260614ie185801h6fe871607afe6615@mail.gmail.com>
References: <b042b23b1001260614ie185801h6fe871607afe6615@mail.gmail.com>
Message-ID: <2C9414EA-0F35-47AA-8B9D-754B693B5D89@gmail.com>

Hi Dan,

What happens when you set the energy-level to something small, such as  
10?

Rob

On Jan 26, 2010, at 8:14 AM, Dan Lane wrote:

> Hi,
>
> I have a conference bridge with a simple web interface that shows who
> is talking using the event api. Unfortunately the users don't like the
> way the first few ms of speech after silence is cut off. Presumably
> this is an unavoidable side-effect of cutting out audio when someone
> isn't speaking.
>
> Is there any way to have a conference where all audio is sent to the
> conference but start/stop talking events are still generated?
> Currently setting energy-level to 0 disables start/stop talking events
> :(
>
> Cheers,
> Dan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From null at invalid.name  Tue Jan 26 07:03:23 2010
From: null at invalid.name (Dan Lane)
Date: Tue, 26 Jan 2010 15:03:23 +0000
Subject: [Freeswitch-users] Conference talk detection
In-Reply-To: <2C9414EA-0F35-47AA-8B9D-754B693B5D89@gmail.com>
References: <b042b23b1001260614ie185801h6fe871607afe6615@mail.gmail.com>
	<2C9414EA-0F35-47AA-8B9D-754B693B5D89@gmail.com>
Message-ID: <b042b23b1001260703r42744b64o8c2196562777b05b@mail.gmail.com>

The same thing, with the energy level set to anything other than 0 the
first few ms of audio is not sent to the conference. Presumably this
is by design as this is the amount of time noise has to be made for
before it's detected as speech and audio is sent to the conference;
including the audio that wasn't sent to the conference would introduce
delay.

What I'd like to be able to do is generate start/stop talk events when
people are talking but without starting or stopping the audio stream.


On Tue, Jan 26, 2010 at 2:30 PM, Rob Forman <rob4manhere at gmail.com> wrote:
> Hi Dan,
>
> What happens when you set the energy-level to something small, such as
> 10?
>
> Rob
>
> On Jan 26, 2010, at 8:14 AM, Dan Lane wrote:
>
>> Hi,
>>
>> I have a conference bridge with a simple web interface that shows who
>> is talking using the event api. Unfortunately the users don't like the
>> way the first few ms of speech after silence is cut off. Presumably
>> this is an unavoidable side-effect of cutting out audio when someone
>> isn't speaking.
>>
>> Is there any way to have a conference where all audio is sent to the
>> conference but start/stop talking events are still generated?
>> Currently setting energy-level to 0 disables start/stop talking events
>> :(
>>
>> Cheers,
>> Dan
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From fvillarroel at yahoo.com  Tue Jan 26 07:15:58 2010
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Tue, 26 Jan 2010 07:15:58 -0800 (PST)
Subject: [Freeswitch-users] default-template
Message-ID: <590217.53161.qm@web34308.mail.mud.yahoo.com>

Dear All.

It?s possible define the default-template in each external gateways, like this:

<include>
    <gateway name="foo">
      <param name="username" value="foo"/>
      <param name="password" value="foo"/>
      <param name="realm" value="190.XXX.XX.XXX"/>
      <param name="register" value="false"/>
      <param name="context" value="public"/>
      <param name="default-template" value="sql1"/>
    </gateway>
</include>

<include>
    <gateway name="foo1">
      <param name="username" value="foo1"/>
      <param name="password" value="foo1"/>
      <param name="realm" value="190.XXX.XX.YYY"/>
      <param name="register" value="false"/>
      <param name="context" value="public"/>
      <param name="default-template" value="sql2"/>
    </gateway>
</include>



Regards


      



From anthony.minessale at gmail.com  Tue Jan 26 07:22:29 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Jan 2010 09:22:29 -0600
Subject: [Freeswitch-users] Conference talk detection
In-Reply-To: <b042b23b1001260703r42744b64o8c2196562777b05b@mail.gmail.com>
References: <b042b23b1001260614ie185801h6fe871607afe6615@mail.gmail.com>
	<2C9414EA-0F35-47AA-8B9D-754B693B5D89@gmail.com>
	<b042b23b1001260703r42744b64o8c2196562777b05b@mail.gmail.com>
Message-ID: <191c3a031001260722u25949cabj973e4c8ced4eb7a8@mail.gmail.com>

We use a conference with energy detection on 12+ hours a day and I don't
recall losing any of the audio.
The instant you breach the level that same packet is sent, and you have to
have several consecutive packets below the level to stop.

My guess is you are talking to something lame like a Sonus who is resetting
its jitterbuffer when you start talking again.

try editing your profile and adding this param

<param name="conference-flags" value="waste-bandwidth"/>

This will make the conference send rtp to the other side of the call even
when you are not talking which typically soothes unruly RTP devices.




On Tue, Jan 26, 2010 at 9:03 AM, Dan Lane <null at invalid.name> wrote:

> The same thing, with the energy level set to anything other than 0 the
> first few ms of audio is not sent to the conference. Presumably this
> is by design as this is the amount of time noise has to be made for
> before it's detected as speech and audio is sent to the conference;
> including the audio that wasn't sent to the conference would introduce
> delay.
>
> What I'd like to be able to do is generate start/stop talk events when
> people are talking but without starting or stopping the audio stream.
>
>
> On Tue, Jan 26, 2010 at 2:30 PM, Rob Forman <rob4manhere at gmail.com> wrote:
> > Hi Dan,
> >
> > What happens when you set the energy-level to something small, such as
> > 10?
> >
> > Rob
> >
> > On Jan 26, 2010, at 8:14 AM, Dan Lane wrote:
> >
> >> Hi,
> >>
> >> I have a conference bridge with a simple web interface that shows who
> >> is talking using the event api. Unfortunately the users don't like the
> >> way the first few ms of speech after silence is cut off. Presumably
> >> this is an unavoidable side-effect of cutting out audio when someone
> >> isn't speaking.
> >>
> >> Is there any way to have a conference where all audio is sent to the
> >> conference but start/stop talking events are still generated?
> >> Currently setting energy-level to 0 disables start/stop talking events
> >> :(
> >>
> >> Cheers,
> >> Dan
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
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From zolotov at altron.ua  Tue Jan 26 07:28:12 2010
From: zolotov at altron.ua (Evgeniy Zolotov)
Date: Tue, 26 Jan 2010 17:28:12 +0200
Subject: [Freeswitch-users] Conference talk detection
In-Reply-To: <b042b23b1001260703r42744b64o8c2196562777b05b@mail.gmail.com>
References: <b042b23b1001260614ie185801h6fe871607afe6615@mail.gmail.com>	<2C9414EA-0F35-47AA-8B9D-754B693B5D89@gmail.com>
	<b042b23b1001260703r42744b64o8c2196562777b05b@mail.gmail.com>
Message-ID: <4B5F0A0C.2060808@altron.ua>

  Try to set flag 'waste' on conference profile.

Dan Lane ?????:
> The same thing, with the energy level set to anything other than 0 the
> first few ms of audio is not sent to the conference. Presumably this
> is by design as this is the amount of time noise has to be made for
> before it's detected as speech and audio is sent to the conference;
> including the audio that wasn't sent to the conference would introduce
> delay.
>
> What I'd like to be able to do is generate start/stop talk events when
> people are talking but without starting or stopping the audio stream.
>
>
> On Tue, Jan 26, 2010 at 2:30 PM, Rob Forman <rob4manhere at gmail.com> wrote:
>   
>> Hi Dan,
>>
>> What happens when you set the energy-level to something small, such as
>> 10?
>>
>> Rob
>>
>> On Jan 26, 2010, at 8:14 AM, Dan Lane wrote:
>>
>>     
>>> Hi,
>>>
>>> I have a conference bridge with a simple web interface that shows who
>>> is talking using the event api. Unfortunately the users don't like the
>>> way the first few ms of speech after silence is cut off. Presumably
>>> this is an unavoidable side-effect of cutting out audio when someone
>>> isn't speaking.
>>>
>>> Is there any way to have a conference where all audio is sent to the
>>> conference but start/stop talking events are still generated?
>>> Currently setting energy-level to 0 disables start/stop talking events
>>> :(
>>>
>>> Cheers,
>>> Dan
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From andrew at hijacked.us  Tue Jan 26 07:28:49 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Tue, 26 Jan 2010 10:28:49 -0500
Subject: [Freeswitch-users] mod_erlang_event: disconnects
In-Reply-To: <241d382f1001260137o535b9e3boe479160378a4a747@mail.gmail.com>
References: <241d382f1001202159l640d9780y6682b35ab4ea55d2@mail.gmail.com>
	<20100121134241.GD1036@hijacked.us>
	<241d382f1001220522g27f67ef5p5ba86ebb3afe63c@mail.gmail.com>
	<20100122154658.GC25693@hijacked.us>
	<241d382f1001260137o535b9e3boe479160378a4a747@mail.gmail.com>
Message-ID: <20100126152849.GG6569@hijacked.us>

On Tue, Jan 26, 2010 at 02:37:18PM +0500, Timur Irmatov wrote:
> 24 hours passed since I have installed this patch. Seems to be working
> fine - I haven't seen a single disconnect between FreeSWITCH and my
> application.
> 
Okay, I'll commit it to trunk then. Thanks for the report.

> Thank you very much! I owe you a beer, if you ever to visit Tashkent,
> Uzbekistan.. :-)
> 
I'll keep that in mind :)

Andrew



From anthony.minessale at gmail.com  Tue Jan 26 07:30:15 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Jan 2010 09:30:15 -0600
Subject: [Freeswitch-users] How to get chat message via event
In-Reply-To: <debff5a81001252351g1befbe63m8d6c9b97dfa31cf5@mail.gmail.com>
References: <7b197bef1001241305o3ee9bc9cq7decdc412f90575c@mail.gmail.com>
	<561160.39572.qm@web33504.mail.mud.yahoo.com>
	<debff5a81001252351g1befbe63m8d6c9b97dfa31cf5@mail.gmail.com>
Message-ID: <191c3a031001260730o7450b95ene02e1b56ccd182b0@mail.gmail.com>

if you are directly controlling a session "myevents <session>" or it was an
outbound socket call
you can send the command "divert_events" to get the events on event_socket
if you are in an embedded script you can get them from the input callback

On Tue, Jan 26, 2010 at 1:51 AM, afshin afzali <a.afzali2003 at gmail.com>wrote:

> As I see in the code, exchanging chat messages don't fire by events at
> least if there is not any sip session between parties.
>
> --afshin
>
> On Mon, Jan 25, 2010 at 11:23 PM, Diego Toro <dftoro at yahoo.com> wrote:
>
>> Hi, try with ESL  (libs/esl).
>>
>>
>> Diego Toro
>> http://lacarretade.blogspot.com/
>>
>>
>> --- On Sun, 1/24/10, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
>>
>> > From: Giovanni Maruzzelli <gmaruzz at celliax.org>
>> > Subject: Re: [Freeswitch-users] How to get chat message via event
>> > To: freeswitch-users at lists.freeswitch.org
>> > Date: Sunday, January 24, 2010, 4:05 PM
>> > Which events you don't get? From
>> > which channel in which circumstances?
>> > (I mean what you do and what do you expect?)
>> >
>> > -giovanni
>> >
>> > On Sun, Jan 24, 2010 at 9:59 PM, afshin afzali <a.afzali2003 at gmail.com>
>> > wrote:
>> > > Hi,
>> > >
>> > > As you say, I've already done and unfortunately did
>> > not get the message
>> > > events although other events are fired as expected :(
>> > >
>> > > -- afshin
>> > >
>> > > On Sun, Jan 24, 2010 at 7:18 PM, Giovanni Maruzzelli
>> > <gmaruzz at celliax.org>
>> > > wrote:
>> > >>
>> > >> you subscribe to them as MESSAGE events
>> > >>
>> > >> eg, from a telnet session:
>> > >>
>> > >> telnet localhost 8021
>> > >> auth ClueCon
>> > >> events plain message
>> > >>
>> > >> then those events will show up in your telnet
>> > session.
>> > >> -gm
>> > >>
>> > >> On Sun, Jan 24, 2010 at 4:39 PM, afshin afzali
>> > <a.afzali2003 at gmail.com>
>> > >> wrote:
>> > >> > Hi,
>> > >> >
>> > >> > It seems that the chat messages don't fire
>> > via events by default and
>> > >> > just
>> > >> > exchange between parties.
>> > >> > Is it true? Is it possible to enable those
>> > via events?
>> > >> >
>> > >> > appreciate all,
>> > >> > -- afshin
>> > >> >
>> > >> >
>> > _______________________________________________
>> > >> > FreeSWITCH-users mailing list
>> > >> > FreeSWITCH-users at lists.freeswitch.org
>> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > >> > http://www.freeswitch.org
>> > >> >
>> > >> >
>> > >>
>> > >>
>> > >>
>> > >> --
>> > >> Sincerely,
>> > >>
>> > >> Giovanni Maruzzelli
>> > >> Cell : +39-347-2665618
>> > >>
>> > >> _______________________________________________
>> > >> FreeSWITCH-users mailing list
>> > >> FreeSWITCH-users at lists.freeswitch.org
>> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > >> http://www.freeswitch.org
>> > >
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> > >
>> >
>> >
>> >
>> > --
>> > Sincerely,
>> >
>> > Giovanni Maruzzelli
>> > Cell : +39-347-2665618
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From freeswitch-users at digitaldan.com  Tue Jan 26 07:34:39 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Tue, 26 Jan 2010 08:34:39 -0700 (MST)
Subject: [Freeswitch-users] Lua: io.popen/read blocking
	in	other	incoming calls
In-Reply-To: <15818864.1.1264519951723.JavaMail.root@zimbra>
Message-ID: <10147006.4.1264520079191.JavaMail.root@zimbra>

Thanks, I'm using os.execute now and piping the output to a temp file. It's not the most elagant solution, but it works. Call volume is light so I don't think there will be any scalability issues. I'll take a look at the sites you mention and see if I can find something a little bit less hacky. 

Thanks again. 
Dan- 

----- Original Message ----- 
From: "Fernando Gregianin Testa" <testa at voicetechnology.com.br> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Sunday, January 24, 2010 12:21:51 PM 
Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls 

You may consider use lua socket.http package as an alternative to popen+wget. Check: 
https://web.tecgraf.puc-rio.br/luasocket/ 
http://www.tecgraf.puc-rio.br/~diego/professional/luasocket/http.html 


Maybe you can be interested also in http://github.com/fertesta/restinlua 




Em 19/01/2010, ?s 21:17, Dan escreveu: 




I would, but I need to post a a wav file that gets recorded, I didn't see a way to supply the location of a file to use as the post data. It looks like you have to url encode the data in the script and pass it all in the call. 


Thanks 
Dan- 
----- Original Message ----- 
From: "Rupa Schomaker" < rupa at rupa.com > 
To: "freeswitch-users" < freeswitch-users at lists.freeswitch.org > 
Sent: Tuesday, January 19, 2010 4:06:36 PM 
Subject: Re: [Freeswitch-users] Lua: io.popen/read blocking in other incoming calls 




On Tue, Jan 19, 2010 at 3:03 PM, Dan < freeswitch-users at digitaldan.com > wrote: 




My lua script is calling wget through lua's io.popen to send and receive data from a web service. While the f:read to wget is running, other incoming calls will block on the same io.popen call until the first call closes the pipe (with f:close()). 


You might want to look at the api that mod_curl exposes to do what you want. No need to do an expensive system call just to call a webservice. 
-- 
-Rupa 

_______________________________________________ 
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FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 
_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 



Fernando Gregianin Testa 
testa at voicetechnology.com.br 
+55 11 35882166 




_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
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From dftoro at yahoo.com  Tue Jan 26 07:55:35 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 26 Jan 2010 07:55:35 -0800 (PST)
Subject: [Freeswitch-users] compilation error on fs_cli (Windows) (latest
	version)
Message-ID: <571042.39380.qm@web33502.mail.mud.yahoo.com>

Hi, I have compilation error "error C2220" on fs_cli project on Windows using VS2008.

FS: latest version (2010/01/26)
VS: VS2008
SO: Windows 7

VS2008 Error log:

Error	1	error C2220: warning treated as error - no 'object' file generated	g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c	106	fs_cli

Warning	2	warning C6385: Invalid data: accessing 'global_profile->console_fnkeys', the readable size is '48' bytes, but '-4' bytes might be read: Lines: 86, 88, 90	g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c	90	fs_cli

Warning	3	warning C6246: Local declaration of 'p' hides declaration of the same name in outer scope. For additional information, see previous declaration at line '844' of 'g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c': Lines: 844	g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c	895	fs_cli

Warning	4	warning C6011: Dereferencing NULL pointer 'cursor': Lines: 839, 840, 841, 842, 843, 844, 846, 849, 853, 857, 868, 870, 871, 884	g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c	884	fs_cli


Thank you

Diego Toro
http://lacarretade.blogspot.com/


      



From dftoro at yahoo.com  Tue Jan 26 07:58:23 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 26 Jan 2010 07:58:23 -0800 (PST)
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <191c3a031001250920y5e622c0ewa7a80b9fe5799388@mail.gmail.com>
Message-ID: <99859.15231.qm@web33505.mail.mud.yahoo.com>

Hi, using \\\\ the is changed also when there is a match with an escape character (\s,\n...)

Thank you

Diego Toro
http://lacarretade.blogspot.com/


--- On Mon, 1/25/10, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] mutiple playback files (unescape_char) Windows
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, January 25, 2010, 12:20 PM
> its possible your string hits the parser
> more than once.
> try using 4 \ 
> 
> \\\\sound
> 
> 
> On Sun, Jan 24, 2010 at 4:03 AM,
> Michael Jerris <mike at jerris.com>
> wrote:
> 
> As noted on that bug, you should be
> able to either use \\ or / for the path separator
> there and it should work.
> 
> 
> 
> 
> Mike
> 
> 
> 
> On Jan 22, 2010, at 9:18 AM, Diego Toro wrote:
> 
> 
> 
> > Hi, with svn r16440 the problem persists, I creted a
> jira report http://jira.freeswitch.org/browse/LBSNDF-8
> this is a minor issue, but activing playback delimiter no
> audio file can be played. On FS the audio files are placed
> in the \sound\ directory, building the path on
> Windows would be \sound '\s' which is
> replaced by 'ound'.
> 
> 
> >
> 
> > Thank you
> 
> >
> 
> > Diego Toro
> 
> > http://lacarretade.blogspot.com/
> 
> >
> 
> 
> 
> 
> 
> _______________________________________________
> 
> FreeSWITCH-users mailing list
> 
> FreeSWITCH-users at lists.freeswitch.org
> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> 
> IRC: irc.freenode.net
> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> 
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From anthony.minessale at gmail.com  Tue Jan 26 08:27:12 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Jan 2010 10:27:12 -0600
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <99859.15231.qm@web33505.mail.mud.yahoo.com>
References: <191c3a031001250920y5e622c0ewa7a80b9fe5799388@mail.gmail.com>
	<99859.15231.qm@web33505.mail.mud.yahoo.com>
Message-ID: <191c3a031001260827j544e1414oaec56f527e116eea@mail.gmail.com>

I didn't understand that

On Tue, Jan 26, 2010 at 9:58 AM, Diego Toro <dftoro at yahoo.com> wrote:

> Hi, using \\\\ the is changed also when there is a match with an escape
> character (\s,\n...)
>
> Thank you
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
> --- On Mon, 1/25/10, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>
> > From: Anthony Minessale <anthony.minessale at gmail.com>
> > Subject: Re: [Freeswitch-users] mutiple playback files (unescape_char)
> Windows
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Monday, January 25, 2010, 12:20 PM
> > its possible your string hits the parser
> > more than once.
> > try using 4 \
> >
> > \\\\sound
> >
> >
> > On Sun, Jan 24, 2010 at 4:03 AM,
> > Michael Jerris <mike at jerris.com>
> > wrote:
> >
> > As noted on that bug, you should be
> > able to either use \\ or / for the path separator
> > there and it should work.
> >
> >
> >
> >
> > Mike
> >
> >
> >
> > On Jan 22, 2010, at 9:18 AM, Diego Toro wrote:
> >
> >
> >
> > > Hi, with svn r16440 the problem persists, I creted a
> > jira report http://jira.freeswitch.org/browse/LBSNDF-8
> > this is a minor issue, but activing playback delimiter no
> > audio file can be played. On FS the audio files are placed
> > in the \sound\ directory, building the path on
> > Windows would be \sound '\s' which is
> > replaced by 'ound'.
> >
> >
> > >
> >
> > > Thank you
> >
> > >
> >
> > > Diego Toro
> >
> > > http://lacarretade.blogspot.com/
> >
> > >
> >
> >
> >
> >
> >
> > _______________________________________________
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net
> > #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> >
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> >
> > -----Inline Attachment Follows-----
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From jeff at jefflenk.com  Tue Jan 26 08:42:54 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 26 Jan 2010 10:42:54 -0600
Subject: [Freeswitch-users] compilation error on fs_cli (Windows)
 (latest	version)
In-Reply-To: <571042.39380.qm@web33502.mail.mud.yahoo.com>
References: <571042.39380.qm@web33502.mail.mud.yahoo.com>
Message-ID: <COL117-W44C42225C0FBBE6BAF2813B95E0@phx.gbl>


I dont see this - do a rebuild all. Are there more errors before these?


 
> Date: Tue, 26 Jan 2010 07:55:35 -0800
> From: dftoro at yahoo.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] compilation error on fs_cli (Windows) (latest version)
> 
> Hi, I have compilation error "error C2220" on fs_cli project on Windows using VS2008.
> 
> FS: latest version (2010/01/26)
> VS: VS2008
> SO: Windows 7
> 
> VS2008 Error log:
> 
> Error 1 error C2220: warning treated as error - no 'object' file generated g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c 106 fs_cli
> 
> Warning 2 warning C6385: Invalid data: accessing 'global_profile->console_fnkeys', the readable size is '48' bytes, but '-4' bytes might be read: Lines: 86, 88, 90 g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c 90 fs_cli
> 
> Warning 3 warning C6246: Local declaration of 'p' hides declaration of the same name in outer scope. For additional information, see previous declaration at line '844' of 'g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c': Lines: 844 g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c 895 fs_cli
> 
> Warning 4 warning C6011: Dereferencing NULL pointer 'cursor': Lines: 839, 840, 841, 842, 843, 844, 846, 849, 853, 857, 868, 870, 871, 884 g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c 884 fs_cli
> 
> 
> Thank you
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
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From mrene_lists at avgs.ca  Tue Jan 26 08:45:49 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 26 Jan 2010 11:45:49 -0500
Subject: [Freeswitch-users] compilation error on fs_cli (Windows)
	(latest version)
In-Reply-To: <571042.39380.qm@web33502.mail.mud.yahoo.com>
References: <571042.39380.qm@web33502.mail.mud.yahoo.com>
Message-ID: <3072AE90-DDEA-4FD0-9B4B-140051730073@avgs.ca>

Looks like the code analyzer is running, this is normally turned off  
when you do a normal build, turn it off and try again.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 26-Jan-10, at 10:55 AM, Diego Toro wrote:

> Hi, I have compilation error "error C2220" on fs_cli project on  
> Windows using VS2008.
>
> FS: latest version (2010/01/26)
> VS: VS2008
> SO: Windows 7
>
> VS2008 Error log:
>
> Error	1	error C2220: warning treated as error - no 'object' file  
> generated	g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl 
> \fs_cli.c	106	fs_cli
>
> Warning	2	warning C6385: Invalid data: accessing 'global_profile- 
> >console_fnkeys', the readable size is '48' bytes, but '-4' bytes  
> might be read: Lines: 86, 88, 90	g:\ftp\incoming\fs 
> \freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c	90	fs_cli
>
> Warning	3	warning C6246: Local declaration of 'p' hides declaration  
> of the same name in outer scope. For additional information, see  
> previous declaration at line '844' of 'g:\ftp\incoming\fs 
> \freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c': Lines: 844	g:\ftp 
> \incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c	895	 
> fs_cli
>
> Warning	4	warning C6011: Dereferencing NULL pointer 'cursor': Lines:  
> 839, 840, 841, 842, 843, 844, 846, 849, 853, 857, 868, 870, 871, 884	 
> g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c	 
> 884	fs_cli
>
>
> Thank you
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From null at invalid.name  Tue Jan 26 08:56:48 2010
From: null at invalid.name (Dan Lane)
Date: Tue, 26 Jan 2010 16:56:48 +0000
Subject: [Freeswitch-users] Conference talk detection
In-Reply-To: <191c3a031001260722u25949cabj973e4c8ced4eb7a8@mail.gmail.com>
References: <b042b23b1001260614ie185801h6fe871607afe6615@mail.gmail.com>
	<2C9414EA-0F35-47AA-8B9D-754B693B5D89@gmail.com>
	<b042b23b1001260703r42744b64o8c2196562777b05b@mail.gmail.com>
	<191c3a031001260722u25949cabj973e4c8ced4eb7a8@mail.gmail.com>
Message-ID: <b042b23b1001260856x21a4c938v619cfe6eb7255561@mail.gmail.com>

On Tue, Jan 26, 2010 at 3:22 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> We use a conference with energy detection on 12+ hours a day and I don't
> recall losing any of the audio.
> The instant you breach the level that same packet is sent, and you have to
> have several consecutive packets below the level to stop.

Interesting could you post the conferences.xml config for that
particular conference? I have yet to find a way of using energy
detection that doesn't miss the first few ms of sound, for example the
word "Testing" often comes out as "esting" perhaps with the tail end
of the T coming across.

It's not a problem when everyone is a native english speaker but when
on a conference containing people with a poor grasp of english or
people who reply with lots of quick utterances such as "yup"

> My guess is you are talking to something lame like a Sonus who is resetting
> its jitterbuffer when you start talking again.

In production it's hitting Cisco gateways but I've experienced this in
a lab environment consisting of a mixture of Snom 360, 820 and Polycom
IP6000 devices.

What would be great is to have start/stop talk events while still
always sending audio from participants to the conference



From anthony.minessale at gmail.com  Tue Jan 26 09:53:55 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Jan 2010 11:53:55 -0600
Subject: [Freeswitch-users] Conference talk detection
In-Reply-To: <b042b23b1001260856x21a4c938v619cfe6eb7255561@mail.gmail.com>
References: <b042b23b1001260614ie185801h6fe871607afe6615@mail.gmail.com>
	<2C9414EA-0F35-47AA-8B9D-754B693B5D89@gmail.com>
	<b042b23b1001260703r42744b64o8c2196562777b05b@mail.gmail.com>
	<191c3a031001260722u25949cabj973e4c8ced4eb7a8@mail.gmail.com>
	<b042b23b1001260856x21a4c938v619cfe6eb7255561@mail.gmail.com>
Message-ID: <191c3a031001260953p4846a541m74da287d1d33897d@mail.gmail.com>

try the param i mentioned and gave the specific example for.


On Tue, Jan 26, 2010 at 10:56 AM, Dan Lane <null at invalid.name> wrote:

> On Tue, Jan 26, 2010 at 3:22 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > We use a conference with energy detection on 12+ hours a day and I don't
> > recall losing any of the audio.
> > The instant you breach the level that same packet is sent, and you have
> to
> > have several consecutive packets below the level to stop.
>
> Interesting could you post the conferences.xml config for that
> particular conference? I have yet to find a way of using energy
> detection that doesn't miss the first few ms of sound, for example the
> word "Testing" often comes out as "esting" perhaps with the tail end
> of the T coming across.
>
> It's not a problem when everyone is a native english speaker but when
> on a conference containing people with a poor grasp of english or
> people who reply with lots of quick utterances such as "yup"
>
> > My guess is you are talking to something lame like a Sonus who is
> resetting
> > its jitterbuffer when you start talking again.
>
> In production it's hitting Cisco gateways but I've experienced this in
> a lab environment consisting of a mixture of Snom 360, 820 and Polycom
> IP6000 devices.
>
> What would be great is to have start/stop talk events while still
> always sending audio from participants to the conference
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jerry.richards at teotech.com  Tue Jan 26 13:02:08 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 26 Jan 2010 13:02:08 -0800
Subject: [Freeswitch-users] Presence Change Distribution
In-Reply-To: <191c3a031001251104p55ba7009g9381841f7de56d65@mail.gmail.com>
References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com><2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com><26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com>
	<191c3a031001251104p55ba7009g9381841f7de56d65@mail.gmail.com>
Message-ID: <BE6F5600B8E54EE297963A8B08F70E5E@greyhawk.tonecommander.com>

Okay, I setup my FS to force SUBSCRIBE Expires to 3600 seconds.  Then I
captured a FS console trace of a Bria softphone changing it's presence state
from 'Busy' to 'Away' (see http://pastebin.freeswitch.org/11953) and
observed that the subscribing Bria softphone did not update to 'Away'.  At
the same time, I executed the sqlite3 app and pasted each of the 3 SQL
select statements I saw in the FS console log, and pasted them below.  I'm
new to sqlite3.  Do you see what my issue is?
 
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'internal' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
sip|5382|192.168.72.79|5401|192.168.72.79|presence|"5382 on 79"
<sip:5382 at 192.168.72.150:34672>|ZTQ2ZWQwZGRlZjRiNTdkYTJjNGM5NTgzOWIyNmIwZmU.
|"5382 on 79" <sip:5382 at 192.168.72.79>;tag=68bb4eb6|SIP/2.0/UDP
192.168.72.150:34672;branch=z9hG4bK-d8754z-eafc60166305eaef-1---d8754z-;rpor
t=34672|1264546204|Teo Softphone release 2.5.4 stamp
55958||internal|Away|away|192.168.72.79|Away|away
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name =
'internal-ipv6' or sip_subscriptions.presence_hosts !=
sip_subscriptions.sub_to_host);
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'external' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
sqlite>

Thanks and Best Regards,
Jerry


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Monday, January 25, 2010 11:05 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


the notify will be instant after the publish
the notify you see are not triggered by the publish or they would be
instant.

Same drill, turn on presence debugging in sofia.conf.xml 
and look at the sql stmts and see why



On Mon, Jan 25, 2010 at 12:30 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:


Okay, I notice that if I reduce the Presence SUBSCRIBE Expires duration
(from 3600 seconds to 60 seconds), then the delay between PUBLISH's and
NOTIFY's is reduced, but FS still waits nearly 45 seconds to send the
NOTIFY's after it receives a PUBLISH.
 
Can a change be made in FS so that NOTIFYs are sent as a direct result of
receipt of the PUBLISH message?  And not tied to the SUBSCRIBE expiration?
I really don't want to configure all my phones to re-subscribe every 30 or
15 seconds.
 
Thanks and Best Regards,
Jerry
 


  _____  

From: RobertT [mailto:siniypin at gmail.com] 
Sent: Tuesday, December 29, 2009 12:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


You can try to reduce your registration time. 
I for one made my client apps send PUBLISH message every minute in addition
to reduced registration time.

Regards, Robert.


2009/12/28 Jerry Richards <jerry.richards at teotech.com>


Is there a setting to control how fast FS distributes presence changes to
subscribers?  Currently, it appears to take several minutes before I see
presence changes.  I would like to see them almost instantaneously, if
possible.

Thanks and Best Regards,
Jerry


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


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From dftoro at yahoo.com  Tue Jan 26 13:02:55 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 26 Jan 2010 13:02:55 -0800 (PST)
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <191c3a031001260827j544e1414oaec56f527e116eea@mail.gmail.com>
Message-ID: <955182.59161.qm@web33504.mail.mud.yahoo.com>

Hi, sorry, I explain better. Using \\\\ is also changed when path matches a character such as \s,\n... My alternative on Windows is to use '/' like path separator.


Diego Toro
http://lacarretade.blogspot.com/


--- On Tue, 1/26/10, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] mutiple playback files (unescape_char) Windows
> To: freeswitch-users at lists.freeswitch.org
> Date: Tuesday, January 26, 2010, 11:27 AM
> I didn't understand that 
> 
> On Tue, Jan 26, 2010 at 9:58 AM,
> Diego Toro <dftoro at yahoo.com>
> wrote:
> 
> Hi, using \\\\ the is changed also when
> there is a match with an escape character
> (\s,\n...)
> 
> 
> 
> Thank you
> 
> 
> 
> Diego Toro
> 
> http://lacarretade.blogspot.com/
> 
> 
> 
> 
> 
> --- On Mon, 1/25/10, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
> 
> 
> 
> > From: Anthony Minessale <anthony.minessale at gmail.com>
> 
> > Subject: Re: [Freeswitch-users]
> mutiple playback files (unescape_char) Windows
> 
> > To: freeswitch-users at lists.freeswitch.org
> 
> > Date: Monday, January 25, 2010, 12:20 PM
> 
> > its possible your
> string hits the parser
> 
> > more than once.
> 
> > try using 4 \
> 
> >
> 
> > \\\\sound
> 
> >
> 
> >
> 
> > On Sun, Jan 24, 2010 at 4:03 AM,
> 
> > Michael Jerris <mike at jerris.com>
> 
> > wrote:
> 
> >
> 
> > As noted on that bug, you should be
> 
> > able to either use \\ or / for the path
> separator
> 
> > there and it should work.
> 
> >
> 
> >
> 
> >
> 
> >
> 
> > Mike
> 
> >
> 
> >
> 
> >
> 
> > On Jan 22, 2010, at 9:18 AM, Diego Toro wrote:
> 
> >
> 
> >
> 
> >
> 
> > > Hi, with svn r16440 the problem persists, I
> creted a
> 
> > jira report http://jira.freeswitch.org/browse/LBSNDF-8
> 
> > this is a minor issue, but activing playback delimiter
> no
> 
> > audio file can be played. On FS the audio files are
> placed
> 
> > in the \sound\ directory, building the path
> on
> 
> > Windows would be \sound '\s' which is
> 
> > replaced by 'ound'.
> 
> >
> 
> >
> 
> > >
> 
> >
> 
> > > Thank you
> 
> >
> 
> > >
> 
> >
> 
> > > Diego Toro
> 
> >
> 
> > > http://lacarretade.blogspot.com/
> 
> >
> 
> > >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> > _______________________________________________
> 
> >
> 
> > FreeSWITCH-users mailing list
> 
> >
> 
> > FreeSWITCH-users at lists.freeswitch.org
> 
> >
> 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> >
> 
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> >
> 
> > http://www.freeswitch.org
> 
> >
> 
> >
> 
> >
> 
> >
> 
> > --
> 
> > Anthony Minessale II
> 
> >
> 
> > FreeSWITCH http://www.freeswitch.org/
> 
> > ClueCon http://www.cluecon.com/
> 
> >
> 
> > Twitter: http://twitter.com/FreeSWITCH_wire
> 
> >
> 
> > AIM: anthm
> 
> > MSN:anthony_minessale at hotmail.com
> 
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> 
> >
> 
> > IRC: irc.freenode.net
> 
> > #freeswitch
> 
> >
> 
> > FreeSWITCH Developer Conference
> 
> > sip:888 at conference.freeswitch.org
> 
> > iax:guest at conference.freeswitch.org/888
> 
> >
> 
> > googletalk:conf+888 at conference.freeswitch.org
> 
> > pstn:+19193869900
> 
> >
> 
> >
> 
> > -----Inline Attachment Follows-----
> 
> >
> 
> > _______________________________________________
> 
> > FreeSWITCH-users mailing list
> 
> > FreeSWITCH-users at lists.freeswitch.org
> 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> > http://www.freeswitch.org
> 
> >
> 
> 
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> 
> FreeSWITCH-users mailing list
> 
> FreeSWITCH-users at lists.freeswitch.org
> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> 
> IRC: irc.freenode.net
> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> 
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From camilin2212 at hotmail.com  Tue Jan 26 13:10:48 2010
From: camilin2212 at hotmail.com (juan camilo ospina quintero)
Date: Tue, 26 Jan 2010 16:10:48 -0500
Subject: [Freeswitch-users] Inbound sip invite from external gateway
Message-ID: <BAY143-W25E4C7D5F62FD42D7BA37BA15E0@phx.gbl>


hi to all

im already do the integration with. Freeswitch sends invite messages to sailfin, in sailfin there is a sip
servlet that acts as a proxy, this means it receives the invite from extension1000 and send the invite back
to freeswitch at extension 1001, but i get the freeswitch messages go to sailfin, but i dont get freeswitch
to understand sailfin messages.

there is my configuration for sending messages and for receiving messages

In /freeswitch/conf/dialplan/default.xml

    <extension name="redirect">
      <condition field="network_addr" expression="^192\.168\.2\.9$"/>    
      <condition field="destination_number" expression="^1(\d+)$">
          <action application="bridge" data="sofia/external/$0 at 127.0.0.1:5070"/>
      </condition>
    </extension>

this works fine, it redirects the messages to sailfin in 127.0.0.1


In /freeswitch/conf/dialplan/public.xml

    <extension name="receive">
      <condition field="network_addr" expression="^127\.0\.0\.1$"/>    
      <condition field="destination_number" expression="^1(\d+)$">
          <action application="bridge" data="sofia/internal/$0 at 192.168.2.9:5060"/>
      </condition>
    </extension>

this doesnt work, i also use <action application="transfer" data="$1001 XML default/> instead <action application="bridge" data="sofia/internal/$0 at 192.168.2.9:5060"/>

but still doesnt work, the invite that sailfin sends appears in the freeswitch console, but the 1001 extension doesnt get it
 		 	   		  
_________________________________________________________________
Keep your friends updated?even when you?re not signed in.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010
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From anthony.minessale at gmail.com  Tue Jan 26 13:21:54 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Jan 2010 15:21:54 -0600
Subject: [Freeswitch-users] Presence Change Distribution
In-Reply-To: <BE6F5600B8E54EE297963A8B08F70E5E@greyhawk.tonecommander.com>
References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com>
	<2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com>
	<26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com>
	<191c3a031001251104p55ba7009g9381841f7de56d65@mail.gmail.com>
	<BE6F5600B8E54EE297963A8B08F70E5E@greyhawk.tonecommander.com>
Message-ID: <191c3a031001261321v2e8ea21cm7da19e01a11b59f9@mail.gmail.com>

its sending a notify to them right away (line 174 of your PB)
the xml in the notify we send looks the same as what they sent except one
thing

They send: <rpid:away/>
We send: <rpid:busy/>

everybody who implements this seems to have their own idea of what to say
here.

This crazy xml presence crap is pure garbage so maybe that's it.



On Tue, Jan 26, 2010 at 3:02 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  Okay, I setup my FS to force SUBSCRIBE Expires to 3600 seconds.  Then I
> captured a FS console trace of a Bria softphone changing it's presence state
> from 'Busy' to 'Away' (see http://pastebin.freeswitch.org/11953) and
> observed that the subscribing Bria softphone did not update to 'Away'.  At
> the same time, I executed the sqlite3 app and pasted each of the 3 SQL
> select statements I saw in the FS console log, and pasted them below.  I'm
> new to sqlite3.  Do you see what my issue is?
>
> sqlite>
> sqlite> select
> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
> from sip_subscriptions left join sip_presence on
> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
> sip_subscriptions.sub_to_host=sip_presence.sip_host and
> sip_subscriptions.profile_name=sip_presence.profile_name) where
> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
> like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'internal' or
> sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
> sip|5382|192.168.72.79|5401|192.168.72.79|presence|"5382 on 79" <
> sip:5382 at 192.168.72.150:34672>|ZTQ2ZWQwZGRlZjRiNTdkYTJjNGM5NTgzOWIyNmIwZmU.|"5382
> on 79" <sip:5382 at 192.168.72.79 <sip%3A5382 at 192.168.72.79>>;tag=68bb4eb6|SIP/2.0/UDP
> 192.168.72.150:34672;branch=z9hG4bK-d8754z-eafc60166305eaef-1---d8754z-;rport=34672|1264546204|Teo
> Softphone release 2.5.4 stamp
> 55958||internal|Away|away|192.168.72.79|Away|away
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite> select
> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
> from sip_subscriptions left join sip_presence on
> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
> sip_subscriptions.sub_to_host=sip_presence.sip_host and
> sip_subscriptions.profile_name=sip_presence.profile_name) where
> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
> like '%192.168.72.79%') and (sip_subscriptions.profile_name =
> 'internal-ipv6' or sip_subscriptions.presence_hosts !=
> sip_subscriptions.sub_to_host);
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite>
> sqlite> select
> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
> from sip_subscriptions left join sip_presence on
> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
> sip_subscriptions.sub_to_host=sip_presence.sip_host and
> sip_subscriptions.profile_name=sip_presence.profile_name) where
> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
> like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'external' or
> sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
> sqlite>
> Thanks and Best Regards,
> Jerry
>
>  ------------------------------
> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
> *Sent:* Monday, January 25, 2010 11:05 AM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>
> the notify will be instant after the publish
> the notify you see are not triggered by the publish or they would be
> instant.
>
> Same drill, turn on presence debugging in sofia.conf.xml
> and look at the sql stmts and see why
>
>
> On Mon, Jan 25, 2010 at 12:30 PM, Jerry Richards <
> jerry.richards at teotech.com> wrote:
>
>>  Okay, I notice that if I reduce the Presence SUBSCRIBE Expires duration
>> (from 3600 seconds to 60 seconds), then the delay between PUBLISH's and
>> NOTIFY's is reduced, but FS still waits nearly 45 seconds to send the
>> NOTIFY's after it receives a PUBLISH.
>>
>> Can a change be made in FS so that NOTIFYs are sent as a direct result of
>> receipt of the PUBLISH message?  And not tied to the SUBSCRIBE expiration?
>> I really don't want to configure all my phones to re-subscribe every 30 or
>> 15 seconds.
>>
>> Thanks and Best Regards,
>> Jerry
>>
>>
>>  ------------------------------
>> *From:* RobertT [mailto:siniypin at gmail.com]
>> *Sent:* Tuesday, December 29, 2009 12:02 AM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>>
>>   You can try to reduce your registration time.
>> I for one made my client apps send PUBLISH message every minute in
>> addition to reduced registration time.
>>
>> Regards, Robert.
>>
>> 2009/12/28 Jerry Richards <jerry.richards at teotech.com>
>>
>>> Is there a setting to control how fast FS distributes presence changes to
>>> subscribers?  Currently, it appears to take several minutes before I see
>>> presence changes.  I would like to see them almost instantaneously, if
>>> possible.
>>>
>>> Thanks and Best Regards,
>>> Jerry
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Tue Jan 26 14:26:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Jan 2010 16:26:51 -0600
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <955182.59161.qm@web33504.mail.mud.yahoo.com>
References: <191c3a031001260827j544e1414oaec56f527e116eea@mail.gmail.com>
	<955182.59161.qm@web33504.mail.mud.yahoo.com>
Message-ID: <191c3a031001261426h7fd87e1fpf95824788d639557@mail.gmail.com>

please update again and try 4 slashes

you need 4 because the expand vars on the data="" will eat the 4 down to 2
then the splitter on ! will turn \\s into \s





On Tue, Jan 26, 2010 at 3:02 PM, Diego Toro <dftoro at yahoo.com> wrote:

> Hi, sorry, I explain better. Using \\\\ is also changed when path matches a
> character such as \s,\n... My alternative on Windows is to use '/' like path
> separator.
>
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
> --- On Tue, 1/26/10, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>
> > From: Anthony Minessale <anthony.minessale at gmail.com>
> > Subject: Re: [Freeswitch-users] mutiple playback files (unescape_char)
> Windows
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Tuesday, January 26, 2010, 11:27 AM
> > I didn't understand that
> >
> > On Tue, Jan 26, 2010 at 9:58 AM,
> > Diego Toro <dftoro at yahoo.com>
> > wrote:
> >
> > Hi, using \\\\ the is changed also when
> > there is a match with an escape character
> > (\s,\n...)
> >
> >
> >
> > Thank you
> >
> >
> >
> > Diego Toro
> >
> > http://lacarretade.blogspot.com/
> >
> >
> >
> >
> >
> > --- On Mon, 1/25/10, Anthony Minessale <anthony.minessale at gmail.com>
> > wrote:
> >
> >
> >
> > > From: Anthony Minessale <anthony.minessale at gmail.com>
> >
> > > Subject: Re: [Freeswitch-users]
> > mutiple playback files (unescape_char) Windows
> >
> > > To: freeswitch-users at lists.freeswitch.org
> >
> > > Date: Monday, January 25, 2010, 12:20 PM
> >
> > > its possible your
> > string hits the parser
> >
> > > more than once.
> >
> > > try using 4 \
> >
> > >
> >
> > > \\\\sound
> >
> > >
> >
> > >
> >
> > > On Sun, Jan 24, 2010 at 4:03 AM,
> >
> > > Michael Jerris <mike at jerris.com>
> >
> > > wrote:
> >
> > >
> >
> > > As noted on that bug, you should be
> >
> > > able to either use \\ or / for the path
> > separator
> >
> > > there and it should work.
> >
> > >
> >
> > >
> >
> > >
> >
> > >
> >
> > > Mike
> >
> > >
> >
> > >
> >
> > >
> >
> > > On Jan 22, 2010, at 9:18 AM, Diego Toro wrote:
> >
> > >
> >
> > >
> >
> > >
> >
> > > > Hi, with svn r16440 the problem persists, I
> > creted a
> >
> > > jira report http://jira.freeswitch.org/browse/LBSNDF-8
> >
> > > this is a minor issue, but activing playback delimiter
> > no
> >
> > > audio file can be played. On FS the audio files are
> > placed
> >
> > > in the \sound\ directory, building the path
> > on
> >
> > > Windows would be \sound '\s' which is
> >
> > > replaced by 'ound'.
> >
> > >
> >
> > >
> >
> > > >
> >
> > >
> >
> > > > Thank you
> >
> > >
> >
> > > >
> >
> > >
> >
> > > > Diego Toro
> >
> > >
> >
> > > > http://lacarretade.blogspot.com/
> >
> > >
> >
> > > >
> >
> > >
> >
> > >
> >
> > >
> >
> > >
> >
> > >
> >
> > > _______________________________________________
> >
> > >
> >
> > > FreeSWITCH-users mailing list
> >
> > >
> >
> > > FreeSWITCH-users at lists.freeswitch.org
> >
> > >
> >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > >
> >
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >
> >
> > > http://www.freeswitch.org
> >
> > >
> >
> > >
> >
> > >
> >
> > >
> >
> > > --
> >
> > > Anthony Minessale II
> >
> > >
> >
> > > FreeSWITCH http://www.freeswitch.org/
> >
> > > ClueCon http://www.cluecon.com/
> >
> > >
> >
> > > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > >
> >
> > > AIM: anthm
> >
> > > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >
> > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >
> > >
> >
> > > IRC: irc.freenode.net
> >
> > > #freeswitch
> >
> > >
> >
> > > FreeSWITCH Developer Conference
> >
> > > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >
> > > iax:guest at conference.freeswitch.org/888
> >
> > >
> >
> > > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >
> > > pstn:+19193869900
> >
> > >
> >
> > >
> >
> > > -----Inline Attachment Follows-----
> >
> > >
> >
> > > _______________________________________________
> >
> > > FreeSWITCH-users mailing list
> >
> > > FreeSWITCH-users at lists.freeswitch.org
> >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > http://www.freeswitch.org
> >
> > >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net
> > #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> >
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> >
> > -----Inline Attachment Follows-----
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jcasale at activenetwerx.com  Tue Jan 26 14:37:55 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 26 Jan 2010 22:37:55 +0000
Subject: [Freeswitch-users] Trouble w/ incoming calls from gateway
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90A4907@prato.activenetwerx.local>

Trying to track down what changed and I am sort of baffled.

I had an SPA3102's FXO port setup as a UA (1004) as I couldn't figure out
how to set it up as "gateway" from fs' perspective. Everything was working fine.

Now an incoming call gets dropped with this:
Hangup sofia/internal/XXXxxxXXXX at 192.168.13.1 [CS_NEW] [INCOMPATIBLE_DESTINATION]

Its set to dial ext 2000 (a group call) and being that its setup as a UA, 1004,
how does this develop the line above?

Thanks,
jlc



From sos at sokhapkin.dyndns.org  Tue Jan 26 14:48:57 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 26 Jan 2010 17:48:57 -0500
Subject: [Freeswitch-users] Trouble w/ incoming calls from gateway
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E90A4907@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E90A4907@prato.activenetwerx.local>
Message-ID: <201001261748.57503.sos@sokhapkin.dyndns.org>

INCOMPATIBLE_DESTINATION usually means codec-related problems. Perhaps SPA 
offers G729 codec only and the codec name in SPA settings is set to G729a?

On Tuesday 26 January 2010, Joseph L. Casale wrote:
> Trying to track down what changed and I am sort of baffled.
>
> I had an SPA3102's FXO port setup as a UA (1004) as I couldn't figure out
> how to set it up as "gateway" from fs' perspective. Everything was working
> fine.
>
> Now an incoming call gets dropped with this:
> Hangup sofia/internal/XXXxxxXXXX at 192.168.13.1 [CS_NEW]
> [INCOMPATIBLE_DESTINATION]
>
> Its set to dial ext 2000 (a group call) and being that its setup as a UA,
> 1004, how does this develop the line above?
>
> Thanks,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From msc at freeswitch.org  Tue Jan 26 14:49:11 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 26 Jan 2010 14:49:11 -0800
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <1b46b4e81001221322w5da04799s2ea86000c5c4a9a@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
	<e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
	<5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>
	<5d2828f1001201406p4154b98ald1af1c5c25f59337@mail.gmail.com>
	<1b46b4e81001210335l42baef16r8a2952aa5b92f6e6@mail.gmail.com>
	<87f2f3b91001211412h7aa0a84ageadb9557b869ca01@mail.gmail.com>
	<1b46b4e81001221322w5da04799s2ea86000c5c4a9a@mail.gmail.com>
Message-ID: <87f2f3b91001261449x3401f48eibae516d2b7abc8d4@mail.gmail.com>

Thanks for your contributions! They are much appreciated.
-MC

On Fri, Jan 22, 2010 at 1:22 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> No problem, here it is:
>
> - http://wiki.freeswitch.org/wiki/Call_retry_based_on_hangup_cause
>
> It is linked from your reference (
> http://wiki.freeswitch.org/wiki/Mod_lua#More_Samples).
>
> Sorry I didn't do it early, I hadn't seen your email.
>
> I also added another, more complete, example here (also linked):
>
> - http://wiki.freeswitch.org/wiki/Bridging_two_calls_with_retry
>
>
>
> On Thu, Jan 21, 2010 at 7:12 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Thu, Jan 21, 2010 at 3:35 AM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>
>>>
>>> On Wed, Jan 20, 2010 at 7:06 PM, Mike van Lammeren <
>>> mike at van.lammeren.net> wrote:
>>>
>>>> So, I've been reading about early media in the wiki, and have made a
>>>> little progress, which leads to more questions.
>>>>
>>>> I understand now why a call is considered connected before one person
>>>> has picked up the phone. I am also able to get my script to wait for the
>>>> phone to be picked up, by setting the ignore_early_media variable when
>>>> starting a new session, like this:
>>>>
>>>> customerSession =
>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/example.com/"
>>>> .. customerPhoneNumber)
>>>>
>>>>
>>>> After that line, the script waits for the other phone to be picked up.
>>>>
>>>> However, now I wonder what to do with calls that don't complete, get
>>>> busy signals, etc.
>>>>
>>>> What do people do in this case? The only related example I can find on
>>>> the web is for a javascript dialer, which doesn't address any of these
>>>> cases.
>>>>
>>>
>>>
>>> I guess it depends on what you want to do. For example I have a lua
>>> script very similar to what you describe, although there is no confirmation
>>> involved. Depending on the hangup cause the session gets, it might try
>>> redialing with a different gateway, try again or just hangup.
>>>
>>> Take a look here http://wiki.freeswitch.org/wiki/Hangup_causes to see
>>> what each hangup cause means. You don't need to have a special case for all
>>> of them, only the ones you are interested in.
>>>
>>> Here's an example in code which retries a call depending on the hangup
>>> cause. It retries max_retries1 times and alternates between 2 different
>>> gateways:
>>>
>>> session1 = null;
>>> max_retries1 = 3;
>>> retries = 0;
>>> ostr = "";
>>> repeat
>>>         retries = retries + 1;
>>>         if (retries % 2) then ostr = originate_str1;
>>>         else ostr = originate_str12; end
>>>         freeswitch.consoleLog("notice", "*********** Dialing Leg1: " ..
>>> ostr .. " - Try: "..retries.." ***********\n");
>>>         session1 = freeswitch.Session(ostr);
>>>         local hcause = session1:hangupCause();
>>>         freeswitch.consoleLog("notice", "*********** Leg1: " .. hcause ..
>>> " - Try: "..retries.." ***********\n");
>>> until not ((hcause == 'NO_ROUTE_DESTINATION' or hcause ==
>>> 'RECOVERY_ON_TIMER_EXPIRE' or hcause == 'INCOMPATIBLE_DESTINATION' or hcause
>>> == 'CALL_REJECTED' or hcause == 'NORMAL_TEMPORARY_FAILURE') and (retries <
>>> max_retriesl1))
>>>
>>>
>>> Note: originate_str1 and originate_str2 are two different dial strings
>>> for 2 different gateways.
>>>
>>>
>> Nicolas,
>>
>> This is really nice. Would you be willing to add this script and a brief
>> explanation to the wiki? You could create a whole new page and just link to
>> it from here: http://wiki.freeswitch.org/wiki/Mod_lua#More_Samples
>>
>> If you have any questions please let me know!
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jcasale at activenetwerx.com  Tue Jan 26 15:04:25 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 26 Jan 2010 23:04:25 +0000
Subject: [Freeswitch-users] Trouble w/ incoming calls from gateway
In-Reply-To: <201001261748.57503.sos@sokhapkin.dyndns.org>
References: <CA5A491E9DEFBE4CB777DE97E21575E90A4907@prato.activenetwerx.local>
	<201001261748.57503.sos@sokhapkin.dyndns.org>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90A49C7@prato.activenetwerx.local>

>INCOMPATIBLE_DESTINATION usually means codec-related problems. Perhaps SPA 
>offers G729 codec only and the codec name in SPA settings is set to G729a?

I think you narrowed it down:) After following the cisco docs for days to get
rid of echo unsuccessfully, we were horsing around and changed the codec from
G711u to G726-40 and the echo disappeared. Blank stares were met with a shrug.
The firmware was updated but no powerdown was done and a test yielded it still
working so we left it. The only change was a powerdown recently.

The output leading up to the hangup is:

2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [G726-40:96:8000:20]/[G7221:115:32000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [G726-40:96:8000:20]/[G7221:107:16000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [G726-40:96:8000:20]/[G722:9:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [G726-40:96:8000:20]/[PCMU:0:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [G726-40:96:8000:20]/[PCMA:8:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [G726-40:96:8000:20]/[GSM:3:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [NSE:100:8000:20]/[G7221:115:32000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [NSE:100:8000:20]/[G7221:107:16000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [NSE:100:8000:20]/[G722:9:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [NSE:100:8000:20]/[PCMU:0:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [NSE:100:8000:20]/[PCMA:8:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [NSE:100:8000:20]/[GSM:3:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3286 Set 2833 dtmf payload to 101
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [telephone-event:101:8000:20]/[G7221:115:32000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [telephone-event:101:8000:20]/[G7221:107:16000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [telephone-event:101:8000:20]/[G722:9:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [telephone-event:101:8000:20]/[PCMU:0:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20]
2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare [telephone-event:101:8000:20]/[GSM:3:8000:20]

So if I gather from this correctly, fs doesn't even have a codec to match against it?

Thanks a ton for the insight Sergey,
jlc



From sos at sokhapkin.dyndns.org  Tue Jan 26 15:22:26 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 26 Jan 2010 18:22:26 -0500
Subject: [Freeswitch-users] Trouble w/ incoming calls from gateway
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E90A49C7@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E90A4907@prato.activenetwerx.local>
	<201001261748.57503.sos@sokhapkin.dyndns.org>
	<CA5A491E9DEFBE4CB777DE97E21575E90A49C7@prato.activenetwerx.local>
Message-ID: <201001261822.26304.sos@sokhapkin.dyndns.org>

Do you have G726-40 enabled in global_codec_prefs in vars.xml?

On Tuesday 26 January 2010, Joseph L. Casale wrote:
> >INCOMPATIBLE_DESTINATION usually means codec-related problems. Perhaps SPA
> >offers G729 codec only and the codec name in SPA settings is set to G729a?
>
> I think you narrowed it down:) After following the cisco docs for days to
> get rid of echo unsuccessfully, we were horsing around and changed the
> codec from G711u to G726-40 and the echo disappeared. Blank stares were met
> with a shrug. The firmware was updated but no powerdown was done and a test
> yielded it still working so we left it. The only change was a powerdown
> recently.
>
> The output leading up to the hangup is:
>
> 2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [G726-40:96:8000:20]/[G7221:115:32000:20] 2010-01-26 16:04:36.682392
> [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [G726-40:96:8000:20]/[G7221:107:16000:20] 2010-01-26 16:04:36.682392
> [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [G726-40:96:8000:20]/[G722:9:8000:20] 2010-01-26 16:04:36.682392 [DEBUG]
> sofia_glue.c:3330 Audio Codec Compare [G726-40:96:8000:20]/[PCMU:0:8000:20]
> 2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [G726-40:96:8000:20]/[PCMA:8:8000:20] 2010-01-26 16:04:36.682392 [DEBUG]
> sofia_glue.c:3330 Audio Codec Compare [G726-40:96:8000:20]/[GSM:3:8000:20]
> 2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [NSE:100:8000:20]/[G7221:115:32000:20] 2010-01-26 16:04:36.682392 [DEBUG]
> sofia_glue.c:3330 Audio Codec Compare
> [NSE:100:8000:20]/[G7221:107:16000:20] 2010-01-26 16:04:36.682392 [DEBUG]
> sofia_glue.c:3330 Audio Codec Compare [NSE:100:8000:20]/[G722:9:8000:20]
> 2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [NSE:100:8000:20]/[PCMU:0:8000:20] 2010-01-26 16:04:36.682392 [DEBUG]
> sofia_glue.c:3330 Audio Codec Compare [NSE:100:8000:20]/[PCMA:8:8000:20]
> 2010-01-26 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [NSE:100:8000:20]/[GSM:3:8000:20] 2010-01-26 16:04:36.682392 [DEBUG]
> sofia_glue.c:3286 Set 2833 dtmf payload to 101 2010-01-26 16:04:36.682392
> [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [telephone-event:101:8000:20]/[G7221:115:32000:20] 2010-01-26
> 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [telephone-event:101:8000:20]/[G7221:107:16000:20] 2010-01-26
> 16:04:36.682392 [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [telephone-event:101:8000:20]/[G722:9:8000:20] 2010-01-26 16:04:36.682392
> [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [telephone-event:101:8000:20]/[PCMU:0:8000:20] 2010-01-26 16:04:36.682392
> [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [telephone-event:101:8000:20]/[PCMA:8:8000:20] 2010-01-26 16:04:36.682392
> [DEBUG] sofia_glue.c:3330 Audio Codec Compare
> [telephone-event:101:8000:20]/[GSM:3:8000:20]
>
> So if I gather from this correctly, fs doesn't even have a codec to match
> against it?
>
> Thanks a ton for the insight Sergey,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From jcasale at activenetwerx.com  Tue Jan 26 15:21:27 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 26 Jan 2010 23:21:27 +0000
Subject: [Freeswitch-users] Trouble w/ incoming calls from gateway
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E90A49C7@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E90A4907@prato.activenetwerx.local>
	<201001261748.57503.sos@sokhapkin.dyndns.org>
	<CA5A491E9DEFBE4CB777DE97E21575E90A49C7@prato.activenetwerx.local>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90A4A0B@prato.activenetwerx.local>

>So if I gather from this correctly, fs doesn't even have a codec to match against it?

Should have read the wiki before replying to that, from the codecs page and
my vars.xml I see it doesn't have that codec configured.

So are there any optimal codecs to choose for this device? I see there is
not an explicitly named module for 726, is that part of the default codecs?

Thanks,
jlc




From jcasale at activenetwerx.com  Tue Jan 26 15:40:00 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 26 Jan 2010 23:40:00 +0000
Subject: [Freeswitch-users] Trouble w/ incoming calls from gateway
In-Reply-To: <201001261822.26304.sos@sokhapkin.dyndns.org>
References: <CA5A491E9DEFBE4CB777DE97E21575E90A4907@prato.activenetwerx.local>
	<201001261748.57503.sos@sokhapkin.dyndns.org>
	<CA5A491E9DEFBE4CB777DE97E21575E90A49C7@prato.activenetwerx.local>
	<201001261822.26304.sos@sokhapkin.dyndns.org>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90A4A6F@prato.activenetwerx.local>

>Do you have G726-40 enabled in global_codec_prefs in vars.xml?

Now I do, but that produced an unbearable call? This 3102 seems
to be a pain, lots of forum posts related to this device hack on
it. Given I have a new TDM410p card, I wish I could get this working
reliably as when it does it works well. Oh well...

Thanks!
jlc



From msc at freeswitch.org  Tue Jan 26 15:54:22 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 26 Jan 2010 15:54:22 -0800
Subject: [Freeswitch-users] FreeSWITCH 1.0.5 and Buying Dinner For The Dev
	Team
Message-ID: <87f2f3b91001261554m2bfbbc45q21fc41c7d0a43b8@mail.gmail.com>

Hello all,

The FreeSWITCH development team is planning to meet in one place during the
week of February 8 to release version 1.0.5! We would like to invite
everyone to show their appreciation by donating a few dollars to help pay
for dinner for the development team. More information is available here:

http://www.freeswitch.org/node/234

Let's all show the team how much we appreciate them by giving them a
well-deserved dinner!

-Michael
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From camilin2212 at hotmail.com  Tue Jan 26 17:30:24 2010
From: camilin2212 at hotmail.com (juan camilo ospina quintero)
Date: Tue, 26 Jan 2010 20:30:24 -0500
Subject: [Freeswitch-users] External Profile Problem
Message-ID: <BAY143-W23698E1AF0CEBDBC04EFBDA15D0@phx.gbl>


Hi, 

im trying to establish a simple conference using freeswitch and sailfin, sailfin is
and application server that works with SipSevlets.
the all thing works as follow.

two softphone register with freeswitch, extension 1000 and 1001
1000 sends and invite to 1001, this invite goes to sailfin, i use this

    <extension name="redirect">
      <condition field="network_addr" expression="^192\.168\.2\.9$"/>    
      <condition field="destination_number" expression="^1(\d+)$">
    <action application="bridge" data="sofia/external/$0 at 127.0.0.1:5070"/>
      </condition>
    </extension>

this is in /freeswitch/conf/dialplan/default.xml


this far all goes well, the servlet receives the invite, and sends back the invite to freeswitch, i put this

    <extension name="receive">
      <condition field="network_addr" expression="^192\.168\.2\.9$"/>
      <condition field="destination_number" expression="^1(\d+)$">
        <action application="transfer" data="$0 XML default"/>
      </condition>
    </extension>

in /freeswitch/conf/dialplan/public.xml, but freeswitch returns this


------------------------------------------------------------------------
2010-01-26 20:14:29.512927 [NOTICE] switch_channel.c:602 New Channel sofia/external/1000 at 192.168.2.9 [5177e93a-0ae1-11df-afc9-db39c681a2f1]
2010-01-26 20:14:29.512927 [NOTICE] sofia.c:3527 Hangup sofia/external/1000 at 192.168.2.9 [CS_NEW] [MANDATORY_IE_MISSING]
send 632 bytes to udp/[192.168.2.9]:5070 at 01:14:29.517927:
   ------------------------------------------------------------------------
   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/UDP 192.168.153.1:5070;branch=z9hG4bKdaacdd64d693615c451ab9db43f9c71c2626;received=192.168.2.9
   From: <sip:1000 at 192.168.2.9>;tag=g4xfbi12-3
   To: <sip:1001 at 192.168.2.9:5080>;tag=4r91165pvcycB
   Call-Id: 192.168.153.1_3_3990383226484831353
   Cseq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING"
   Content-Length: 0
   
   ------------------------------------------------------------------------
2010-01-26 20:14:29.525646 [NOTICE] switch_core_session.c:1086 Session 9 (sofia/external/1000 at 192.168.2.9) Ended
2010-01-26 20:14:29.525646 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/1000 at 192.168.2.9 [CS_DESTROY]

i dont understand why i doesnt work if in public.xml, i tell to transfer the call if extension starts with 1 and the caller ip address is 192.168.2.9

please if someone can help me
 		 	   		  
_________________________________________________________________
Windows Live: Keep your friends up to date with what you do online.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_1:092010
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From frank at carmickle.com  Tue Jan 26 19:09:29 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Tue, 26 Jan 2010 22:09:29 -0500
Subject: [Freeswitch-users] External Profile Problem
In-Reply-To: <BAY143-W23698E1AF0CEBDBC04EFBDA15D0@phx.gbl>
References: <BAY143-W23698E1AF0CEBDBC04EFBDA15D0@phx.gbl>
Message-ID: <20100127030929.GC3841@base.carmickle.com>

On Tue, Jan 26, juan camilo ospina quintero wrote:
> 
> Hi, 
> 
> im trying to establish a simple conference using freeswitch and sailfin, sailfin is
> and application server that works with SipSevlets.
> the all thing works as follow.
> 
> two softphone register with freeswitch, extension 1000 and 1001
> 1000 sends and invite to 1001, this invite goes to sailfin, i use this
> 
>     <extension name="redirect">
>       <condition field="network_addr" expression="^192\.168\.2\.9$"/>    
>       <condition field="destination_number" expression="^1(\d+)$">
>     <action application="bridge" data="sofia/external/$0 at 127.0.0.1:5070"/>

And what is the external profile listening on?  Probably not the loopback address.  Set up another profile listening on 127.0.0.1 and bridge to that.

I could be off base here because you haven't given us very much info about your freeswitch configurations.

--FC



From dftoro at yahoo.com  Tue Jan 26 19:16:24 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 26 Jan 2010 19:16:24 -0800 (PST)
Subject: [Freeswitch-users] compilation error on fs_cli (Windows)
	(latest version)
In-Reply-To: <3072AE90-DDEA-4FD0-9B4B-140051730073@avgs.ca>
Message-ID: <761142.19511.qm@web33506.mail.mud.yahoo.com>


yes, code analyzer is active. when I turn it off fs_cli project compiled fine. Before, this project compiled fine, why I need turn off analyzer code now ?

Thank you

Diego Toro
http://lacarretade.blogspot.com/


--- On Tue, 1/26/10, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> From: Mathieu Rene <mrene_lists at avgs.ca>
> Subject: Re: [Freeswitch-users] compilation error on fs_cli (Windows) (latest version)
> To: freeswitch-users at lists.freeswitch.org
> Date: Tuesday, January 26, 2010, 11:45 AM
> Looks like the code analyzer is
> running, this is normally turned off? 
> when you do a normal build, turn it off and try again.
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 
> 
> 
> On 26-Jan-10, at 10:55 AM, Diego Toro wrote:
> 
> > Hi, I have compilation error "error C2220" on fs_cli
> project on? 
> > Windows using VS2008.
> >
> > FS: latest version (2010/01/26)
> > VS: VS2008
> > SO: Windows 7
> >
> > VS2008 Error log:
> >
> > Error??? 1??? error
> C2220: warning treated as error - no 'object' file? 
> > generated???
> g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl 
> > \fs_cli.c??? 106???
> fs_cli
> >
> > Warning??? 2??? warning
> C6385: Invalid data: accessing 'global_profile- 
> > >console_fnkeys', the readable size is '48' bytes,
> but '-4' bytes? 
> > might be read: Lines: 86, 88, 90???
> g:\ftp\incoming\fs 
> >
> \freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c???
> 90??? fs_cli
> >
> > Warning??? 3??? warning
> C6246: Local declaration of 'p' hides declaration? 
> > of the same name in outer scope. For additional
> information, see? 
> > previous declaration at line '844' of
> 'g:\ftp\incoming\fs 
> > \freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c':
> Lines: 844??? g:\ftp 
> >
> \incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c???
> 895?????
> > fs_cli
> >
> > Warning??? 4??? warning
> C6011: Dereferencing NULL pointer 'cursor': Lines:? 
> > 839, 840, 841, 842, 843, 844, 846, 849, 853, 857, 868,
> 870, 871, 884?????
> >
> g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c?????
> > 884??? fs_cli
> >
> >
> > Thank you
> >
> > Diego Toro
> > http://lacarretade.blogspot.com/
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From airsignal at wavecable.com  Tue Jan 26 00:57:34 2010
From: airsignal at wavecable.com (Airsignal)
Date: Tue, 26 Jan 2010 00:57:34 -0800
Subject: [Freeswitch-users] NAT keep alive
Message-ID: <8657153B18DB4AB3B17DB7A4BAAFF862@Terminal>

Good Evening:

I am trying to get my switch to send keep alives to the ata's in the field.

<param name="nat-options-ping" value="true"/>   seems to be meant for this.

where should it go?  I can little documentation discussing this...


Thanks!
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From matomoya at yahoo.co.jp  Tue Jan 26 04:07:10 2010
From: matomoya at yahoo.co.jp (=?ISO-2022-JP?B?GyRCJF4kRCRQJGkbKEIgGyRCJEgkYiRkGyhC?=)
Date: Tue, 26 Jan 2010 21:07:10 +0900 (JST)
Subject: [Freeswitch-users] mod_spidermonkey memory leak?
Message-ID: <20100126120711.92321.qmail@web10512.mail.ogk.yahoo.co.jp>

I have tested the SVN trunk 15341. 
When the following scripts are executed, it seems to do the memory leak. 
Is the mistake found in the following scripts?
Or Is there a problem in mod_spidermonkey?

-- test script --
session.execute("ring_ready");
session.answer();
session.setVariable("ringback", "%(1000, 2000, 440, 460)");

var bleg = new Session();
var sound_wav = "sounds/test01.wav";
var sound_leg = "both";
var op = "api_on_answer=uuid_broadcast "+session.uuid+" "+sound_wav+"
"+sound_leg;
var ret = bleg.originate(session, "{"+op+"}" +
"sofia/gateway/profile0_gateway1/1000");
if(!session.ready()){
  return;
}


if(!ret){ // bleg not answered.
  var sound_wav = "sounds/test02.wav";
  session.streamFile(sound_wav);
  if(session.ready()){
    session.hangup();
  }
  return;
}

if(bleg.ready()){
  bridge(session, bleg);
}




From codeghar at gmail.com  Tue Jan 26 17:38:17 2010
From: codeghar at gmail.com (Code Ghar)
Date: Tue, 26 Jan 2010 19:38:17 -0600
Subject: [Freeswitch-users] Replace Internal IP with External IP in From
	Header
Message-ID: <4571ff701001261738w4f51b33dqee1d19b8d0e2236d@mail.gmail.com>

I followed the example in Freeswitch behind NAT (
http://wiki.freeswitch.org/wiki/NAT_Traversal#Freeswitch_behind_NAT). In the
Contact header of invite sent to an external gateway, I see
sip:extension at ExternalIP:port but in the From header I see
sip:extension at InternalIP. How can I change the From header of SIP message so
that it displays the external IP instead of internal IP? The reason for
doing this is that the external gateway authenticates and authorizes call
based on the IP in From header. They expect an external IP and not an
internal IP.
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From matsubara_tomoya at intec.co.jp  Tue Jan 26 19:35:33 2010
From: matsubara_tomoya at intec.co.jp (Tomoya Matsubara)
Date: Wed, 27 Jan 2010 12:35:33 +0900
Subject: [Freeswitch-users] Question about javascript
Message-ID: <20100127123533.602850e1.matsubara_tomoya@intec.co.jp>

Hello,

When the following scripts were tested, it seems to do the memory leak. 
Please teach when there is a problem in this script. 

-- test script --
session.execute("ring_ready");
session.answer();
session.setVariable("ringback", "%(1000, 2000, 440, 460)");

var bleg = new Session();
var sound_wav = "sounds/test01.wav";
var sound_leg = "both";
var op = "api_on_answer=uuid_broadcast "+session.uuid+" "+sound_wav+" "+sound_leg;
var ret = bleg.originate(session, "{"+op+"}" + "sofia/gateway/profile0_gateway1/1000");
if(!session.ready()){
  return;
}

if(!ret){ // bleg not answered.
  var sound_wav = "sounds/test02.wav";
  session.streamFile(sound_wav);
  if(session.ready()){
    session.hangup();
  }
  return;
}

if(bleg.ready()){
  bridge(session, bleg);
}



From j4szczur at gmail.com  Tue Jan 26 15:03:13 2010
From: j4szczur at gmail.com (michal kalinowski)
Date: Wed, 27 Jan 2010 00:03:13 +0100
Subject: [Freeswitch-users] mod_fax ECM
Message-ID: <7c74f5761001261503g457879f9x1447ee8dab1e9104@mail.gmail.com>

Hello,

What mean ECM and for what is used ?

I found in my fax.conf parameter:
<param name="use-ecm"           value="true"/>

BR,
Micha?



From camilin2212 at hotmail.com  Tue Jan 26 20:06:12 2010
From: camilin2212 at hotmail.com (juan camilo ospina quintero)
Date: Tue, 26 Jan 2010 23:06:12 -0500
Subject: [Freeswitch-users] External Profile Problem
In-Reply-To: <20100127030929.GC3841@base.carmickle.com>
References: <BAY143-W23698E1AF0CEBDBC04EFBDA15D0@phx.gbl>,
	<20100127030929.GC3841@base.carmickle.com>
Message-ID: <BAY143-W20A3102CC2FF36EF1C170FA15D0@phx.gbl>


Hi

This works fine

<extension name="redirect">
 <condition field="network_addr" expression="^192\.168\.2\.9$"/>    
 <condition field="destination_number" expression="^1(\d+)$">
 <action application="bridge" data="sofia/external/$0 at 127.0.0.1:5070"/>
</extension>

this redirects from freeswitch to sailfin (127.0.0.1:5070), and is in default.xml, in the dialplan.

the problem is this

    <extension name="receive">
      <condition field="network_addr" expression="^127\.0\.0\.1$"/>
      <condition field="destination_number" expression="^1(\d+)$">
        <action application="transfer" data="$0 XML default"/>
      </condition>
    </extension>
this doesnt work, this configuration can be found in public.xml in the dialplan, the idea of
this is that when a sip invite comes from sailfin (127.0.0.1) transfer the invite to the destination number

the both configurations above are the only configuration i have change from  the default instalation of 
freeswitch.

i would like to have some hep with this thanks

here is the trace log again

2010-01-26 20:14:29.512927 [NOTICE] switch_channel.c:602 New Channel sofia/external/1000 at 192.168.2.9 [5177e93a-0ae1-11df-afc9-db39c681a2f1]
2010-01-26 20:14:29.512927 [NOTICE] sofia.c:3527 Hangup sofia/external/1000 at 192.168.2.9 [CS_NEW] [MANDATORY_IE_MISSING]
send 632 bytes to udp/[192.168.2.9]:5070 at 01:14:29.517927:
   ------------------------------------------------------------------------
   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/UDP 192.168.153.1:5070;branch=z9hG4bKdaacdd64d693615c451ab9db43f9c71c2626;received=192.168.2.9
   From: <sip:1000 at 192.168.2.9>;tag=g4xfbi12-3
   To: <sip:1001 at 192.168.2.9:5080>;tag=4r91165pvcycB
   Call-Id: 192.168.153.1_3_3990383226484831353
   Cseq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING"
   Content-Length: 0
   
   ------------------------------------------------------------------------
2010-01-26 20:14:29.525646 [NOTICE] switch_core_session.c:1086 Session 9 (sofia/external/1000 at 192.168.2.9) Ended
2010-01-26 20:14:29.525646 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/1000 at 192.168.2.9 [CS_DESTROY]



> Date: Tue, 26 Jan 2010 22:09:29 -0500
> From: frank at carmickle.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] External Profile Problem
> 
> On Tue, Jan 26, juan camilo ospina quintero wrote:
> > 
> > Hi, 
> > 
> > im trying to establish a simple conference using freeswitch and sailfin, sailfin is
> > and application server that works with SipSevlets.
> > the all thing works as follow.
> > 
> > two softphone register with freeswitch, extension 1000 and 1001
> > 1000 sends and invite to 1001, this invite goes to sailfin, i use this
> > 
> >     <extension name="redirect">
> >       <condition field="network_addr" expression="^192\.168\.2\.9$"/>    
> >       <condition field="destination_number" expression="^1(\d+)$">
> >     <action application="bridge" data="sofia/external/$0 at 127.0.0.1:5070"/>
> 
> And what is the external profile listening on?  Probably not the loopback address.  Set up another profile listening on 127.0.0.1 and bridge to that.
> 
> I could be off base here because you haven't given us very much info about your freeswitch configurations.
> 
> --FC
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_3:092010
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From jcasale at activenetwerx.com  Tue Jan 26 20:51:13 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 27 Jan 2010 04:51:13 +0000
Subject: [Freeswitch-users] mod_fax ECM
In-Reply-To: <7c74f5761001261503g457879f9x1447ee8dab1e9104@mail.gmail.com>
References: <7c74f5761001261503g457879f9x1447ee8dab1e9104@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90A4F15@prato.activenetwerx.local>

>What mean ECM and for what is used ?
>
>I found in my fax.conf parameter:
><param name="use-ecm"           value="true"/>

Error correction mode, first hit on google -> "fax ecm" :)



From freeswitch-users at digitaldan.com  Tue Jan 26 21:15:37 2010
From: freeswitch-users at digitaldan.com (Dan)
Date: Tue, 26 Jan 2010 22:15:37 -0700 (MST)
Subject: [Freeswitch-users] FreeSWITCH 1.0.5 and Buying Dinner For The
 Dev	Team
In-Reply-To: <87f2f3b91001261554m2bfbbc45q21fc41c7d0a43b8@mail.gmail.com>
Message-ID: <14340419.35.1264569337594.JavaMail.root@zimbra>

bon appetit! 
----- Original Message ----- 
From: "Michael Collins" <msc at freeswitch.org> 
To: freeswitch-users at lists.freeswitch.org, freeswitch-dev at lists.freeswitch.org 
Sent: Tuesday, January 26, 2010 4:54:22 PM 
Subject: [Freeswitch-users] FreeSWITCH 1.0.5 and Buying Dinner For The Dev Team 

Hello all, 

The FreeSWITCH development team is planning to meet in one place during the week of February 8 to release version 1.0.5! We would like to invite everyone to show their appreciation by donating a few dollars to help pay for dinner for the development team. More information is available here: 

http://www.freeswitch.org/node/234 

Let's all show the team how much we appreciate them by giving them a well-deserved dinner! 

-Michael 

_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 
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From jason at jasonjgw.net  Tue Jan 26 22:28:22 2010
From: jason at jasonjgw.net (Jason White)
Date: Wed, 27 Jan 2010 17:28:22 +1100
Subject: [Freeswitch-users] External Profile Problem
In-Reply-To: <BAY143-W20A3102CC2FF36EF1C170FA15D0@phx.gbl>
References: <BAY143-W23698E1AF0CEBDBC04EFBDA15D0@phx.gbl>
	<20100127030929.GC3841@base.carmickle.com>
	<BAY143-W20A3102CC2FF36EF1C170FA15D0@phx.gbl>
Message-ID: <20100127062822.GA27365@jdc.jasonjgw.net>

juan camilo ospina quintero <camilin2212 at hotmail.com> wrote:
> 2010-01-26 20:14:29.512927 [NOTICE] switch_channel.c:602 New Channel sofia/external/1000 at 192.168.2.9 [5177e93a-0ae1-11df-afc9-db39c681a2f1]
> 2010-01-26 20:14:29.512927 [NOTICE] sofia.c:3527 Hangup sofia/external/1000 at 192.168.2.9 [CS_NEW] [MANDATORY_IE_MISSING]

I haven't been following the thread, but the above error is usually an
authentication problem, for example, the destination isn't accepting the call.
If the destination is a FreeSWITCH instance, make sure that auth-calls is set
appropriately and that the ACLs aren't causing any problems.




From Prometheus001 at gmx.net  Wed Jan 27 02:14:07 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 27 Jan 2010 11:14:07 +0100
Subject: [Freeswitch-users] Wrong RTP port submitted?
Message-ID: <4B6011EF.6090706@gmx.net>

I have defined the rtp port range for 12000-12100 in switch.conf.xml.
However Freeswitch is offering a port 48320 in the invite message. The
result is, that the incoming RTP stream is blocked by the firewall (I
can see a reject for UDP 48320).
Any hint how to solve this?

Best regards Peter

See config and invite message:
    <!--RTP port range -->
    <param name="rtp-start-port" value="12000"/>-->
    <param name="rtp-end-port" value="12100"/>-->

Invite:
INVITE sip:027xxxxxxxx at sip.itsp.de SIP/2.0.
Via: SIP/2.0/UDP 217.24.xx.xxx:5080;rport;branch=z9hG4bKjD923NvctXaFm.
Max-Forwards: 69.
From: "0608xxxxxxx"
<sip:4918xxxxxxxxxxxx at sip.itsp.de;transport=udp>;tag=0Kp4tvU44UmXp.
To: <sip:027xxxxxxxx at sip.itsp.de>.
Call-ID: 30c86b94-85ca-122d-f88e-080027e51f59.
CSeq: 126174137 INVITE.
Contact: <sip:gw+tgnet3 at 217.24.xx.xxx:5080;transport=udp;gw=tgnet3>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16032.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 320.
P-Key-Flags: keys="3".
X-FS-Support: update_display.
Remote-Party-ID: "0608xxxxxxx"
<sip:0608xxxxxxx at sip.itsp.de>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1264536651 1264536652 IN IP4 217.24.xx.xxx.
s=FreeSWITCH.
c=IN IP4 217.24.xx.xxx.
t=0 0.
m=audio 48320 RTP/AVP 8 0 98 3 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:98 SPEEX/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.



From codecomplete at free.fr  Wed Jan 27 04:38:20 2010
From: codecomplete at free.fr (Fred-145)
Date: Wed, 27 Jan 2010 04:38:20 -0800 (PST)
Subject: [Freeswitch-users] Equivalent to Asterisk's "directrtpsetup=yes"?
Message-ID: <27338355.post@talk.nabble.com>


Hello

Thanks to "directrtpsetup=yes", it appears that Asterisk can act as an SIP
proxy and a PBX at the same time, ie. RTP packets will flow directly between
the two SIP end-points with Asterisk still being able to provide PBX
services like call transfer, MoH, etc.
The point is to lower latency for packets to reach their final destination
and lower CPU load on the Freeswitch server.

Does Freeswitch offer the same feature, or must RTP packets always go
through the Freeswitch servers?

Thank you.
-- 
View this message in context: http://old.nabble.com/Equivalent-to-Asterisk%27s-%22directrtpsetup%3Dyes%22--tp27338355p27338355.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From codecomplete at free.fr  Wed Jan 27 04:38:50 2010
From: codecomplete at free.fr (Fred-145)
Date: Wed, 27 Jan 2010 04:38:50 -0800 (PST)
Subject: [Freeswitch-users] Equivalent to Asterisk's "directrtpsetup=yes"?
Message-ID: <27338355.post@talk.nabble.com>


Hello

Thanks to "directrtpsetup=yes", it appears that Asterisk can act as an SIP
proxy and a PBX at the same time, ie. RTP packets will flow directly between
the two SIP end-points with Asterisk still being able to provide PBX
services like call transfer, MoH, etc.
The point is lower latency for packets to reach their final destination and
lower CPU load on the Freeswitch server.

Does Freeswitch offer the same feature, or must RTP packets always go
through the Freeswitch servers?

Thank you.
-- 
View this message in context: http://old.nabble.com/Equivalent-to-Asterisk%27s-%22directrtpsetup%3Dyes%22--tp27338355p27338355.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From Russell.Mosemann at cune.org  Wed Jan 27 04:47:28 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Wed, 27 Jan 2010 06:47:28 -0600
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
In-Reply-To: <27338355.post@talk.nabble.com>
References: <27338355.post@talk.nabble.com>
Message-ID: <9BB5BC82B9F54466ACFA5BA610669FD7@cune.pri>

Fred-145 was wondering:

> Does Freeswitch offer the same feature, or must RTP packets always go
> through the Freeswitch servers?

http://wiki.freeswitch.org/wiki/Bypass_Media

> Thank you.

You're welcome.

--
Russell Mosemann




From rob4manhere at gmail.com  Wed Jan 27 04:49:43 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Wed, 27 Jan 2010 06:49:43 -0600
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
In-Reply-To: <27338355.post@talk.nabble.com>
References: <27338355.post@talk.nabble.com>
Message-ID: <C70CDDCF-AB2E-4D10-A4EC-0E6205193A31@gmail.com>

Hi Fred,

Check out bypass_media mode:
http://wiki.freeswitch.org/wiki/Bypass_Media

Cheers,
Rob

On Jan 27, 2010, at 6:38 AM, Fred-145 wrote:

>
> Hello
>
> Thanks to "directrtpsetup=yes", it appears that Asterisk can act as  
> an SIP
> proxy and a PBX at the same time, ie. RTP packets will flow directly  
> between
> the two SIP end-points with Asterisk still being able to provide PBX
> services like call transfer, MoH, etc.
> The point is lower latency for packets to reach their final  
> destination and
> lower CPU load on the Freeswitch server.
>
> Does Freeswitch offer the same feature, or must RTP packets always go
> through the Freeswitch servers?
>
> Thank you.
> -- 
> View this message in context: http://old.nabble.com/Equivalent-to-Asterisk%27s-%22directrtpsetup%3Dyes%22--tp27338355p27338355.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From sos at sokhapkin.dyndns.org  Wed Jan 27 04:50:16 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 27 Jan 2010 07:50:16 -0500
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
In-Reply-To: <27338355.post@talk.nabble.com>
References: <27338355.post@talk.nabble.com>
Message-ID: <201001270750.16772.sos@sokhapkin.dyndns.org>

set bypass_media=true

On Wednesday 27 January 2010, Fred-145 wrote:
> Hello
>
> Thanks to "directrtpsetup=yes", it appears that Asterisk can act as an SIP
> proxy and a PBX at the same time, ie. RTP packets will flow directly
> between the two SIP end-points with Asterisk still being able to provide
> PBX services like call transfer, MoH, etc.
> The point is lower latency for packets to reach their final destination and
> lower CPU load on the Freeswitch server.
>
> Does Freeswitch offer the same feature, or must RTP packets always go
> through the Freeswitch servers?
>
> Thank you.





From mcampbellsmith at gmail.com  Wed Jan 27 05:06:52 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 28 Jan 2010 00:06:52 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
Message-ID: <33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>

Thanks guys.  I have this working except for one user who is
registered like this:

Call-ID:        2ff39277-fd9a6ab1 at 10.0.0.1
User:           2000 at 192.168.1.120
Contact:        2000
<sip:2000 at 124.xxx.xxx.xxx:5075;transport=tls;fs_nat=yes;fs_path=sip%3A2000%40124.xxx.xxx.xxx%3A10281%3Btransport%3Dtls>
Agent:          Linksys/SPA3102-5.1.10(GW)
Status:         Registered(TLS-NAT)(unknown) EXP(2010-01-28 00:29:34)
Host:           freeswitch
IP:             124.xxx.xxx.xxx
Port:           10281
Auth-User:      2000
Auth-Realm:     mydns.dyndns.org
MWI-Account:    2000 at 192.168.1.120

When I do the following commands via the telnet socket, no notify
command is sent to user 2000:

sendevent NOTIFY
profile: internal
content-type: application/simple-message-summary
event-string: reboot_now
user: 2000
host: 192.168.1.120
content-length: 0

However, if I do exactly the same thing with user 2001 it works.  2001
is registered as:

Contact:        2001 <sip:2001 at 124.yyy.yyy.yyy:5066;transport=tls>

Any ideas why that would be?

On Mon, Jan 25, 2010 at 11:50 AM, Ognjen Seslija <oseslija at gmail.com> wrote:
> The phone is asking FS to authenticate prior then accepting a NOTIFY from
> it.
> The authentication of notify's from spa endpoints work (afaik) only with
> Linksys SPA9000 PBX , and FS doesn't have the code for that for obvious
> reasons.
> If you have SPA9000 maybe you can collect SIP traces.
>
> Ognjen
>
>
>
>
>
> On Mon, Jan 25, 2010 at 1:29 AM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>>
>> Hi Ognjen,
>>
>> Thanks for the tip on the resync under the EXT tab. ?It now works
>> using mod_event_socket and the following:
>>
>> sendevent NOTIFY
>> profile: internal
>> event-string: resync
>> user: 1000
>> host: 192.168.1.121
>> content-type: application/simple-message-summary
>>
>> However, if AUTH is required, why does FS send the wrong information to
>> the SPA?
>>
>> On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija <oseslija at gmail.com>
>> wrote:
>> > You? should not authenticate those NOTIFYs (this will work only with
>> > SPA9000
>> > afaik). The option to change for this is in EXT tabs:
>> >
>> > Auth Resync-Reboot: No
>> >
>> > Also, FSs code will do a reboot of a phone, not resync (it sends
>> > reboot_now
>> > event). For that to work a patch is required.
>> >
>> > I've just tried to reboot my 942 (rev 16506) and it definitely works.
>> >
>> > Regards,
>> > Ognjen
>> >
>> >
>> > On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
>> > <mcampbellsmith at gmail.com> wrote:
>> >>
>> >> Thanks Brian.. this still does not work. ?Maybe I need to open a Jira?
>> >> ?Notice the username in the authorization field. ?It should be 1000.
>> >>
>> >> Cheers
>> >> Mark
>> >>
>> >> freeswitch at internal> sofia status profile internal user
>> >> 1000 at 192.168.1.120
>> >>
>> >> Registrations:
>> >>
>> >>
>> >> =================================================================================================
>> >> Call-ID: ? ? ? ?bd783b73-66877627 at 192.168.1.121
>> >> User: ? ? ? ? ? 1000 at 192.168.1.120
>> >> Contact: ? ? ? ?1000 <sip:1000 at 192.168.1.121:5060>
>> >> Agent: ? ? ? ? ?Linksys/PAP2T-5.1.6(LS)
>> >> Status: ? ? ? ? Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
>> >> Host: ? ? ? ? ? freeswitch
>> >> IP: ? ? ? ? ? ? 192.168.1.121
>> >> Port: ? ? ? ? ? 5060
>> >> Auth-User: ? ? ?1000
>> >> Auth-Realm: ? ? 192.168.1.120
>> >> MWI-Account: ? ?1000 at 192.168.1.120
>> >>
>> >>
>> >>
>> >> =================================================================================================
>> >>
>> >> freeswitch at internal> sofia profile internal flush_inbound_reg
>> >> bd783b73-66877627 at 192.168.1.121 reboot
>> >> +OK rebooting all registrations matching specified call_id
>> >>
>> >> freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060 at
>> >> 23:55:49.012627:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> >> ? Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
>> >> ? Max-Forwards: 70
>> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> ? To: <sip:1000 at 192.168.1.120>
>> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> ? CSeq: 126070338 NOTIFY
>> >> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> >> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> ? Supported: timer, precondition, path, replaces
>> >> ? Event: reboot_now
>> >> ? Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> >> include-session-description, presence.winfo, message-summary, refer
>> >> ? Subscription-State: terminated;reason=timeout
>> >> ? Content-Type: application/simple-message-summary
>> >> ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? SIP/2.0 401 Unauthorized
>> >> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> ? CSeq: 126070338 NOTIFY
>> >> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
>> >> ? Server: Linksys/PAP2T-5.1.6(LS)
>> >> ? WWW-Authenticate: Digest realm="192.168.1.120", nonce="8e54805b",
>> >> qop="auth", algorithm=md5
>> >> ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> >> ? Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
>> >> ? Max-Forwards: 70
>> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> ? To: <sip:1000 at 192.168.1.120>
>> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> ? CSeq: 126070339 NOTIFY
>> >> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> >> ? Expires: 3590
>> >> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> ? Supported: timer, precondition, path, replaces
>> >> ? Event: reboot_now
>> >> ? Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> >> include-session-description, presence.winfo, message-summary, refer
>> >> ? Subscription-State: terminated;reason=timeout
>> >> ? Authorization: Digest username="1115633124", realm="192.168.1.120",
>> >> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
>> >> uri="sip:1000 at 192.168.1.121:5060",
>> >> response="747b4d04544c84535dbbd987f2999ca7", qop=auth, nc=00000001
>> >> ? Content-Type: application/simple-message-summary
>> >> ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
>> >>
>> >> ------------------------------------------------------------------------
>> >> ? SIP/2.0 401 Unauthorized
>> >> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> ? CSeq: 126070339 NOTIFY
>> >> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
>> >> ? Server: Linksys/PAP2T-5.1.6(LS)
>> >> ? WWW-Authenticate: Digest realm="192.168.1.120", nonce="5339c7ba",
>> >> qop="auth", algorithm=md5
>> >> ? Content-Length: 0
>> >>
>> >>
>> >> ------------------------------------------------------------------------
>> >>
>> >> On Mon, Jan 25, 2010 at 4:46 AM, Brian West <brian at freeswitch.org>
>> >> wrote:
>> >> > or sofia profile xxx flush_inbound_reg callid reboot
>> >> > callid you can get from sofia status profile xxx
>> >> > /b
>> >> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
>> >> >
>> >> > Actually I just
>> >> > found?http://wiki.freeswitch.org/wiki/Mod_event_socket
>> >> >
>> >> > If I telnet to FS as described
>> >> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do I
>> >> > just need to enter somthing like:
>> >> >
>> >> > sendevent NOTIFY
>> >> > profile: internal
>> >> > event-string: resync
>> >> > user: 1000
>> >> > host: 192.168.1.121
>> >> > content-type: application/simple-message-summary
>> >> >
>> >> > where 192.168.1.121 is the ip address of one of the Linksys devices?
>> >> >
>> >> > I don't see any messages sent when I do this. ?What am I doing wrong?
>> >> >
>> >> > Thanks
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From nagalenoj at gmail.com  Wed Jan 27 05:14:47 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 27 Jan 2010 18:44:47 +0530
Subject: [Freeswitch-users] Event socket: filter delete isn't working
Message-ID: <fe67a3341001270514g2715de2ag6aa2e3b099fa7816@mail.gmail.com>

Dear friends,
   I've tried to delete the filter which I applied for an unique id. But, it
doesn't work. After executing 'filter delete', I am receiving the events
from that uuid.
   I used the command as 'filter delete unique-id
c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565'.

I did the following operations.
    Made call to the event socket.
    Registered events for all. (events plain all).
    Applied filter for the uuid. (filter unique-id
aa3cb8ea-0b2f-11df-9e84-fb15c3cd8565).
    I've got a new uuid by using create_uuid.
    Applied filter for this new uuid. (filter unique-id
c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565)
    Originated a call with that uuid.
    Now, I could receive events from both uuids. (Tested by giving DTMFs in
both end and checked unique-id in event header).
    Then, I wanted to delete a uuid from the filter. (filter delete
unique-id c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565).
    I thought, i won't receive the events from this deleted unique-id. But,
I received the dtmfs from both unique-id.

I'm using 'FreeSWITCH Version 1.0.trunk (15982)'.
-- 
Regards,
Nagalenoj H.
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From dftoro at yahoo.com  Wed Jan 27 05:27:38 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Wed, 27 Jan 2010 05:27:38 -0800 (PST)
Subject: [Freeswitch-users] External Profile Problem
In-Reply-To: <BAY143-W20A3102CC2FF36EF1C170FA15D0@phx.gbl>
Message-ID: <443888.41110.qm@web33505.mail.mud.yahoo.com>

Hi, 

<action application="transfer" data="$0 XML default"/>

You must take into account that transfer application not "transfer" a call to destination, only transfer it to a new context

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer


Diego Toro
http://lacarretade.blogspot.com/


--- On Tue, 1/26/10, juan camilo ospina quintero <camilin2212 at hotmail.com> wrote:

> From: juan camilo ospina quintero <camilin2212 at hotmail.com>
> Subject: Re: [Freeswitch-users] External Profile Problem
> To: "freeswitch" <freeswitch-users at lists.freeswitch.org>
> Date: Tuesday, January 26, 2010, 11:06 PM
> 
> 
> 
> 
>  
> Hi
> 
> This works fine
> 
> <extension name="redirect">
> ?<condition field="network_addr"
> expression="^192\.168\.2\.9$"/>   
> 
> ?<condition field="destination_number"
> expression="^1(\d+)$">
> ?<action application="bridge"
> data="sofia/external/$0 at 127.0.0.1:5070"/>
> </extension>
> 
> this redirects from freeswitch to sailfin (127.0.0.1:5070),
> and is in default.xml, in the dialplan.
> 
> the problem is this
> 
>     <extension name="receive">
>       <condition field="network_addr"
> expression="^127\.0\.0\.1$"/>
>       <condition field="destination_number"
> expression="^1(\d+)$">
>         <action application="transfer"
> data="$0 XML default"/>
>       </condition>
>     </extension>
> this doesnt work, this configuration can be found in
> public.xml in the dialplan, the idea of
> this is that when a sip invite comes from sailfin
> (127.0.0.1) transfer the invite to the destination number
> 
> the both configurations above are the only configuration i
> have change from? the default instalation of 
> freeswitch.
> 
> i would like to have some hep with this thanks
> 
> here is the trace log again
> 
> 2010-01-26 20:14:29.512927 [NOTICE]
> switch_channel.c:602 New Channel sofia/external/1000
> at 192.168.2.9 [5177e93a-0ae1-11df-afc9-db39c681a2f1]
> 2010-01-26 20:14:29.512927 [NOTICE] sofia.c:3527 Hangup
> sofia/external/1000
> at 192.168.2.9 [CS_NEW] [MANDATORY_IE_MISSING]
> send 632 bytes to udp/[192.168.2.9]:5070 at
> 01:14:29.517927:
>   
> ------------------------------------------------------------------------
>    SIP/2.0 480 Temporarily Unavailable
>    Via: SIP/2.0/UDP
> 192.168.153.1:5070;branch=z9hG4bKdaacdd64d693615c451ab9db43f9c71c2626;received=192.168.2.9
>    From: <sip:1000
> at 192.168.2.9>;tag=g4xfbi12-3
>    To: <sip:1001
> at 192.168.2.9:5080>;tag=4r91165pvcycB
>    Call-Id: 192.168.153.1_3_3990383226484831353
>    Cseq: 1 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
> MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, refer
>    Reason:
> Q.850;cause=96;text="MANDATORY_IE_MISSING"
>    Content-Length: 0
>    
>   
> ------------------------------------------------------------------------
> 2010-01-26 20:14:29.525646 [NOTICE]
> switch_core_session.c:1086 Session 9 (sofia/external/1000
> at 192.168.2.9) Ended
> 2010-01-26 20:14:29.525646 [NOTICE]
> switch_core_session.c:1088 Close Channel sofia/external/1000
> at 192.168.2.9 [CS_DESTROY]
> 
> 
> 
> > Date: Tue, 26 Jan 2010 22:09:29 -0500
> > From: frank at carmickle.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] External Profile
> Problem
> > 
> > On Tue, Jan 26, juan camilo ospina quintero wrote:
> > > 
> > > Hi, 
> > > 
> > > im trying to establish a simple conference using
> freeswitch and sailfin, sailfin is
> > > and application server that works with
> SipSevlets.
> > > the all thing works as follow.
> > > 
> > > two softphone register with freeswitch, extension
> 1000 and 1001
> > > 1000 sends and invite to 1001, this invite goes
> to sailfin, i use this
> > > 
> > >     <extension name="redirect">
> > >       <condition
> field="network_addr"
> expression="^192\.168\.2\.9$"/>   
> 
> > >       <condition
> field="destination_number"
> expression="^1(\d+)$">
> > >     <action application="bridge"
> data="sofia/external/$0 at 127.0.0.1:5070"/>
> > 
> > And what is the external profile listening on? 
> Probably not the loopback address.  Set up another profile
> listening on 127.0.0.1 and bridge to that.
> > 
> > I could be off base here because you haven't given
> us very much info about your freeswitch configurations.
> > 
> > --FC
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>  		 	   		  
> Windows Live:  Friends
> get your Flickr, Yelp, and Digg updates when they e-mail
> you. 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From Prometheus001 at gmx.net  Wed Jan 27 07:01:47 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 27 Jan 2010 16:01:47 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
Message-ID: <4B60555B.2020004@gmx.net>

I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
to all contributors, excellent work!

In order to have more than 8 channels working, I have followed the
instructions in
http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
ist Debian 5.0R3)
It compiled well however when I start snd-dummy I only have
one-way-audio and my logs show

Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
/usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
working on a machine with 250HZ kernel
Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
obsolete setsockopt SO_BSDCOMPAT
Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
suppressed
Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
obsolete setsockopt SO_BSDCOMPAT
Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
suppressed
If I reinstall alsa from deb everything sworks fine again (of course
with the current limitations).

First question: Has anybody had this issue before? How can I solve this?

Second question:
As I do not need 64 channels or more: how do I manage, that Skype
instances 9..15 use a second instance of snd-dummy as addressed in the wiki?

Best regards
Peter



From jeff at jefflenk.com  Wed Jan 27 07:07:18 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 27 Jan 2010 09:07:18 -0600
Subject: [Freeswitch-users] Polycom buddy watch
Message-ID: <COL117-W58990B36F9ECF38B0E23FEB95D0@phx.gbl>


Hello,

 

Do the Polycom buddy watch presence updates work for "away" "busy" "Dnd" etc. I am running SIP 3.1.3.

 

FS is sending the notifies for when the other phones are in use and that works fine.

 

The Polys are not sending publish events to FS at all for the extended presence states as seen with "sofia profile internal siptrace on". Is this normal?

 

Thanks

 

Jeff
 		 	   		  
_________________________________________________________________
Your E-mail and More On-the-Go. Get Windows Live Hotmail Free.
http://clk.atdmt.com/GBL/go/196390709/direct/01/
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From codecomplete at free.fr  Wed Jan 27 07:20:23 2010
From: codecomplete at free.fr (Fred-145)
Date: Wed, 27 Jan 2010 07:20:23 -0800 (PST)
Subject: [Freeswitch-users]  Investigating one-way sound?
Message-ID: <27341219.post@talk.nabble.com>


Hello

With both a PC+XLite and a Siemens A580IP phone on the same LAN, sound is OK
both ways when I call from the XLite application, but when calling from the
Siemens, I can't hear sound coming from the XLite application. FWIW, both
phones are configured to use G711a and G711u, in this order.

To investigate, what kind of error message should I pay attention to in all
the messages that scroll through the console?

Thank you.
-- 
View this message in context: http://old.nabble.com/Investigating-one-way-sound--tp27341219p27341219.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From kristian.kielhofner at gmail.com  Wed Jan 27 07:20:14 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 27 Jan 2010 10:20:14 -0500
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
In-Reply-To: <27338355.post@talk.nabble.com>
References: <27338355.post@talk.nabble.com>
Message-ID: <2d9149cd1001270720q63ec12bfof4173343a26026b8@mail.gmail.com>

On Wed, Jan 27, 2010 at 7:38 AM, Fred-145 <codecomplete at free.fr> wrote:
>
> Hello
>
> Thanks to "directrtpsetup=yes", it appears that Asterisk can act as an SIP
> proxy and a PBX at the same time, ie. RTP packets will flow directly between
> the two SIP end-points with Asterisk still being able to provide PBX
> services like call transfer, MoH, etc.
> The point is to lower latency for packets to reach their final destination
> and lower CPU load on the Freeswitch server.
>
> Does Freeswitch offer the same feature, or must RTP packets always go
> through the Freeswitch servers?

  This isn't exactly true...  If you do some research you'll find that
OEJ (the author of the Asterisk SIP channel driver) does NOT recommend
the use of directrtpsetup because its use hasn't been tested with many
scenarios (including some you describe, I'm sure).  AFAIK it's still
marked "experimental".  The last time this came up on Asterisk-users
here was the exchange:

Kevin P. Fleming wrote:
> Kristian Kielhofner wrote:
>
>
>> What version of Asterisk is this?  Last I heard (from Olle) this
>> option was very experimental and should not be used on production
>> systems.
>>
>
> He even helpfully documented it that way in the sip.conf.sample file,
> along with a list of (known) cases where it will fail, although there
> are probably plenty more.
>
>

  So Kevin Fleming agrees.

  Needless to say FreeSWITCH has bypass_media.  Last I heard
FreeSWITCH will re-INVITE itself back in the media path if you put a
call on hold (for example) but it won't go back to bypass_media when
you take the call off hold.  Anthony said it would probably take about
$500 in bounty to get that functionality.  Then again, maybe he just
decided to do anyway it because he thought it was cool.  That's been
known to happen too.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From brian at freeswitch.org  Wed Jan 27 07:22:53 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 27 Jan 2010 09:22:53 -0600
Subject: [Freeswitch-users] Polycom buddy watch
In-Reply-To: <COL117-W58990B36F9ECF38B0E23FEB95D0@phx.gbl>
References: <COL117-W58990B36F9ECF38B0E23FEB95D0@phx.gbl>
Message-ID: <3D42D858-4933-4DBC-89F7-14A83A77726B@freeswitch.org>

I'm going to guess we are missing some type of outbound subscription similar to how we do it in sofia_sla.c for the presence events.

/b

On Jan 27, 2010, at 9:07 AM, Jeff Lenk wrote:

> Hello,
>  
> Do the Polycom buddy watch presence updates work for "away" "busy" "Dnd" etc. I am running SIP 3.1.3.
>  
> FS is sending the notifies for when the other phones are in use and that works fine.
>  
> The Polys are not sending publish events to FS at all for the extended presence states as seen with "sofia profile internal siptrace on". Is this normal?
>  
> Thanks
>  
> Jeff

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From gmaruzz at celliax.org  Wed Jan 27 07:26:08 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 27 Jan 2010 16:26:08 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B60555B.2020004@gmx.net>
References: <4B60555B.2020004@gmx.net>
Message-ID: <7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com>

Ciao Peter

one instance of snd-dummy "customized" is enough for 64 instances of
skype clients, no need (and do not works) with more instances of
snd-dummy-customized.

Maybe you got the one-way problem because of kernel at 250HZ (don't
know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
8.04).

Or maybe you have to check and modify which sound devices the skype
clients are using (try to check that with snd-summy-custom loaded,
maybe with the ssh -X trick (as in the wiki page).

To load more than one snd-dummy-original (the non modified one), you
do this with the modprobe command, as in:

rmmod snd-dummy
modprobe snd-dummy enable=1,1,1

this command will enable three instances of snd-dummy original, so
you'll have three fake soundcards, and you'll have to setup each group
of 8 skype instances to use sound devices from one fake soundcard, RG:
no more than 8 skype client instances can use one instance of fake
soundcard.

Also, please update the mod_skypiax code (svn up in its directory) I
just committed some improvements.

If you have any other doubts, or need more info, don't hesitate to
write the mailing list again,

ciao for now,

-giovanni



On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
> to all contributors, excellent work!
>
> In order to have more than 8 channels working, I have followed the
> instructions in
> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
> ist Debian 5.0R3)
> It compiled well however when I start snd-dummy I only have
> one-way-audio and my logs show
>
> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
> working on a machine with 250HZ kernel
> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
> obsolete setsockopt SO_BSDCOMPAT
> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
> suppressed
> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
> obsolete setsockopt SO_BSDCOMPAT
> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
> suppressed
> If I reinstall alsa from deb everything sworks fine again (of course
> with the current limitations).
>
> First question: Has anybody had this issue before? How can I solve this?
>
> Second question:
> As I do not need 64 channels or more: how do I manage, that Skype
> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From brian at freeswitch.org  Wed Jan 27 07:30:41 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 27 Jan 2010 09:30:41 -0600
Subject: [Freeswitch-users] Investigating one-way sound?
In-Reply-To: <27341219.post@talk.nabble.com>
References: <27341219.post@talk.nabble.com>
Message-ID: <22DD670E-B920-4328-9939-56447375D5C7@freeswitch.org>

Looking at the SIP traffic, paying special attention to the SDP.  I'm going to guess right off the X-Lite is putting its public IP and since maybe your NAT router can't hair pin the media you get on way media.  If you go into the settings make sure not to be using the Globally Discovered IP and use the Local IP in the network options.

/b

On Jan 27, 2010, at 9:20 AM, Fred-145 wrote:

> To investigate, what kind of error message should I pay attention to in all
> the messages that scroll through the console?




From mouncifbb at gmail.com  Wed Jan 27 07:31:45 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Wed, 27 Jan 2010 10:31:45 -0500
Subject: [Freeswitch-users] Call limits (time)
In-Reply-To: <5F37CA2E-129F-4082-B304-D50D0E1A4FAF@freeswitch.org>
References: <201001052125.06909.sos@sokhapkin.dyndns.org>
	<4B4417B3.9090807@aastral.net>
	<201001060620.13735.sos@sokhapkin.dyndns.org>
	<8ccbff061001060859p59aa6b5bw6f52275650f9138@mail.gmail.com>
	<ebbc2d551001061416l25c17143gafe62de35cc6dee0@mail.gmail.com>
	<5F37CA2E-129F-4082-B304-D50D0E1A4FAF@freeswitch.org>
Message-ID: <ade741d61001270731w4925d05mb5f3e87e8dec94c9@mail.gmail.com>

yeah it will be nice if mod_nibble can do call minimum and rounding, same
way opensource a2billing doing for asterisk.
I think without this feature the module will not be useful for business use.
a common bill minimum/increments should in a form of: 6/6 for US , 6/30 for
A-Z and 60/60 for Mexico.
1/1 type is not offered by Tier1 Telecom providers.

BTW Freeswitch Rocks Rocks!!!!! It's The best IP telephony App I ever used!!



On Wed, Jan 6, 2010 at 5:33 PM, Brian West <brian at freeswitch.org> wrote:

> Or better yet take over the maint. of the module if its making you money
> give back by providing some help to the project... its the ultimate way to
> give back.
>
> /b
>
> On Jan 6, 2010, at 4:16 PM, Jo?o Mesquita wrote:
>
> > Why can't someone just sponsor some love to the module? The author has
> his email on the header or open a Jira asking stuff so we have nibblebill
> more mature.
> >
> > Jo?o Mesquita
> > FreeSWITCH? Solutions
> > t: +1 (646) 4959927
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From camilin2212 at hotmail.com  Wed Jan 27 07:48:56 2010
From: camilin2212 at hotmail.com (juan camilo ospina quintero)
Date: Wed, 27 Jan 2010 10:48:56 -0500
Subject: [Freeswitch-users] External Profile Problem
In-Reply-To: <443888.41110.qm@web33505.mail.mud.yahoo.com>
References: <BAY143-W20A3102CC2FF36EF1C170FA15D0@phx.gbl>,
	<443888.41110.qm@web33505.mail.mud.yahoo.com>
Message-ID: <BAY143-W267C2F235ED9FCFA619FA1A15D0@phx.gbl>


hi thanks

sorry but i dont really understand what a context is.

so, when i put <action application="transfer" data="$0 XML default"/>
what does it really does, what it means that transfer to new context,


bye

> Date: Wed, 27 Jan 2010 05:27:38 -0800
> From: dftoro at yahoo.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] External Profile Problem
> 
> Hi, 
> 
> <action application="transfer" data="$0 XML default"/>
> 
> You must take into account that transfer application not "transfer" a call to destination, only transfer it to a new context
> 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer
> 
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 
> 
> --- On Tue, 1/26/10, juan camilo ospina quintero <camilin2212 at hotmail.com> wrote:
> 
> > From: juan camilo ospina quintero <camilin2212 at hotmail.com>
> > Subject: Re: [Freeswitch-users] External Profile Problem
> > To: "freeswitch" <freeswitch-users at lists.freeswitch.org>
> > Date: Tuesday, January 26, 2010, 11:06 PM
> > 
> > 
> > 
> > 
> >  
> > Hi
> > 
> > This works fine
> > 
> > <extension name="redirect">
> >  <condition field="network_addr"
> > expression="^192\.168\.2\.9$"/>   
> > 
> >  <condition field="destination_number"
> > expression="^1(\d+)$">
> >  <action application="bridge"
> > data="sofia/external/$0 at 127.0.0.1:5070"/>
> > </extension>
> > 
> > this redirects from freeswitch to sailfin (127.0.0.1:5070),
> > and is in default.xml, in the dialplan.
> > 
> > the problem is this
> > 
> >     <extension name="receive">
> >       <condition field="network_addr"
> > expression="^127\.0\.0\.1$"/>
> >       <condition field="destination_number"
> > expression="^1(\d+)$">
> >         <action application="transfer"
> > data="$0 XML default"/>
> >       </condition>
> >     </extension>
> > this doesnt work, this configuration can be found in
> > public.xml in the dialplan, the idea of
> > this is that when a sip invite comes from sailfin
> > (127.0.0.1) transfer the invite to the destination number
> > 
> > the both configurations above are the only configuration i
> > have change from  the default instalation of 
> > freeswitch.
> > 
> > i would like to have some hep with this thanks
> > 
> > here is the trace log again
> > 
> > 2010-01-26 20:14:29.512927 [NOTICE]
> > switch_channel.c:602 New Channel sofia/external/1000
> > at 192.168.2.9 [5177e93a-0ae1-11df-afc9-db39c681a2f1]
> > 2010-01-26 20:14:29.512927 [NOTICE] sofia.c:3527 Hangup
> > sofia/external/1000
> > at 192.168.2.9 [CS_NEW] [MANDATORY_IE_MISSING]
> > send 632 bytes to udp/[192.168.2.9]:5070 at
> > 01:14:29.517927:
> >   
> > ------------------------------------------------------------------------
> >    SIP/2.0 480 Temporarily Unavailable
> >    Via: SIP/2.0/UDP
> > 192.168.153.1:5070;branch=z9hG4bKdaacdd64d693615c451ab9db43f9c71c2626;received=192.168.2.9
> >    From: <sip:1000
> > at 192.168.2.9>;tag=g4xfbi12-3
> >    To: <sip:1001
> > at 192.168.2.9:5080>;tag=4r91165pvcycB
> >    Call-Id: 192.168.153.1_3_3990383226484831353
> >    Cseq: 1 INVITE
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
> >    Accept: application/sdp
> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
> > MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
> >    Supported: timer, precondition, path, replaces
> >    Allow-Events: talk, refer
> >    Reason:
> > Q.850;cause=96;text="MANDATORY_IE_MISSING"
> >    Content-Length: 0
> >    
> >   
> > ------------------------------------------------------------------------
> > 2010-01-26 20:14:29.525646 [NOTICE]
> > switch_core_session.c:1086 Session 9 (sofia/external/1000
> > at 192.168.2.9) Ended
> > 2010-01-26 20:14:29.525646 [NOTICE]
> > switch_core_session.c:1088 Close Channel sofia/external/1000
> > at 192.168.2.9 [CS_DESTROY]
> > 
> > 
> > 
> > > Date: Tue, 26 Jan 2010 22:09:29 -0500
> > > From: frank at carmickle.com
> > > To: freeswitch-users at lists.freeswitch.org
> > > Subject: Re: [Freeswitch-users] External Profile
> > Problem
> > > 
> > > On Tue, Jan 26, juan camilo ospina quintero wrote:
> > > > 
> > > > Hi, 
> > > > 
> > > > im trying to establish a simple conference using
> > freeswitch and sailfin, sailfin is
> > > > and application server that works with
> > SipSevlets.
> > > > the all thing works as follow.
> > > > 
> > > > two softphone register with freeswitch, extension
> > 1000 and 1001
> > > > 1000 sends and invite to 1001, this invite goes
> > to sailfin, i use this
> > > > 
> > > >     <extension name="redirect">
> > > >       <condition
> > field="network_addr"
> > expression="^192\.168\.2\.9$"/>   
> > 
> > > >       <condition
> > field="destination_number"
> > expression="^1(\d+)$">
> > > >     <action application="bridge"
> > data="sofia/external/$0 at 127.0.0.1:5070"/>
> > > 
> > > And what is the external profile listening on? 
> > Probably not the loopback address.  Set up another profile
> > listening on 127.0.0.1 and bridge to that.
> > > 
> > > I could be off base here because you haven't given
> > us very much info about your freeswitch configurations.
> > > 
> > > --FC
> > > 
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >  		 	   		  
> > Windows Live:  Friends
> > get your Flickr, Yelp, and Digg updates when they e-mail
> > you. 
> > 
> > -----Inline Attachment Follows-----
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> 
> 
>       
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From gmaruzz at celliax.org  Wed Jan 27 08:23:15 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 27 Jan 2010 17:23:15 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com>
Message-ID: <7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com>

This warning is harmless:

Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
obsolete setsockopt SO_BSDCOMPAT

On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> Ciao Peter
>
> one instance of snd-dummy "customized" is enough for 64 instances of
> skype clients, no need (and do not works) with more instances of
> snd-dummy-customized.
>
> Maybe you got the one-way problem because of kernel at 250HZ (don't
> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
> 8.04).
>
> Or maybe you have to check and modify which sound devices the skype
> clients are using (try to check that with snd-summy-custom loaded,
> maybe with the ssh -X trick (as in the wiki page).
>
> To load more than one snd-dummy-original (the non modified one), you
> do this with the modprobe command, as in:
>
> rmmod snd-dummy
> modprobe snd-dummy enable=1,1,1
>
> this command will enable three instances of snd-dummy original, so
> you'll have three fake soundcards, and you'll have to setup each group
> of 8 skype instances to use sound devices from one fake soundcard, RG:
> no more than 8 skype client instances can use one instance of fake
> soundcard.
>
> Also, please update the mod_skypiax code (svn up in its directory) I
> just committed some improvements.
>
> If you have any other doubts, or need more info, don't hesitate to
> write the mailing list again,
>
> ciao for now,
>
> -giovanni
>
>
>
> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>> to all contributors, excellent work!
>>
>> In order to have more than 8 channels working, I have followed the
>> instructions in
>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>> ist Debian 5.0R3)
>> It compiled well however when I start snd-dummy I only have
>> one-way-audio and my logs show
>>
>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>> working on a machine with 250HZ kernel
>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>> obsolete setsockopt SO_BSDCOMPAT
>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>> suppressed
>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>> obsolete setsockopt SO_BSDCOMPAT
>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>> suppressed
>> If I reinstall alsa from deb everything sworks fine again (of course
>> with the current limitations).
>>
>> First question: Has anybody had this issue before? How can I solve this?
>>
>> Second question:
>> As I do not need 64 channels or more: how do I manage, that Skype
>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>
>> Best regards
>> Peter
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From anthony.minessale at gmail.com  Wed Jan 27 08:45:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 10:45:56 -0600
Subject: [Freeswitch-users] mod_spidermonkey memory leak?
In-Reply-To: <20100126120711.92321.qmail@web10512.mail.ogk.yahoo.co.jp>
References: <20100126120711.92321.qmail@web10512.mail.ogk.yahoo.co.jp>
Message-ID: <191c3a031001270845v20f30563w64d7ce9b0510f21a@mail.gmail.com>

if you suspect a memory leak
can you please update to the latest SVN trunk (over 1000 revs higher than
yours)
then reproduce the problem under valgrind and send the report in a jira
http://jira.freeswitch.org

valgrind --tool=memcheck --log-file=vg.log --leak-check=full
--leak-resolution=high --show-reachable=yes
/usr/local/freeswitch/bin/freeswitch -vg



2010/1/26 ???? ??? <matomoya at yahoo.co.jp>

> I have tested the SVN trunk 15341.
> When the following scripts are executed, it seems to do the memory leak.
> Is the mistake found in the following scripts?
> Or Is there a problem in mod_spidermonkey?
>
> -- test script --
> session.execute("ring_ready");
> session.answer();
> session.setVariable("ringback", "%(1000, 2000, 440, 460)");
>
> var bleg = new Session();
> var sound_wav = "sounds/test01.wav";
> var sound_leg = "both";
> var op = "api_on_answer=uuid_broadcast "+session.uuid+" "+sound_wav+"
> "+sound_leg;
> var ret = bleg.originate(session, "{"+op+"}" +
> "sofia/gateway/profile0_gateway1/1000");
> if(!session.ready()){
>  return;
> }
>
>
> if(!ret){ // bleg not answered.
>  var sound_wav = "sounds/test02.wav";
>  session.streamFile(sound_wav);
>  if(session.ready()){
>    session.hangup();
>  }
>  return;
> }
>
> if(bleg.ready()){
>  bridge(session, bleg);
> }
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From Prometheus001 at gmx.net  Wed Jan 27 08:58:16 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 27 Jan 2010 17:58:16 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com>
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com>
Message-ID: <4B6070A8.6050607@gmx.net>

Thanks Giovanni,

I think there may be the problem, that I have 2 sound devices now:
- Dummy PCM (hw0:0) (this is from debian install)
- Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
since I compiled alsa newly)

I tried both, but both do not work. How do I get rid of the old alsa device?
By the way: I uninstalled Alsa before I installed the new driver
(apt-get remove alsa-utils alsa-base).

Best regards
Peter


Giovanni Maruzzelli schrieb:
> This warning is harmless:
>
> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
> obsolete setsockopt SO_BSDCOMPAT
>
> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
> <gmaruzz at celliax.org> wrote:
>   
>> Ciao Peter
>>
>> one instance of snd-dummy "customized" is enough for 64 instances of
>> skype clients, no need (and do not works) with more instances of
>> snd-dummy-customized.
>>
>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>> 8.04).
>>
>> Or maybe you have to check and modify which sound devices the skype
>> clients are using (try to check that with snd-summy-custom loaded,
>> maybe with the ssh -X trick (as in the wiki page).
>>
>> To load more than one snd-dummy-original (the non modified one), you
>> do this with the modprobe command, as in:
>>
>> rmmod snd-dummy
>> modprobe snd-dummy enable=1,1,1
>>
>> this command will enable three instances of snd-dummy original, so
>> you'll have three fake soundcards, and you'll have to setup each group
>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>> no more than 8 skype client instances can use one instance of fake
>> soundcard.
>>
>> Also, please update the mod_skypiax code (svn up in its directory) I
>> just committed some improvements.
>>
>> If you have any other doubts, or need more info, don't hesitate to
>> write the mailing list again,
>>
>> ciao for now,
>>
>> -giovanni
>>
>>
>>
>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>     
>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>> to all contributors, excellent work!
>>>
>>> In order to have more than 8 channels working, I have followed the
>>> instructions in
>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>> ist Debian 5.0R3)
>>> It compiled well however when I start snd-dummy I only have
>>> one-way-audio and my logs show
>>>
>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>> working on a machine with 250HZ kernel
>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>> obsolete setsockopt SO_BSDCOMPAT
>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>> suppressed
>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>> obsolete setsockopt SO_BSDCOMPAT
>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>> suppressed
>>> If I reinstall alsa from deb everything sworks fine again (of course
>>> with the current limitations).
>>>
>>> First question: Has anybody had this issue before? How can I solve this?
>>>
>>> Second question:
>>> As I do not need 64 channels or more: how do I manage, that Skype
>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>
>>> Best regards
>>> Peter
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>>     
>
>
>
>   



From gmaruzz at celliax.org  Wed Jan 27 09:07:54 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 27 Jan 2010 18:07:54 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B6070A8.6050607@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com> 
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com> 
	<4B6070A8.6050607@gmx.net>
Message-ID: <7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com>

I don't think you got two snd-dummy loaded (but maybe yes)
what's the output of:

aplay -l

?

If instead you are referring to the choices that skype clients offers
you in the "set audio devices" window, choose Dummy PCM (hw0:0)

Eg: not the "default", but the "hardware" one


On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Thanks Giovanni,
>
> I think there may be the problem, that I have 2 sound devices now:
> - Dummy PCM (hw0:0) (this is from debian install)
> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
> since I compiled alsa newly)
>
> I tried both, but both do not work. How do I get rid of the old alsa device?
> By the way: I uninstalled Alsa before I installed the new driver
> (apt-get remove alsa-utils alsa-base).
>
> Best regards
> Peter
>
>
> Giovanni Maruzzelli schrieb:
>> This warning is harmless:
>>
>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>> obsolete setsockopt SO_BSDCOMPAT
>>
>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>> <gmaruzz at celliax.org> wrote:
>>
>>> Ciao Peter
>>>
>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>> skype clients, no need (and do not works) with more instances of
>>> snd-dummy-customized.
>>>
>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>> 8.04).
>>>
>>> Or maybe you have to check and modify which sound devices the skype
>>> clients are using (try to check that with snd-summy-custom loaded,
>>> maybe with the ssh -X trick (as in the wiki page).
>>>
>>> To load more than one snd-dummy-original (the non modified one), you
>>> do this with the modprobe command, as in:
>>>
>>> rmmod snd-dummy
>>> modprobe snd-dummy enable=1,1,1
>>>
>>> this command will enable three instances of snd-dummy original, so
>>> you'll have three fake soundcards, and you'll have to setup each group
>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>> no more than 8 skype client instances can use one instance of fake
>>> soundcard.
>>>
>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>> just committed some improvements.
>>>
>>> If you have any other doubts, or need more info, don't hesitate to
>>> write the mailing list again,
>>>
>>> ciao for now,
>>>
>>> -giovanni
>>>
>>>
>>>
>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>> to all contributors, excellent work!
>>>>
>>>> In order to have more than 8 channels working, I have followed the
>>>> instructions in
>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>> ist Debian 5.0R3)
>>>> It compiled well however when I start snd-dummy I only have
>>>> one-way-audio and my logs show
>>>>
>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>> working on a machine with 250HZ kernel
>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>> obsolete setsockopt SO_BSDCOMPAT
>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>> suppressed
>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>> obsolete setsockopt SO_BSDCOMPAT
>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>> suppressed
>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>> with the current limitations).
>>>>
>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>
>>>> Second question:
>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>>
>>
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From jeff at jefflenk.com  Wed Jan 27 09:11:00 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 27 Jan 2010 11:11:00 -0600
Subject: [Freeswitch-users] Polycom buddy watch
In-Reply-To: <3D42D858-4933-4DBC-89F7-14A83A77726B@freeswitch.org>
References: <COL117-W58990B36F9ECF38B0E23FEB95D0@phx.gbl>,
	<3D42D858-4933-4DBC-89F7-14A83A77726B@freeswitch.org>
Message-ID: <COL117-W31E21D3B622F7F39375F70B95D0@phx.gbl>


Thanks Brian,

 

I was just wondering if anyone else sees the same thing so I can eliminate whether I have it configured wrong.

 

Do others use the Buddy watch feature with Polycom and have the "Busy" "DND" or "Away" extended statuses working?

 

-Jeff
 


From: brian at freeswitch.org
Date: Wed, 27 Jan 2010 09:22:53 -0600
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom buddy watch

I'm going to guess we are missing some type of outbound subscription similar to how we do it in sofia_sla.c for the presence events.


/b



On Jan 27, 2010, at 9:07 AM, Jeff Lenk wrote:
Hello,
 
Do the Polycom buddy watch presence updates work for "away" "busy" "Dnd" etc. I am running SIP 3.1.3.
 
FS is sending the notifies for when the other phones are in use and that works fine.
 
The Polys are not sending publish events to FS at all for the extended presence states as seen with "sofia profile internal siptrace on". Is this normal?
 
Thanks
 
Jeff
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with powerful SPAM protection.
http://clk.atdmt.com/GBL/go/196390707/direct/01/
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From jcasale at activenetwerx.com  Wed Jan 27 09:19:47 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 27 Jan 2010 17:19:47 +0000
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90B9E04@prato.activenetwerx.local>

Anyone running this with their latest 2.2.1 release successfully
using Digium analog TDM cards? Is it known to work fine, or possibly
not been tested yet?

Thanks,
jlc



From freeswitch at aastral.net  Wed Jan 27 09:31:11 2010
From: freeswitch at aastral.net (Bill W)
Date: Wed, 27 Jan 2010 12:31:11 -0500
Subject: [Freeswitch-users] Strategies for reliably detecting nat on B-leg?
Message-ID: <4B60785F.6030505@aastral.net>

Hey All,

I know FreeSWITCH has the agressive-nat-detection parameter for sofia 
configs which will detect NAT on an incoming call.  So we know if the 
A-leg is natted.  The question is, are there any reliable ways to detect 
nat at the destination before bridging that call?

One could assume that if the destination is a gateway, then it would be 
okay to bypass-media.

Also, one could check the sofia registry and assume that if an endpoint 
is registered to FreeSWITCH that it is NATted and therefore no point in 
trying to bypass media.

But neither of these options seems 100% reliable.

Thoughts?  Suggestions?

Thanks!
Bill W.



From Prometheus001 at gmx.net  Wed Jan 27 09:35:00 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 27 Jan 2010 18:35:00 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com>
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com>
	<4B6070A8.6050607@gmx.net>
	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com>
Message-ID: <4B607944.4040700@gmx.net>

Her's the output:

skype:~# aplay -l
bash: aplay: command not found

Giovanni Maruzzelli schrieb:
> I don't think you got two snd-dummy loaded (but maybe yes)
> what's the output of:
>
> aplay -l
>
> ?
>
> If instead you are referring to the choices that skype clients offers
> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>
> Eg: not the "default", but the "hardware" one
>
>
> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> Thanks Giovanni,
>>
>> I think there may be the problem, that I have 2 sound devices now:
>> - Dummy PCM (hw0:0) (this is from debian install)
>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>> since I compiled alsa newly)
>>
>> I tried both, but both do not work. How do I get rid of the old alsa device?
>> By the way: I uninstalled Alsa before I installed the new driver
>> (apt-get remove alsa-utils alsa-base).
>>
>> Best regards
>> Peter
>>
>>
>> Giovanni Maruzzelli schrieb:
>>     
>>> This warning is harmless:
>>>
>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>> obsolete setsockopt SO_BSDCOMPAT
>>>
>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>> <gmaruzz at celliax.org> wrote:
>>>
>>>       
>>>> Ciao Peter
>>>>
>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>> skype clients, no need (and do not works) with more instances of
>>>> snd-dummy-customized.
>>>>
>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>> 8.04).
>>>>
>>>> Or maybe you have to check and modify which sound devices the skype
>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>
>>>> To load more than one snd-dummy-original (the non modified one), you
>>>> do this with the modprobe command, as in:
>>>>
>>>> rmmod snd-dummy
>>>> modprobe snd-dummy enable=1,1,1
>>>>
>>>> this command will enable three instances of snd-dummy original, so
>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>> no more than 8 skype client instances can use one instance of fake
>>>> soundcard.
>>>>
>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>> just committed some improvements.
>>>>
>>>> If you have any other doubts, or need more info, don't hesitate to
>>>> write the mailing list again,
>>>>
>>>> ciao for now,
>>>>
>>>> -giovanni
>>>>
>>>>
>>>>
>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>
>>>>         
>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>> to all contributors, excellent work!
>>>>>
>>>>> In order to have more than 8 channels working, I have followed the
>>>>> instructions in
>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>> ist Debian 5.0R3)
>>>>> It compiled well however when I start snd-dummy I only have
>>>>> one-way-audio and my logs show
>>>>>
>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>> working on a machine with 250HZ kernel
>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>> suppressed
>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>> suppressed
>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>> with the current limitations).
>>>>>
>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>
>>>>> Second question:
>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>           
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>>
>>>>         
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   



From moises.silva at gmail.com  Wed Jan 27 09:46:15 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 27 Jan 2010 12:46:15 -0500
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E90B9E04@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E90B9E04@prato.activenetwerx.local>
Message-ID: <c4d05cbe1001270946r39532b7al8290f64442415f4@mail.gmail.com>

On Wed, Jan 27, 2010 at 12:19 PM, Joseph L. Casale <
jcasale at activenetwerx.com> wrote:

> Anyone running this with their latest 2.2.1 release successfully
> using Digium analog TDM cards? Is it known to work fine, or possibly
> not been tested yet?


To my knowledge, neither one. The best answer I can give you is "it should
work", the dahdi driver API do not change often and previous versions of
DAHDI were tested just fine.

 --
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From brian at freeswitch.org  Wed Jan 27 09:50:59 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 27 Jan 2010 11:50:59 -0600
Subject: [Freeswitch-users] Strategies for reliably detecting nat on
	B-leg?
In-Reply-To: <4B60785F.6030505@aastral.net>
References: <4B60785F.6030505@aastral.net>
Message-ID: <FB139E6B-F685-491C-A672-16844878200E@freeswitch.org>

update to trunk.  and don't use agressive-nat, set local-network-acl, set the ext-rtp-ip and ext-sip-ip to autonat:x.x.x.x or if you're behind a natpmp or upnp router set it to auto-nat.  

It should just work.  Again you have no real way to know if the far end client never lies to you.  Which it should never do anyway.  Endpoints should know how to traverse their own nat and not leave it up to the registrar to figure it out.

/b

On Jan 27, 2010, at 11:31 AM, Bill W wrote:

> Thoughts?  Suggestions?




From Russell.Mosemann at cune.org  Wed Jan 27 10:00:58 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Wed, 27 Jan 2010 18:00:58 -0000
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E90B9E04@prato.activenetwerx.local>
Message-ID: <20100127180058.29470216DEA@cuneorg-email.cune.pri>

"Joseph L. Casale" <jcasale at activenetwerx.com> said:

> Anyone running this with their latest 2.2.1 release successfully
> using Digium analog TDM cards? Is it known to work fine, or possibly
> not been tested yet?

It's working OK here. We're actually using 2.2.1RC2 with a TE110P in a VZ
environment.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From Prometheus001 at gmx.net  Wed Jan 27 10:04:04 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 27 Jan 2010 19:04:04 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <4B607944.4040700@gmx.net>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com>
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com>
	<4B6070A8.6050607@gmx.net>
	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com>
	<4B607944.4040700@gmx.net>
Message-ID: <4B608014.4030902@gmx.net>

I installed alsa-utile,

now I get:

skype:/var/cache/apt/archives# aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
  Subdevices: 127/128
  Subdevice #0: subdevice #0
  Subdevice #1: subdevice #1
  Subdevice #2: subdevice #2
  Subdevice #3: subdevice #3
  Subdevice #4: subdevice #4
  Subdevice #5: subdevice #5
  Subdevice #6: subdevice #6
  Subdevice #7: subdevice #7
  Subdevice #8: subdevice #8
  Subdevice #9: subdevice #9
  Subdevice #10: subdevice #10
  Subdevice #11: subdevice #11
  Subdevice #12: subdevice #12
  Subdevice #13: subdevice #13
  Subdevice #14: subdevice #14
  Subdevice #15: subdevice #15
  Subdevice #16: subdevice #16
  Subdevice #17: subdevice #17
  Subdevice #18: subdevice #18
  Subdevice #19: subdevice #19
  Subdevice #20: subdevice #20
  Subdevice #21: subdevice #21
  Subdevice #22: subdevice #22
  Subdevice #23: subdevice #23
  Subdevice #24: subdevice #24
  Subdevice #25: subdevice #25
  Subdevice #26: subdevice #26
  Subdevice #27: subdevice #27
  Subdevice #28: subdevice #28
  Subdevice #29: subdevice #29
  Subdevice #30: subdevice #30
  Subdevice #31: subdevice #31
  Subdevice #32: subdevice #32
  Subdevice #33: subdevice #33
  Subdevice #34: subdevice #34
  Subdevice #35: subdevice #35
  Subdevice #36: subdevice #36
  Subdevice #37: subdevice #37
  Subdevice #38: subdevice #38
  Subdevice #39: subdevice #39
  Subdevice #40: subdevice #40
  Subdevice #41: subdevice #41
  Subdevice #42: subdevice #42
  Subdevice #43: subdevice #43
  Subdevice #44: subdevice #44
  Subdevice #45: subdevice #45
  Subdevice #46: subdevice #46
  Subdevice #47: subdevice #47
  Subdevice #48: subdevice #48
  Subdevice #49: subdevice #49
  Subdevice #50: subdevice #50
  Subdevice #51: subdevice #51
  Subdevice #52: subdevice #52
  Subdevice #53: subdevice #53
  Subdevice #54: subdevice #54
  Subdevice #55: subdevice #55
  Subdevice #56: subdevice #56
  Subdevice #57: subdevice #57
  Subdevice #58: subdevice #58
  Subdevice #59: subdevice #59
  Subdevice #60: subdevice #60
  Subdevice #61: subdevice #61
  Subdevice #62: subdevice #62
  Subdevice #63: subdevice #63
  Subdevice #64: subdevice #64
  Subdevice #65: subdevice #65
  Subdevice #66: subdevice #66
  Subdevice #67: subdevice #67
  Subdevice #68: subdevice #68
  Subdevice #69: subdevice #69
  Subdevice #70: subdevice #70
  Subdevice #71: subdevice #71
  Subdevice #72: subdevice #72
  Subdevice #73: subdevice #73
  Subdevice #74: subdevice #74
  Subdevice #75: subdevice #75
  Subdevice #76: subdevice #76
  Subdevice #77: subdevice #77
  Subdevice #78: subdevice #78
  Subdevice #79: subdevice #79
  Subdevice #80: subdevice #80
  Subdevice #81: subdevice #81
  Subdevice #82: subdevice #82
  Subdevice #83: subdevice #83
  Subdevice #84: subdevice #84
  Subdevice #85: subdevice #85
  Subdevice #86: subdevice #86
  Subdevice #87: subdevice #87
  Subdevice #88: subdevice #88
  Subdevice #89: subdevice #89
  Subdevice #90: subdevice #90
  Subdevice #91: subdevice #91
  Subdevice #92: subdevice #92
  Subdevice #93: subdevice #93
  Subdevice #94: subdevice #94
  Subdevice #95: subdevice #95
  Subdevice #96: subdevice #96
  Subdevice #97: subdevice #97
  Subdevice #98: subdevice #98
  Subdevice #99: subdevice #99
  Subdevice #100: subdevice #100
  Subdevice #101: subdevice #101
  Subdevice #102: subdevice #102
  Subdevice #103: subdevice #103
  Subdevice #104: subdevice #104
  Subdevice #105: subdevice #105
  Subdevice #106: subdevice #106
  Subdevice #107: subdevice #107
  Subdevice #108: subdevice #108
  Subdevice #109: subdevice #109
  Subdevice #110: subdevice #110
  Subdevice #111: subdevice #111
  Subdevice #112: subdevice #112
  Subdevice #113: subdevice #113
  Subdevice #114: subdevice #114
  Subdevice #115: subdevice #115
  Subdevice #116: subdevice #116
  Subdevice #117: subdevice #117
  Subdevice #118: subdevice #118
  Subdevice #119: subdevice #119
  Subdevice #120: subdevice #120
  Subdevice #121: subdevice #121
  Subdevice #122: subdevice #122
  Subdevice #123: subdevice #123
  Subdevice #124: subdevice #124
  Subdevice #125: subdevice #125
  Subdevice #126: subdevice #126
  Subdevice #127: subdevice #127


Peter P GMX schrieb:
> Her's the output:
>
> skype:~# aplay -l
> bash: aplay: command not found
>
> Giovanni Maruzzelli schrieb:
>   
>> I don't think you got two snd-dummy loaded (but maybe yes)
>> what's the output of:
>>
>> aplay -l
>>
>> ?
>>
>> If instead you are referring to the choices that skype clients offers
>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>
>> Eg: not the "default", but the "hardware" one
>>
>>
>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>   
>>     
>>> Thanks Giovanni,
>>>
>>> I think there may be the problem, that I have 2 sound devices now:
>>> - Dummy PCM (hw0:0) (this is from debian install)
>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>> since I compiled alsa newly)
>>>
>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>> By the way: I uninstalled Alsa before I installed the new driver
>>> (apt-get remove alsa-utils alsa-base).
>>>
>>> Best regards
>>> Peter
>>>
>>>
>>> Giovanni Maruzzelli schrieb:
>>>     
>>>       
>>>> This warning is harmless:
>>>>
>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>
>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>> <gmaruzz at celliax.org> wrote:
>>>>
>>>>       
>>>>         
>>>>> Ciao Peter
>>>>>
>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>> skype clients, no need (and do not works) with more instances of
>>>>> snd-dummy-customized.
>>>>>
>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>> 8.04).
>>>>>
>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>
>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>> do this with the modprobe command, as in:
>>>>>
>>>>> rmmod snd-dummy
>>>>> modprobe snd-dummy enable=1,1,1
>>>>>
>>>>> this command will enable three instances of snd-dummy original, so
>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>> no more than 8 skype client instances can use one instance of fake
>>>>> soundcard.
>>>>>
>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>> just committed some improvements.
>>>>>
>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>> write the mailing list again,
>>>>>
>>>>> ciao for now,
>>>>>
>>>>> -giovanni
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>
>>>>>         
>>>>>           
>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>> to all contributors, excellent work!
>>>>>>
>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>> instructions in
>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>> ist Debian 5.0R3)
>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>> one-way-audio and my logs show
>>>>>>
>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>> working on a machine with 250HZ kernel
>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>> suppressed
>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>> suppressed
>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>> with the current limitations).
>>>>>>
>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>
>>>>>> Second question:
>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>
>>>>>> Best regards
>>>>>> Peter
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>           
>>>>>>             
>>>>> --
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> Cell : +39-347-2665618
>>>>>
>>>>>
>>>>>         
>>>>>           
>>>>       
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>     
>>>       
>>
>>   
>>     
>
>   



From anthony.minessale at gmail.com  Wed Jan 27 10:05:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 12:05:10 -0600
Subject: [Freeswitch-users] Strategies for reliably detecting nat on
	B-leg?
In-Reply-To: <FB139E6B-F685-491C-A672-16844878200E@freeswitch.org>
References: <4B60785F.6030505@aastral.net>
	<FB139E6B-F685-491C-A672-16844878200E@freeswitch.org>
Message-ID: <191c3a031001271005t62631e77sf6d9ca406054ba00@mail.gmail.com>

also you can set
sip_sticky_contact=true
channel var which will make that session turn on nat lock in the b leg so
they can't change the contact to a nat addr

add it in {} to your dial string like

{sip_sticky_contact=true}sofia/internal/foo at bar.com




On Wed, Jan 27, 2010 at 11:50 AM, Brian West <brian at freeswitch.org> wrote:

> update to trunk.  and don't use agressive-nat, set local-network-acl, set
> the ext-rtp-ip and ext-sip-ip to autonat:x.x.x.x or if you're behind a
> natpmp or upnp router set it to auto-nat.
>
> It should just work.  Again you have no real way to know if the far end
> client never lies to you.  Which it should never do anyway.  Endpoints
> should know how to traverse their own nat and not leave it up to the
> registrar to figure it out.
>
> /b
>
> On Jan 27, 2010, at 11:31 AM, Bill W wrote:
>
> > Thoughts?  Suggestions?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From freeswitch at aastral.net  Wed Jan 27 10:44:23 2010
From: freeswitch at aastral.net (Bill W)
Date: Wed, 27 Jan 2010 13:44:23 -0500
Subject: [Freeswitch-users] Strategies for reliably detecting nat
	on	B-leg?
In-Reply-To: <191c3a031001271005t62631e77sf6d9ca406054ba00@mail.gmail.com>
References: <4B60785F.6030505@aastral.net>	<FB139E6B-F685-491C-A672-16844878200E@freeswitch.org>
	<191c3a031001271005t62631e77sf6d9ca406054ba00@mail.gmail.com>
Message-ID: <4B608987.9090606@aastral.net>

Thanks for the reply!

Just to make sure we're on the same page,  my FreeSWITCH sesrver has a 
public IP, and I'm trying to bypass media whenever possible to reduce my 
bandwidth usage.   My concern is trying to bypass media when one of the 
remote endpoints (b-leg) is behind NAT (since I can reliably detect nat 
with aggressive-nat on the A-leg).

Isn't local-network-acl and autonat:x.x.x.x for FS behind nat?

Does this new information change your responses?

Thanks again!
Bill



Anthony Minessale wrote:
> also you can set
> sip_sticky_contact=true
> channel var which will make that session turn on nat lock in the b leg 
> so they can't change the contact to a nat addr
> 
> add it in {} to your dial string like
> 
> {sip_sticky_contact=true}sofia/internal/foo at bar.com <mailto:foo at bar.com>
> 
> 
> 
> 
> On Wed, Jan 27, 2010 at 11:50 AM, Brian West <brian at freeswitch.org 
> <mailto:brian at freeswitch.org>> wrote:
> 
>     update to trunk.  and don't use agressive-nat, set
>     local-network-acl, set the ext-rtp-ip and ext-sip-ip to
>     autonat:x.x.x.x or if you're behind a natpmp or upnp router set it
>     to auto-nat.
> 
>     It should just work.  Again you have no real way to know if the far
>     end client never lies to you.  Which it should never do anyway.
>      Endpoints should know how to traverse their own nat and not leave
>     it up to the registrar to figure it out.
> 
>     /b
> 
>     On Jan 27, 2010, at 11:31 AM, Bill W wrote:
> 
>      > Thoughts?  Suggestions?
> 
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From nicolas at medularis.com  Wed Jan 27 10:49:19 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Wed, 27 Jan 2010 15:49:19 -0300
Subject: [Freeswitch-users] Question about Lua script: How do I detect
	when someone picks up the phone?
In-Reply-To: <87f2f3b91001261449x3401f48eibae516d2b7abc8d4@mail.gmail.com>
References: <5d2828f1001201318s15e2b75fpcb48ad99cad85749@mail.gmail.com>
	<e15d328c1001201332v79efe644v33a11161f476b8ef@mail.gmail.com>
	<5d2828f1001201345t4bcfdbecyf6ce3dc1210acb8c@mail.gmail.com>
	<5d2828f1001201406p4154b98ald1af1c5c25f59337@mail.gmail.com>
	<1b46b4e81001210335l42baef16r8a2952aa5b92f6e6@mail.gmail.com>
	<87f2f3b91001211412h7aa0a84ageadb9557b869ca01@mail.gmail.com>
	<1b46b4e81001221322w5da04799s2ea86000c5c4a9a@mail.gmail.com>
	<87f2f3b91001261449x3401f48eibae516d2b7abc8d4@mail.gmail.com>
Message-ID: <1b46b4e81001271049u527be075n3439028e916181af@mail.gmail.com>

On the contrary, thank you and the whole FreeSWITCH team. Your software and
support are awesome, thank you very much!

Here's something else, a little guide on creating a simple click to call
application with FreeSWITCH:

- Part I: http://www.guayal.com/how-to-bridge-two-calls-with-freeswitch

- Part II: http://www.guayal.com/how-to-create-a-basic-click-to-call-app

I still have to write Part III and see if I keep going.


On Tue, Jan 26, 2010 at 7:49 PM, Michael Collins <msc at freeswitch.org> wrote:

> Thanks for your contributions! They are much appreciated.
> -MC
>
>
> On Fri, Jan 22, 2010 at 1:22 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> No problem, here it is:
>>
>> - http://wiki.freeswitch.org/wiki/Call_retry_based_on_hangup_cause
>>
>> It is linked from your reference (
>> http://wiki.freeswitch.org/wiki/Mod_lua#More_Samples).
>>
>> Sorry I didn't do it early, I hadn't seen your email.
>>
>> I also added another, more complete, example here (also linked):
>>
>> - http://wiki.freeswitch.org/wiki/Bridging_two_calls_with_retry
>>
>>
>>
>> On Thu, Jan 21, 2010 at 7:12 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Thu, Jan 21, 2010 at 3:35 AM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>>
>>>>
>>>> On Wed, Jan 20, 2010 at 7:06 PM, Mike van Lammeren <
>>>> mike at van.lammeren.net> wrote:
>>>>
>>>>> So, I've been reading about early media in the wiki, and have made a
>>>>> little progress, which leads to more questions.
>>>>>
>>>>> I understand now why a call is considered connected before one person
>>>>> has picked up the phone. I am also able to get my script to wait for the
>>>>> phone to be picked up, by setting the ignore_early_media variable when
>>>>> starting a new session, like this:
>>>>>
>>>>> customerSession =
>>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/
>>>>> example.com/" .. customerPhoneNumber)
>>>>>
>>>>>
>>>>> After that line, the script waits for the other phone to be picked up.
>>>>>
>>>>> However, now I wonder what to do with calls that don't complete, get
>>>>> busy signals, etc.
>>>>>
>>>>> What do people do in this case? The only related example I can find on
>>>>> the web is for a javascript dialer, which doesn't address any of these
>>>>> cases.
>>>>>
>>>>
>>>>
>>>> I guess it depends on what you want to do. For example I have a lua
>>>> script very similar to what you describe, although there is no confirmation
>>>> involved. Depending on the hangup cause the session gets, it might try
>>>> redialing with a different gateway, try again or just hangup.
>>>>
>>>> Take a look here http://wiki.freeswitch.org/wiki/Hangup_causes to see
>>>> what each hangup cause means. You don't need to have a special case for all
>>>> of them, only the ones you are interested in.
>>>>
>>>> Here's an example in code which retries a call depending on the hangup
>>>> cause. It retries max_retries1 times and alternates between 2 different
>>>> gateways:
>>>>
>>>> session1 = null;
>>>> max_retries1 = 3;
>>>> retries = 0;
>>>> ostr = "";
>>>> repeat
>>>>         retries = retries + 1;
>>>>         if (retries % 2) then ostr = originate_str1;
>>>>         else ostr = originate_str12; end
>>>>         freeswitch.consoleLog("notice", "*********** Dialing Leg1: " ..
>>>> ostr .. " - Try: "..retries.." ***********\n");
>>>>         session1 = freeswitch.Session(ostr);
>>>>         local hcause = session1:hangupCause();
>>>>         freeswitch.consoleLog("notice", "*********** Leg1: " .. hcause
>>>> .. " - Try: "..retries.." ***********\n");
>>>> until not ((hcause == 'NO_ROUTE_DESTINATION' or hcause ==
>>>> 'RECOVERY_ON_TIMER_EXPIRE' or hcause == 'INCOMPATIBLE_DESTINATION' or hcause
>>>> == 'CALL_REJECTED' or hcause == 'NORMAL_TEMPORARY_FAILURE') and (retries <
>>>> max_retriesl1))
>>>>
>>>>
>>>> Note: originate_str1 and originate_str2 are two different dial strings
>>>> for 2 different gateways.
>>>>
>>>>
>>> Nicolas,
>>>
>>> This is really nice. Would you be willing to add this script and a brief
>>> explanation to the wiki? You could create a whole new page and just link to
>>> it from here: http://wiki.freeswitch.org/wiki/Mod_lua#More_Samples
>>>
>>> If you have any questions please let me know!
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Jan 27 10:50:53 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 27 Jan 2010 12:50:53 -0600
Subject: [Freeswitch-users] Strategies for reliably detecting nat on
	B-leg?
In-Reply-To: <4B608987.9090606@aastral.net>
References: <4B60785F.6030505@aastral.net>	<FB139E6B-F685-491C-A672-16844878200E@freeswitch.org>
	<191c3a031001271005t62631e77sf6d9ca406054ba00@mail.gmail.com>
	<4B608987.9090606@aastral.net>
Message-ID: <3206F734-534B-478B-87A7-A51C2B999E03@freeswitch.org>


On Jan 27, 2010, at 12:44 PM, Bill W wrote:

> 
> Isn't local-network-acl and autonat:x.x.x.x for FS behind nat?

Yes Yes... sorry my misunderstanding.  ;)

> 
> Does this new information change your responses?
> 
> Thanks again!
> Bill




From jcasale at activenetwerx.com  Wed Jan 27 10:54:05 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 27 Jan 2010 18:54:05 +0000
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
In-Reply-To: <20100127180058.29470216DEA@cuneorg-email.cune.pri>
References: <CA5A491E9DEFBE4CB777DE97E21575E90B9E04@prato.activenetwerx.local>
	<20100127180058.29470216DEA@cuneorg-email.cune.pri>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90BC47F@prato.activenetwerx.local>

>It's working OK here. We're actually using 2.2.1RC2 with a TE110P in a VZ
>environment.

Russell,
Appreciate you letting me know this. I had some trouble getting 2.1.0 working
with my TDM410P, not sure how oz handles the difference between these two cards
but I am going to try again.

I was getting all sorts of intermittent behavior of calls working/not working?
CID working, then not working... That was on CentOS 5.4 though before you guys
informed me of the issues, so it likely had nothing to do with dahdi then. I
plan to give this another go now with my 5.3 setup.

Thanks a lot!
jlc



From jonas.gauffin at gmail.com  Wed Jan 27 11:26:00 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Wed, 27 Jan 2010 20:26:00 +0100
Subject: [Freeswitch-users] Call hanging up after 30 minutes (exactly)
Message-ID: <d021d3be1001271126y7a40e209t13e3d5b71af0d1a4@mail.gmail.com>

Hello,

I have problems with calls being hung up after 30 minutes.
I do not know if the problem is with FreeSWITCH or my sip provider.

I've got a log here: http://pastebin.freeswitch.org/11962

Can you please point me in the right direction?

Also, why is a second invite sent after 15 minutes?

Regards,
  Jonas
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From brian at freeswitch.org  Wed Jan 27 11:40:01 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 27 Jan 2010 13:40:01 -0600
Subject: [Freeswitch-users] Call hanging up after 30 minutes (exactly)
In-Reply-To: <d021d3be1001271126y7a40e209t13e3d5b71af0d1a4@mail.gmail.com>
References: <d021d3be1001271126y7a40e209t13e3d5b71af0d1a4@mail.gmail.com>
Message-ID: <ADD964DC-4B88-4DFA-A330-D391BE786F78@freeswitch.org>

Its a session timer.

/b

On Jan 27, 2010, at 1:26 PM, Jonas Gauffin wrote:

> Also, why is a second invite sent after 15 minutes?




From jcasale at activenetwerx.com  Wed Jan 27 11:48:10 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 27 Jan 2010 19:48:10 +0000
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
In-Reply-To: <20100127180058.29470216DEA@cuneorg-email.cune.pri>
References: <CA5A491E9DEFBE4CB777DE97E21575E90B9E04@prato.activenetwerx.local>
	<20100127180058.29470216DEA@cuneorg-email.cune.pri>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90BD7F4@prato.activenetwerx.local>

>It's working OK here. We're actually using 2.2.1RC2 with a TE110P in a VZ
>environment.

Russell,
On that note, what OS are you on?




From jerry.richards at teotech.com  Wed Jan 27 11:56:53 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 27 Jan 2010 11:56:53 -0800
Subject: [Freeswitch-users] Presence Change Distribution
In-Reply-To: <191c3a031001261321v2e8ea21cm7da19e01a11b59f9@mail.gmail.com>
References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com><2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com><26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com><191c3a031001251104p55ba7009g9381841f7de56d65@mail.gmail.com><BE6F5600B8E54EE297963A8B08F70E5E@greyhawk.tonecommander.com>
	<191c3a031001261321v2e8ea21cm7da19e01a11b59f9@mail.gmail.com>
Message-ID: <591B9C113F064880993543272B16ADF3@greyhawk.tonecommander.com>

There are two places in the XML body that are diffierent:
 
FS Rcvd PUBLISH has:  <rpid:away/> and <dm:note>Away</dm:note>
FS Sent NOTIFY   has:  <rpid:busy/> and <dm:note>Busy</dm:note>
 
This behavior (above) is why I'm not seeing the published presence at the
subscribing softphone.  FS should be sending the new Away status in the
NOTIFY message.
 
I did notice there is an "[ERR] sofia_presnece.c:674" FS log between the
PUBLISH and the NOTIFY (please see Line 89 of
http://pastebin.freeswitch.org/11953).  Line 674 is in the
sofia_presence_event_thread_run() function where it calls
switch_mutex_unlock(mod_sofia_globals.mutex).  Do you think this [ERR] is
related to why FS sends the previous status and not updated status?
 
Thanks And Best Regards,
Jerry
 

  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Tuesday, January 26, 2010 1:22 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


its sending a notify to them right away (line 174 of your PB)
the xml in the notify we send looks the same as what they sent except one
thing 

They send: <rpid:away/>
We send: <rpid:busy/>

everybody who implements this seems to have their own idea of what to say
here.

This crazy xml presence crap is pure garbage so maybe that's it.

 


On Tue, Jan 26, 2010 at 3:02 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


Okay, I setup my FS to force SUBSCRIBE Expires to 3600 seconds.  Then I
captured a FS console trace of a Bria softphone changing it's presence state
from 'Busy' to 'Away' (see http://pastebin.freeswitch.org/11953) and
observed that the subscribing Bria softphone did not update to 'Away'.  At
the same time, I executed the sqlite3 app and pasted each of the 3 SQL
select statements I saw in the FS console log, and pasted them below.  I'm
new to sqlite3.  Do you see what my issue is?
 
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'internal' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
sip|5382|192.168.72.79|5401|192.168.72.79|presence|"5382 on 79"
<sip:5382 at 192.168.72.150:34672>|ZTQ2ZWQwZGRlZjRiNTdkYTJjNGM5NTgzOWIyNmIwZmU.
|"5382 on 79" <sip:5382 at 192.168.72.79 <mailto:sip%3A5382 at 192.168.72.79>
>;tag=68bb4eb6|SIP/2.0/UDP
192.168.72.150:34672;branch=z9hG4bK-d8754z-eafc60166305eaef-1---d8754z-;rpor
t=34672|1264546204|Teo Softphone release 2.5.4 stamp
55958||internal|Away|away|192.168.72.79|Away|away
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name =
'internal-ipv6' or sip_subscriptions.presence_hosts !=
sip_subscriptions.sub_to_host);
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite>
sqlite> select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
sip_subscriptions.expires > -1 and (event='presence' or event='presence')
and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'external' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
sqlite>

Thanks and Best Regards,
Jerry


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Monday, January 25, 2010 11:05 AM 

To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


the notify will be instant after the publish
the notify you see are not triggered by the publish or they would be
instant.

Same drill, turn on presence debugging in sofia.conf.xml 
and look at the sql stmts and see why



On Mon, Jan 25, 2010 at 12:30 PM, Jerry Richards
<jerry.richards at teotech.com> wrote:


Okay, I notice that if I reduce the Presence SUBSCRIBE Expires duration
(from 3600 seconds to 60 seconds), then the delay between PUBLISH's and
NOTIFY's is reduced, but FS still waits nearly 45 seconds to send the
NOTIFY's after it receives a PUBLISH.
 
Can a change be made in FS so that NOTIFYs are sent as a direct result of
receipt of the PUBLISH message?  And not tied to the SUBSCRIBE expiration?
I really don't want to configure all my phones to re-subscribe every 30 or
15 seconds.
 
Thanks and Best Regards,
Jerry
 


  _____  

From: RobertT [mailto:siniypin at gmail.com] 
Sent: Tuesday, December 29, 2009 12:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Presence Change Distribution


You can try to reduce your registration time. 
I for one made my client apps send PUBLISH message every minute in addition
to reduced registration time.

Regards, Robert.


2009/12/28 Jerry Richards <jerry.richards at teotech.com>


Is there a setting to control how fast FS distributes presence changes to
subscribers?  Currently, it appears to take several minutes before I see
presence changes.  I would like to see them almost instantaneously, if
possible.

Thanks and Best Regards,
Jerry


_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900



_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


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From jonas.gauffin at gmail.com  Wed Jan 27 11:58:37 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Wed, 27 Jan 2010 20:58:37 +0100
Subject: [Freeswitch-users] Call hanging up after 30 minutes (exactly)
In-Reply-To: <ADD964DC-4B88-4DFA-A330-D391BE786F78@freeswitch.org>
References: <d021d3be1001271126y7a40e209t13e3d5b71af0d1a4@mail.gmail.com>
	<ADD964DC-4B88-4DFA-A330-D391BE786F78@freeswitch.org>
Message-ID: <d021d3be1001271158h352b9ffcn4da299003aeb0f87@mail.gmail.com>

Ok. And it's my sip provider and not FreeSWITCH that do not refresh
properly?


On Wed, Jan 27, 2010 at 8:40 PM, Brian West <brian at freeswitch.org> wrote:

> Its a session timer.
>
> /b
>
> On Jan 27, 2010, at 1:26 PM, Jonas Gauffin wrote:
>
> > Also, why is a second invite sent after 15 minutes?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From troy at tlainvestments.com  Wed Jan 27 12:05:08 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Wed, 27 Jan 2010 13:05:08 -0700
Subject: [Freeswitch-users] Call Dropping with SIP 503 status
In-Reply-To: <ADD964DC-4B88-4DFA-A330-D391BE786F78@freeswitch.org>
References: <d021d3be1001271126y7a40e209t13e3d5b71af0d1a4@mail.gmail.com>
	<ADD964DC-4B88-4DFA-A330-D391BE786F78@freeswitch.org>
Message-ID: <BD4CD56A-D488-4CCD-955D-0A838CB3CFD1@tlainvestments.com>

We are experiencing an odd issue.  We have many calls that don't drop, but some do after being up a minute or two.  

The reason code is NORMAL_TEMPORARY_FAILURE and the sip status that is triggering that is 503 (Service Unavailable).  With only one or two calls up at a time, I don't think it's a session limit issue (set to 1000).

Here is the console log from just before the 503 status - any help is greatly appreciated!

2010-01-27 12:49:11.879251 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
2010-01-27 12:49:11.899334 [INFO] sofia.c:597 Update Callee ID to "400" <400>
2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [ready][200]
2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4011 Duplicate SDP
v=0
o=- 1264621687 1264621687 IN IP4 192.168.0.46
s=Polycom IP Phone
c=IN IP4 192.168.0.46
t=0 0
a=sendrecv
m=audio 2222 RTP/AVP 0 8 18 127
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000

2010-01-27 12:50:06.068999 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [ready][200]
2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4011 Duplicate SDP
v=0
o=- 1264621687 1264621687 IN IP4 192.168.0.46
s=Polycom IP Phone
c=IN IP4 192.168.0.46
t=0 0
a=sendrecv
m=audio 2222 RTP/AVP 0 8 18 127
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000

2010-01-27 12:51:05.259614 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [ready][200]
2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4011 Duplicate SDP
v=0
o=- 1264621687 1264621687 IN IP4 192.168.0.46
s=Polycom IP Phone
c=IN IP4 192.168.0.46
t=0 0
a=sendrecv
m=audio 2222 RTP/AVP 0 8 18 127
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000

2010-01-27 12:52:05.369138 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [ready][200]
2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4011 Duplicate SDP
v=0
o=- 1264621687 1264621687 IN IP4 192.168.0.46
s=Polycom IP Phone
c=IN IP4 192.168.0.46
t=0 0
a=sendrecv
m=audio 2222 RTP/AVP 0 8 18 127
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000

2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [terminating][503]
2010-01-27 12:53:09.599069 [NOTICE] sofia.c:4647 Hangup sofia/internal/400 at 192.168.0.31 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
2010-01-27 12:53:09.599069 [DEBUG] switch_ivr_bridge.c:466 sofia/internal/400 at 192.168.0.31 ending bridge by request from write function

	




From gmaruzz at celliax.org  Wed Jan 27 12:15:30 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 27 Jan 2010 21:15:30 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B608014.4030902@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com> 
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com> 
	<4B6070A8.6050607@gmx.net>
	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com> 
	<4B607944.4040700@gmx.net> <4B608014.4030902@gmx.net>
Message-ID: <7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>

good, so you have only one sound device, the right one.

Use the one with hw:0 in the window that skype gives you to set sound devices

-gm

On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> I installed alsa-utile,
>
> now I get:
>
> skype:/var/cache/apt/archives# aplay -l
> **** List of PLAYBACK Hardware Devices ****
> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
> ?Subdevices: 127/128
> ?Subdevice #0: subdevice #0
> ?Subdevice #1: subdevice #1
> ?Subdevice #2: subdevice #2
> ?Subdevice #3: subdevice #3
> ?Subdevice #4: subdevice #4
> ?Subdevice #5: subdevice #5
> ?Subdevice #6: subdevice #6
> ?Subdevice #7: subdevice #7
> ?Subdevice #8: subdevice #8
> ?Subdevice #9: subdevice #9
> ?Subdevice #10: subdevice #10
> ?Subdevice #11: subdevice #11
> ?Subdevice #12: subdevice #12
> ?Subdevice #13: subdevice #13
> ?Subdevice #14: subdevice #14
> ?Subdevice #15: subdevice #15
> ?Subdevice #16: subdevice #16
> ?Subdevice #17: subdevice #17
> ?Subdevice #18: subdevice #18
> ?Subdevice #19: subdevice #19
> ?Subdevice #20: subdevice #20
> ?Subdevice #21: subdevice #21
> ?Subdevice #22: subdevice #22
> ?Subdevice #23: subdevice #23
> ?Subdevice #24: subdevice #24
> ?Subdevice #25: subdevice #25
> ?Subdevice #26: subdevice #26
> ?Subdevice #27: subdevice #27
> ?Subdevice #28: subdevice #28
> ?Subdevice #29: subdevice #29
> ?Subdevice #30: subdevice #30
> ?Subdevice #31: subdevice #31
> ?Subdevice #32: subdevice #32
> ?Subdevice #33: subdevice #33
> ?Subdevice #34: subdevice #34
> ?Subdevice #35: subdevice #35
> ?Subdevice #36: subdevice #36
> ?Subdevice #37: subdevice #37
> ?Subdevice #38: subdevice #38
> ?Subdevice #39: subdevice #39
> ?Subdevice #40: subdevice #40
> ?Subdevice #41: subdevice #41
> ?Subdevice #42: subdevice #42
> ?Subdevice #43: subdevice #43
> ?Subdevice #44: subdevice #44
> ?Subdevice #45: subdevice #45
> ?Subdevice #46: subdevice #46
> ?Subdevice #47: subdevice #47
> ?Subdevice #48: subdevice #48
> ?Subdevice #49: subdevice #49
> ?Subdevice #50: subdevice #50
> ?Subdevice #51: subdevice #51
> ?Subdevice #52: subdevice #52
> ?Subdevice #53: subdevice #53
> ?Subdevice #54: subdevice #54
> ?Subdevice #55: subdevice #55
> ?Subdevice #56: subdevice #56
> ?Subdevice #57: subdevice #57
> ?Subdevice #58: subdevice #58
> ?Subdevice #59: subdevice #59
> ?Subdevice #60: subdevice #60
> ?Subdevice #61: subdevice #61
> ?Subdevice #62: subdevice #62
> ?Subdevice #63: subdevice #63
> ?Subdevice #64: subdevice #64
> ?Subdevice #65: subdevice #65
> ?Subdevice #66: subdevice #66
> ?Subdevice #67: subdevice #67
> ?Subdevice #68: subdevice #68
> ?Subdevice #69: subdevice #69
> ?Subdevice #70: subdevice #70
> ?Subdevice #71: subdevice #71
> ?Subdevice #72: subdevice #72
> ?Subdevice #73: subdevice #73
> ?Subdevice #74: subdevice #74
> ?Subdevice #75: subdevice #75
> ?Subdevice #76: subdevice #76
> ?Subdevice #77: subdevice #77
> ?Subdevice #78: subdevice #78
> ?Subdevice #79: subdevice #79
> ?Subdevice #80: subdevice #80
> ?Subdevice #81: subdevice #81
> ?Subdevice #82: subdevice #82
> ?Subdevice #83: subdevice #83
> ?Subdevice #84: subdevice #84
> ?Subdevice #85: subdevice #85
> ?Subdevice #86: subdevice #86
> ?Subdevice #87: subdevice #87
> ?Subdevice #88: subdevice #88
> ?Subdevice #89: subdevice #89
> ?Subdevice #90: subdevice #90
> ?Subdevice #91: subdevice #91
> ?Subdevice #92: subdevice #92
> ?Subdevice #93: subdevice #93
> ?Subdevice #94: subdevice #94
> ?Subdevice #95: subdevice #95
> ?Subdevice #96: subdevice #96
> ?Subdevice #97: subdevice #97
> ?Subdevice #98: subdevice #98
> ?Subdevice #99: subdevice #99
> ?Subdevice #100: subdevice #100
> ?Subdevice #101: subdevice #101
> ?Subdevice #102: subdevice #102
> ?Subdevice #103: subdevice #103
> ?Subdevice #104: subdevice #104
> ?Subdevice #105: subdevice #105
> ?Subdevice #106: subdevice #106
> ?Subdevice #107: subdevice #107
> ?Subdevice #108: subdevice #108
> ?Subdevice #109: subdevice #109
> ?Subdevice #110: subdevice #110
> ?Subdevice #111: subdevice #111
> ?Subdevice #112: subdevice #112
> ?Subdevice #113: subdevice #113
> ?Subdevice #114: subdevice #114
> ?Subdevice #115: subdevice #115
> ?Subdevice #116: subdevice #116
> ?Subdevice #117: subdevice #117
> ?Subdevice #118: subdevice #118
> ?Subdevice #119: subdevice #119
> ?Subdevice #120: subdevice #120
> ?Subdevice #121: subdevice #121
> ?Subdevice #122: subdevice #122
> ?Subdevice #123: subdevice #123
> ?Subdevice #124: subdevice #124
> ?Subdevice #125: subdevice #125
> ?Subdevice #126: subdevice #126
> ?Subdevice #127: subdevice #127
>
>
> Peter P GMX schrieb:
>> Her's the output:
>>
>> skype:~# aplay -l
>> bash: aplay: command not found
>>
>> Giovanni Maruzzelli schrieb:
>>
>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>> what's the output of:
>>>
>>> aplay -l
>>>
>>> ?
>>>
>>> If instead you are referring to the choices that skype clients offers
>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>
>>> Eg: not the "default", but the "hardware" one
>>>
>>>
>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>
>>>> Thanks Giovanni,
>>>>
>>>> I think there may be the problem, that I have 2 sound devices now:
>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>> since I compiled alsa newly)
>>>>
>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>> (apt-get remove alsa-utils alsa-base).
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>>
>>>> Giovanni Maruzzelli schrieb:
>>>>
>>>>
>>>>> This warning is harmless:
>>>>>
>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>
>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>> <gmaruzz at celliax.org> wrote:
>>>>>
>>>>>
>>>>>
>>>>>> Ciao Peter
>>>>>>
>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>> snd-dummy-customized.
>>>>>>
>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>> 8.04).
>>>>>>
>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>
>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>> do this with the modprobe command, as in:
>>>>>>
>>>>>> rmmod snd-dummy
>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>
>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>> soundcard.
>>>>>>
>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>> just committed some improvements.
>>>>>>
>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>> write the mailing list again,
>>>>>>
>>>>>> ciao for now,
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>
>>>>>>
>>>>>>
>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>> to all contributors, excellent work!
>>>>>>>
>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>> instructions in
>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>> ist Debian 5.0R3)
>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>> one-way-audio and my logs show
>>>>>>>
>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>> working on a machine with 250HZ kernel
>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>> suppressed
>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>> suppressed
>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>> with the current limitations).
>>>>>>>
>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>
>>>>>>> Second question:
>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>
>>>>>>> Best regards
>>>>>>> Peter
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>> --
>>>>>> Sincerely,
>>>>>>
>>>>>> Giovanni Maruzzelli
>>>>>> Cell : +39-347-2665618
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>
>>>
>>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From Russell.Mosemann at cune.org  Wed Jan 27 13:08:36 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Wed, 27 Jan 2010 21:08:36 -0000
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E90BD7F4@prato.activenetwerx.local>
Message-ID: <20100127210836.5ED5D2E3B2D@cuneorg-email.cune.pri>

"Joseph L. Casale" <jcasale at activenetwerx.com> said:

> On that note, what OS are you on?

Debian 5.0.3

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From anthony.minessale at gmail.com  Wed Jan 27 13:08:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 15:08:38 -0600
Subject: [Freeswitch-users] Presence Change Distribution
In-Reply-To: <591B9C113F064880993543272B16ADF3@greyhawk.tonecommander.com>
References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com>
	<2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com>
	<26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com>
	<191c3a031001251104p55ba7009g9381841f7de56d65@mail.gmail.com>
	<BE6F5600B8E54EE297963A8B08F70E5E@greyhawk.tonecommander.com>
	<191c3a031001261321v2e8ea21cm7da19e01a11b59f9@mail.gmail.com>
	<591B9C113F064880993543272B16ADF3@greyhawk.tonecommander.com>
Message-ID: <191c3a031001271308l5c0c4eedw925e7660fbc2069d@mail.gmail.com>

Try latest trunk.
I tried forcing the db update in real-time to avoid a race on the event.





On Wed, Jan 27, 2010 at 1:56 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  There are two places in the XML body that are diffierent:
>
> FS Rcvd PUBLISH has:  <rpid:away/> and <dm:note>Away</dm:note>
> FS Sent NOTIFY   has:  <rpid:busy/> and <dm:note>Busy</dm:note>
>
> This behavior (above) is why I'm not seeing the published presence at the
> subscribing softphone.  FS should be sending the new Away status in the
> NOTIFY message.
>
> I did notice there is an "[ERR] sofia_presnece.c:674" FS log between the
> PUBLISH and the NOTIFY (please see Line 89 of
> http://pastebin.freeswitch.org/11953).  Line 674 is in the
> sofia_presence_event_thread_run() function where it calls
> switch_mutex_unlock(mod_sofia_globals.mutex).  Do you think this [ERR] is
> related to why FS sends the previous status and not updated status?
>
> Thanks And Best Regards,
> Jerry
>
>
>  ------------------------------
> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
> *Sent:* Tuesday, January 26, 2010 1:22 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>
> its sending a notify to them right away (line 174 of your PB)
> the xml in the notify we send looks the same as what they sent except one
> thing
>
> They send: <rpid:away/>
> We send: <rpid:busy/>
>
> everybody who implements this seems to have their own idea of what to say
> here.
>
> This crazy xml presence crap is pure garbage so maybe that's it.
>
>
>
> On Tue, Jan 26, 2010 at 3:02 PM, Jerry Richards <
> jerry.richards at teotech.com> wrote:
>
>>  Okay, I setup my FS to force SUBSCRIBE Expires to 3600 seconds.  Then I
>> captured a FS console trace of a Bria softphone changing it's presence state
>> from 'Busy' to 'Away' (see http://pastebin.freeswitch.org/11953) and
>> observed that the subscribing Bria softphone did not update to 'Away'.  At
>> the same time, I executed the sqlite3 app and pasted each of the 3 SQL
>> select statements I saw in the FS console log, and pasted them below.  I'm
>> new to sqlite3.  Do you see what my issue is?
>>
>> sqlite>
>> sqlite> select
>> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
>> from sip_subscriptions left join sip_presence on
>> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
>> sip_subscriptions.sub_to_host=sip_presence.sip_host and
>> sip_subscriptions.profile_name=sip_presence.profile_name) where
>> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
>> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
>> like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'internal' or
>> sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
>> sip|5382|192.168.72.79|5401|192.168.72.79|presence|"5382 on 79" <
>> sip:5382 at 192.168.72.150:34672>|ZTQ2ZWQwZGRlZjRiNTdkYTJjNGM5NTgzOWIyNmIwZmU.|"5382
>> on 79" <sip:5382 at 192.168.72.79 <sip%3A5382 at 192.168.72.79>>;tag=68bb4eb6|SIP/2.0/UDP
>> 192.168.72.150:34672;branch=z9hG4bK-d8754z-eafc60166305eaef-1---d8754z-;rport=34672|1264546204|Teo
>> Softphone release 2.5.4 stamp
>> 55958||internal|Away|away|192.168.72.79|Away|away
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite> select
>> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
>> from sip_subscriptions left join sip_presence on
>> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
>> sip_subscriptions.sub_to_host=sip_presence.sip_host and
>> sip_subscriptions.profile_name=sip_presence.profile_name) where
>> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
>> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
>> like '%192.168.72.79%') and (sip_subscriptions.profile_name =
>> 'internal-ipv6' or sip_subscriptions.presence_hosts !=
>> sip_subscriptions.sub_to_host);
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite> select
>> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
>> from sip_subscriptions left join sip_presence on
>> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
>> sip_subscriptions.sub_to_host=sip_presence.sip_host and
>> sip_subscriptions.profile_name=sip_presence.profile_name) where
>> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
>> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
>> like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'external' or
>> sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
>> sqlite>
>>  Thanks and Best Regards,
>> Jerry
>>
>>  ------------------------------
>> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
>> *Sent:* Monday, January 25, 2010 11:05 AM
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>>
>>   the notify will be instant after the publish
>> the notify you see are not triggered by the publish or they would be
>> instant.
>>
>> Same drill, turn on presence debugging in sofia.conf.xml
>> and look at the sql stmts and see why
>>
>>
>> On Mon, Jan 25, 2010 at 12:30 PM, Jerry Richards <
>> jerry.richards at teotech.com> wrote:
>>
>>>  Okay, I notice that if I reduce the Presence SUBSCRIBE Expires duration
>>> (from 3600 seconds to 60 seconds), then the delay between PUBLISH's and
>>> NOTIFY's is reduced, but FS still waits nearly 45 seconds to send the
>>> NOTIFY's after it receives a PUBLISH.
>>>
>>> Can a change be made in FS so that NOTIFYs are sent as a direct result of
>>> receipt of the PUBLISH message?  And not tied to the SUBSCRIBE expiration?
>>> I really don't want to configure all my phones to re-subscribe every 30 or
>>> 15 seconds.
>>>
>>> Thanks and Best Regards,
>>> Jerry
>>>
>>>
>>>  ------------------------------
>>> *From:* RobertT [mailto:siniypin at gmail.com]
>>> *Sent:* Tuesday, December 29, 2009 12:02 AM
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>>>
>>>   You can try to reduce your registration time.
>>> I for one made my client apps send PUBLISH message every minute in
>>> addition to reduced registration time.
>>>
>>> Regards, Robert.
>>>
>>> 2009/12/28 Jerry Richards <jerry.richards at teotech.com>
>>>
>>>> Is there a setting to control how fast FS distributes presence changes
>>>> to
>>>> subscribers?  Currently, it appears to take several minutes before I see
>>>> presence changes.  I would like to see them almost instantaneously, if
>>>> possible.
>>>>
>>>> Thanks and Best Regards,
>>>> Jerry
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Jan 27 14:31:04 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 16:31:04 -0600
Subject: [Freeswitch-users] Call Dropping with SIP 503 status
In-Reply-To: <BD4CD56A-D488-4CCD-955D-0A838CB3CFD1@tlainvestments.com>
References: <d021d3be1001271126y7a40e209t13e3d5b71af0d1a4@mail.gmail.com>
	<ADD964DC-4B88-4DFA-A330-D391BE786F78@freeswitch.org>
	<BD4CD56A-D488-4CCD-955D-0A838CB3CFD1@tlainvestments.com>
Message-ID: <191c3a031001271431re6cfad2w1389e96db4e6c1a0@mail.gmail.com>

try turning on sip trace as well to see the sip traffic

sofia profile internal siptrace on (from cli)
probably its something that said it could do session timers but was lying


On Wed, Jan 27, 2010 at 2:05 PM, Troy Anderson <troy at tlainvestments.com>wrote:

> We are experiencing an odd issue.  We have many calls that don't drop, but
> some do after being up a minute or two.
>
> The reason code is NORMAL_TEMPORARY_FAILURE and the sip status that is
> triggering that is 503 (Service Unavailable).  With only one or two calls up
> at a time, I don't think it's a session limit issue (set to 1000).
>
> Here is the console log from just before the 503 status - any help is
> greatly appreciated!
>
> 2010-01-27 12:49:11.879251 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:49:11.899334 [INFO] sofia.c:597 Update Callee ID to "400"
> <400>
> 2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [ready][200]
> 2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4011 Duplicate SDP
> v=0
> o=- 1264621687 1264621687 IN IP4 192.168.0.46
> s=Polycom IP Phone
> c=IN IP4 192.168.0.46
> t=0 0
> a=sendrecv
> m=audio 2222 RTP/AVP 0 8 18 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
>
> 2010-01-27 12:50:06.068999 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [ready][200]
> 2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4011 Duplicate SDP
> v=0
> o=- 1264621687 1264621687 IN IP4 192.168.0.46
> s=Polycom IP Phone
> c=IN IP4 192.168.0.46
> t=0 0
> a=sendrecv
> m=audio 2222 RTP/AVP 0 8 18 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
>
> 2010-01-27 12:51:05.259614 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [ready][200]
> 2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4011 Duplicate SDP
> v=0
> o=- 1264621687 1264621687 IN IP4 192.168.0.46
> s=Polycom IP Phone
> c=IN IP4 192.168.0.46
> t=0 0
> a=sendrecv
> m=audio 2222 RTP/AVP 0 8 18 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
>
> 2010-01-27 12:52:05.369138 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [ready][200]
> 2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4011 Duplicate SDP
> v=0
> o=- 1264621687 1264621687 IN IP4 192.168.0.46
> s=Polycom IP Phone
> c=IN IP4 192.168.0.46
> t=0 0
> a=sendrecv
> m=audio 2222 RTP/AVP 0 8 18 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
>
> 2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/
> 400 at 192.168.0.31 entering state [terminating][503]
> 2010-01-27 12:53:09.599069 [NOTICE] sofia.c:4647 Hangup sofia/internal/
> 400 at 192.168.0.31 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
> 2010-01-27 12:53:09.599069 [DEBUG] switch_ivr_bridge.c:466 sofia/internal/
> 400 at 192.168.0.31 ending bridge by request from write function
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From wiltingtree at gmail.com  Wed Jan 27 16:15:48 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Wed, 27 Jan 2010 19:15:48 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
Message-ID: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>

Hi, I followed the instructions in the Lua documentation for setting up
luasql, but when I try to run my script I get:

2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module
'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
        /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment prot
after reloc: Permission denied
stack traceback:
        [C]: ?
        [C]: in function 'require'
        /usr/local/freeswitch/scripts/l.lua:2: in main chunk

I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.

I changed the permissions for mysql.so and for my script to 777, so I'm not
sure where the permission problem could be.

I'd appreciate any suggestions.

Thanks,
Adam
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From anthony.minessale at gmail.com  Wed Jan 27 17:02:44 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 19:02:44 -0600
Subject: [Freeswitch-users] Event socket: filter delete isn't working
In-Reply-To: <fe67a3341001270514g2715de2ag6aa2e3b099fa7816@mail.gmail.com>
References: <fe67a3341001270514g2715de2ag6aa2e3b099fa7816@mail.gmail.com>
Message-ID: <191c3a031001271702l3e77c952lc8aba35bacbf3d58@mail.gmail.com>

in the future please report issues to jira http://jira.freeswitch.org

please try svn trunk 16527 or higher

This was not a bug but I made it work the way you describe since it made
sense.

you should have done

filter delete unique-id

which would have delete all the unique-id filters that was the only option


you should be able to now say

filter delete unique-id <value>

To delete entry with specific value

or

filter delete unique-id

to delete all entries with matching key





On Wed, Jan 27, 2010 at 7:14 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Dear friends,
>    I've tried to delete the filter which I applied for an unique id. But,
> it doesn't work. After executing 'filter delete', I am receiving the events
> from that uuid.
>    I used the command as 'filter delete unique-id
> c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565'.
>
> I did the following operations.
>     Made call to the event socket.
>     Registered events for all. (events plain all).
>     Applied filter for the uuid. (filter unique-id
> aa3cb8ea-0b2f-11df-9e84-fb15c3cd8565).
>     I've got a new uuid by using create_uuid.
>     Applied filter for this new uuid. (filter unique-id
> c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565)
>     Originated a call with that uuid.
>     Now, I could receive events from both uuids. (Tested by giving DTMFs in
> both end and checked unique-id in event header).
>     Then, I wanted to delete a uuid from the filter. (filter delete
> unique-id c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565).
>     I thought, i won't receive the events from this deleted unique-id. But,
> I received the dtmfs from both unique-id.
>
> I'm using 'FreeSWITCH Version 1.0.trunk (15982)'.
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From david.villasmil.work at gmail.com  Wed Jan 27 17:09:33 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 28 Jan 2010 02:09:33 +0100
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
Message-ID: <9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>

I got the same error, my script was working with no problems before an
update to trunk.

David

On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com> wrote:
> Hi, I followed the instructions in the Lua documentation for setting up
> luasql, but when I try to run my script I get:
> 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module
> 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
> ?? ? ? ?/usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment prot
> after reloc: Permission denied
> stack traceback:
> ?? ? ? ?[C]: ?
> ?? ? ? ?[C]: in function 'require'
> ?? ? ? ?/usr/local/freeswitch/scripts/l.lua:2: in main chunk
> I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
> I changed the permissions for mysql.so and for my script to 777, so I'm not
> sure where the permission problem could be.
> I'd appreciate any suggestions.
> Thanks,
> Adam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From anthony.minessale at gmail.com  Wed Jan 27 17:09:41 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 19:09:41 -0600
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
Message-ID: <191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>

user and host have to match too

On Wed, Jan 27, 2010 at 7:06 AM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Thanks guys.  I have this working except for one user who is
> registered like this:
>
> Call-ID:        2ff39277-fd9a6ab1 at 10.0.0.1
> User:           2000 at 192.168.1.120
> Contact:        2000
> <sip:2000 at 124.xxx.xxx.xxx
> :5075;transport=tls;fs_nat=yes;fs_path=sip%3A2000%40124.xxx.xxx.xxx%3A10281%3Btransport%3Dtls>
> Agent:          Linksys/SPA3102-5.1.10(GW)
> Status:         Registered(TLS-NAT)(unknown) EXP(2010-01-28 00:29:34)
> Host:           freeswitch
> IP:             124.xxx.xxx.xxx
> Port:           10281
> Auth-User:      2000
> Auth-Realm:     mydns.dyndns.org
> MWI-Account:    2000 at 192.168.1.120
>
> When I do the following commands via the telnet socket, no notify
> command is sent to user 2000:
>
> sendevent NOTIFY
> profile: internal
> content-type: application/simple-message-summary
> event-string: reboot_now
> user: 2000
> host: 192.168.1.120
> content-length: 0
>
> However, if I do exactly the same thing with user 2001 it works.  2001
> is registered as:
>
> Contact:        2001 <sip:2001 at 124.yyy.yyy.yyy:5066;transport=tls>
>
> Any ideas why that would be?
>
> On Mon, Jan 25, 2010 at 11:50 AM, Ognjen Seslija <oseslija at gmail.com>
> wrote:
> > The phone is asking FS to authenticate prior then accepting a NOTIFY from
> > it.
> > The authentication of notify's from spa endpoints work (afaik) only with
> > Linksys SPA9000 PBX , and FS doesn't have the code for that for obvious
> > reasons.
> > If you have SPA9000 maybe you can collect SIP traces.
> >
> > Ognjen
> >
> >
> >
> >
> >
> > On Mon, Jan 25, 2010 at 1:29 AM, Mark Campbell-Smith
> > <mcampbellsmith at gmail.com> wrote:
> >>
> >> Hi Ognjen,
> >>
> >> Thanks for the tip on the resync under the EXT tab.  It now works
> >> using mod_event_socket and the following:
> >>
> >> sendevent NOTIFY
> >> profile: internal
> >> event-string: resync
> >> user: 1000
> >> host: 192.168.1.121
> >> content-type: application/simple-message-summary
> >>
> >> However, if AUTH is required, why does FS send the wrong information to
> >> the SPA?
> >>
> >> On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija <oseslija at gmail.com>
> >> wrote:
> >> > You  should not authenticate those NOTIFYs (this will work only with
> >> > SPA9000
> >> > afaik). The option to change for this is in EXT tabs:
> >> >
> >> > Auth Resync-Reboot: No
> >> >
> >> > Also, FSs code will do a reboot of a phone, not resync (it sends
> >> > reboot_now
> >> > event). For that to work a patch is required.
> >> >
> >> > I've just tried to reboot my 942 (rev 16506) and it definitely works.
> >> >
> >> > Regards,
> >> > Ognjen
> >> >
> >> >
> >> > On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
> >> > <mcampbellsmith at gmail.com> wrote:
> >> >>
> >> >> Thanks Brian.. this still does not work.  Maybe I need to open a
> Jira?
> >> >>  Notice the username in the authorization field.  It should be 1000.
> >> >>
> >> >> Cheers
> >> >> Mark
> >> >>
> >> >> freeswitch at internal> sofia status profile internal user
> >> >> 1000 at 192.168.1.120
> >> >>
> >> >> Registrations:
> >> >>
> >> >>
> >> >>
> =================================================================================================
> >> >> Call-ID:        bd783b73-66877627 at 192.168.1.121
> >> >> User:           1000 at 192.168.1.120
> >> >> Contact:        1000 <sip:1000 at 192.168.1.121:5060>
> >> >> Agent:          Linksys/PAP2T-5.1.6(LS)
> >> >> Status:         Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
> >> >> Host:           freeswitch
> >> >> IP:             192.168.1.121
> >> >> Port:           5060
> >> >> Auth-User:      1000
> >> >> Auth-Realm:     192.168.1.120
> >> >> MWI-Account:    1000 at 192.168.1.120
> >> >>
> >> >>
> >> >>
> >> >>
> =================================================================================================
> >> >>
> >> >> freeswitch at internal> sofia profile internal flush_inbound_reg
> >> >> bd783b73-66877627 at 192.168.1.121 reboot
> >> >> +OK rebooting all registrations matching specified call_id
> >> >>
> >> >> freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060 at
> >> >> 23:55:49.012627:
> >> >>
> >> >>
> ------------------------------------------------------------------------
> >> >>   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
> >> >>   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
> >> >>   Max-Forwards: 70
> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >>   CSeq: 126070338 NOTIFY
> >> >>   Contact: <sip:mod_sofia at 192.168.1.120:5060>
> >> >>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
> >> >>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >>   Supported: timer, precondition, path, replaces
> >> >>   Event: reboot_now
> >> >>   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >> >> include-session-description, presence.winfo, message-summary, refer
> >> >>   Subscription-State: terminated;reason=timeout
> >> >>   Content-Type: application/simple-message-summary
> >> >>   Content-Length: 0
> >> >>
> >> >>
> >> >>
> ------------------------------------------------------------------------
> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
> >> >>
> >> >>
> ------------------------------------------------------------------------
> >> >>   SIP/2.0 401 Unauthorized
> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >>   CSeq: 126070338 NOTIFY
> >> >>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
> >> >>   Server: Linksys/PAP2T-5.1.6(LS)
> >> >>   WWW-Authenticate: Digest realm="192.168.1.120", nonce="8e54805b",
> >> >> qop="auth", algorithm=md5
> >> >>   Content-Length: 0
> >> >>
> >> >>
> >> >>
> ------------------------------------------------------------------------
> >> >> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
> >> >>
> >> >>
> ------------------------------------------------------------------------
> >> >>   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
> >> >>   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
> >> >>   Max-Forwards: 70
> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >>   CSeq: 126070339 NOTIFY
> >> >>   Contact: <sip:mod_sofia at 192.168.1.120:5060>
> >> >>   Expires: 3590
> >> >>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
> >> >>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >>   Supported: timer, precondition, path, replaces
> >> >>   Event: reboot_now
> >> >>   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >> >> include-session-description, presence.winfo, message-summary, refer
> >> >>   Subscription-State: terminated;reason=timeout
> >> >>   Authorization: Digest username="1115633124", realm="192.168.1.120",
> >> >> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
> >> >> uri="sip:1000 at 192.168.1.121:5060",
> >> >> response="747b4d04544c84535dbbd987f2999ca7", qop=auth, nc=00000001
> >> >>   Content-Type: application/simple-message-summary
> >> >>   Content-Length: 0
> >> >>
> >> >>
> >> >>
> ------------------------------------------------------------------------
> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
> >> >>
> >> >>
> ------------------------------------------------------------------------
> >> >>   SIP/2.0 401 Unauthorized
> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >>   CSeq: 126070339 NOTIFY
> >> >>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
> >> >>   Server: Linksys/PAP2T-5.1.6(LS)
> >> >>   WWW-Authenticate: Digest realm="192.168.1.120", nonce="5339c7ba",
> >> >> qop="auth", algorithm=md5
> >> >>   Content-Length: 0
> >> >>
> >> >>
> >> >>
> ------------------------------------------------------------------------
> >> >>
> >> >> On Mon, Jan 25, 2010 at 4:46 AM, Brian West <brian at freeswitch.org>
> >> >> wrote:
> >> >> > or sofia profile xxx flush_inbound_reg callid reboot
> >> >> > callid you can get from sofia status profile xxx
> >> >> > /b
> >> >> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
> >> >> >
> >> >> > Actually I just
> >> >> > found http://wiki.freeswitch.org/wiki/Mod_event_socket
> >> >> >
> >> >> > If I telnet to FS as described
> >> >> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do
> I
> >> >> > just need to enter somthing like:
> >> >> >
> >> >> > sendevent NOTIFY
> >> >> > profile: internal
> >> >> > event-string: resync
> >> >> > user: 1000
> >> >> > host: 192.168.1.121
> >> >> > content-type: application/simple-message-summary
> >> >> >
> >> >> > where 192.168.1.121 is the ip address of one of the Linksys
> devices?
> >> >> >
> >> >> > I don't see any messages sent when I do this.  What am I doing
> wrong?
> >> >> >
> >> >> > Thanks
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at van.lammeren.net  Wed Jan 27 17:13:36 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Wed, 27 Jan 2010 20:13:36 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
Message-ID: <5d2828f1001271713x5a86d8cjc11e5609bddd5b43@mail.gmail.com>

For me, the mysql.so library didn't work until I ran ldconfig on the
directory that contained it.

Mike van Lammeren


On Wed, Jan 27, 2010 at 7:15 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Hi, I followed the instructions in the Lua documentation for setting up
> luasql, but when I try to run my script I get:
>
> 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module
> 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
>         /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment prot
> after reloc: Permission denied
> stack traceback:
>         [C]: ?
>         [C]: in function 'require'
>         /usr/local/freeswitch/scripts/l.lua:2: in main chunk
>
> I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
>
> I changed the permissions for mysql.so and for my script to 777, so I'm not
> sure where the permission problem could be.
>
> I'd appreciate any suggestions.
>
> Thanks,
> Adam
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Wed Jan 27 17:26:28 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 28 Jan 2010 12:26:28 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
Message-ID: <33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>

Thanks Anthony,

I think user matches (ie the extension 2000 or 2001).   What should
host be?  In the sofia printout, it says 'freeswitch' (freeswitch has
ip address 192.168.1.120).

However, if I try to use 'freeswitch' as the host for user 2001,
nothing is sent.  But using 192.168.1.120 does.

If I do exactly the same thing for 2000, the NOTIFY message is not
sent.  Are there logs I can send to show you or any ideas what I am
doing wrong?



On Thu, Jan 28, 2010 at 12:09 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> user and host have to match too
>
> On Wed, Jan 27, 2010 at 7:06 AM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>>
>> Thanks guys. ?I have this working except for one user who is
>> registered like this:
>>
>> Call-ID: ? ? ? ?2ff39277-fd9a6ab1 at 10.0.0.1
>> User: ? ? ? ? ? 2000 at 192.168.1.120
>> Contact: ? ? ? ?2000
>>
>> <sip:2000 at 124.xxx.xxx.xxx:5075;transport=tls;fs_nat=yes;fs_path=sip%3A2000%40124.xxx.xxx.xxx%3A10281%3Btransport%3Dtls>
>> Agent: ? ? ? ? ?Linksys/SPA3102-5.1.10(GW)
>> Status: ? ? ? ? Registered(TLS-NAT)(unknown) EXP(2010-01-28 00:29:34)
>> Host: ? ? ? ? ? freeswitch
>> IP: ? ? ? ? ? ? 124.xxx.xxx.xxx
>> Port: ? ? ? ? ? 10281
>> Auth-User: ? ? ?2000
>> Auth-Realm: ? ? mydns.dyndns.org
>> MWI-Account: ? ?2000 at 192.168.1.120
>>
>> When I do the following commands via the telnet socket, no notify
>> command is sent to user 2000:
>>
>> sendevent NOTIFY
>> profile: internal
>> content-type: application/simple-message-summary
>> event-string: reboot_now
>> user: 2000
>> host: 192.168.1.120
>> content-length: 0
>>
>> However, if I do exactly the same thing with user 2001 it works. ?2001
>> is registered as:
>>
>> Contact: ? ? ? ?2001 <sip:2001 at 124.yyy.yyy.yyy:5066;transport=tls>
>>
>> Any ideas why that would be?
>>
>> On Mon, Jan 25, 2010 at 11:50 AM, Ognjen Seslija <oseslija at gmail.com>
>> wrote:
>> > The phone is asking FS to authenticate prior then accepting a NOTIFY
>> > from
>> > it.
>> > The authentication of notify's from spa endpoints work (afaik) only with
>> > Linksys SPA9000 PBX , and FS doesn't have the code for that for obvious
>> > reasons.
>> > If you have SPA9000 maybe you can collect SIP traces.
>> >
>> > Ognjen
>> >
>> >
>> >
>> >
>> >
>> > On Mon, Jan 25, 2010 at 1:29 AM, Mark Campbell-Smith
>> > <mcampbellsmith at gmail.com> wrote:
>> >>
>> >> Hi Ognjen,
>> >>
>> >> Thanks for the tip on the resync under the EXT tab. ?It now works
>> >> using mod_event_socket and the following:
>> >>
>> >> sendevent NOTIFY
>> >> profile: internal
>> >> event-string: resync
>> >> user: 1000
>> >> host: 192.168.1.121
>> >> content-type: application/simple-message-summary
>> >>
>> >> However, if AUTH is required, why does FS send the wrong information to
>> >> the SPA?
>> >>
>> >> On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija <oseslija at gmail.com>
>> >> wrote:
>> >> > You? should not authenticate those NOTIFYs (this will work only with
>> >> > SPA9000
>> >> > afaik). The option to change for this is in EXT tabs:
>> >> >
>> >> > Auth Resync-Reboot: No
>> >> >
>> >> > Also, FSs code will do a reboot of a phone, not resync (it sends
>> >> > reboot_now
>> >> > event). For that to work a patch is required.
>> >> >
>> >> > I've just tried to reboot my 942 (rev 16506) and it definitely works.
>> >> >
>> >> > Regards,
>> >> > Ognjen
>> >> >
>> >> >
>> >> > On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
>> >> > <mcampbellsmith at gmail.com> wrote:
>> >> >>
>> >> >> Thanks Brian.. this still does not work. ?Maybe I need to open a
>> >> >> Jira?
>> >> >> ?Notice the username in the authorization field. ?It should be 1000.
>> >> >>
>> >> >> Cheers
>> >> >> Mark
>> >> >>
>> >> >> freeswitch at internal> sofia status profile internal user
>> >> >> 1000 at 192.168.1.120
>> >> >>
>> >> >> Registrations:
>> >> >>
>> >> >>
>> >> >>
>> >> >> =================================================================================================
>> >> >> Call-ID: ? ? ? ?bd783b73-66877627 at 192.168.1.121
>> >> >> User: ? ? ? ? ? 1000 at 192.168.1.120
>> >> >> Contact: ? ? ? ?1000 <sip:1000 at 192.168.1.121:5060>
>> >> >> Agent: ? ? ? ? ?Linksys/PAP2T-5.1.6(LS)
>> >> >> Status: ? ? ? ? Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
>> >> >> Host: ? ? ? ? ? freeswitch
>> >> >> IP: ? ? ? ? ? ? 192.168.1.121
>> >> >> Port: ? ? ? ? ? 5060
>> >> >> Auth-User: ? ? ?1000
>> >> >> Auth-Realm: ? ? 192.168.1.120
>> >> >> MWI-Account: ? ?1000 at 192.168.1.120
>> >> >>
>> >> >>
>> >> >>
>> >> >>
>> >> >> =================================================================================================
>> >> >>
>> >> >> freeswitch at internal> sofia profile internal flush_inbound_reg
>> >> >> bd783b73-66877627 at 192.168.1.121 reboot
>> >> >> +OK rebooting all registrations matching specified call_id
>> >> >>
>> >> >> freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060 at
>> >> >> 23:55:49.012627:
>> >> >>
>> >> >>
>> >> >> ------------------------------------------------------------------------
>> >> >> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> >> >> ? Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
>> >> >> ? Max-Forwards: 70
>> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> ? To: <sip:1000 at 192.168.1.120>
>> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> ? CSeq: 126070338 NOTIFY
>> >> >> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> >> >> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> >> >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> >> ? Supported: timer, precondition, path, replaces
>> >> >> ? Event: reboot_now
>> >> >> ? Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> >> >> include-session-description, presence.winfo, message-summary, refer
>> >> >> ? Subscription-State: terminated;reason=timeout
>> >> >> ? Content-Type: application/simple-message-summary
>> >> >> ? Content-Length: 0
>> >> >>
>> >> >>
>> >> >>
>> >> >> ------------------------------------------------------------------------
>> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
>> >> >>
>> >> >>
>> >> >> ------------------------------------------------------------------------
>> >> >> ? SIP/2.0 401 Unauthorized
>> >> >> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> ? CSeq: 126070338 NOTIFY
>> >> >> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
>> >> >> ? Server: Linksys/PAP2T-5.1.6(LS)
>> >> >> ? WWW-Authenticate: Digest realm="192.168.1.120", nonce="8e54805b",
>> >> >> qop="auth", algorithm=md5
>> >> >> ? Content-Length: 0
>> >> >>
>> >> >>
>> >> >>
>> >> >> ------------------------------------------------------------------------
>> >> >> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
>> >> >>
>> >> >>
>> >> >> ------------------------------------------------------------------------
>> >> >> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> >> >> ? Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
>> >> >> ? Max-Forwards: 70
>> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> ? To: <sip:1000 at 192.168.1.120>
>> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> ? CSeq: 126070339 NOTIFY
>> >> >> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> >> >> ? Expires: 3590
>> >> >> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> >> >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> >> ? Supported: timer, precondition, path, replaces
>> >> >> ? Event: reboot_now
>> >> >> ? Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
>> >> >> include-session-description, presence.winfo, message-summary, refer
>> >> >> ? Subscription-State: terminated;reason=timeout
>> >> >> ? Authorization: Digest username="1115633124",
>> >> >> realm="192.168.1.120",
>> >> >> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
>> >> >> uri="sip:1000 at 192.168.1.121:5060",
>> >> >> response="747b4d04544c84535dbbd987f2999ca7", qop=auth, nc=00000001
>> >> >> ? Content-Type: application/simple-message-summary
>> >> >> ? Content-Length: 0
>> >> >>
>> >> >>
>> >> >>
>> >> >> ------------------------------------------------------------------------
>> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
>> >> >>
>> >> >>
>> >> >> ------------------------------------------------------------------------
>> >> >> ? SIP/2.0 401 Unauthorized
>> >> >> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> ? CSeq: 126070339 NOTIFY
>> >> >> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
>> >> >> ? Server: Linksys/PAP2T-5.1.6(LS)
>> >> >> ? WWW-Authenticate: Digest realm="192.168.1.120", nonce="5339c7ba",
>> >> >> qop="auth", algorithm=md5
>> >> >> ? Content-Length: 0
>> >> >>
>> >> >>
>> >> >>
>> >> >> ------------------------------------------------------------------------
>> >> >>
>> >> >> On Mon, Jan 25, 2010 at 4:46 AM, Brian West <brian at freeswitch.org>
>> >> >> wrote:
>> >> >> > or sofia profile xxx flush_inbound_reg callid reboot
>> >> >> > callid you can get from sofia status profile xxx
>> >> >> > /b
>> >> >> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
>> >> >> >
>> >> >> > Actually I just
>> >> >> > found?http://wiki.freeswitch.org/wiki/Mod_event_socket
>> >> >> >
>> >> >> > If I telnet to FS as described
>> >> >> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client, do
>> >> >> > I
>> >> >> > just need to enter somthing like:
>> >> >> >
>> >> >> > sendevent NOTIFY
>> >> >> > profile: internal
>> >> >> > event-string: resync
>> >> >> > user: 1000
>> >> >> > host: 192.168.1.121
>> >> >> > content-type: application/simple-message-summary
>> >> >> >
>> >> >> > where 192.168.1.121 is the ip address of one of the Linksys
>> >> >> > devices?
>> >> >> >
>> >> >> > I don't see any messages sent when I do this. ?What am I doing
>> >> >> > wrong?
>> >> >> >
>> >> >> > Thanks
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From anthony.minessale at gmail.com  Wed Jan 27 17:41:36 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 19:41:36 -0600
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001240328q7228ba76vea0b44477dbf0341@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
Message-ID: <191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>

the host is not resolved it has to be an exact string match with the host
that is in the db.
if you want to normalize it set force-reg-domain and force-reg-db-domain to
the same val


On Wed, Jan 27, 2010 at 7:26 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Thanks Anthony,
>
> I think user matches (ie the extension 2000 or 2001).   What should
> host be?  In the sofia printout, it says 'freeswitch' (freeswitch has
> ip address 192.168.1.120).
>
> However, if I try to use 'freeswitch' as the host for user 2001,
> nothing is sent.  But using 192.168.1.120 does.
>
> If I do exactly the same thing for 2000, the NOTIFY message is not
> sent.  Are there logs I can send to show you or any ideas what I am
> doing wrong?
>
>
>
> On Thu, Jan 28, 2010 at 12:09 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > user and host have to match too
> >
> > On Wed, Jan 27, 2010 at 7:06 AM, Mark Campbell-Smith
> > <mcampbellsmith at gmail.com> wrote:
> >>
> >> Thanks guys.  I have this working except for one user who is
> >> registered like this:
> >>
> >> Call-ID:        2ff39277-fd9a6ab1 at 10.0.0.1
> >> User:           2000 at 192.168.1.120
> >> Contact:        2000
> >>
> >> <sip:2000 at 124.xxx.xxx.xxx
> :5075;transport=tls;fs_nat=yes;fs_path=sip%3A2000%40124.xxx.xxx.xxx%3A10281%3Btransport%3Dtls>
> >> Agent:          Linksys/SPA3102-5.1.10(GW)
> >> Status:         Registered(TLS-NAT)(unknown) EXP(2010-01-28 00:29:34)
> >> Host:           freeswitch
> >> IP:             124.xxx.xxx.xxx
> >> Port:           10281
> >> Auth-User:      2000
> >> Auth-Realm:     mydns.dyndns.org
> >> MWI-Account:    2000 at 192.168.1.120
> >>
> >> When I do the following commands via the telnet socket, no notify
> >> command is sent to user 2000:
> >>
> >> sendevent NOTIFY
> >> profile: internal
> >> content-type: application/simple-message-summary
> >> event-string: reboot_now
> >> user: 2000
> >> host: 192.168.1.120
> >> content-length: 0
> >>
> >> However, if I do exactly the same thing with user 2001 it works.  2001
> >> is registered as:
> >>
> >> Contact:        2001 <sip:2001 at 124.yyy.yyy.yyy:5066;transport=tls>
> >>
> >> Any ideas why that would be?
> >>
> >> On Mon, Jan 25, 2010 at 11:50 AM, Ognjen Seslija <oseslija at gmail.com>
> >> wrote:
> >> > The phone is asking FS to authenticate prior then accepting a NOTIFY
> >> > from
> >> > it.
> >> > The authentication of notify's from spa endpoints work (afaik) only
> with
> >> > Linksys SPA9000 PBX , and FS doesn't have the code for that for
> obvious
> >> > reasons.
> >> > If you have SPA9000 maybe you can collect SIP traces.
> >> >
> >> > Ognjen
> >> >
> >> >
> >> >
> >> >
> >> >
> >> > On Mon, Jan 25, 2010 at 1:29 AM, Mark Campbell-Smith
> >> > <mcampbellsmith at gmail.com> wrote:
> >> >>
> >> >> Hi Ognjen,
> >> >>
> >> >> Thanks for the tip on the resync under the EXT tab.  It now works
> >> >> using mod_event_socket and the following:
> >> >>
> >> >> sendevent NOTIFY
> >> >> profile: internal
> >> >> event-string: resync
> >> >> user: 1000
> >> >> host: 192.168.1.121
> >> >> content-type: application/simple-message-summary
> >> >>
> >> >> However, if AUTH is required, why does FS send the wrong information
> to
> >> >> the SPA?
> >> >>
> >> >> On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija <oseslija at gmail.com
> >
> >> >> wrote:
> >> >> > You  should not authenticate those NOTIFYs (this will work only
> with
> >> >> > SPA9000
> >> >> > afaik). The option to change for this is in EXT tabs:
> >> >> >
> >> >> > Auth Resync-Reboot: No
> >> >> >
> >> >> > Also, FSs code will do a reboot of a phone, not resync (it sends
> >> >> > reboot_now
> >> >> > event). For that to work a patch is required.
> >> >> >
> >> >> > I've just tried to reboot my 942 (rev 16506) and it definitely
> works.
> >> >> >
> >> >> > Regards,
> >> >> > Ognjen
> >> >> >
> >> >> >
> >> >> > On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
> >> >> > <mcampbellsmith at gmail.com> wrote:
> >> >> >>
> >> >> >> Thanks Brian.. this still does not work.  Maybe I need to open a
> >> >> >> Jira?
> >> >> >>  Notice the username in the authorization field.  It should be
> 1000.
> >> >> >>
> >> >> >> Cheers
> >> >> >> Mark
> >> >> >>
> >> >> >> freeswitch at internal> sofia status profile internal user
> >> >> >> 1000 at 192.168.1.120
> >> >> >>
> >> >> >> Registrations:
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> =================================================================================================
> >> >> >> Call-ID:        bd783b73-66877627 at 192.168.1.121
> >> >> >> User:           1000 at 192.168.1.120
> >> >> >> Contact:        1000 <sip:1000 at 192.168.1.121:5060>
> >> >> >> Agent:          Linksys/PAP2T-5.1.6(LS)
> >> >> >> Status:         Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
> >> >> >> Host:           freeswitch
> >> >> >> IP:             192.168.1.121
> >> >> >> Port:           5060
> >> >> >> Auth-User:      1000
> >> >> >> Auth-Realm:     192.168.1.120
> >> >> >> MWI-Account:    1000 at 192.168.1.120
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> =================================================================================================
> >> >> >>
> >> >> >> freeswitch at internal> sofia profile internal flush_inbound_reg
> >> >> >> bd783b73-66877627 at 192.168.1.121 reboot
> >> >> >> +OK rebooting all registrations matching specified call_id
> >> >> >>
> >> >> >> freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060
> at
> >> >> >> 23:55:49.012627:
> >> >> >>
> >> >> >>
> >> >> >>
> ------------------------------------------------------------------------
> >> >> >>   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
> >> >> >>   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
> >> >> >>   Max-Forwards: 70
> >> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
> >> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >> >>   CSeq: 126070338 NOTIFY
> >> >> >>   Contact: <sip:mod_sofia at 192.168.1.120:5060>
> >> >> >>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
> >> >> >>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >> >>   Supported: timer, precondition, path, replaces
> >> >> >>   Event: reboot_now
> >> >> >>   Allow-Events: talk, presence, dialog, line-seize, call-info,
> sla,
> >> >> >> include-session-description, presence.winfo, message-summary,
> refer
> >> >> >>   Subscription-State: terminated;reason=timeout
> >> >> >>   Content-Type: application/simple-message-summary
> >> >> >>   Content-Length: 0
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
> >> >> >>
> >> >> >>
> >> >> >>
> ------------------------------------------------------------------------
> >> >> >>   SIP/2.0 401 Unauthorized
> >> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
> >> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >> >>   CSeq: 126070338 NOTIFY
> >> >> >>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
> >> >> >>   Server: Linksys/PAP2T-5.1.6(LS)
> >> >> >>   WWW-Authenticate: Digest realm="192.168.1.120",
> nonce="8e54805b",
> >> >> >> qop="auth", algorithm=md5
> >> >> >>   Content-Length: 0
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
> >> >> >>
> >> >> >>
> >> >> >>
> ------------------------------------------------------------------------
> >> >> >>   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
> >> >> >>   Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
> >> >> >>   Max-Forwards: 70
> >> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
> >> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >> >>   CSeq: 126070339 NOTIFY
> >> >> >>   Contact: <sip:mod_sofia at 192.168.1.120:5060>
> >> >> >>   Expires: 3590
> >> >> >>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
> >> >> >>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >> >>   Supported: timer, precondition, path, replaces
> >> >> >>   Event: reboot_now
> >> >> >>   Allow-Events: talk, presence, dialog, line-seize, call-info,
> sla,
> >> >> >> include-session-description, presence.winfo, message-summary,
> refer
> >> >> >>   Subscription-State: terminated;reason=timeout
> >> >> >>   Authorization: Digest username="1115633124",
> >> >> >> realm="192.168.1.120",
> >> >> >> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
> >> >> >> uri="sip:1000 at 192.168.1.121:5060",
> >> >> >> response="747b4d04544c84535dbbd987f2999ca7", qop=auth, nc=00000001
> >> >> >>   Content-Type: application/simple-message-summary
> >> >> >>   Content-Length: 0
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
> >> >> >>
> >> >> >>
> >> >> >>
> ------------------------------------------------------------------------
> >> >> >>   SIP/2.0 401 Unauthorized
> >> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
> >> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >> >>   CSeq: 126070339 NOTIFY
> >> >> >>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
> >> >> >>   Server: Linksys/PAP2T-5.1.6(LS)
> >> >> >>   WWW-Authenticate: Digest realm="192.168.1.120",
> nonce="5339c7ba",
> >> >> >> qop="auth", algorithm=md5
> >> >> >>   Content-Length: 0
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> ------------------------------------------------------------------------
> >> >> >>
> >> >> >> On Mon, Jan 25, 2010 at 4:46 AM, Brian West <brian at freeswitch.org
> >
> >> >> >> wrote:
> >> >> >> > or sofia profile xxx flush_inbound_reg callid reboot
> >> >> >> > callid you can get from sofia status profile xxx
> >> >> >> > /b
> >> >> >> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
> >> >> >> >
> >> >> >> > Actually I just
> >> >> >> > found http://wiki.freeswitch.org/wiki/Mod_event_socket
> >> >> >> >
> >> >> >> > If I telnet to FS as described
> >> >> >> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client,
> do
> >> >> >> > I
> >> >> >> > just need to enter somthing like:
> >> >> >> >
> >> >> >> > sendevent NOTIFY
> >> >> >> > profile: internal
> >> >> >> > event-string: resync
> >> >> >> > user: 1000
> >> >> >> > host: 192.168.1.121
> >> >> >> > content-type: application/simple-message-summary
> >> >> >> >
> >> >> >> > where 192.168.1.121 is the ip address of one of the Linksys
> >> >> >> > devices?
> >> >> >> >
> >> >> >> > I don't see any messages sent when I do this.  What am I doing
> >> >> >> > wrong?
> >> >> >> >
> >> >> >> > Thanks
> >> >> >> >
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jcasale at activenetwerx.com  Wed Jan 27 18:09:06 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Thu, 28 Jan 2010 02:09:06 +0000
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
In-Reply-To: <20100127210836.5ED5D2E3B2D@cuneorg-email.cune.pri>
References: <CA5A491E9DEFBE4CB777DE97E21575E90BD7F4@prato.activenetwerx.local>
	<20100127210836.5ED5D2E3B2D@cuneorg-email.cune.pri>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90BF91C@prato.activenetwerx.local>

>Debian 5.0.3

Well, given the time I had tonight, I tried on my CentOS 5.3 box.
The incoming log is the first block, and an outgoing log is the
second block at http://pastebin.freeswitch.org/11965

When I call in, I can hear it get answered, as I play the wav file
I hear the tone go very low, but no sound.

When I try to call out, nothing happens?

Is there anything in the log that might standout from your perspective?

Thanks everyone!
jlc



From mcampbellsmith at gmail.com  Wed Jan 27 18:34:14 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 28 Jan 2010 13:34:14 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<99181A2B-1950-43F0-A076-32525C441490@freeswitch.org>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
Message-ID: <33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>

If it works with 2001 doesn't that mean I am using the correct host?

Both 2001 and 2000 register with exactly the same data, except
username and password ....



On Thu, Jan 28, 2010 at 12:41 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> the host is not resolved it has to be an exact string match with the host
> that is in the db.
> if you want to normalize it set force-reg-domain and force-reg-db-domain to
> the same val
>
>
> On Wed, Jan 27, 2010 at 7:26 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>>
>> Thanks Anthony,
>>
>> I think user matches (ie the extension 2000 or 2001). ? What should
>> host be? ?In the sofia printout, it says 'freeswitch' (freeswitch has
>> ip address 192.168.1.120).
>>
>> However, if I try to use 'freeswitch' as the host for user 2001,
>> nothing is sent. ?But using 192.168.1.120 does.
>>
>> If I do exactly the same thing for 2000, the NOTIFY message is not
>> sent. ?Are there logs I can send to show you or any ideas what I am
>> doing wrong?
>>
>>
>>
>> On Thu, Jan 28, 2010 at 12:09 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > user and host have to match too
>> >
>> > On Wed, Jan 27, 2010 at 7:06 AM, Mark Campbell-Smith
>> > <mcampbellsmith at gmail.com> wrote:
>> >>
>> >> Thanks guys. ?I have this working except for one user who is
>> >> registered like this:
>> >>
>> >> Call-ID: ? ? ? ?2ff39277-fd9a6ab1 at 10.0.0.1
>> >> User: ? ? ? ? ? 2000 at 192.168.1.120
>> >> Contact: ? ? ? ?2000
>> >>
>> >>
>> >> <sip:2000 at 124.xxx.xxx.xxx:5075;transport=tls;fs_nat=yes;fs_path=sip%3A2000%40124.xxx.xxx.xxx%3A10281%3Btransport%3Dtls>
>> >> Agent: ? ? ? ? ?Linksys/SPA3102-5.1.10(GW)
>> >> Status: ? ? ? ? Registered(TLS-NAT)(unknown) EXP(2010-01-28 00:29:34)
>> >> Host: ? ? ? ? ? freeswitch
>> >> IP: ? ? ? ? ? ? 124.xxx.xxx.xxx
>> >> Port: ? ? ? ? ? 10281
>> >> Auth-User: ? ? ?2000
>> >> Auth-Realm: ? ? mydns.dyndns.org
>> >> MWI-Account: ? ?2000 at 192.168.1.120
>> >>
>> >> When I do the following commands via the telnet socket, no notify
>> >> command is sent to user 2000:
>> >>
>> >> sendevent NOTIFY
>> >> profile: internal
>> >> content-type: application/simple-message-summary
>> >> event-string: reboot_now
>> >> user: 2000
>> >> host: 192.168.1.120
>> >> content-length: 0
>> >>
>> >> However, if I do exactly the same thing with user 2001 it works. ?2001
>> >> is registered as:
>> >>
>> >> Contact: ? ? ? ?2001 <sip:2001 at 124.yyy.yyy.yyy:5066;transport=tls>
>> >>
>> >> Any ideas why that would be?
>> >>
>> >> On Mon, Jan 25, 2010 at 11:50 AM, Ognjen Seslija <oseslija at gmail.com>
>> >> wrote:
>> >> > The phone is asking FS to authenticate prior then accepting a NOTIFY
>> >> > from
>> >> > it.
>> >> > The authentication of notify's from spa endpoints work (afaik) only
>> >> > with
>> >> > Linksys SPA9000 PBX , and FS doesn't have the code for that for
>> >> > obvious
>> >> > reasons.
>> >> > If you have SPA9000 maybe you can collect SIP traces.
>> >> >
>> >> > Ognjen
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > On Mon, Jan 25, 2010 at 1:29 AM, Mark Campbell-Smith
>> >> > <mcampbellsmith at gmail.com> wrote:
>> >> >>
>> >> >> Hi Ognjen,
>> >> >>
>> >> >> Thanks for the tip on the resync under the EXT tab. ?It now works
>> >> >> using mod_event_socket and the following:
>> >> >>
>> >> >> sendevent NOTIFY
>> >> >> profile: internal
>> >> >> event-string: resync
>> >> >> user: 1000
>> >> >> host: 192.168.1.121
>> >> >> content-type: application/simple-message-summary
>> >> >>
>> >> >> However, if AUTH is required, why does FS send the wrong information
>> >> >> to
>> >> >> the SPA?
>> >> >>
>> >> >> On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija
>> >> >> <oseslija at gmail.com>
>> >> >> wrote:
>> >> >> > You? should not authenticate those NOTIFYs (this will work only
>> >> >> > with
>> >> >> > SPA9000
>> >> >> > afaik). The option to change for this is in EXT tabs:
>> >> >> >
>> >> >> > Auth Resync-Reboot: No
>> >> >> >
>> >> >> > Also, FSs code will do a reboot of a phone, not resync (it sends
>> >> >> > reboot_now
>> >> >> > event). For that to work a patch is required.
>> >> >> >
>> >> >> > I've just tried to reboot my 942 (rev 16506) and it definitely
>> >> >> > works.
>> >> >> >
>> >> >> > Regards,
>> >> >> > Ognjen
>> >> >> >
>> >> >> >
>> >> >> > On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
>> >> >> > <mcampbellsmith at gmail.com> wrote:
>> >> >> >>
>> >> >> >> Thanks Brian.. this still does not work. ?Maybe I need to open a
>> >> >> >> Jira?
>> >> >> >> ?Notice the username in the authorization field. ?It should be
>> >> >> >> 1000.
>> >> >> >>
>> >> >> >> Cheers
>> >> >> >> Mark
>> >> >> >>
>> >> >> >> freeswitch at internal> sofia status profile internal user
>> >> >> >> 1000 at 192.168.1.120
>> >> >> >>
>> >> >> >> Registrations:
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> =================================================================================================
>> >> >> >> Call-ID: ? ? ? ?bd783b73-66877627 at 192.168.1.121
>> >> >> >> User: ? ? ? ? ? 1000 at 192.168.1.120
>> >> >> >> Contact: ? ? ? ?1000 <sip:1000 at 192.168.1.121:5060>
>> >> >> >> Agent: ? ? ? ? ?Linksys/PAP2T-5.1.6(LS)
>> >> >> >> Status: ? ? ? ? Registered(UDP)(unknown) EXP(2010-01-25 11:25:05)
>> >> >> >> Host: ? ? ? ? ? freeswitch
>> >> >> >> IP: ? ? ? ? ? ? 192.168.1.121
>> >> >> >> Port: ? ? ? ? ? 5060
>> >> >> >> Auth-User: ? ? ?1000
>> >> >> >> Auth-Realm: ? ? 192.168.1.120
>> >> >> >> MWI-Account: ? ?1000 at 192.168.1.120
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> =================================================================================================
>> >> >> >>
>> >> >> >> freeswitch at internal> sofia profile internal flush_inbound_reg
>> >> >> >> bd783b73-66877627 at 192.168.1.121 reboot
>> >> >> >> +OK rebooting all registrations matching specified call_id
>> >> >> >>
>> >> >> >> freeswitch at internal> send 804 bytes to udp/[192.168.1.121]:5060
>> >> >> >> at
>> >> >> >> 23:55:49.012627:
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> ------------------------------------------------------------------------
>> >> >> >> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> >> >> >> ? Via: SIP/2.0/UDP
>> >> >> >> 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
>> >> >> >> ? Max-Forwards: 70
>> >> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> >> ? To: <sip:1000 at 192.168.1.120>
>> >> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> >> ? CSeq: 126070338 NOTIFY
>> >> >> >> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> >> >> >> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> >> >> >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>> >> >> >> INFO,
>> >> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> >> >> ? Supported: timer, precondition, path, replaces
>> >> >> >> ? Event: reboot_now
>> >> >> >> ? Allow-Events: talk, presence, dialog, line-seize, call-info,
>> >> >> >> sla,
>> >> >> >> include-session-description, presence.winfo, message-summary,
>> >> >> >> refer
>> >> >> >> ? Subscription-State: terminated;reason=timeout
>> >> >> >> ? Content-Type: application/simple-message-summary
>> >> >> >> ? Content-Length: 0
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> ------------------------------------------------------------------------
>> >> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.045267:
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> ------------------------------------------------------------------------
>> >> >> >> ? SIP/2.0 401 Unauthorized
>> >> >> >> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> >> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> >> ? CSeq: 126070338 NOTIFY
>> >> >> >> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
>> >> >> >> ? Server: Linksys/PAP2T-5.1.6(LS)
>> >> >> >> ? WWW-Authenticate: Digest realm="192.168.1.120",
>> >> >> >> nonce="8e54805b",
>> >> >> >> qop="auth", algorithm=md5
>> >> >> >> ? Content-Length: 0
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> ------------------------------------------------------------------------
>> >> >> >> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> ------------------------------------------------------------------------
>> >> >> >> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> >> >> >> ? Via: SIP/2.0/UDP
>> >> >> >> 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
>> >> >> >> ? Max-Forwards: 70
>> >> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> >> ? To: <sip:1000 at 192.168.1.120>
>> >> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> >> ? CSeq: 126070339 NOTIFY
>> >> >> >> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> >> >> >> ? Expires: 3590
>> >> >> >> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> >> >> >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>> >> >> >> INFO,
>> >> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> >> >> ? Supported: timer, precondition, path, replaces
>> >> >> >> ? Event: reboot_now
>> >> >> >> ? Allow-Events: talk, presence, dialog, line-seize, call-info,
>> >> >> >> sla,
>> >> >> >> include-session-description, presence.winfo, message-summary,
>> >> >> >> refer
>> >> >> >> ? Subscription-State: terminated;reason=timeout
>> >> >> >> ? Authorization: Digest username="1115633124",
>> >> >> >> realm="192.168.1.120",
>> >> >> >> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q", algorithm=MD5,
>> >> >> >> uri="sip:1000 at 192.168.1.121:5060",
>> >> >> >> response="747b4d04544c84535dbbd987f2999ca7", qop=auth,
>> >> >> >> nc=00000001
>> >> >> >> ? Content-Type: application/simple-message-summary
>> >> >> >> ? Content-Length: 0
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> ------------------------------------------------------------------------
>> >> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at 23:55:49.086375:
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> ------------------------------------------------------------------------
>> >> >> >> ? SIP/2.0 401 Unauthorized
>> >> >> >> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> >> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> >> ? CSeq: 126070339 NOTIFY
>> >> >> >> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
>> >> >> >> ? Server: Linksys/PAP2T-5.1.6(LS)
>> >> >> >> ? WWW-Authenticate: Digest realm="192.168.1.120",
>> >> >> >> nonce="5339c7ba",
>> >> >> >> qop="auth", algorithm=md5
>> >> >> >> ? Content-Length: 0
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> ------------------------------------------------------------------------
>> >> >> >>
>> >> >> >> On Mon, Jan 25, 2010 at 4:46 AM, Brian West
>> >> >> >> <brian at freeswitch.org>
>> >> >> >> wrote:
>> >> >> >> > or sofia profile xxx flush_inbound_reg callid reboot
>> >> >> >> > callid you can get from sofia status profile xxx
>> >> >> >> > /b
>> >> >> >> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
>> >> >> >> >
>> >> >> >> > Actually I just
>> >> >> >> > found?http://wiki.freeswitch.org/wiki/Mod_event_socket
>> >> >> >> >
>> >> >> >> > If I telnet to FS as described
>> >> >> >> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client,
>> >> >> >> > do
>> >> >> >> > I
>> >> >> >> > just need to enter somthing like:
>> >> >> >> >
>> >> >> >> > sendevent NOTIFY
>> >> >> >> > profile: internal
>> >> >> >> > event-string: resync
>> >> >> >> > user: 1000
>> >> >> >> > host: 192.168.1.121
>> >> >> >> > content-type: application/simple-message-summary
>> >> >> >> >
>> >> >> >> > where 192.168.1.121 is the ip address of one of the Linksys
>> >> >> >> > devices?
>> >> >> >> >
>> >> >> >> > I don't see any messages sent when I do this. ?What am I doing
>> >> >> >> > wrong?
>> >> >> >> >
>> >> >> >> > Thanks
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > _______________________________________________
>> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> > http://www.freeswitch.org
>> >> >> >> >
>> >> >> >> >
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From anthony.minessale at gmail.com  Wed Jan 27 19:25:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 21:25:17 -0600
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001241558g5e616492s489e40b390d78e7d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
	<33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
Message-ID: <191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>

you said one was the word freeswitch and one was an ip didn't you?


On Wed, Jan 27, 2010 at 8:34 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> If it works with 2001 doesn't that mean I am using the correct host?
>
> Both 2001 and 2000 register with exactly the same data, except
> username and password ....
>
>
>
> On Thu, Jan 28, 2010 at 12:41 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > the host is not resolved it has to be an exact string match with the host
> > that is in the db.
> > if you want to normalize it set force-reg-domain and force-reg-db-domain
> to
> > the same val
> >
> >
> > On Wed, Jan 27, 2010 at 7:26 PM, Mark Campbell-Smith
> > <mcampbellsmith at gmail.com> wrote:
> >>
> >> Thanks Anthony,
> >>
> >> I think user matches (ie the extension 2000 or 2001).   What should
> >> host be?  In the sofia printout, it says 'freeswitch' (freeswitch has
> >> ip address 192.168.1.120).
> >>
> >> However, if I try to use 'freeswitch' as the host for user 2001,
> >> nothing is sent.  But using 192.168.1.120 does.
> >>
> >> If I do exactly the same thing for 2000, the NOTIFY message is not
> >> sent.  Are there logs I can send to show you or any ideas what I am
> >> doing wrong?
> >>
> >>
> >>
> >> On Thu, Jan 28, 2010 at 12:09 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >> > user and host have to match too
> >> >
> >> > On Wed, Jan 27, 2010 at 7:06 AM, Mark Campbell-Smith
> >> > <mcampbellsmith at gmail.com> wrote:
> >> >>
> >> >> Thanks guys.  I have this working except for one user who is
> >> >> registered like this:
> >> >>
> >> >> Call-ID:        2ff39277-fd9a6ab1 at 10.0.0.1
> >> >> User:           2000 at 192.168.1.120
> >> >> Contact:        2000
> >> >>
> >> >>
> >> >> <sip:2000 at 124.xxx.xxx.xxx
> :5075;transport=tls;fs_nat=yes;fs_path=sip%3A2000%40124.xxx.xxx.xxx%3A10281%3Btransport%3Dtls>
> >> >> Agent:          Linksys/SPA3102-5.1.10(GW)
> >> >> Status:         Registered(TLS-NAT)(unknown) EXP(2010-01-28 00:29:34)
> >> >> Host:           freeswitch
> >> >> IP:             124.xxx.xxx.xxx
> >> >> Port:           10281
> >> >> Auth-User:      2000
> >> >> Auth-Realm:     mydns.dyndns.org
> >> >> MWI-Account:    2000 at 192.168.1.120
> >> >>
> >> >> When I do the following commands via the telnet socket, no notify
> >> >> command is sent to user 2000:
> >> >>
> >> >> sendevent NOTIFY
> >> >> profile: internal
> >> >> content-type: application/simple-message-summary
> >> >> event-string: reboot_now
> >> >> user: 2000
> >> >> host: 192.168.1.120
> >> >> content-length: 0
> >> >>
> >> >> However, if I do exactly the same thing with user 2001 it works.
>  2001
> >> >> is registered as:
> >> >>
> >> >> Contact:        2001 <sip:2001 at 124.yyy.yyy.yyy:5066;transport=tls>
> >> >>
> >> >> Any ideas why that would be?
> >> >>
> >> >> On Mon, Jan 25, 2010 at 11:50 AM, Ognjen Seslija <oseslija at gmail.com
> >
> >> >> wrote:
> >> >> > The phone is asking FS to authenticate prior then accepting a
> NOTIFY
> >> >> > from
> >> >> > it.
> >> >> > The authentication of notify's from spa endpoints work (afaik) only
> >> >> > with
> >> >> > Linksys SPA9000 PBX , and FS doesn't have the code for that for
> >> >> > obvious
> >> >> > reasons.
> >> >> > If you have SPA9000 maybe you can collect SIP traces.
> >> >> >
> >> >> > Ognjen
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > On Mon, Jan 25, 2010 at 1:29 AM, Mark Campbell-Smith
> >> >> > <mcampbellsmith at gmail.com> wrote:
> >> >> >>
> >> >> >> Hi Ognjen,
> >> >> >>
> >> >> >> Thanks for the tip on the resync under the EXT tab.  It now works
> >> >> >> using mod_event_socket and the following:
> >> >> >>
> >> >> >> sendevent NOTIFY
> >> >> >> profile: internal
> >> >> >> event-string: resync
> >> >> >> user: 1000
> >> >> >> host: 192.168.1.121
> >> >> >> content-type: application/simple-message-summary
> >> >> >>
> >> >> >> However, if AUTH is required, why does FS send the wrong
> information
> >> >> >> to
> >> >> >> the SPA?
> >> >> >>
> >> >> >> On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija
> >> >> >> <oseslija at gmail.com>
> >> >> >> wrote:
> >> >> >> > You  should not authenticate those NOTIFYs (this will work only
> >> >> >> > with
> >> >> >> > SPA9000
> >> >> >> > afaik). The option to change for this is in EXT tabs:
> >> >> >> >
> >> >> >> > Auth Resync-Reboot: No
> >> >> >> >
> >> >> >> > Also, FSs code will do a reboot of a phone, not resync (it sends
> >> >> >> > reboot_now
> >> >> >> > event). For that to work a patch is required.
> >> >> >> >
> >> >> >> > I've just tried to reboot my 942 (rev 16506) and it definitely
> >> >> >> > works.
> >> >> >> >
> >> >> >> > Regards,
> >> >> >> > Ognjen
> >> >> >> >
> >> >> >> >
> >> >> >> > On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
> >> >> >> > <mcampbellsmith at gmail.com> wrote:
> >> >> >> >>
> >> >> >> >> Thanks Brian.. this still does not work.  Maybe I need to open
> a
> >> >> >> >> Jira?
> >> >> >> >>  Notice the username in the authorization field.  It should be
> >> >> >> >> 1000.
> >> >> >> >>
> >> >> >> >> Cheers
> >> >> >> >> Mark
> >> >> >> >>
> >> >> >> >> freeswitch at internal> sofia status profile internal user
> >> >> >> >> 1000 at 192.168.1.120
> >> >> >> >>
> >> >> >> >> Registrations:
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> =================================================================================================
> >> >> >> >> Call-ID:        bd783b73-66877627 at 192.168.1.121
> >> >> >> >> User:           1000 at 192.168.1.120
> >> >> >> >> Contact:        1000 <sip:1000 at 192.168.1.121:5060>
> >> >> >> >> Agent:          Linksys/PAP2T-5.1.6(LS)
> >> >> >> >> Status:         Registered(UDP)(unknown) EXP(2010-01-25
> 11:25:05)
> >> >> >> >> Host:           freeswitch
> >> >> >> >> IP:             192.168.1.121
> >> >> >> >> Port:           5060
> >> >> >> >> Auth-User:      1000
> >> >> >> >> Auth-Realm:     192.168.1.120
> >> >> >> >> MWI-Account:    1000 at 192.168.1.120
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> =================================================================================================
> >> >> >> >>
> >> >> >> >> freeswitch at internal> sofia profile internal flush_inbound_reg
> >> >> >> >> bd783b73-66877627 at 192.168.1.121 reboot
> >> >> >> >> +OK rebooting all registrations matching specified call_id
> >> >> >> >>
> >> >> >> >> freeswitch at internal> send 804 bytes to
> udp/[192.168.1.121]:5060
> >> >> >> >> at
> >> >> >> >> 23:55:49.012627:
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> >>   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
> >> >> >> >>   Via: SIP/2.0/UDP
> >> >> >> >> 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
> >> >> >> >>   Max-Forwards: 70
> >> >> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
> >> >> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >> >> >>   CSeq: 126070338 NOTIFY
> >> >> >> >>   Contact: <sip:mod_sofia at 192.168.1.120:5060>
> >> >> >> >>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
> >> >> >> >>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
> >> >> >> >> INFO,
> >> >> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >> >> >>   Supported: timer, precondition, path, replaces
> >> >> >> >>   Event: reboot_now
> >> >> >> >>   Allow-Events: talk, presence, dialog, line-seize, call-info,
> >> >> >> >> sla,
> >> >> >> >> include-session-description, presence.winfo, message-summary,
> >> >> >> >> refer
> >> >> >> >>   Subscription-State: terminated;reason=timeout
> >> >> >> >>   Content-Type: application/simple-message-summary
> >> >> >> >>   Content-Length: 0
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at
> 23:55:49.045267:
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> >>   SIP/2.0 401 Unauthorized
> >> >> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
> >> >> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >> >> >>   CSeq: 126070338 NOTIFY
> >> >> >> >>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
> >> >> >> >>   Server: Linksys/PAP2T-5.1.6(LS)
> >> >> >> >>   WWW-Authenticate: Digest realm="192.168.1.120",
> >> >> >> >> nonce="8e54805b",
> >> >> >> >> qop="auth", algorithm=md5
> >> >> >> >>   Content-Length: 0
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> >> send 1056 bytes to udp/[192.168.1.121]:5060 at 23:55:49.060073:
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> >>   NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
> >> >> >> >>   Via: SIP/2.0/UDP
> >> >> >> >> 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
> >> >> >> >>   Max-Forwards: 70
> >> >> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>>
> >> >> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >> >> >>   CSeq: 126070339 NOTIFY
> >> >> >> >>   Contact: <sip:mod_sofia at 192.168.1.120:5060>
> >> >> >> >>   Expires: 3590
> >> >> >> >>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
> >> >> >> >>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
> >> >> >> >> INFO,
> >> >> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >> >> >> >>   Supported: timer, precondition, path, replaces
> >> >> >> >>   Event: reboot_now
> >> >> >> >>   Allow-Events: talk, presence, dialog, line-seize, call-info,
> >> >> >> >> sla,
> >> >> >> >> include-session-description, presence.winfo, message-summary,
> >> >> >> >> refer
> >> >> >> >>   Subscription-State: terminated;reason=timeout
> >> >> >> >>   Authorization: Digest username="1115633124",
> >> >> >> >> realm="192.168.1.120",
> >> >> >> >> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q",
> algorithm=MD5,
> >> >> >> >> uri="sip:1000 at 192.168.1.121:5060",
> >> >> >> >> response="747b4d04544c84535dbbd987f2999ca7", qop=auth,
> >> >> >> >> nc=00000001
> >> >> >> >>   Content-Type: application/simple-message-summary
> >> >> >> >>   Content-Length: 0
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at
> 23:55:49.086375:
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> >>   SIP/2.0 401 Unauthorized
> >> >> >> >>   To: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=3300b5853719f35di0
> >> >> >> >>   From: <sip:1000 at 192.168.1.120 <sip%3A1000 at 192.168.1.120>
> >;tag=Z440t7e61ND0g
> >> >> >> >>   Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
> >> >> >> >>   CSeq: 126070339 NOTIFY
> >> >> >> >>   Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
> >> >> >> >>   Server: Linksys/PAP2T-5.1.6(LS)
> >> >> >> >>   WWW-Authenticate: Digest realm="192.168.1.120",
> >> >> >> >> nonce="5339c7ba",
> >> >> >> >> qop="auth", algorithm=md5
> >> >> >> >>   Content-Length: 0
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> ------------------------------------------------------------------------
> >> >> >> >>
> >> >> >> >> On Mon, Jan 25, 2010 at 4:46 AM, Brian West
> >> >> >> >> <brian at freeswitch.org>
> >> >> >> >> wrote:
> >> >> >> >> > or sofia profile xxx flush_inbound_reg callid reboot
> >> >> >> >> > callid you can get from sofia status profile xxx
> >> >> >> >> > /b
> >> >> >> >> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
> >> >> >> >> >
> >> >> >> >> > Actually I just
> >> >> >> >> > found http://wiki.freeswitch.org/wiki/Mod_event_socket
> >> >> >> >> >
> >> >> >> >> > If I telnet to FS as described
> >> >> >> >> >
> http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client,
> >> >> >> >> > do
> >> >> >> >> > I
> >> >> >> >> > just need to enter somthing like:
> >> >> >> >> >
> >> >> >> >> > sendevent NOTIFY
> >> >> >> >> > profile: internal
> >> >> >> >> > event-string: resync
> >> >> >> >> > user: 1000
> >> >> >> >> > host: 192.168.1.121
> >> >> >> >> > content-type: application/simple-message-summary
> >> >> >> >> >
> >> >> >> >> > where 192.168.1.121 is the ip address of one of the Linksys
> >> >> >> >> > devices?
> >> >> >> >> >
> >> >> >> >> > I don't see any messages sent when I do this.  What am I
> doing
> >> >> >> >> > wrong?
> >> >> >> >> >
> >> >> >> >> > Thanks
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > _______________________________________________
> >> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> > http://www.freeswitch.org
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >>
> >> >> >> >> _______________________________________________
> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mcampbellsmith at gmail.com  Wed Jan 27 19:39:54 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 28 Jan 2010 14:39:54 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
	<33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
	<191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>
Message-ID: <33c87fa31001271939i668e9676y1f687dc182b4ca89@mail.gmail.com>

Apologies Anthony...

This is the printout from sofia.  2000 does not work sending custom
NOTIFY messages and 2001 does.  Is it that 2001 is NAT'd, the port
10281 out of range or the contact incorrect or something?

Thanks!

Call-ID:        a7c8c53f-c18596ef at 192.168.1.3
User:           2001 at 192.168.1.120
Contact:        2001 <sip:2001 at 124.yyy.yyy.yyy:5072;transport=tls>
Agent:          Linksys/SPA3102-5.1.10(GW)
Status:         Registered(TLS)(unknown) EXP(2010-01-28 12:54:31)
Host:           freeswitch
IP:             124.yyy.yyy.yyy
Port:           5072
Auth-User:      2001
Auth-Realm:     mydns.dyndns.org
MWI-Account:    2001 at 192.168.1.120

Call-ID:        2ff39277-fd9a6ab1 at 10.0.0.1
User:           2000 at 192.168.1.120
Contact:        2000
<sip:2000 at 124.xxx.xxx.xxx:5075;transport=tls;fs_nat=yes;fs_path=sip%3A2000%40124.xxx.xxx.xxx%3A10281%3Btransport%3Dtls>
Agent:          Linksys/SPA3102-5.1.10(GW)
Status:         Registered(TLS-NAT)(unknown) EXP(2010-01-28 00:29:34)
Host:           freeswitch
IP:             124.xxx.xxx.xxx
Port:           10281
Auth-User:      2000
Auth-Realm:     mydns.dyndns.org
MWI-Account:    2000 at 192.168.1.120

On Thu, Jan 28, 2010 at 2:25 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> you said one was the word freeswitch and one was an ip didn't you?
>
>
> On Wed, Jan 27, 2010 at 8:34 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>>
>> If it works with 2001 doesn't that mean I am using the correct host?
>>
>> Both 2001 and 2000 register with exactly the same data, except
>> username and password ....
>>
>>
>>
>> On Thu, Jan 28, 2010 at 12:41 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > the host is not resolved it has to be an exact string match with the
>> > host
>> > that is in the db.
>> > if you want to normalize it set force-reg-domain and force-reg-db-domain
>> > to
>> > the same val
>> >
>> >
>> > On Wed, Jan 27, 2010 at 7:26 PM, Mark Campbell-Smith
>> > <mcampbellsmith at gmail.com> wrote:
>> >>
>> >> Thanks Anthony,
>> >>
>> >> I think user matches (ie the extension 2000 or 2001). ? What should
>> >> host be? ?In the sofia printout, it says 'freeswitch' (freeswitch has
>> >> ip address 192.168.1.120).
>> >>
>> >> However, if I try to use 'freeswitch' as the host for user 2001,
>> >> nothing is sent. ?But using 192.168.1.120 does.
>> >>
>> >> If I do exactly the same thing for 2000, the NOTIFY message is not
>> >> sent. ?Are there logs I can send to show you or any ideas what I am
>> >> doing wrong?
>> >>
>> >>
>> >>
>> >> On Thu, Jan 28, 2010 at 12:09 PM, Anthony Minessale
>> >> <anthony.minessale at gmail.com> wrote:
>> >> > user and host have to match too
>> >> >
>> >> > On Wed, Jan 27, 2010 at 7:06 AM, Mark Campbell-Smith
>> >> > <mcampbellsmith at gmail.com> wrote:
>> >> >>
>> >> >> Thanks guys. ?I have this working except for one user who is
>> >> >> registered like this:
>> >> >>
>> >> >> Call-ID: ? ? ? ?2ff39277-fd9a6ab1 at 10.0.0.1
>> >> >> User: ? ? ? ? ? 2000 at 192.168.1.120
>> >> >> Contact: ? ? ? ?2000
>> >> >>
>> >> >>
>> >> >>
>> >> >> <sip:2000 at 124.xxx.xxx.xxx:5075;transport=tls;fs_nat=yes;fs_path=sip%3A2000%40124.xxx.xxx.xxx%3A10281%3Btransport%3Dtls>
>> >> >> Agent: ? ? ? ? ?Linksys/SPA3102-5.1.10(GW)
>> >> >> Status: ? ? ? ? Registered(TLS-NAT)(unknown) EXP(2010-01-28
>> >> >> 00:29:34)
>> >> >> Host: ? ? ? ? ? freeswitch
>> >> >> IP: ? ? ? ? ? ? 124.xxx.xxx.xxx
>> >> >> Port: ? ? ? ? ? 10281
>> >> >> Auth-User: ? ? ?2000
>> >> >> Auth-Realm: ? ? mydns.dyndns.org
>> >> >> MWI-Account: ? ?2000 at 192.168.1.120
>> >> >>
>> >> >> When I do the following commands via the telnet socket, no notify
>> >> >> command is sent to user 2000:
>> >> >>
>> >> >> sendevent NOTIFY
>> >> >> profile: internal
>> >> >> content-type: application/simple-message-summary
>> >> >> event-string: reboot_now
>> >> >> user: 2000
>> >> >> host: 192.168.1.120
>> >> >> content-length: 0
>> >> >>
>> >> >> However, if I do exactly the same thing with user 2001 it works.
>> >> >> ?2001
>> >> >> is registered as:
>> >> >>
>> >> >> Contact: ? ? ? ?2001 <sip:2001 at 124.yyy.yyy.yyy:5066;transport=tls>
>> >> >>
>> >> >> Any ideas why that would be?
>> >> >>
>> >> >> On Mon, Jan 25, 2010 at 11:50 AM, Ognjen Seslija
>> >> >> <oseslija at gmail.com>
>> >> >> wrote:
>> >> >> > The phone is asking FS to authenticate prior then accepting a
>> >> >> > NOTIFY
>> >> >> > from
>> >> >> > it.
>> >> >> > The authentication of notify's from spa endpoints work (afaik)
>> >> >> > only
>> >> >> > with
>> >> >> > Linksys SPA9000 PBX , and FS doesn't have the code for that for
>> >> >> > obvious
>> >> >> > reasons.
>> >> >> > If you have SPA9000 maybe you can collect SIP traces.
>> >> >> >
>> >> >> > Ognjen
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > On Mon, Jan 25, 2010 at 1:29 AM, Mark Campbell-Smith
>> >> >> > <mcampbellsmith at gmail.com> wrote:
>> >> >> >>
>> >> >> >> Hi Ognjen,
>> >> >> >>
>> >> >> >> Thanks for the tip on the resync under the EXT tab. ?It now works
>> >> >> >> using mod_event_socket and the following:
>> >> >> >>
>> >> >> >> sendevent NOTIFY
>> >> >> >> profile: internal
>> >> >> >> event-string: resync
>> >> >> >> user: 1000
>> >> >> >> host: 192.168.1.121
>> >> >> >> content-type: application/simple-message-summary
>> >> >> >>
>> >> >> >> However, if AUTH is required, why does FS send the wrong
>> >> >> >> information
>> >> >> >> to
>> >> >> >> the SPA?
>> >> >> >>
>> >> >> >> On Mon, Jan 25, 2010 at 11:15 AM, Ognjen Seslija
>> >> >> >> <oseslija at gmail.com>
>> >> >> >> wrote:
>> >> >> >> > You? should not authenticate those NOTIFYs (this will work only
>> >> >> >> > with
>> >> >> >> > SPA9000
>> >> >> >> > afaik). The option to change for this is in EXT tabs:
>> >> >> >> >
>> >> >> >> > Auth Resync-Reboot: No
>> >> >> >> >
>> >> >> >> > Also, FSs code will do a reboot of a phone, not resync (it
>> >> >> >> > sends
>> >> >> >> > reboot_now
>> >> >> >> > event). For that to work a patch is required.
>> >> >> >> >
>> >> >> >> > I've just tried to reboot my 942 (rev 16506) and it definitely
>> >> >> >> > works.
>> >> >> >> >
>> >> >> >> > Regards,
>> >> >> >> > Ognjen
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > On Mon, Jan 25, 2010 at 12:58 AM, Mark Campbell-Smith
>> >> >> >> > <mcampbellsmith at gmail.com> wrote:
>> >> >> >> >>
>> >> >> >> >> Thanks Brian.. this still does not work. ?Maybe I need to open
>> >> >> >> >> a
>> >> >> >> >> Jira?
>> >> >> >> >> ?Notice the username in the authorization field. ?It should be
>> >> >> >> >> 1000.
>> >> >> >> >>
>> >> >> >> >> Cheers
>> >> >> >> >> Mark
>> >> >> >> >>
>> >> >> >> >> freeswitch at internal> sofia status profile internal user
>> >> >> >> >> 1000 at 192.168.1.120
>> >> >> >> >>
>> >> >> >> >> Registrations:
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> =================================================================================================
>> >> >> >> >> Call-ID: ? ? ? ?bd783b73-66877627 at 192.168.1.121
>> >> >> >> >> User: ? ? ? ? ? 1000 at 192.168.1.120
>> >> >> >> >> Contact: ? ? ? ?1000 <sip:1000 at 192.168.1.121:5060>
>> >> >> >> >> Agent: ? ? ? ? ?Linksys/PAP2T-5.1.6(LS)
>> >> >> >> >> Status: ? ? ? ? Registered(UDP)(unknown) EXP(2010-01-25
>> >> >> >> >> 11:25:05)
>> >> >> >> >> Host: ? ? ? ? ? freeswitch
>> >> >> >> >> IP: ? ? ? ? ? ? 192.168.1.121
>> >> >> >> >> Port: ? ? ? ? ? 5060
>> >> >> >> >> Auth-User: ? ? ?1000
>> >> >> >> >> Auth-Realm: ? ? 192.168.1.120
>> >> >> >> >> MWI-Account: ? ?1000 at 192.168.1.120
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> =================================================================================================
>> >> >> >> >>
>> >> >> >> >> freeswitch at internal> sofia profile internal flush_inbound_reg
>> >> >> >> >> bd783b73-66877627 at 192.168.1.121 reboot
>> >> >> >> >> +OK rebooting all registrations matching specified call_id
>> >> >> >> >>
>> >> >> >> >> freeswitch at internal> send 804 bytes to
>> >> >> >> >> udp/[192.168.1.121]:5060
>> >> >> >> >> at
>> >> >> >> >> 23:55:49.012627:
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> ------------------------------------------------------------------------
>> >> >> >> >> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> >> >> >> >> ? Via: SIP/2.0/UDP
>> >> >> >> >> 192.168.1.120;rport;branch=z9hG4bK1DKgFmj8QDp4g
>> >> >> >> >> ? Max-Forwards: 70
>> >> >> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> >> >> ? To: <sip:1000 at 192.168.1.120>
>> >> >> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> >> >> ? CSeq: 126070338 NOTIFY
>> >> >> >> >> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> >> >> >> >> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> >> >> >> >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>> >> >> >> >> INFO,
>> >> >> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> >> >> >> ? Supported: timer, precondition, path, replaces
>> >> >> >> >> ? Event: reboot_now
>> >> >> >> >> ? Allow-Events: talk, presence, dialog, line-seize, call-info,
>> >> >> >> >> sla,
>> >> >> >> >> include-session-description, presence.winfo, message-summary,
>> >> >> >> >> refer
>> >> >> >> >> ? Subscription-State: terminated;reason=timeout
>> >> >> >> >> ? Content-Type: application/simple-message-summary
>> >> >> >> >> ? Content-Length: 0
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> ------------------------------------------------------------------------
>> >> >> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at
>> >> >> >> >> 23:55:49.045267:
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> ------------------------------------------------------------------------
>> >> >> >> >> ? SIP/2.0 401 Unauthorized
>> >> >> >> >> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> >> >> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> >> >> ? CSeq: 126070338 NOTIFY
>> >> >> >> >> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK1DKgFmj8QDp4g
>> >> >> >> >> ? Server: Linksys/PAP2T-5.1.6(LS)
>> >> >> >> >> ? WWW-Authenticate: Digest realm="192.168.1.120",
>> >> >> >> >> nonce="8e54805b",
>> >> >> >> >> qop="auth", algorithm=md5
>> >> >> >> >> ? Content-Length: 0
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> ------------------------------------------------------------------------
>> >> >> >> >> send 1056 bytes to udp/[192.168.1.121]:5060 at
>> >> >> >> >> 23:55:49.060073:
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> ------------------------------------------------------------------------
>> >> >> >> >> ? NOTIFY sip:1000 at 192.168.1.121:5060 SIP/2.0
>> >> >> >> >> ? Via: SIP/2.0/UDP
>> >> >> >> >> 192.168.1.120;rport;branch=z9hG4bK2pc9gF3BNpcQc
>> >> >> >> >> ? Max-Forwards: 70
>> >> >> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> >> >> ? To: <sip:1000 at 192.168.1.120>
>> >> >> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> >> >> ? CSeq: 126070339 NOTIFY
>> >> >> >> >> ? Contact: <sip:mod_sofia at 192.168.1.120:5060>
>> >> >> >> >> ? Expires: 3590
>> >> >> >> >> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16256
>> >> >> >> >> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>> >> >> >> >> INFO,
>> >> >> >> >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >> >> >> >> ? Supported: timer, precondition, path, replaces
>> >> >> >> >> ? Event: reboot_now
>> >> >> >> >> ? Allow-Events: talk, presence, dialog, line-seize, call-info,
>> >> >> >> >> sla,
>> >> >> >> >> include-session-description, presence.winfo, message-summary,
>> >> >> >> >> refer
>> >> >> >> >> ? Subscription-State: terminated;reason=timeout
>> >> >> >> >> ? Authorization: Digest username="1115633124",
>> >> >> >> >> realm="192.168.1.120",
>> >> >> >> >> nonce="8e54805b", cnonce="1mWxHoPmEi2pewDgTAMS6Q",
>> >> >> >> >> algorithm=MD5,
>> >> >> >> >> uri="sip:1000 at 192.168.1.121:5060",
>> >> >> >> >> response="747b4d04544c84535dbbd987f2999ca7", qop=auth,
>> >> >> >> >> nc=00000001
>> >> >> >> >> ? Content-Type: application/simple-message-summary
>> >> >> >> >> ? Content-Length: 0
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> ------------------------------------------------------------------------
>> >> >> >> >> recv 407 bytes from udp/[192.168.1.121]:5060 at
>> >> >> >> >> 23:55:49.086375:
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> ------------------------------------------------------------------------
>> >> >> >> >> ? SIP/2.0 401 Unauthorized
>> >> >> >> >> ? To: <sip:1000 at 192.168.1.120>;tag=3300b5853719f35di0
>> >> >> >> >> ? From: <sip:1000 at 192.168.1.120>;tag=Z440t7e61ND0g
>> >> >> >> >> ? Call-ID: d65f2a5a-83e6-122d-7ba9-00e04c0312e9
>> >> >> >> >> ? CSeq: 126070339 NOTIFY
>> >> >> >> >> ? Via: SIP/2.0/UDP 192.168.1.120;branch=z9hG4bK2pc9gF3BNpcQc
>> >> >> >> >> ? Server: Linksys/PAP2T-5.1.6(LS)
>> >> >> >> >> ? WWW-Authenticate: Digest realm="192.168.1.120",
>> >> >> >> >> nonce="5339c7ba",
>> >> >> >> >> qop="auth", algorithm=md5
>> >> >> >> >> ? Content-Length: 0
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> ------------------------------------------------------------------------
>> >> >> >> >>
>> >> >> >> >> On Mon, Jan 25, 2010 at 4:46 AM, Brian West
>> >> >> >> >> <brian at freeswitch.org>
>> >> >> >> >> wrote:
>> >> >> >> >> > or sofia profile xxx flush_inbound_reg callid reboot
>> >> >> >> >> > callid you can get from sofia status profile xxx
>> >> >> >> >> > /b
>> >> >> >> >> > On Jan 24, 2010, at 5:28 AM, Mark Campbell-Smith wrote:
>> >> >> >> >> >
>> >> >> >> >> > Actually I just
>> >> >> >> >> > found?http://wiki.freeswitch.org/wiki/Mod_event_socket
>> >> >> >> >> >
>> >> >> >> >> > If I telnet to FS as described
>> >> >> >> >> >
>> >> >> >> >> > http://wiki.freeswitch.org/wiki/Mod_event_socket#Telnet_Client,
>> >> >> >> >> > do
>> >> >> >> >> > I
>> >> >> >> >> > just need to enter somthing like:
>> >> >> >> >> >
>> >> >> >> >> > sendevent NOTIFY
>> >> >> >> >> > profile: internal
>> >> >> >> >> > event-string: resync
>> >> >> >> >> > user: 1000
>> >> >> >> >> > host: 192.168.1.121
>> >> >> >> >> > content-type: application/simple-message-summary
>> >> >> >> >> >
>> >> >> >> >> > where 192.168.1.121 is the ip address of one of the Linksys
>> >> >> >> >> > devices?
>> >> >> >> >> >
>> >> >> >> >> > I don't see any messages sent when I do this. ?What am I
>> >> >> >> >> > doing
>> >> >> >> >> > wrong?
>> >> >> >> >> >
>> >> >> >> >> > Thanks
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> > _______________________________________________
>> >> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> >> >
>> >> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> >> > http://www.freeswitch.org
>> >> >> >> >> >
>> >> >> >> >> >
>> >> >> >> >>
>> >> >> >> >> _______________________________________________
>> >> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> >> http://www.freeswitch.org
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > _______________________________________________
>> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> > http://www.freeswitch.org
>> >> >> >> >
>> >> >> >> >
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> > Anthony Minessale II
>> >> >
>> >> > FreeSWITCH http://www.freeswitch.org/
>> >> > ClueCon http://www.cluecon.com/
>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >
>> >> > AIM: anthm
>> >> > MSN:anthony_minessale at hotmail.com
>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> > IRC: irc.freenode.net #freeswitch
>> >> >
>> >> > FreeSWITCH Developer Conference
>> >> > sip:888 at conference.freeswitch.org
>> >> > iax:guest at conference.freeswitch.org/888
>> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> > pstn:+19193869900
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Wed Jan 27 19:54:07 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 27 Jan 2010 21:54:07 -0600
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001271939i668e9676y1f687dc182b4ca89@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<4468a6771001241615r1e7291dci5a85db14017b3c97@mail.gmail.com>
	<33c87fa31001241629g1961a8d2m6e76f62641270b7a@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
	<33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
	<191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>
	<33c87fa31001271939i668e9676y1f687dc182b4ca89@mail.gmail.com>
Message-ID: <CF0D5382-EE44-48CB-BF14-27A443014336@freeswitch.org>

I'm suspecting the code just isn't honoring the fs_path and sending it to the right place do you have a sip trace of this?

/b

On Jan 27, 2010, at 9:39 PM, Mark Campbell-Smith wrote:

> This is the printout from sofia.  2000 does not work sending custom
> NOTIFY messages and 2001 does.  Is it that 2001 is NAT'd, the port
> 10281 out of range or the contact incorrect or something?




From mcampbellsmith at gmail.com  Wed Jan 27 20:35:47 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 28 Jan 2010 15:35:47 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <CF0D5382-EE44-48CB-BF14-27A443014336@freeswitch.org>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<4468a6771001241650n74a033e0ne79ae3e137d3de44@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
	<33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
	<191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>
	<33c87fa31001271939i668e9676y1f687dc182b4ca89@mail.gmail.com>
	<CF0D5382-EE44-48CB-BF14-27A443014336@freeswitch.org>
Message-ID: <33c87fa31001272035t7c46e054t7ce2dd598fa40ebb@mail.gmail.com>

Hi Brian,

 I've previously enabled siptrace for internal profile, but I see
nothing sent and nothing received.

On Thu, Jan 28, 2010 at 2:54 PM, Brian West <brian at freeswitch.org> wrote:
> I'm suspecting the code just isn't honoring the fs_path and sending it to the right place do you have a sip trace of this?
>
> /b
>
> On Jan 27, 2010, at 9:39 PM, Mark Campbell-Smith wrote:
>
>> This is the printout from sofia. ?2000 does not work sending custom
>> NOTIFY messages and 2001 does. ?Is it that 2001 is NAT'd, the port
>> 10281 out of range or the contact incorrect or something?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From wiltingtree at gmail.com  Wed Jan 27 20:47:35 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Wed, 27 Jan 2010 23:47:35 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
Message-ID: <eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>

I tried running ldconfig on the directory containing mysql.so, but it did
not help.
So it sounds like there could be a bug in the latter versions?


On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> I got the same error, my script was working with no problems before an
> update to trunk.
>
> David
>
> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com> wrote:
> > Hi, I followed the instructions in the Lua documentation for setting up
> > luasql, but when I try to run my script I get:
> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module
> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
> >         /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment
> prot
> > after reloc: Permission denied
> > stack traceback:
> >         [C]: ?
> >         [C]: in function 'require'
> >         /usr/local/freeswitch/scripts/l.lua:2: in main chunk
> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
> > I changed the permissions for mysql.so and for my script to 777, so I'm
> not
> > sure where the permission problem could be.
> > I'd appreciate any suggestions.
> > Thanks,
> > Adam
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Wed Jan 27 21:18:51 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 28 Jan 2010 00:18:51 -0500
Subject: [Freeswitch-users] compilation error on fs_cli (Windows)
	(latest version)
In-Reply-To: <761142.19511.qm@web33506.mail.mud.yahoo.com>
References: <761142.19511.qm@web33506.mail.mud.yahoo.com>
Message-ID: <CC043BFD-5F5E-439E-AA16-EBCA411EA3C1@jerris.com>

These are real issues we need to fix.  Please open a bug on jira for these (even better with a patch to fix them).

Mie

On Jan 26, 2010, at 10:16 PM, Diego Toro wrote:

> 
> yes, code analyzer is active. when I turn it off fs_cli project compiled fine. Before, this project compiled fine, why I need turn off analyzer code now ?
> 
> Thank you
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 
> 
> --- On Tue, 1/26/10, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> 
>> From: Mathieu Rene <mrene_lists at avgs.ca>
>> Subject: Re: [Freeswitch-users] compilation error on fs_cli (Windows) (latest version)
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Tuesday, January 26, 2010, 11:45 AM
>> Looks like the code analyzer is
>> running, this is normally turned off  
>> when you do a normal build, turn it off and try again.
>> 
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>> 
>> 
>> 
>> 
>> On 26-Jan-10, at 10:55 AM, Diego Toro wrote:
>> 
>>> Hi, I have compilation error "error C2220" on fs_cli
>> project on  
>>> Windows using VS2008.
>>> 
>>> FS: latest version (2010/01/26)
>>> VS: VS2008
>>> SO: Windows 7
>>> 
>>> VS2008 Error log:
>>> 
>>> Error    1    error
>> C2220: warning treated as error - no 'object' file  
>>> generated   
>> g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl 
>>> \fs_cli.c    106   
>> fs_cli
>>> 
>>> Warning    2    warning
>> C6385: Invalid data: accessing 'global_profile- 
>>>> console_fnkeys', the readable size is '48' bytes,
>> but '-4' bytes  
>>> might be read: Lines: 86, 88, 90   
>> g:\ftp\incoming\fs 
>>> 
>> \freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c   
>> 90    fs_cli
>>> 
>>> Warning    3    warning
>> C6246: Local declaration of 'p' hides declaration  
>>> of the same name in outer scope. For additional
>> information, see  
>>> previous declaration at line '844' of
>> 'g:\ftp\incoming\fs 
>>> \freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c':
>> Lines: 844    g:\ftp 
>>> 
>> \incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c   
>> 895     
>>> fs_cli
>>> 
>>> Warning    4    warning
>> C6011: Dereferencing NULL pointer 'cursor': Lines:  
>>> 839, 840, 841, 842, 843, 844, 846, 849, 853, 857, 868,
>> 870, 871, 884     
>>> 
>> g:\ftp\incoming\fs\freeswitch-1.0.5-20100126-0400\libs\esl\fs_cli.c     
>>> 884    fs_cli




From anthony.minessale at gmail.com  Wed Jan 27 21:25:16 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Jan 2010 23:25:16 -0600
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001272035t7c46e054t7ce2dd598fa40ebb@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001270506s5628c407k26ff0b2942818a48@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
	<33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
	<191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>
	<33c87fa31001271939i668e9676y1f687dc182b4ca89@mail.gmail.com>
	<CF0D5382-EE44-48CB-BF14-27A443014336@freeswitch.org>
	<33c87fa31001272035t7c46e054t7ce2dd598fa40ebb@mail.gmail.com>
Message-ID: <191c3a031001272125t2b19d134lbb65dfebbb7a52b9@mail.gmail.com>

You have to look in the sql db and compare the specified vals with the ones
looked up from the event again the user and host need to match the db

On Jan 27, 2010 10:41 PM, "Mark Campbell-Smith" <mcampbellsmith at gmail.com>
wrote:

Hi Brian,

 I've previously enabled siptrace for internal profile, but I see
nothing sent and nothing received.


On Thu, Jan 28, 2010 at 2:54 PM, Brian West <brian at freeswitch.org> wrote:
> I'm suspecting the code...
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From mike at van.lammeren.net  Wed Jan 27 22:27:45 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 28 Jan 2010 01:27:45 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
Message-ID: <5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>

Have you tried running a Lua script that includes the library from outside
of FreeSWITCH? What does that do?


On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> I tried running ldconfig on the directory containing mysql.so, but it did
> not help.
> So it sounds like there could be a bug in the latter versions?
>
>
> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> I got the same error, my script was working with no problems before an
>> update to trunk.
>>
>> David
>>
>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com> wrote:
>> > Hi, I followed the instructions in the Lua documentation for setting up
>> > luasql, but when I try to run my script I get:
>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module
>> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
>> >         /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment
>> prot
>> > after reloc: Permission denied
>> > stack traceback:
>> >         [C]: ?
>> >         [C]: in function 'require'
>> >         /usr/local/freeswitch/scripts/l.lua:2: in main chunk
>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
>> > I changed the permissions for mysql.so and for my script to 777, so I'm
>> not
>> > sure where the permission problem could be.
>> > I'd appreciate any suggestions.
>> > Thanks,
>> > Adam
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ranjtech at gmail.com  Wed Jan 27 23:11:29 2010
From: ranjtech at gmail.com (RR)
Date: Thu, 28 Jan 2010 02:11:29 -0500
Subject: [Freeswitch-users] 409 Conflict When registering FS with Softswitch
Message-ID: <020c01ca9fe9$1d5952f0$580bf8d0$@com>

Gentlemen,

 

I have a probably a simple problem but I have no idea why it's occurring. I
am beyond novice/new to configuring FS and as my first try, tried to
configure it to register with our softswitch so I could have the basic
registration to happen between FS and it. However, I keep getting the "409
Conflict" message when that username is not actually registered with the
softswitch. The profile / config file under sip_profiles/external is
configured like so

 

<include>

  <gateway name="Test-Inbound">

  <param name="username" value="user1"/>-->

  <param name="password" value="user1pwd"/>-->

  <param name="realm" value="mysoftswitch.net.au"/>-->

  <param name="extension" value="1000"/>

  <param name="from-user" value="user1"/>-->

  <param name="expire-seconds" value="60"/>-->

  <param name="register" value="true"/>-->

  </gateway>-->

</include>

 

Any ideas why? And how to resolve it or what kind of traces I can send to
you guys to help me resolve it? Like I said I am so new to FS that I don't
even know what I can turn on in FS to see what's going on

 

Thanks a lot

\RR

 

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From david.villasmil.work at gmail.com  Wed Jan 27 23:43:59 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 28 Jan 2010 08:43:59 +0100
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
Message-ID: <9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>

Hello,

That works fine:

box:~# lua testdb.lua
box:~#


David

On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
<mike at van.lammeren.net> wrote:
> Have you tried running a Lua script that includes the library from outside
> of FreeSWITCH? What does that do?
>
> On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
>>
>> I tried running ldconfig on the directory containing mysql.so, but it did
>> not help.
>> So it sounds like there could be a bug in the latter versions?
>>
>> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
>> <david.villasmil.work at gmail.com> wrote:
>>>
>>> I got the same error, my script was working with no problems before an
>>> update to trunk.
>>>
>>> David
>>>
>>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com> wrote:
>>> > Hi, I followed the instructions in the Lua documentation for setting up
>>> > luasql, but when I try to run my script I get:
>>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module
>>> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
>>> > ?? ? ? ?/usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment
>>> > prot
>>> > after reloc: Permission denied
>>> > stack traceback:
>>> > ?? ? ? ?[C]: ?
>>> > ?? ? ? ?[C]: in function 'require'
>>> > ?? ? ? ?/usr/local/freeswitch/scripts/l.lua:2: in main chunk
>>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
>>> > I changed the permissions for mysql.so and for my script to 777, so I'm
>>> > not
>>> > sure where the permission problem could be.
>>> > I'd appreciate any suggestions.
>>> > Thanks,
>>> > Adam
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Wed Jan 27 23:49:56 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 28 Jan 2010 01:49:56 -0600
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
Message-ID: <1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>

Can you provide a SIP Trace?

/b

On Jan 28, 2010, at 1:11 AM, RR wrote:

> Gentlemen,
>  
> I have a probably a simple problem but I have no idea why it?s occurring. I am beyond novice/new to configuring FS and as my first try, tried to configure it to register with our softswitch so I could have the basic registration to happen between FS and it. However, I keep getting the ?409 Conflict? message when that username is not actually registered with the softswitch. The profile / config file under sip_profiles/external is configured like so
>  
> <include>
>   <gateway name="Test-Inbound">
>   <param name="username" value="user1"/>-->
>   <param name="password" value="user1pwd"/>-->
>   <param name="realm" value="mysoftswitch.net.au"/>-->
>   <param name="extension" value="1000"/>
>   <param name="from-user" value="user1"/>-->
>   <param name="expire-seconds" value="60"/>-->
>   <param name="register" value="true"/>-->
>   </gateway>-->
> </include>
>  
> Any ideas why? And how to resolve it or what kind of traces I can send to you guys to help me resolve it? Like I said I am so new to FS that I don?t even know what I can turn on in FS to see what?s going on
>  
> Thanks a lot
> \RR
>  
> 
> 
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4812 (20100128) __________
> 
> The message was checked by ESET NOD32 Antivirus.
> 
> http://www.eset.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From siniypin at gmail.com  Thu Jan 28 00:09:18 2010
From: siniypin at gmail.com (RobertT)
Date: Thu, 28 Jan 2010 11:09:18 +0300
Subject: [Freeswitch-users] Presence Change Distribution
In-Reply-To: <591B9C113F064880993543272B16ADF3@greyhawk.tonecommander.com>
References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com>
	<2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com>
	<26B8578C14BA4BE18F2D1278B0C9561B@greyhawk.tonecommander.com>
	<191c3a031001251104p55ba7009g9381841f7de56d65@mail.gmail.com>
	<BE6F5600B8E54EE297963A8B08F70E5E@greyhawk.tonecommander.com>
	<191c3a031001261321v2e8ea21cm7da19e01a11b59f9@mail.gmail.com>
	<591B9C113F064880993543272B16ADF3@greyhawk.tonecommander.com>
Message-ID: <2160023e1001280009v65d9b3ees78a8cb319205649a@mail.gmail.com>

You can send your own custom notes in xml. That is what I do to make
presence a little bit reliable.

Best regards, Robert.

2010/1/27 Jerry Richards <jerry.richards at teotech.com>

>  There are two places in the XML body that are diffierent:
>
> FS Rcvd PUBLISH has:  <rpid:away/> and <dm:note>Away</dm:note>
> FS Sent NOTIFY   has:  <rpid:busy/> and <dm:note>Busy</dm:note>
>
> This behavior (above) is why I'm not seeing the published presence at the
> subscribing softphone.  FS should be sending the new Away status in the
> NOTIFY message.
>
> I did notice there is an "[ERR] sofia_presnece.c:674" FS log between the
> PUBLISH and the NOTIFY (please see Line 89 of
> http://pastebin.freeswitch.org/11953).  Line 674 is in the
> sofia_presence_event_thread_run() function where it calls
> switch_mutex_unlock(mod_sofia_globals.mutex).  Do you think this [ERR] is
> related to why FS sends the previous status and not updated status?
>
> Thanks And Best Regards,
> Jerry
>
>
>  ------------------------------
> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
> *Sent:* Tuesday, January 26, 2010 1:22 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>
> its sending a notify to them right away (line 174 of your PB)
> the xml in the notify we send looks the same as what they sent except one
> thing
>
> They send: <rpid:away/>
> We send: <rpid:busy/>
>
> everybody who implements this seems to have their own idea of what to say
> here.
>
> This crazy xml presence crap is pure garbage so maybe that's it.
>
>
>
> On Tue, Jan 26, 2010 at 3:02 PM, Jerry Richards <
> jerry.richards at teotech.com> wrote:
>
>>  Okay, I setup my FS to force SUBSCRIBE Expires to 3600 seconds.  Then I
>> captured a FS console trace of a Bria softphone changing it's presence state
>> from 'Busy' to 'Away' (see http://pastebin.freeswitch.org/11953) and
>> observed that the subscribing Bria softphone did not update to 'Away'.  At
>> the same time, I executed the sqlite3 app and pasted each of the 3 SQL
>> select statements I saw in the FS console log, and pasted them below.  I'm
>> new to sqlite3.  Do you see what my issue is?
>>
>> sqlite>
>> sqlite> select
>> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
>> from sip_subscriptions left join sip_presence on
>> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
>> sip_subscriptions.sub_to_host=sip_presence.sip_host and
>> sip_subscriptions.profile_name=sip_presence.profile_name) where
>> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
>> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
>> like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'internal' or
>> sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
>> sip|5382|192.168.72.79|5401|192.168.72.79|presence|"5382 on 79" <
>> sip:5382 at 192.168.72.150:34672>|ZTQ2ZWQwZGRlZjRiNTdkYTJjNGM5NTgzOWIyNmIwZmU.|"5382
>> on 79" <sip:5382 at 192.168.72.79 <sip%3A5382 at 192.168.72.79>>;tag=68bb4eb6|SIP/2.0/UDP
>> 192.168.72.150:34672;branch=z9hG4bK-d8754z-eafc60166305eaef-1---d8754z-;rport=34672|1264546204|Teo
>> Softphone release 2.5.4 stamp
>> 55958||internal|Away|away|192.168.72.79|Away|away
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite> select
>> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
>> from sip_subscriptions left join sip_presence on
>> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
>> sip_subscriptions.sub_to_host=sip_presence.sip_host and
>> sip_subscriptions.profile_name=sip_presence.profile_name) where
>> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
>> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
>> like '%192.168.72.79%') and (sip_subscriptions.profile_name =
>> 'internal-ipv6' or sip_subscriptions.presence_hosts !=
>> sip_subscriptions.sub_to_host);
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite>
>> sqlite> select
>> sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','192.168.72.79',sip_presence.status,sip_presence.rpid
>> from sip_subscriptions left join sip_presence on
>> (sip_subscriptions.sub_to_user=sip_presence.sip_user and
>> sip_subscriptions.sub_to_host=sip_presence.sip_host and
>> sip_subscriptions.profile_name=sip_presence.profile_name) where
>> sip_subscriptions.expires > -1 and (event='presence' or event='presence')
>> and sub_to_user='5401' and (sub_to_host='192.168.72.79' or presence_hosts
>> like '%192.168.72.79%') and (sip_subscriptions.profile_name = 'external' or
>> sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host);
>> sqlite>
>>  Thanks and Best Regards,
>> Jerry
>>
>>  ------------------------------
>> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
>> *Sent:* Monday, January 25, 2010 11:05 AM
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>>
>>   the notify will be instant after the publish
>> the notify you see are not triggered by the publish or they would be
>> instant.
>>
>> Same drill, turn on presence debugging in sofia.conf.xml
>> and look at the sql stmts and see why
>>
>>
>> On Mon, Jan 25, 2010 at 12:30 PM, Jerry Richards <
>> jerry.richards at teotech.com> wrote:
>>
>>>  Okay, I notice that if I reduce the Presence SUBSCRIBE Expires duration
>>> (from 3600 seconds to 60 seconds), then the delay between PUBLISH's and
>>> NOTIFY's is reduced, but FS still waits nearly 45 seconds to send the
>>> NOTIFY's after it receives a PUBLISH.
>>>
>>> Can a change be made in FS so that NOTIFYs are sent as a direct result of
>>> receipt of the PUBLISH message?  And not tied to the SUBSCRIBE expiration?
>>> I really don't want to configure all my phones to re-subscribe every 30 or
>>> 15 seconds.
>>>
>>> Thanks and Best Regards,
>>> Jerry
>>>
>>>
>>>  ------------------------------
>>> *From:* RobertT [mailto:siniypin at gmail.com]
>>> *Sent:* Tuesday, December 29, 2009 12:02 AM
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Subject:* Re: [Freeswitch-users] Presence Change Distribution
>>>
>>>   You can try to reduce your registration time.
>>> I for one made my client apps send PUBLISH message every minute in
>>> addition to reduced registration time.
>>>
>>> Regards, Robert.
>>>
>>> 2009/12/28 Jerry Richards <jerry.richards at teotech.com>
>>>
>>>> Is there a setting to control how fast FS distributes presence changes
>>>> to
>>>> subscribers?  Currently, it appears to take several minutes before I see
>>>> presence changes.  I would like to see them almost instantaneously, if
>>>> possible.
>>>>
>>>> Thanks and Best Regards,
>>>> Jerry
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From codecomplete at free.fr  Thu Jan 28 02:24:07 2010
From: codecomplete at free.fr (Fred-145)
Date: Thu, 28 Jan 2010 11:24:07 +0100
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
References: <27338355.post@talk.nabble.com>
	<27338355.post-WJuSqJV8a7jJsTRiRinrng@public.gmane.org>
	<9BB5BC82B9F54466ACFA5BA610669FD7@cune.pri>
Message-ID: <mcp2m55d16dlaq51kt5a0k7m4iut33c1k1@4ax.com>

On Wed, 27 Jan 2010 06:47:28 -0600, "Russell Mosemann"
<Russell.Mosemann at cune.org> wrote:
>http://wiki.freeswitch.org/wiki/Bypass_Media

Thanks everyone. I'll experiment with Freeswitch in the LAN behind a
NAT router and check if performance is good enough to avoid using the
bypass-media option.




From codecomplete at free.fr  Thu Jan 28 03:04:46 2010
From: codecomplete at free.fr (Fred-145)
Date: Thu, 28 Jan 2010 12:04:46 +0100
Subject: [Freeswitch-users] Investigating one-way sound?
References: <27341219.post@talk.nabble.com>
	<27341219.post-WJuSqJV8a7jJsTRiRinrng@public.gmane.org>
	<22DD670E-B920-4328-9939-56447375D5C7@freeswitch.org>
Message-ID: <jfp2m55k5vedrabmubv801lgd404s5769j@4ax.com>

On Wed, 27 Jan 2010 09:30:41 -0600, Brian West
<brian at freeswitch.org> wrote:

> I'm going to guess right off the X-Lite is putting its public IP and
> since maybe your NAT router can't hair pin the media you get on way media.

Thanks Brian for the tip. I tried this, but it didn't work. Turns out
it's a bug in how XLite and the Siemens negotiate codecs, and it's now
working.




From codecomplete at free.fr  Thu Jan 28 03:54:16 2010
From: codecomplete at free.fr (Fred-145)
Date: Thu, 28 Jan 2010 12:54:16 +0100
Subject: [Freeswitch-users] [fs_cli] Only display SIP messages?
Message-ID: <mku2m5lrafrcvq4aqr0pclu1p9530hs73j@4ax.com>

Hello

Before I learn how to use Wireshark and filter stuff... can the fs_cli
console be configured so that only SIP messages are displayed? I'd
like to do this so I can learn more about what goes on when I play
with SIP clients.

Thank you.




From lakindia89 at gmail.com  Thu Jan 28 04:11:02 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 28 Jan 2010 17:41:02 +0530
Subject: [Freeswitch-users] Multiple DTMF on group_confirm_key
Message-ID: <7d79b3931001280411u6262f627xca723c64de5e118d@mail.gmail.com>

Hi all,

I've experimented with group confirm key and group confirm file. It works
great. However, I was unable to give multiple DTMF digits to get the
confirmation.

I've set group_confirm_key=1234, I thought it will ask the 4 digits from the
user. But it simply taken 1 and when the user presses 1, the call got
bridged.

Is there any way to specify multiple dtmf to be confirmed??
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From javieraristizabal at gmail.com  Thu Jan 28 05:57:14 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Thu, 28 Jan 2010 08:57:14 -0500
Subject: [Freeswitch-users] [fs_cli] Only display SIP messages?
In-Reply-To: <mku2m5lrafrcvq4aqr0pclu1p9530hs73j@4ax.com>
References: <mku2m5lrafrcvq4aqr0pclu1p9530hs73j@4ax.com>
Message-ID: <b6f11f301001280557w57114acbn39c4e749b43b2d0@mail.gmail.com>

Hello. Try:

sofia profile internal siptrace on

Replace internal for the profile that you need.


Javier

On Thu, Jan 28, 2010 at 6:54 AM, Fred-145 <codecomplete at free.fr> wrote:

> Hello
>
> Before I learn how to use Wireshark and filter stuff... can the fs_cli
> console be configured so that only SIP messages are displayed? I'd
> like to do this so I can learn more about what goes on when I play
> with SIP clients.
>
> Thank you.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From wiltingtree at gmail.com  Thu Jan 28 06:41:42 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Thu, 28 Jan 2010 09:41:42 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
Message-ID: <eb7797d71001280641j48a0d93bt7da6c27aa8a317ee@mail.gmail.com>

Man, I'm in the process of switching from Python to Lua because of bugs in
the Python support.
If Lua doesn't work with odbc, that's a big problem.
Does anybody have any other advice?


On Thu, Jan 28, 2010 at 2:43 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello,
>
> That works fine:
>
> box:~# lua testdb.lua
> box:~#
>
>
> David
>
> On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
> <mike at van.lammeren.net> wrote:
> > Have you tried running a Lua script that includes the library from
> outside
> > of FreeSWITCH? What does that do?
> >
> > On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com>
> wrote:
> >>
> >> I tried running ldconfig on the directory containing mysql.so, but it
> did
> >> not help.
> >> So it sounds like there could be a bug in the latter versions?
> >>
> >> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
> >> <david.villasmil.work at gmail.com> wrote:
> >>>
> >>> I got the same error, my script was working with no problems before an
> >>> update to trunk.
> >>>
> >>> David
> >>>
> >>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com>
> wrote:
> >>> > Hi, I followed the instructions in the Lua documentation for setting
> up
> >>> > luasql, but when I try to run my script I get:
> >>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module
> >>> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
> >>> >         /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore
> segment
> >>> > prot
> >>> > after reloc: Permission denied
> >>> > stack traceback:
> >>> >         [C]: ?
> >>> >         [C]: in function 'require'
> >>> >         /usr/local/freeswitch/scripts/l.lua:2: in main chunk
> >>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
> >>> > I changed the permissions for mysql.so and for my script to 777, so
> I'm
> >>> > not
> >>> > sure where the permission problem could be.
> >>> > I'd appreciate any suggestions.
> >>> > Thanks,
> >>> > Adam
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From codecomplete at free.fr  Thu Jan 28 07:09:55 2010
From: codecomplete at free.fr (Fred-145)
Date: Thu, 28 Jan 2010 16:09:55 +0100
Subject: [Freeswitch-users] [fs_cli] Only display SIP messages?
References: <mku2m5lrafrcvq4aqr0pclu1p9530hs73j@4ax.com>
	<mku2m5lrafrcvq4aqr0pclu1p9530hs73j-e09XROE/p8c@public.gmane.org>
	<b6f11f301001280557w57114acbn39c4e749b43b2d0@mail.gmail.com>
Message-ID: <p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi@4ax.com>

On Thu, 28 Jan 2010 08:57:14 -0500, Javier Aristiz?bal
<javieraristizabal at gmail.com> wrote:
>sofia profile internal siptrace on

Thanks for the tip, but even with this command, I get a lot more data
than just the SIP dialog:

==========
Dialplan: sofia/internal/1004 at 192.168.0.7 Regex (FAIL)
[global-intercept] destination_number(1001) =~ /^886$/ break=on-false
Dialplan: sofia/internal/1004 at 192.168.0.7 parsing
[default->group-intercept] continue=false
[...]
2010-01-28 16:07:34.149826 [DEBUG] switch_core_state_machine.c:122
(sofia/internal/1004 at 192.168.0.7) State Change CS_ROUTING ->
CS_EXECUTE
2010-01-28 16:07:34.149826 [DEBUG] switch_core_session.c:1013 Send
signal sofia/internal/1004 at 192.168.0.7 [BREAK]
[...]
send 1369 bytes to udp/[192.168.0.1]:19354 at 15:07:34.017297:

------------------------------------------------------------------------
   INVITE sip:1001 at 192.168.0.1:19354;rinstance=804198d1614a592d
SIP/2.0
   Via: SIP/2.0/UDP 192.168.0.7;rport;branch=z9hG4bKpDgjj09Qv54Xj
   Max-Forwards: 69
   From: "Extension 1004" <sip:1004 at 192.168.0.7>;tag=Nra5SSp7aat4S
   To: <sip:1001 at 192.168.0.1:19354;rinstance=804198d1614a592d>
   Call-ID: b455d480-86c1-122d-f48c-00242116bb98
   CSeq: 126227291 INVITE
   Contact: <sip:mod_sofia at 192.168.0.7:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16456
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 425
   X-FS-Support: update_display
   Remote-Party-ID: "Extension 1004"
<sip:1004 at 192.168.0.7>;party=calling;screen=yes;privacy=off
   
   v=0
   o=FreeSWITCH 1264662214 1264662215 IN IP4 192.168.0.7
   s=FreeSWITCH
   c=IN IP4 192.168.0.7
   t=0 0
   m=audio 29040 RTP/AVP 8 115 107 9 0 3 101 13
   a=rtpmap:8 PCMA/8000
==========

Can fs_cli be configured to filter everything but the SIP protocol
itself?




From john at acsol.net  Thu Jan 28 07:26:43 2010
From: john at acsol.net (John)
Date: Thu, 28 Jan 2010 08:26:43 -0700
Subject: [Freeswitch-users] Voicemail via web interface
Message-ID: <4B61ACB3.50903@acsol.net>

Hello,
Can you point me to any additional information about the voice mail via 
web interface? I have it up and running; however if you click the play 
button there is no playback, if you click download it will play in MS 
media player. Thanks John



From rupa at rupa.com  Thu Jan 28 07:44:07 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 28 Jan 2010 09:44:07 -0600
Subject: [Freeswitch-users] [fs_cli] Only display SIP messages?
In-Reply-To: <p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi@4ax.com>
References: <mku2m5lrafrcvq4aqr0pclu1p9530hs73j@4ax.com>
	<mku2m5lrafrcvq4aqr0pclu1p9530hs73j-e09XROE/p8c@public.gmane.org>
	<b6f11f301001280557w57114acbn39c4e749b43b2d0@mail.gmail.com>
	<p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi@4ax.com>
Message-ID: <b39273531001280744j29f596c9qad879db501cac509@mail.gmail.com>

I would suggest using ngrep or sipgrep to get a capture of the sip
dialog.  Or going full bore and using wireshark.  I often use ngrep or
sipgrep when I want to follow a sip dialog.

On Thu, Jan 28, 2010 at 9:09 AM, Fred-145 <codecomplete at free.fr> wrote:
> On Thu, 28 Jan 2010 08:57:14 -0500, Javier Aristiz?bal
> <javieraristizabal at gmail.com> wrote:
>>sofia profile internal siptrace on
>
> Thanks for the tip, but even with this command, I get a lot more data
> than just the SIP dialog:
>
> ==========
> Dialplan: sofia/internal/1004 at 192.168.0.7 Regex (FAIL)
> [global-intercept] destination_number(1001) =~ /^886$/ break=on-false
> Dialplan: sofia/internal/1004 at 192.168.0.7 parsing
> [default->group-intercept] continue=false
> [...]
> 2010-01-28 16:07:34.149826 [DEBUG] switch_core_state_machine.c:122
> (sofia/internal/1004 at 192.168.0.7) State Change CS_ROUTING ->
> CS_EXECUTE
> 2010-01-28 16:07:34.149826 [DEBUG] switch_core_session.c:1013 Send
> signal sofia/internal/1004 at 192.168.0.7 [BREAK]
> [...]
> send 1369 bytes to udp/[192.168.0.1]:19354 at 15:07:34.017297:
>
> ------------------------------------------------------------------------
> ? INVITE sip:1001 at 192.168.0.1:19354;rinstance=804198d1614a592d
> SIP/2.0
> ? Via: SIP/2.0/UDP 192.168.0.7;rport;branch=z9hG4bKpDgjj09Qv54Xj
> ? Max-Forwards: 69
> ? From: "Extension 1004" <sip:1004 at 192.168.0.7>;tag=Nra5SSp7aat4S
> ? To: <sip:1001 at 192.168.0.1:19354;rinstance=804198d1614a592d>
> ? Call-ID: b455d480-86c1-122d-f48c-00242116bb98
> ? CSeq: 126227291 INVITE
> ? Contact: <sip:mod_sofia at 192.168.0.7:5060>
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16456
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> ? Supported: timer, precondition, path, replaces
> ? Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> ? Content-Type: application/sdp
> ? Content-Disposition: session
> ? Content-Length: 425
> ? X-FS-Support: update_display
> ? Remote-Party-ID: "Extension 1004"
> <sip:1004 at 192.168.0.7>;party=calling;screen=yes;privacy=off
>
> ? v=0
> ? o=FreeSWITCH 1264662214 1264662215 IN IP4 192.168.0.7
> ? s=FreeSWITCH
> ? c=IN IP4 192.168.0.7
> ? t=0 0
> ? m=audio 29040 RTP/AVP 8 115 107 9 0 3 101 13
> ? a=rtpmap:8 PCMA/8000
> ==========
>
> Can fs_cli be configured to filter everything but the SIP protocol
> itself?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa



From codecomplete at free.fr  Thu Jan 28 08:22:14 2010
From: codecomplete at free.fr (Fred-145)
Date: Thu, 28 Jan 2010 17:22:14 +0100
Subject: [Freeswitch-users] [fs_cli] Only display SIP messages?
References: <mku2m5lrafrcvq4aqr0pclu1p9530hs73j@4ax.com>
	<mku2m5lrafrcvq4aqr0pclu1p9530hs73j-e09XROE/p8c@public.gmane.org>
	<b6f11f301001280557w57114acbn39c4e749b43b2d0@mail.gmail.com>
	<p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi@4ax.com>
	<p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi-e09XROE/p8c@public.gmane.org>
	<b39273531001280744j29f596c9qad879db501cac509@mail.gmail.com>
Message-ID: <uce3m59t58cs4o8r51cjvslbljmdv266pt@4ax.com>

On Thu, 28 Jan 2010 09:44:07 -0600, Rupa Schomaker
<rupa at rupa.com> wrote:
> I often use ngrep or >sipgrep when I want to follow a sip dialog.

Thanks for the tip.




From anthony.minessale at gmail.com  Thu Jan 28 08:28:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 Jan 2010 10:28:56 -0600
Subject: [Freeswitch-users] [fs_cli] Only display SIP messages?
In-Reply-To: <p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi@4ax.com>
References: <mku2m5lrafrcvq4aqr0pclu1p9530hs73j@4ax.com>
	<mku2m5lrafrcvq4aqr0pclu1p9530hs73j-e09XROE/p8c@public.gmane.org>
	<b6f11f301001280557w57114acbn39c4e749b43b2d0@mail.gmail.com>
	<p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi@4ax.com>
Message-ID: <191c3a031001280828m785a077fm5bccc7eb7abe2a6f@mail.gmail.com>

console loglevel 0
as well

On Thu, Jan 28, 2010 at 9:09 AM, Fred-145 <codecomplete at free.fr> wrote:

> On Thu, 28 Jan 2010 08:57:14 -0500, Javier Aristiz?bal
> <javieraristizabal at gmail.com> wrote:
> >sofia profile internal siptrace on
>
> Thanks for the tip, but even with this command, I get a lot more data
> than just the SIP dialog:
>
> ==========
> Dialplan: sofia/internal/1004 at 192.168.0.7 Regex (FAIL)
> [global-intercept] destination_number(1001) =~ /^886$/ break=on-false
> Dialplan: sofia/internal/1004 at 192.168.0.7 parsing
> [default->group-intercept] continue=false
> [...]
> 2010-01-28 16:07:34.149826 [DEBUG] switch_core_state_machine.c:122
> (sofia/internal/1004 at 192.168.0.7) State Change CS_ROUTING ->
> CS_EXECUTE
> 2010-01-28 16:07:34.149826 [DEBUG] switch_core_session.c:1013 Send
> signal sofia/internal/1004 at 192.168.0.7 [BREAK]
> [...]
> send 1369 bytes to udp/[192.168.0.1]:19354 at 15:07:34.017297:
>
> ------------------------------------------------------------------------
>   INVITE sip:1001 at 192.168.0.1:19354;rinstance=804198d1614a592d
> SIP/2.0
>   Via: SIP/2.0/UDP 192.168.0.7;rport;branch=z9hG4bKpDgjj09Qv54Xj
>   Max-Forwards: 69
>   From: "Extension 1004" <sip:1004 at 192.168.0.7 <sip%3A1004 at 192.168.0.7>
> >;tag=Nra5SSp7aat4S
>   To: <sip:1001 at 192.168.0.1:19354;rinstance=804198d1614a592d>
>   Call-ID: b455d480-86c1-122d-f48c-00242116bb98
>   CSeq: 126227291 INVITE
>   Contact: <sip:mod_sofia at 192.168.0.7:5060>
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16456
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>   Content-Type: application/sdp
>   Content-Disposition: session
>   Content-Length: 425
>   X-FS-Support: update_display
>   Remote-Party-ID: "Extension 1004"
> <sip:1004 at 192.168.0.7 <sip%3A1004 at 192.168.0.7>
> >;party=calling;screen=yes;privacy=off
>
>   v=0
>   o=FreeSWITCH 1264662214 1264662215 IN IP4 192.168.0.7
>   s=FreeSWITCH
>   c=IN IP4 192.168.0.7
>   t=0 0
>   m=audio 29040 RTP/AVP 8 115 107 9 0 3 101 13
>   a=rtpmap:8 PCMA/8000
> ==========
>
> Can fs_cli be configured to filter everything but the SIP protocol
> itself?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From codecomplete at free.fr  Thu Jan 28 08:37:31 2010
From: codecomplete at free.fr (Fred)
Date: Thu, 28 Jan 2010 17:37:31 +0100
Subject: [Freeswitch-users] [fs_cli] Only display SIP messages?
Message-ID: <7.0.1.0.2.20100128173441.0266b8e0@free.fr>

At 17:17 28/01/2010, Diego Viola <diego.viola at gmail.com> wrote:
>You could try this from fs_cli:
>
>fsctl loglevel 0
>sofia profile internal siptrace on

Works great. Thank you.  




From codecomplete at free.fr  Thu Jan 28 08:41:00 2010
From: codecomplete at free.fr (Fred-145)
Date: Thu, 28 Jan 2010 17:41:00 +0100
Subject: [Freeswitch-users] [fs_cli] Only display SIP messages?
References: <mku2m5lrafrcvq4aqr0pclu1p9530hs73j@4ax.com>
	<mku2m5lrafrcvq4aqr0pclu1p9530hs73j-e09XROE/p8c@public.gmane.org>
	<b6f11f301001280557w57114acbn39c4e749b43b2d0@mail.gmail.com>
	<p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi@4ax.com>
	<p3a3m5p9bo39kqioq5hm2cr1l7bcfgdqbi-e09XROE/p8c@public.gmane.org>
	<191c3a031001280828m785a077fm5bccc7eb7abe2a6f@mail.gmail.com>
Message-ID: <1gf3m55r2oc346agcop9fp5m89tt13g1j8@4ax.com>

On Thu, 28 Jan 2010 10:28:56 -0600, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
>console loglevel 0
>as well

Thanks, just what I was looking for.




From troy at tlainvestments.com  Thu Jan 28 08:49:20 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 28 Jan 2010 09:49:20 -0700
Subject: [Freeswitch-users] Call Dropping with SIP 503 status
In-Reply-To: <191c3a031001271431re6cfad2w1389e96db4e6c1a0@mail.gmail.com>
References: <d021d3be1001271126y7a40e209t13e3d5b71af0d1a4@mail.gmail.com>
	<ADD964DC-4B88-4DFA-A330-D391BE786F78@freeswitch.org>
	<BD4CD56A-D488-4CCD-955D-0A838CB3CFD1@tlainvestments.com>
	<191c3a031001271431re6cfad2w1389e96db4e6c1a0@mail.gmail.com>
Message-ID: <910698FF-EDB7-43B1-B776-8CC7E69E8023@tlainvestments.com>

Of course the error didn't show up in the 4 hours I had the sip trace on...

I downgraded the firmware on the Polycom 301's to 3.3.1RevB in stead of 3.2.2 and don't seem to be having the problem any more.

If it comes back, we'll break out sip trance again to see what's up.

Thanks!
-Troy

On Jan 27, 2010, at 3:31 PM, Anthony Minessale wrote:

> try turning on sip trace as well to see the sip traffic
> 
> sofia profile internal siptrace on (from cli)
> probably its something that said it could do session timers but was lying
> 
> 
> On Wed, Jan 27, 2010 at 2:05 PM, Troy Anderson <troy at tlainvestments.com> wrote:
> We are experiencing an odd issue.  We have many calls that don't drop, but some do after being up a minute or two.
> 
> The reason code is NORMAL_TEMPORARY_FAILURE and the sip status that is triggering that is 503 (Service Unavailable).  With only one or two calls up at a time, I don't think it's a session limit issue (set to 1000).
> 
> Here is the console log from just before the 503 status - any help is greatly appreciated!
> 
> 2010-01-27 12:49:11.879251 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:49:11.899334 [INFO] sofia.c:597 Update Callee ID to "400" <400>
> 2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [ready][200]
> 2010-01-27 12:49:11.919018 [DEBUG] sofia.c:4011 Duplicate SDP
> v=0
> o=- 1264621687 1264621687 IN IP4 192.168.0.46
> s=Polycom IP Phone
> c=IN IP4 192.168.0.46
> t=0 0
> a=sendrecv
> m=audio 2222 RTP/AVP 0 8 18 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
> 
> 2010-01-27 12:50:06.068999 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [ready][200]
> 2010-01-27 12:50:06.108877 [DEBUG] sofia.c:4011 Duplicate SDP
> v=0
> o=- 1264621687 1264621687 IN IP4 192.168.0.46
> s=Polycom IP Phone
> c=IN IP4 192.168.0.46
> t=0 0
> a=sendrecv
> m=audio 2222 RTP/AVP 0 8 18 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
> 
> 2010-01-27 12:51:05.259614 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [ready][200]
> 2010-01-27 12:51:05.298972 [DEBUG] sofia.c:4011 Duplicate SDP
> v=0
> o=- 1264621687 1264621687 IN IP4 192.168.0.46
> s=Polycom IP Phone
> c=IN IP4 192.168.0.46
> t=0 0
> a=sendrecv
> m=audio 2222 RTP/AVP 0 8 18 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
> 
> 2010-01-27 12:52:05.369138 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [ready][200]
> 2010-01-27 12:52:05.399251 [DEBUG] sofia.c:4011 Duplicate SDP
> v=0
> o=- 1264621687 1264621687 IN IP4 192.168.0.46
> s=Polycom IP Phone
> c=IN IP4 192.168.0.46
> t=0 0
> a=sendrecv
> m=audio 2222 RTP/AVP 0 8 18 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
> 
> 2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [calling][0]
> 2010-01-27 12:53:09.599069 [DEBUG] sofia.c:4003 Channel sofia/internal/400 at 192.168.0.31 entering state [terminating][503]
> 2010-01-27 12:53:09.599069 [NOTICE] sofia.c:4647 Hangup sofia/internal/400 at 192.168.0.31 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
> 2010-01-27 12:53:09.599069 [DEBUG] switch_ivr_bridge.c:466 sofia/internal/400 at 192.168.0.31 ending bridge by request from write function
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Thu Jan 28 09:14:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 Jan 2010 11:14:33 -0600
Subject: [Freeswitch-users] Multiple DTMF on group_confirm_key
In-Reply-To: <7d79b3931001280411u6262f627xca723c64de5e118d@mail.gmail.com>
References: <7d79b3931001280411u6262f627xca723c64de5e118d@mail.gmail.com>
Message-ID: <191c3a031001280914l56a035s74fd6d4b00dd9b3d@mail.gmail.com>

you have to use a script (See the wiki for executing a script)
then you can read in as many digits as you want and do what you need.


On Thu, Jan 28, 2010 at 6:11 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi all,
>
> I've experimented with group confirm key and group confirm file. It works
> great. However, I was unable to give multiple DTMF digits to get the
> confirmation.
>
> I've set group_confirm_key=1234, I thought it will ask the 4 digits from
> the user. But it simply taken 1 and when the user presses 1, the call got
> bridged.
>
> Is there any way to specify multiple dtmf to be confirmed??
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From david.villasmil.work at gmail.com  Thu Jan 28 09:49:31 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 28 Jan 2010 18:49:31 +0100
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <eb7797d71001280641j48a0d93bt7da6c27aa8a317ee@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
	<eb7797d71001280641j48a0d93bt7da6c27aa8a317ee@mail.gmail.com>
Message-ID: <9853f4ff1001280949u41336092j6b3ed6f3d3b6545@mail.gmail.com>

It does support SQL. Not sure about ODBC specifically. I know it does
support PostgreSQL, MySQL and Oracle
(http://www.keplerproject.org/luasql/)

And for it was working, it is actually still working on 1 box.

David

On Thu, Jan 28, 2010 at 3:41 PM, Adam Wilt <wiltingtree at gmail.com> wrote:
> Man, I'm in the process of switching from Python to Lua because of bugs in
> the Python support.
> If Lua doesn't work with odbc, that's a big problem.
> Does anybody have any other advice?
>
> On Thu, Jan 28, 2010 at 2:43 AM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> Hello,
>>
>> That works fine:
>>
>> box:~# lua testdb.lua
>> box:~#
>>
>>
>> David
>>
>> On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
>> <mike at van.lammeren.net> wrote:
>> > Have you tried running a Lua script that includes the library from
>> > outside
>> > of FreeSWITCH? What does that do?
>> >
>> > On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com>
>> > wrote:
>> >>
>> >> I tried running ldconfig on the directory containing mysql.so, but it
>> >> did
>> >> not help.
>> >> So it sounds like there could be a bug in the latter versions?
>> >>
>> >> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
>> >> <david.villasmil.work at gmail.com> wrote:
>> >>>
>> >>> I got the same error, my script was working with no problems before an
>> >>> update to trunk.
>> >>>
>> >>> David
>> >>>
>> >>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com>
>> >>> wrote:
>> >>> > Hi, I followed the instructions in the Lua documentation for setting
>> >>> > up
>> >>> > luasql, but when I try to run my script I get:
>> >>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading
>> >>> > module
>> >>> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
>> >>> > ?? ? ? ?/usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore
>> >>> > segment
>> >>> > prot
>> >>> > after reloc: Permission denied
>> >>> > stack traceback:
>> >>> > ?? ? ? ?[C]: ?
>> >>> > ?? ? ? ?[C]: in function 'require'
>> >>> > ?? ? ? ?/usr/local/freeswitch/scripts/l.lua:2: in main chunk
>> >>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
>> >>> > I changed the permissions for mysql.so and for my script to 777, so
>> >>> > I'm
>> >>> > not
>> >>> > sure where the permission problem could be.
>> >>> > I'd appreciate any suggestions.
>> >>> > Thanks,
>> >>> > Adam
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> >
>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From john at acsol.net  Thu Jan 28 10:43:16 2010
From: john at acsol.net (John)
Date: Thu, 28 Jan 2010 11:43:16 -0700
Subject: [Freeswitch-users] Voicemail in MP3
Message-ID: <4B61DAC4.2030301@acsol.net>

I have installed mod_shout and edited the modules.conf.xml to installed 
it as well. I have updated the user.xml to include <variable 
name="vm_message_ext" value="mp3"/>.... Messages are still being saved 
in WAV format. Missing Step? Thanks





From ranjtech at gmail.com  Thu Jan 28 10:46:56 2010
From: ranjtech at gmail.com (RR)
Date: Thu, 28 Jan 2010 13:46:56 -0500
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
Message-ID: <022701caa04a$44f60b80$cee22280$@com>

Hi brian,

 

Do you need the sip trace from within FS or just any packet capture like
from snoop/tcpdump etc will do? I don't know how to do a sip trace from
within FS

 

\R

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, January 28, 2010 2:50 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 409 Conflict When registering FS with
Softswitch

 

Can you provide a SIP Trace?

 

/b

 

On Jan 28, 2010, at 1:11 AM, RR wrote:





Gentlemen,

 

I have a probably a simple problem but I have no idea why it's occurring. I
am beyond novice/new to configuring FS and as my first try, tried to
configure it to register with our softswitch so I could have the basic
registration to happen between FS and it. However, I keep getting the "409
Conflict" message when that username is not actually registered with the
softswitch. The profile / config file under sip_profiles/external is
configured like so

 

<include>

  <gateway name="Test-Inbound">

  <param name="username" value="user1"/>-->

  <param name="password" value="user1pwd"/>-->

  <param name="realm" value="mysoftswitch.net.au"/>-->

  <param name="extension" value="1000"/>

  <param name="from-user" value="user1"/>-->

  <param name="expire-seconds" value="60"/>-->

  <param name="register" value="true"/>-->

  </gateway>-->

</include>

 

Any ideas why? And how to resolve it or what kind of traces I can send to
you guys to help me resolve it? Like I said I am so new to FS that I don't
even know what I can turn on in FS to see what's going on

 

Thanks a lot

\RR

 



__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4812 (20100128) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 



__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4815 (20100128) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From brian at freeswitch.org  Thu Jan 28 10:59:04 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 28 Jan 2010 12:59:04 -0600
Subject: [Freeswitch-users] Voicemail in MP3
In-Reply-To: <4B61DAC4.2030301@acsol.net>
References: <4B61DAC4.2030301@acsol.net>
Message-ID: <FB99FBD9-0E9B-4D8C-9B9E-C512EF23554B@freeswitch.org>

yes you're not loading mod_shout

/b

On Jan 28, 2010, at 12:43 PM, John wrote:

> I have installed mod_shout and edited the modules.conf.xml to installed 
> it as well. I have updated the user.xml to include <variable 
> name="vm_message_ext" value="mp3"/>.... Messages are still being saved 
> in WAV format. Missing Step? Thanks




From brian at freeswitch.org  Thu Jan 28 10:59:50 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 28 Jan 2010 12:59:50 -0600
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <022701caa04a$44f60b80$cee22280$@com>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
Message-ID: <697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>

Any sip trace would work.. text.. or what not.. 



/b

On Jan 28, 2010, at 12:46 PM, RR wrote:

> Hi brian,
>  
> Do you need the sip trace from within FS or just any packet capture like from snoop/tcpdump etc will do? I don?t know how to do a sip trace from within FS
>  
> \R

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From john at acsol.net  Thu Jan 28 11:11:11 2010
From: john at acsol.net (John)
Date: Thu, 28 Jan 2010 12:11:11 -0700
Subject: [Freeswitch-users] Voicemail in MP3
In-Reply-To: <FB99FBD9-0E9B-4D8C-9B9E-C512EF23554B@freeswitch.org>
References: <4B61DAC4.2030301@acsol.net>
	<FB99FBD9-0E9B-4D8C-9B9E-C512EF23554B@freeswitch.org>
Message-ID: <4B61E14F.4090801@acsol.net>

It shows that it's loaded without issues. I did uncomment the line in 
modules.conf.xml . Any way to know beyond that? Could anything else be 
wrong?
thanks




On 1/28/2010 11:59 AM, Brian West wrote:
> yes you're not loading mod_shout
>
> /b
>
> On Jan 28, 2010, at 12:43 PM, John wrote:
>
>    
>> I have installed mod_shout and edited the modules.conf.xml to installed
>> it as well. I have updated the user.xml to include<variable
>> name="vm_message_ext" value="mp3"/>.... Messages are still being saved
>> in WAV format. Missing Step? Thanks
>>      
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    




From jeff at jefflenk.com  Thu Jan 28 11:12:53 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 28 Jan 2010 13:12:53 -0600
Subject: [Freeswitch-users] Voicemail in MP3
In-Reply-To: <FB99FBD9-0E9B-4D8C-9B9E-C512EF23554B@freeswitch.org>
References: <4B61DAC4.2030301@acsol.net>,
	<FB99FBD9-0E9B-4D8C-9B9E-C512EF23554B@freeswitch.org>
Message-ID: <COL117-W304F063A2308794AE83AD3B95C0@phx.gbl>


There is a small problem with the logic using the callers setup rather than the callee. Been to busy to submit fix. If someone doesnt beat me to it I will submit fix later today.


 
> From: brian at freeswitch.org
> Date: Thu, 28 Jan 2010 12:59:04 -0600
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Voicemail in MP3
> 
> yes you're not loading mod_shout
> 
> /b
> 
> On Jan 28, 2010, at 12:43 PM, John wrote:
> 
> > I have installed mod_shout and edited the modules.conf.xml to installed 
> > it as well. I have updated the user.xml to include <variable 
> > name="vm_message_ext" value="mp3"/>.... Messages are still being saved 
> > in WAV format. Missing Step? Thanks
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Hotmail: Free, trusted and rich email service.
http://clk.atdmt.com/GBL/go/196390708/direct/01/
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From robert.hadley at teotech.com  Thu Jan 28 11:24:29 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 28 Jan 2010 11:24:29 -0800
Subject: [Freeswitch-users] Voicemail in MP3
In-Reply-To: <4B61DAC4.2030301@acsol.net>
References: <4B61DAC4.2030301@acsol.net>
Message-ID: <63E67C0D94CA4805822B2126EBA7D483@greyhawk.tonecommander.com>

A couple of things to check:

1. Make sure to enable mod_shout in the runtime installation folder:
   e.g. /usr/local/freeswitch/conf/autoload_conf/modules.conf.xml
2. I found the variable vm_message_ext to apply to sent emails, not
received.  For an internal extensions test set this variable for both ends.

-RobertH

-----Original Message-----
From: John [mailto:john at acsol.net] 
Sent: Thursday, January 28, 2010 10:43 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Voicemail in MP3

I have installed mod_shout and edited the modules.conf.xml to installed 
it as well. I have updated the user.xml to include <variable 
name="vm_message_ext" value="mp3"/>.... Messages are still being saved 
in WAV format. Missing Step? Thanks








From robert.hadley at teotech.com  Thu Jan 28 11:28:05 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 28 Jan 2010 11:28:05 -0800
Subject: [Freeswitch-users] Voicemail via web interface
In-Reply-To: <4B61ACB3.50903@acsol.net>
References: <4B61ACB3.50903@acsol.net>
Message-ID: <090DDB5F675E4FC48512CA3FC52FA096@greyhawk.tonecommander.com>


Using Firefox I was asked to install the latest Flash plugin and then I
could play the messages from the webpage directly.  IE8 never asked to add
the plugin that I noticed.
-RobertH


-----Original Message-----
From: John [mailto:john at acsol.net] 
Sent: Thursday, January 28, 2010 7:27 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Voicemail via web interface

Hello,
Can you point me to any additional information about the voice mail via 
web interface? I have it up and running; however if you click the play 
button there is no playback, if you click download it will play in MS 
media player. Thanks John






From troy at tlainvestments.com  Thu Jan 28 11:49:10 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 28 Jan 2010 12:49:10 -0700
Subject: [Freeswitch-users] ERR root tag missing
In-Reply-To: <090DDB5F675E4FC48512CA3FC52FA096@greyhawk.tonecommander.com>
References: <4B61ACB3.50903@acsol.net>
	<090DDB5F675E4FC48512CA3FC52FA096@greyhawk.tonecommander.com>
Message-ID: <F5D03273-03A1-4829-BDFA-A1173BA6CEEC@tlainvestments.com>

I'm seeing this error quite often on my systems:
	2010-01-28 12:35:46.703112 [ERR] switch_xml.c:1571 Error[[error near line 1]: root tag missing]

I've looked at freeswitch.xml.fsxml to see if I could find some kind of malformed XML, but with no luck.  Which Is line 1is it referring to?

Line 1 of freeswitch.xml.fsxml is a comment, with the first actual XML line like:
	<document type="freeswitch/xml">

This error always happens right after after a mod_dialplan_xml.c:408 log message, so I'm led to believe my dialplan XML is messed up, but I cannot see where.

In freeswitch.xml.fsxml near the dialplan section, this is what I have:

...
<section name="configuration" description="Various Configuration">
	<!-- CUT a bunch of stuff ... CUT -->
</section>
<section name="dialplan" description="Regex/XML Dialplan">
	<context name="my_office">
		<extension name="unloop">
			<condition field="${unroll_loops}" expression="^true$"/>
			<condition field="${sip_looped_call}" expression="^true$">
				<action application="deflect" data="${destination_number}"/>
			</condition>
		</extension>
		<extension name="set_domain" continue="true">
			<condition field="${domain_name}" expression="^$"/>
			<condition field="source" expression="mod_sofia"/>
			<condition field="${sip_auth_realm}" expression="^$">
				<action application="set" data="domain_name=10.0.0.120"/>
				<anti-action application="set" data="domain_name=${sip_auth_realm}"/>
			</condition>
		</extension>
...
</section>
...

Thanks for any ideas!

-Troy




From john at acsol.net  Thu Jan 28 11:52:04 2010
From: john at acsol.net (John)
Date: Thu, 28 Jan 2010 12:52:04 -0700
Subject: [Freeswitch-users] Voicemail via web interface
In-Reply-To: <090DDB5F675E4FC48512CA3FC52FA096@greyhawk.tonecommander.com>
References: <4B61ACB3.50903@acsol.net>
	<090DDB5F675E4FC48512CA3FC52FA096@greyhawk.tonecommander.com>
Message-ID: <4B61EAE4.2070607@acsol.net>

Thanks Robert. I believe the issue is probably because our files are in 
WAV format and not MP3.

On 1/28/2010 12:28 PM, Robert Hadley wrote:
> Using Firefox I was asked to install the latest Flash plugin and then I
> could play the messages from the webpage directly.  IE8 never asked to add
> the plugin that I noticed.
> -RobertH
>
>
> -----Original Message-----
> From: John [mailto:john at acsol.net]
> Sent: Thursday, January 28, 2010 7:27 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Voicemail via web interface
>
> Hello,
> Can you point me to any additional information about the voice mail via
> web interface? I have it up and running; however if you click the play
> button there is no playback, if you click download it will play in MS
> media player. Thanks John
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    




From john at acsol.net  Thu Jan 28 11:53:03 2010
From: john at acsol.net (John)
Date: Thu, 28 Jan 2010 12:53:03 -0700
Subject: [Freeswitch-users] Voicemail in MP3
In-Reply-To: <63E67C0D94CA4805822B2126EBA7D483@greyhawk.tonecommander.com>
References: <4B61DAC4.2030301@acsol.net>
	<63E67C0D94CA4805822B2126EBA7D483@greyhawk.tonecommander.com>
Message-ID: <4B61EB1F.2070105@acsol.net>

Would I setup the vm_message_ext in the dialplan then? Thanks

On 1/28/2010 12:24 PM, Robert Hadley wrote:
> A couple of things to check:
>
> 1. Make sure to enable mod_shout in the runtime installation folder:
>     e.g. /usr/local/freeswitch/conf/autoload_conf/modules.conf.xml
> 2. I found the variable vm_message_ext to apply to sent emails, not
> received.  For an internal extensions test set this variable for both ends.
>
> -RobertH
>
> -----Original Message-----
> From: John [mailto:john at acsol.net]
> Sent: Thursday, January 28, 2010 10:43 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Voicemail in MP3
>
> I have installed mod_shout and edited the modules.conf.xml to installed
> it as well. I have updated the user.xml to include<variable
> name="vm_message_ext" value="mp3"/>.... Messages are still being saved
> in WAV format. Missing Step? Thanks
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    




From Prometheus001 at gmx.net  Thu Jan 28 12:00:32 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 28 Jan 2010 21:00:32 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com>
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com>
	<4B6070A8.6050607@gmx.net>	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com>
	<4B607944.4040700@gmx.net> <4B608014.4030902@gmx.net>
	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>
Message-ID: <4B61ECE0.10409@gmx.net>

Hello Giovanni,

I did so but the same problem again.

Did you ever test in on Debian 5.0?

Best reards
Peter

Giovanni Maruzzelli schrieb:
> good, so you have only one sound device, the right one.
>
> Use the one with hw:0 in the window that skype gives you to set sound devices
>
> -gm
>
> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> I installed alsa-utile,
>>
>> now I get:
>>
>> skype:/var/cache/apt/archives# aplay -l
>> **** List of PLAYBACK Hardware Devices ****
>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>  Subdevices: 127/128
>>  Subdevice #0: subdevice #0
>>  Subdevice #1: subdevice #1
>>  Subdevice #2: subdevice #2
>>  Subdevice #3: subdevice #3
>>  Subdevice #4: subdevice #4
>>  Subdevice #5: subdevice #5
>>  Subdevice #6: subdevice #6
>>  Subdevice #7: subdevice #7
>>  Subdevice #8: subdevice #8
>>  Subdevice #9: subdevice #9
>>  Subdevice #10: subdevice #10
>>  Subdevice #11: subdevice #11
>>  Subdevice #12: subdevice #12
>>  Subdevice #13: subdevice #13
>>  Subdevice #14: subdevice #14
>>  Subdevice #15: subdevice #15
>>  Subdevice #16: subdevice #16
>>  Subdevice #17: subdevice #17
>>  Subdevice #18: subdevice #18
>>  Subdevice #19: subdevice #19
>>  Subdevice #20: subdevice #20
>>  Subdevice #21: subdevice #21
>>  Subdevice #22: subdevice #22
>>  Subdevice #23: subdevice #23
>>  Subdevice #24: subdevice #24
>>  Subdevice #25: subdevice #25
>>  Subdevice #26: subdevice #26
>>  Subdevice #27: subdevice #27
>>  Subdevice #28: subdevice #28
>>  Subdevice #29: subdevice #29
>>  Subdevice #30: subdevice #30
>>  Subdevice #31: subdevice #31
>>  Subdevice #32: subdevice #32
>>  Subdevice #33: subdevice #33
>>  Subdevice #34: subdevice #34
>>  Subdevice #35: subdevice #35
>>  Subdevice #36: subdevice #36
>>  Subdevice #37: subdevice #37
>>  Subdevice #38: subdevice #38
>>  Subdevice #39: subdevice #39
>>  Subdevice #40: subdevice #40
>>  Subdevice #41: subdevice #41
>>  Subdevice #42: subdevice #42
>>  Subdevice #43: subdevice #43
>>  Subdevice #44: subdevice #44
>>  Subdevice #45: subdevice #45
>>  Subdevice #46: subdevice #46
>>  Subdevice #47: subdevice #47
>>  Subdevice #48: subdevice #48
>>  Subdevice #49: subdevice #49
>>  Subdevice #50: subdevice #50
>>  Subdevice #51: subdevice #51
>>  Subdevice #52: subdevice #52
>>  Subdevice #53: subdevice #53
>>  Subdevice #54: subdevice #54
>>  Subdevice #55: subdevice #55
>>  Subdevice #56: subdevice #56
>>  Subdevice #57: subdevice #57
>>  Subdevice #58: subdevice #58
>>  Subdevice #59: subdevice #59
>>  Subdevice #60: subdevice #60
>>  Subdevice #61: subdevice #61
>>  Subdevice #62: subdevice #62
>>  Subdevice #63: subdevice #63
>>  Subdevice #64: subdevice #64
>>  Subdevice #65: subdevice #65
>>  Subdevice #66: subdevice #66
>>  Subdevice #67: subdevice #67
>>  Subdevice #68: subdevice #68
>>  Subdevice #69: subdevice #69
>>  Subdevice #70: subdevice #70
>>  Subdevice #71: subdevice #71
>>  Subdevice #72: subdevice #72
>>  Subdevice #73: subdevice #73
>>  Subdevice #74: subdevice #74
>>  Subdevice #75: subdevice #75
>>  Subdevice #76: subdevice #76
>>  Subdevice #77: subdevice #77
>>  Subdevice #78: subdevice #78
>>  Subdevice #79: subdevice #79
>>  Subdevice #80: subdevice #80
>>  Subdevice #81: subdevice #81
>>  Subdevice #82: subdevice #82
>>  Subdevice #83: subdevice #83
>>  Subdevice #84: subdevice #84
>>  Subdevice #85: subdevice #85
>>  Subdevice #86: subdevice #86
>>  Subdevice #87: subdevice #87
>>  Subdevice #88: subdevice #88
>>  Subdevice #89: subdevice #89
>>  Subdevice #90: subdevice #90
>>  Subdevice #91: subdevice #91
>>  Subdevice #92: subdevice #92
>>  Subdevice #93: subdevice #93
>>  Subdevice #94: subdevice #94
>>  Subdevice #95: subdevice #95
>>  Subdevice #96: subdevice #96
>>  Subdevice #97: subdevice #97
>>  Subdevice #98: subdevice #98
>>  Subdevice #99: subdevice #99
>>  Subdevice #100: subdevice #100
>>  Subdevice #101: subdevice #101
>>  Subdevice #102: subdevice #102
>>  Subdevice #103: subdevice #103
>>  Subdevice #104: subdevice #104
>>  Subdevice #105: subdevice #105
>>  Subdevice #106: subdevice #106
>>  Subdevice #107: subdevice #107
>>  Subdevice #108: subdevice #108
>>  Subdevice #109: subdevice #109
>>  Subdevice #110: subdevice #110
>>  Subdevice #111: subdevice #111
>>  Subdevice #112: subdevice #112
>>  Subdevice #113: subdevice #113
>>  Subdevice #114: subdevice #114
>>  Subdevice #115: subdevice #115
>>  Subdevice #116: subdevice #116
>>  Subdevice #117: subdevice #117
>>  Subdevice #118: subdevice #118
>>  Subdevice #119: subdevice #119
>>  Subdevice #120: subdevice #120
>>  Subdevice #121: subdevice #121
>>  Subdevice #122: subdevice #122
>>  Subdevice #123: subdevice #123
>>  Subdevice #124: subdevice #124
>>  Subdevice #125: subdevice #125
>>  Subdevice #126: subdevice #126
>>  Subdevice #127: subdevice #127
>>
>>
>> Peter P GMX schrieb:
>>     
>>> Her's the output:
>>>
>>> skype:~# aplay -l
>>> bash: aplay: command not found
>>>
>>> Giovanni Maruzzelli schrieb:
>>>
>>>       
>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>> what's the output of:
>>>>
>>>> aplay -l
>>>>
>>>> ?
>>>>
>>>> If instead you are referring to the choices that skype clients offers
>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>
>>>> Eg: not the "default", but the "hardware" one
>>>>
>>>>
>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>
>>>>
>>>>         
>>>>> Thanks Giovanni,
>>>>>
>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>> since I compiled alsa newly)
>>>>>
>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>>
>>>>> Giovanni Maruzzelli schrieb:
>>>>>
>>>>>
>>>>>           
>>>>>> This warning is harmless:
>>>>>>
>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>
>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>
>>>>>>
>>>>>>
>>>>>>             
>>>>>>> Ciao Peter
>>>>>>>
>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>> snd-dummy-customized.
>>>>>>>
>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>> 8.04).
>>>>>>>
>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>
>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>> do this with the modprobe command, as in:
>>>>>>>
>>>>>>> rmmod snd-dummy
>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>
>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>> soundcard.
>>>>>>>
>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>> just committed some improvements.
>>>>>>>
>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>> write the mailing list again,
>>>>>>>
>>>>>>> ciao for now,
>>>>>>>
>>>>>>> -giovanni
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>> to all contributors, excellent work!
>>>>>>>>
>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>> instructions in
>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>> ist Debian 5.0R3)
>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>> one-way-audio and my logs show
>>>>>>>>
>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>> suppressed
>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>> suppressed
>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>> with the current limitations).
>>>>>>>>
>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>
>>>>>>>> Second question:
>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>
>>>>>>>> Best regards
>>>>>>>> Peter
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>> --
>>>>>>> Sincerely,
>>>>>>>
>>>>>>> Giovanni Maruzzelli
>>>>>>> Cell : +39-347-2665618
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>             
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>           
>>>>
>>>>         
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   



From mike at van.lammeren.net  Thu Jan 28 12:05:11 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 28 Jan 2010 15:05:11 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
Message-ID: <5d2828f1001281205u2eccc999re6f25a55b07156c7@mail.gmail.com>

And you can make queries against your MySQL database, and get results, etc.?

On Thu, Jan 28, 2010 at 2:43 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello,
>
> That works fine:
>
> box:~# lua testdb.lua
> box:~#
>
>
> David
>
> On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
> <mike at van.lammeren.net> wrote:
> > Have you tried running a Lua script that includes the library from
> outside
> > of FreeSWITCH? What does that do?
> >
> > On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com>
> wrote:
> >>
> >> I tried running ldconfig on the directory containing mysql.so, but it
> did
> >> not help.
> >> So it sounds like there could be a bug in the latter versions?
> >>
> >> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
> >> <david.villasmil.work at gmail.com> wrote:
> >>>
> >>> I got the same error, my script was working with no problems before an
> >>> update to trunk.
> >>>
> >>> David
> >>>
> >>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com>
> wrote:
> >>> > Hi, I followed the instructions in the Lua documentation for setting
> up
> >>> > luasql, but when I try to run my script I get:
> >>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module
> >>> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
> >>> >         /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore
> segment
> >>> > prot
> >>> > after reloc: Permission denied
> >>> > stack traceback:
> >>> >         [C]: ?
> >>> >         [C]: in function 'require'
> >>> >         /usr/local/freeswitch/scripts/l.lua:2: in main chunk
> >>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
> >>> > I changed the permissions for mysql.so and for my script to 777, so
> I'm
> >>> > not
> >>> > sure where the permission problem could be.
> >>> > I'd appreciate any suggestions.
> >>> > Thanks,
> >>> > Adam
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at celliax.org  Thu Jan 28 12:10:51 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 28 Jan 2010 21:10:51 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B61ECE0.10409@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com> 
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com> 
	<4B6070A8.6050607@gmx.net>
	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com> 
	<4B607944.4040700@gmx.net> <4B608014.4030902@gmx.net>
	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com> 
	<4B61ECE0.10409@gmx.net>
Message-ID: <7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>

Ciao Peter,

Never tested on Debian 5.

When you write "same problem" you are referring to the audio going one
way only (btw, which way?) with the custom audio driver?

Have you tried with multiple instances of the regular Debian
snd-dummy, as I wrote in a mail before?

-gm



On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello Giovanni,
>
> I did so but the same problem again.
>
> Did you ever test in on Debian 5.0?
>
> Best reards
> Peter
>
> Giovanni Maruzzelli schrieb:
>> good, so you have only one sound device, the right one.
>>
>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>
>> -gm
>>
>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>
>>> I installed alsa-utile,
>>>
>>> now I get:
>>>
>>> skype:/var/cache/apt/archives# aplay -l
>>> **** List of PLAYBACK Hardware Devices ****
>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>> ?Subdevices: 127/128
>>> ?Subdevice #0: subdevice #0
>>> ?Subdevice #1: subdevice #1
>>> ?Subdevice #2: subdevice #2
>>> ?Subdevice #3: subdevice #3
>>> ?Subdevice #4: subdevice #4
>>> ?Subdevice #5: subdevice #5
>>> ?Subdevice #6: subdevice #6
>>> ?Subdevice #7: subdevice #7
>>> ?Subdevice #8: subdevice #8
>>> ?Subdevice #9: subdevice #9
>>> ?Subdevice #10: subdevice #10
>>> ?Subdevice #11: subdevice #11
>>> ?Subdevice #12: subdevice #12
>>> ?Subdevice #13: subdevice #13
>>> ?Subdevice #14: subdevice #14
>>> ?Subdevice #15: subdevice #15
>>> ?Subdevice #16: subdevice #16
>>> ?Subdevice #17: subdevice #17
>>> ?Subdevice #18: subdevice #18
>>> ?Subdevice #19: subdevice #19
>>> ?Subdevice #20: subdevice #20
>>> ?Subdevice #21: subdevice #21
>>> ?Subdevice #22: subdevice #22
>>> ?Subdevice #23: subdevice #23
>>> ?Subdevice #24: subdevice #24
>>> ?Subdevice #25: subdevice #25
>>> ?Subdevice #26: subdevice #26
>>> ?Subdevice #27: subdevice #27
>>> ?Subdevice #28: subdevice #28
>>> ?Subdevice #29: subdevice #29
>>> ?Subdevice #30: subdevice #30
>>> ?Subdevice #31: subdevice #31
>>> ?Subdevice #32: subdevice #32
>>> ?Subdevice #33: subdevice #33
>>> ?Subdevice #34: subdevice #34
>>> ?Subdevice #35: subdevice #35
>>> ?Subdevice #36: subdevice #36
>>> ?Subdevice #37: subdevice #37
>>> ?Subdevice #38: subdevice #38
>>> ?Subdevice #39: subdevice #39
>>> ?Subdevice #40: subdevice #40
>>> ?Subdevice #41: subdevice #41
>>> ?Subdevice #42: subdevice #42
>>> ?Subdevice #43: subdevice #43
>>> ?Subdevice #44: subdevice #44
>>> ?Subdevice #45: subdevice #45
>>> ?Subdevice #46: subdevice #46
>>> ?Subdevice #47: subdevice #47
>>> ?Subdevice #48: subdevice #48
>>> ?Subdevice #49: subdevice #49
>>> ?Subdevice #50: subdevice #50
>>> ?Subdevice #51: subdevice #51
>>> ?Subdevice #52: subdevice #52
>>> ?Subdevice #53: subdevice #53
>>> ?Subdevice #54: subdevice #54
>>> ?Subdevice #55: subdevice #55
>>> ?Subdevice #56: subdevice #56
>>> ?Subdevice #57: subdevice #57
>>> ?Subdevice #58: subdevice #58
>>> ?Subdevice #59: subdevice #59
>>> ?Subdevice #60: subdevice #60
>>> ?Subdevice #61: subdevice #61
>>> ?Subdevice #62: subdevice #62
>>> ?Subdevice #63: subdevice #63
>>> ?Subdevice #64: subdevice #64
>>> ?Subdevice #65: subdevice #65
>>> ?Subdevice #66: subdevice #66
>>> ?Subdevice #67: subdevice #67
>>> ?Subdevice #68: subdevice #68
>>> ?Subdevice #69: subdevice #69
>>> ?Subdevice #70: subdevice #70
>>> ?Subdevice #71: subdevice #71
>>> ?Subdevice #72: subdevice #72
>>> ?Subdevice #73: subdevice #73
>>> ?Subdevice #74: subdevice #74
>>> ?Subdevice #75: subdevice #75
>>> ?Subdevice #76: subdevice #76
>>> ?Subdevice #77: subdevice #77
>>> ?Subdevice #78: subdevice #78
>>> ?Subdevice #79: subdevice #79
>>> ?Subdevice #80: subdevice #80
>>> ?Subdevice #81: subdevice #81
>>> ?Subdevice #82: subdevice #82
>>> ?Subdevice #83: subdevice #83
>>> ?Subdevice #84: subdevice #84
>>> ?Subdevice #85: subdevice #85
>>> ?Subdevice #86: subdevice #86
>>> ?Subdevice #87: subdevice #87
>>> ?Subdevice #88: subdevice #88
>>> ?Subdevice #89: subdevice #89
>>> ?Subdevice #90: subdevice #90
>>> ?Subdevice #91: subdevice #91
>>> ?Subdevice #92: subdevice #92
>>> ?Subdevice #93: subdevice #93
>>> ?Subdevice #94: subdevice #94
>>> ?Subdevice #95: subdevice #95
>>> ?Subdevice #96: subdevice #96
>>> ?Subdevice #97: subdevice #97
>>> ?Subdevice #98: subdevice #98
>>> ?Subdevice #99: subdevice #99
>>> ?Subdevice #100: subdevice #100
>>> ?Subdevice #101: subdevice #101
>>> ?Subdevice #102: subdevice #102
>>> ?Subdevice #103: subdevice #103
>>> ?Subdevice #104: subdevice #104
>>> ?Subdevice #105: subdevice #105
>>> ?Subdevice #106: subdevice #106
>>> ?Subdevice #107: subdevice #107
>>> ?Subdevice #108: subdevice #108
>>> ?Subdevice #109: subdevice #109
>>> ?Subdevice #110: subdevice #110
>>> ?Subdevice #111: subdevice #111
>>> ?Subdevice #112: subdevice #112
>>> ?Subdevice #113: subdevice #113
>>> ?Subdevice #114: subdevice #114
>>> ?Subdevice #115: subdevice #115
>>> ?Subdevice #116: subdevice #116
>>> ?Subdevice #117: subdevice #117
>>> ?Subdevice #118: subdevice #118
>>> ?Subdevice #119: subdevice #119
>>> ?Subdevice #120: subdevice #120
>>> ?Subdevice #121: subdevice #121
>>> ?Subdevice #122: subdevice #122
>>> ?Subdevice #123: subdevice #123
>>> ?Subdevice #124: subdevice #124
>>> ?Subdevice #125: subdevice #125
>>> ?Subdevice #126: subdevice #126
>>> ?Subdevice #127: subdevice #127
>>>
>>>
>>> Peter P GMX schrieb:
>>>
>>>> Her's the output:
>>>>
>>>> skype:~# aplay -l
>>>> bash: aplay: command not found
>>>>
>>>> Giovanni Maruzzelli schrieb:
>>>>
>>>>
>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>> what's the output of:
>>>>>
>>>>> aplay -l
>>>>>
>>>>> ?
>>>>>
>>>>> If instead you are referring to the choices that skype clients offers
>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>
>>>>> Eg: not the "default", but the "hardware" one
>>>>>
>>>>>
>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>
>>>>>
>>>>>
>>>>>> Thanks Giovanni,
>>>>>>
>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>> since I compiled alsa newly)
>>>>>>
>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>
>>>>>> Best regards
>>>>>> Peter
>>>>>>
>>>>>>
>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>
>>>>>>
>>>>>>
>>>>>>> This warning is harmless:
>>>>>>>
>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>
>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> Ciao Peter
>>>>>>>>
>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>> snd-dummy-customized.
>>>>>>>>
>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>> 8.04).
>>>>>>>>
>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>
>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>
>>>>>>>> rmmod snd-dummy
>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>
>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>> soundcard.
>>>>>>>>
>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>> just committed some improvements.
>>>>>>>>
>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>> write the mailing list again,
>>>>>>>>
>>>>>>>> ciao for now,
>>>>>>>>
>>>>>>>> -giovanni
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>
>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>> instructions in
>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>
>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>> suppressed
>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>> suppressed
>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>> with the current limitations).
>>>>>>>>>
>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>
>>>>>>>>> Second question:
>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>
>>>>>>>>> Best regards
>>>>>>>>> Peter
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>> --
>>>>>>>> Sincerely,
>>>>>>>>
>>>>>>>> Giovanni Maruzzelli
>>>>>>>> Cell : +39-347-2665618
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From robert.hadley at teotech.com  Thu Jan 28 12:12:18 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 28 Jan 2010 12:12:18 -0800
Subject: [Freeswitch-users] Voicemail in MP3
In-Reply-To: <4B61EB1F.2070105@acsol.net>
References: <4B61DAC4.2030301@acsol.net><63E67C0D94CA4805822B2126EBA7D483@greyhawk.tonecommander.com>
	<4B61EB1F.2070105@acsol.net>
Message-ID: <9C598BC10A7142BF8791A37C7ABFFF38@greyhawk.tonecommander.com>


No, I was using the default dialplan.  If 1000 is calling 1007 and leaving
1007 a voicmail, add the <variable> name="vm_message_ext" value="mp3"/> to
the conf/directory/default/1000.xml.  All of the other email "params"
changes should be made to 1007.xml.  Another email suggested that the caller
vs. callee behavior may be fixed in future.

-Robert

-----Original Message-----
From: John [mailto:john at acsol.net] 
Sent: Thursday, January 28, 2010 11:53 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Voicemail in MP3

Would I setup the vm_message_ext in the dialplan then? Thanks

On 1/28/2010 12:24 PM, Robert Hadley wrote:
> A couple of things to check:
>
> 1. Make sure to enable mod_shout in the runtime installation folder:
>     e.g. /usr/local/freeswitch/conf/autoload_conf/modules.conf.xml
> 2. I found the variable vm_message_ext to apply to sent emails, not
> received.  For an internal extensions test set this variable for both
ends.
>
> -RobertH
>
> -----Original Message-----
> From: John [mailto:john at acsol.net]
> Sent: Thursday, January 28, 2010 10:43 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Voicemail in MP3
>
> I have installed mod_shout and edited the modules.conf.xml to installed
> it as well. I have updated the user.xml to include<variable
> name="vm_message_ext" value="mp3"/>.... Messages are still being saved
> in WAV format. Missing Step? Thanks
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    







From john at acsol.net  Thu Jan 28 12:16:59 2010
From: john at acsol.net (John)
Date: Thu, 28 Jan 2010 13:16:59 -0700
Subject: [Freeswitch-users] Voicemail in MP3
In-Reply-To: <63E67C0D94CA4805822B2126EBA7D483@greyhawk.tonecommander.com>
References: <4B61DAC4.2030301@acsol.net>
	<63E67C0D94CA4805822B2126EBA7D483@greyhawk.tonecommander.com>
Message-ID: <4B61F0BB.4070904@acsol.net>

Robert - You were correct, I added the vm_message_ext to the sending 
extension and now it works.
Thanks
On 1/28/2010 12:24 PM, Robert Hadley wrote:
> A couple of things to check:
>
> 1. Make sure to enable mod_shout in the runtime installation folder:
>     e.g. /usr/local/freeswitch/conf/autoload_conf/modules.conf.xml
> 2. I found the variable vm_message_ext to apply to sent emails, not
> received.  For an internal extensions test set this variable for both ends.
>
> -RobertH
>
> -----Original Message-----
> From: John [mailto:john at acsol.net]
> Sent: Thursday, January 28, 2010 10:43 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Voicemail in MP3
>
> I have installed mod_shout and edited the modules.conf.xml to installed
> it as well. I have updated the user.xml to include<variable
> name="vm_message_ext" value="mp3"/>.... Messages are still being saved
> in WAV format. Missing Step? Thanks
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    




From Prometheus001 at gmx.net  Thu Jan 28 13:07:55 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 28 Jan 2010 22:07:55 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001270726o60a5bf32pdb3e5ccedb25375e@mail.gmail.com>
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com>
	<4B6070A8.6050607@gmx.net>	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com>
	<4B607944.4040700@gmx.net>
	<4B608014.4030902@gmx.net>	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>
	<4B61ECE0.10409@gmx.net>
	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>
Message-ID: <4B61FCAB.5040707@gmx.net>

I crated 3 instances of snd-dummy, this worked. I assigned then Instance
#2 to the Skype accounts. Still no sound.
On the frist call there is one way audio, on the following calls there
is no audio at all.
This is weird.

Best regards
Peter

Giovanni Maruzzelli schrieb:
> Ciao Peter,
>
> Never tested on Debian 5.
>
> When you write "same problem" you are referring to the audio going one
> way only (btw, which way?) with the custom audio driver?
>
> Have you tried with multiple instances of the regular Debian
> snd-dummy, as I wrote in a mail before?
>
> -gm
>
>
>
> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> Hello Giovanni,
>>
>> I did so but the same problem again.
>>
>> Did you ever test in on Debian 5.0?
>>
>> Best reards
>> Peter
>>
>> Giovanni Maruzzelli schrieb:
>>     
>>> good, so you have only one sound device, the right one.
>>>
>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>
>>> -gm
>>>
>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>       
>>>> I installed alsa-utile,
>>>>
>>>> now I get:
>>>>
>>>> skype:/var/cache/apt/archives# aplay -l
>>>> **** List of PLAYBACK Hardware Devices ****
>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>  Subdevices: 127/128
>>>>  Subdevice #0: subdevice #0
>>>>  Subdevice #1: subdevice #1
>>>>  Subdevice #2: subdevice #2
>>>>  Subdevice #3: subdevice #3
>>>>  Subdevice #4: subdevice #4
>>>>  Subdevice #5: subdevice #5
>>>>  Subdevice #6: subdevice #6
>>>>  Subdevice #7: subdevice #7
>>>>  Subdevice #8: subdevice #8
>>>>  Subdevice #9: subdevice #9
>>>>  Subdevice #10: subdevice #10
>>>>  Subdevice #11: subdevice #11
>>>>  Subdevice #12: subdevice #12
>>>>  Subdevice #13: subdevice #13
>>>>  Subdevice #14: subdevice #14
>>>>  Subdevice #15: subdevice #15
>>>>  Subdevice #16: subdevice #16
>>>>  Subdevice #17: subdevice #17
>>>>  Subdevice #18: subdevice #18
>>>>  Subdevice #19: subdevice #19
>>>>  Subdevice #20: subdevice #20
>>>>  Subdevice #21: subdevice #21
>>>>  Subdevice #22: subdevice #22
>>>>  Subdevice #23: subdevice #23
>>>>  Subdevice #24: subdevice #24
>>>>  Subdevice #25: subdevice #25
>>>>  Subdevice #26: subdevice #26
>>>>  Subdevice #27: subdevice #27
>>>>  Subdevice #28: subdevice #28
>>>>  Subdevice #29: subdevice #29
>>>>  Subdevice #30: subdevice #30
>>>>  Subdevice #31: subdevice #31
>>>>  Subdevice #32: subdevice #32
>>>>  Subdevice #33: subdevice #33
>>>>  Subdevice #34: subdevice #34
>>>>  Subdevice #35: subdevice #35
>>>>  Subdevice #36: subdevice #36
>>>>  Subdevice #37: subdevice #37
>>>>  Subdevice #38: subdevice #38
>>>>  Subdevice #39: subdevice #39
>>>>  Subdevice #40: subdevice #40
>>>>  Subdevice #41: subdevice #41
>>>>  Subdevice #42: subdevice #42
>>>>  Subdevice #43: subdevice #43
>>>>  Subdevice #44: subdevice #44
>>>>  Subdevice #45: subdevice #45
>>>>  Subdevice #46: subdevice #46
>>>>  Subdevice #47: subdevice #47
>>>>  Subdevice #48: subdevice #48
>>>>  Subdevice #49: subdevice #49
>>>>  Subdevice #50: subdevice #50
>>>>  Subdevice #51: subdevice #51
>>>>  Subdevice #52: subdevice #52
>>>>  Subdevice #53: subdevice #53
>>>>  Subdevice #54: subdevice #54
>>>>  Subdevice #55: subdevice #55
>>>>  Subdevice #56: subdevice #56
>>>>  Subdevice #57: subdevice #57
>>>>  Subdevice #58: subdevice #58
>>>>  Subdevice #59: subdevice #59
>>>>  Subdevice #60: subdevice #60
>>>>  Subdevice #61: subdevice #61
>>>>  Subdevice #62: subdevice #62
>>>>  Subdevice #63: subdevice #63
>>>>  Subdevice #64: subdevice #64
>>>>  Subdevice #65: subdevice #65
>>>>  Subdevice #66: subdevice #66
>>>>  Subdevice #67: subdevice #67
>>>>  Subdevice #68: subdevice #68
>>>>  Subdevice #69: subdevice #69
>>>>  Subdevice #70: subdevice #70
>>>>  Subdevice #71: subdevice #71
>>>>  Subdevice #72: subdevice #72
>>>>  Subdevice #73: subdevice #73
>>>>  Subdevice #74: subdevice #74
>>>>  Subdevice #75: subdevice #75
>>>>  Subdevice #76: subdevice #76
>>>>  Subdevice #77: subdevice #77
>>>>  Subdevice #78: subdevice #78
>>>>  Subdevice #79: subdevice #79
>>>>  Subdevice #80: subdevice #80
>>>>  Subdevice #81: subdevice #81
>>>>  Subdevice #82: subdevice #82
>>>>  Subdevice #83: subdevice #83
>>>>  Subdevice #84: subdevice #84
>>>>  Subdevice #85: subdevice #85
>>>>  Subdevice #86: subdevice #86
>>>>  Subdevice #87: subdevice #87
>>>>  Subdevice #88: subdevice #88
>>>>  Subdevice #89: subdevice #89
>>>>  Subdevice #90: subdevice #90
>>>>  Subdevice #91: subdevice #91
>>>>  Subdevice #92: subdevice #92
>>>>  Subdevice #93: subdevice #93
>>>>  Subdevice #94: subdevice #94
>>>>  Subdevice #95: subdevice #95
>>>>  Subdevice #96: subdevice #96
>>>>  Subdevice #97: subdevice #97
>>>>  Subdevice #98: subdevice #98
>>>>  Subdevice #99: subdevice #99
>>>>  Subdevice #100: subdevice #100
>>>>  Subdevice #101: subdevice #101
>>>>  Subdevice #102: subdevice #102
>>>>  Subdevice #103: subdevice #103
>>>>  Subdevice #104: subdevice #104
>>>>  Subdevice #105: subdevice #105
>>>>  Subdevice #106: subdevice #106
>>>>  Subdevice #107: subdevice #107
>>>>  Subdevice #108: subdevice #108
>>>>  Subdevice #109: subdevice #109
>>>>  Subdevice #110: subdevice #110
>>>>  Subdevice #111: subdevice #111
>>>>  Subdevice #112: subdevice #112
>>>>  Subdevice #113: subdevice #113
>>>>  Subdevice #114: subdevice #114
>>>>  Subdevice #115: subdevice #115
>>>>  Subdevice #116: subdevice #116
>>>>  Subdevice #117: subdevice #117
>>>>  Subdevice #118: subdevice #118
>>>>  Subdevice #119: subdevice #119
>>>>  Subdevice #120: subdevice #120
>>>>  Subdevice #121: subdevice #121
>>>>  Subdevice #122: subdevice #122
>>>>  Subdevice #123: subdevice #123
>>>>  Subdevice #124: subdevice #124
>>>>  Subdevice #125: subdevice #125
>>>>  Subdevice #126: subdevice #126
>>>>  Subdevice #127: subdevice #127
>>>>
>>>>
>>>> Peter P GMX schrieb:
>>>>
>>>>         
>>>>> Her's the output:
>>>>>
>>>>> skype:~# aplay -l
>>>>> bash: aplay: command not found
>>>>>
>>>>> Giovanni Maruzzelli schrieb:
>>>>>
>>>>>
>>>>>           
>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>> what's the output of:
>>>>>>
>>>>>> aplay -l
>>>>>>
>>>>>> ?
>>>>>>
>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>
>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>
>>>>>>
>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>
>>>>>>
>>>>>>
>>>>>>             
>>>>>>> Thanks Giovanni,
>>>>>>>
>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>> since I compiled alsa newly)
>>>>>>>
>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>
>>>>>>> Best regards
>>>>>>> Peter
>>>>>>>
>>>>>>>
>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>>> This warning is harmless:
>>>>>>>>
>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>
>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>>>> Ciao Peter
>>>>>>>>>
>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>> snd-dummy-customized.
>>>>>>>>>
>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>> 8.04).
>>>>>>>>>
>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>
>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>
>>>>>>>>> rmmod snd-dummy
>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>
>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>> soundcard.
>>>>>>>>>
>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>> just committed some improvements.
>>>>>>>>>
>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>> write the mailing list again,
>>>>>>>>>
>>>>>>>>> ciao for now,
>>>>>>>>>
>>>>>>>>> -giovanni
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>
>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>> instructions in
>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>
>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>> suppressed
>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>> suppressed
>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>> with the current limitations).
>>>>>>>>>>
>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>
>>>>>>>>>> Second question:
>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>
>>>>>>>>>> Best regards
>>>>>>>>>> Peter
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>                     
>>>>>>>>> --
>>>>>>>>> Sincerely,
>>>>>>>>>
>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>             
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   



From fvillarroel at yahoo.com  Thu Jan 28 13:14:37 2010
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Thu, 28 Jan 2010 13:14:37 -0800 (PST)
Subject: [Freeswitch-users] prefix on exten
Message-ID: <968062.52038.qm@web34307.mail.mud.yahoo.com>

Dear.

If i receive a call from a customer with some prefx like 1234

How i can do in order to forward this call with out prefix like Asterisk

{ENTEN:4}

Regards.


      



From gmaruzz at celliax.org  Thu Jan 28 13:41:16 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 28 Jan 2010 22:41:16 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B61FCAB.5040707@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com> 
	<4B6070A8.6050607@gmx.net>
	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com> 
	<4B607944.4040700@gmx.net> <4B608014.4030902@gmx.net>
	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com> 
	<4B61ECE0.10409@gmx.net>
	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com> 
	<4B61FCAB.5040707@gmx.net>
Message-ID: <7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>

with three instances you will assign the hw:0 device to skype client
0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
Must work. Pay attention to assign the same device name to all devices
needed by a skype instance (sound devices window): playback, capture
AND ring.

Or maybe is a bug of ALSA on Debian...

-giovanni

On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
> #2 to the Skype accounts. Still no sound.
> On the frist call there is one way audio, on the following calls there
> is no audio at all.
> This is weird.
>
> Best regards
> Peter
>
> Giovanni Maruzzelli schrieb:
>> Ciao Peter,
>>
>> Never tested on Debian 5.
>>
>> When you write "same problem" you are referring to the audio going one
>> way only (btw, which way?) with the custom audio driver?
>>
>> Have you tried with multiple instances of the regular Debian
>> snd-dummy, as I wrote in a mail before?
>>
>> -gm
>>
>>
>>
>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>
>>> Hello Giovanni,
>>>
>>> I did so but the same problem again.
>>>
>>> Did you ever test in on Debian 5.0?
>>>
>>> Best reards
>>> Peter
>>>
>>> Giovanni Maruzzelli schrieb:
>>>
>>>> good, so you have only one sound device, the right one.
>>>>
>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>
>>>> -gm
>>>>
>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>
>>>>
>>>>> I installed alsa-utile,
>>>>>
>>>>> now I get:
>>>>>
>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>> ?Subdevices: 127/128
>>>>> ?Subdevice #0: subdevice #0
>>>>> ?Subdevice #1: subdevice #1
>>>>> ?Subdevice #2: subdevice #2
>>>>> ?Subdevice #3: subdevice #3
>>>>> ?Subdevice #4: subdevice #4
>>>>> ?Subdevice #5: subdevice #5
>>>>> ?Subdevice #6: subdevice #6
>>>>> ?Subdevice #7: subdevice #7
>>>>> ?Subdevice #8: subdevice #8
>>>>> ?Subdevice #9: subdevice #9
>>>>> ?Subdevice #10: subdevice #10
>>>>> ?Subdevice #11: subdevice #11
>>>>> ?Subdevice #12: subdevice #12
>>>>> ?Subdevice #13: subdevice #13
>>>>> ?Subdevice #14: subdevice #14
>>>>> ?Subdevice #15: subdevice #15
>>>>> ?Subdevice #16: subdevice #16
>>>>> ?Subdevice #17: subdevice #17
>>>>> ?Subdevice #18: subdevice #18
>>>>> ?Subdevice #19: subdevice #19
>>>>> ?Subdevice #20: subdevice #20
>>>>> ?Subdevice #21: subdevice #21
>>>>> ?Subdevice #22: subdevice #22
>>>>> ?Subdevice #23: subdevice #23
>>>>> ?Subdevice #24: subdevice #24
>>>>> ?Subdevice #25: subdevice #25
>>>>> ?Subdevice #26: subdevice #26
>>>>> ?Subdevice #27: subdevice #27
>>>>> ?Subdevice #28: subdevice #28
>>>>> ?Subdevice #29: subdevice #29
>>>>> ?Subdevice #30: subdevice #30
>>>>> ?Subdevice #31: subdevice #31
>>>>> ?Subdevice #32: subdevice #32
>>>>> ?Subdevice #33: subdevice #33
>>>>> ?Subdevice #34: subdevice #34
>>>>> ?Subdevice #35: subdevice #35
>>>>> ?Subdevice #36: subdevice #36
>>>>> ?Subdevice #37: subdevice #37
>>>>> ?Subdevice #38: subdevice #38
>>>>> ?Subdevice #39: subdevice #39
>>>>> ?Subdevice #40: subdevice #40
>>>>> ?Subdevice #41: subdevice #41
>>>>> ?Subdevice #42: subdevice #42
>>>>> ?Subdevice #43: subdevice #43
>>>>> ?Subdevice #44: subdevice #44
>>>>> ?Subdevice #45: subdevice #45
>>>>> ?Subdevice #46: subdevice #46
>>>>> ?Subdevice #47: subdevice #47
>>>>> ?Subdevice #48: subdevice #48
>>>>> ?Subdevice #49: subdevice #49
>>>>> ?Subdevice #50: subdevice #50
>>>>> ?Subdevice #51: subdevice #51
>>>>> ?Subdevice #52: subdevice #52
>>>>> ?Subdevice #53: subdevice #53
>>>>> ?Subdevice #54: subdevice #54
>>>>> ?Subdevice #55: subdevice #55
>>>>> ?Subdevice #56: subdevice #56
>>>>> ?Subdevice #57: subdevice #57
>>>>> ?Subdevice #58: subdevice #58
>>>>> ?Subdevice #59: subdevice #59
>>>>> ?Subdevice #60: subdevice #60
>>>>> ?Subdevice #61: subdevice #61
>>>>> ?Subdevice #62: subdevice #62
>>>>> ?Subdevice #63: subdevice #63
>>>>> ?Subdevice #64: subdevice #64
>>>>> ?Subdevice #65: subdevice #65
>>>>> ?Subdevice #66: subdevice #66
>>>>> ?Subdevice #67: subdevice #67
>>>>> ?Subdevice #68: subdevice #68
>>>>> ?Subdevice #69: subdevice #69
>>>>> ?Subdevice #70: subdevice #70
>>>>> ?Subdevice #71: subdevice #71
>>>>> ?Subdevice #72: subdevice #72
>>>>> ?Subdevice #73: subdevice #73
>>>>> ?Subdevice #74: subdevice #74
>>>>> ?Subdevice #75: subdevice #75
>>>>> ?Subdevice #76: subdevice #76
>>>>> ?Subdevice #77: subdevice #77
>>>>> ?Subdevice #78: subdevice #78
>>>>> ?Subdevice #79: subdevice #79
>>>>> ?Subdevice #80: subdevice #80
>>>>> ?Subdevice #81: subdevice #81
>>>>> ?Subdevice #82: subdevice #82
>>>>> ?Subdevice #83: subdevice #83
>>>>> ?Subdevice #84: subdevice #84
>>>>> ?Subdevice #85: subdevice #85
>>>>> ?Subdevice #86: subdevice #86
>>>>> ?Subdevice #87: subdevice #87
>>>>> ?Subdevice #88: subdevice #88
>>>>> ?Subdevice #89: subdevice #89
>>>>> ?Subdevice #90: subdevice #90
>>>>> ?Subdevice #91: subdevice #91
>>>>> ?Subdevice #92: subdevice #92
>>>>> ?Subdevice #93: subdevice #93
>>>>> ?Subdevice #94: subdevice #94
>>>>> ?Subdevice #95: subdevice #95
>>>>> ?Subdevice #96: subdevice #96
>>>>> ?Subdevice #97: subdevice #97
>>>>> ?Subdevice #98: subdevice #98
>>>>> ?Subdevice #99: subdevice #99
>>>>> ?Subdevice #100: subdevice #100
>>>>> ?Subdevice #101: subdevice #101
>>>>> ?Subdevice #102: subdevice #102
>>>>> ?Subdevice #103: subdevice #103
>>>>> ?Subdevice #104: subdevice #104
>>>>> ?Subdevice #105: subdevice #105
>>>>> ?Subdevice #106: subdevice #106
>>>>> ?Subdevice #107: subdevice #107
>>>>> ?Subdevice #108: subdevice #108
>>>>> ?Subdevice #109: subdevice #109
>>>>> ?Subdevice #110: subdevice #110
>>>>> ?Subdevice #111: subdevice #111
>>>>> ?Subdevice #112: subdevice #112
>>>>> ?Subdevice #113: subdevice #113
>>>>> ?Subdevice #114: subdevice #114
>>>>> ?Subdevice #115: subdevice #115
>>>>> ?Subdevice #116: subdevice #116
>>>>> ?Subdevice #117: subdevice #117
>>>>> ?Subdevice #118: subdevice #118
>>>>> ?Subdevice #119: subdevice #119
>>>>> ?Subdevice #120: subdevice #120
>>>>> ?Subdevice #121: subdevice #121
>>>>> ?Subdevice #122: subdevice #122
>>>>> ?Subdevice #123: subdevice #123
>>>>> ?Subdevice #124: subdevice #124
>>>>> ?Subdevice #125: subdevice #125
>>>>> ?Subdevice #126: subdevice #126
>>>>> ?Subdevice #127: subdevice #127
>>>>>
>>>>>
>>>>> Peter P GMX schrieb:
>>>>>
>>>>>
>>>>>> Her's the output:
>>>>>>
>>>>>> skype:~# aplay -l
>>>>>> bash: aplay: command not found
>>>>>>
>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>
>>>>>>
>>>>>>
>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>> what's the output of:
>>>>>>>
>>>>>>> aplay -l
>>>>>>>
>>>>>>> ?
>>>>>>>
>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>
>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> Thanks Giovanni,
>>>>>>>>
>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>> since I compiled alsa newly)
>>>>>>>>
>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>
>>>>>>>> Best regards
>>>>>>>> Peter
>>>>>>>>
>>>>>>>>
>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>> This warning is harmless:
>>>>>>>>>
>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>
>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>> Ciao Peter
>>>>>>>>>>
>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>
>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>> 8.04).
>>>>>>>>>>
>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>
>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>
>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>
>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>> soundcard.
>>>>>>>>>>
>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>> just committed some improvements.
>>>>>>>>>>
>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>> write the mailing list again,
>>>>>>>>>>
>>>>>>>>>> ciao for now,
>>>>>>>>>>
>>>>>>>>>> -giovanni
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>
>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>> instructions in
>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>
>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>> suppressed
>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>> suppressed
>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>
>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>
>>>>>>>>>>> Second question:
>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>
>>>>>>>>>>> Best regards
>>>>>>>>>>> Peter
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>> Sincerely,
>>>>>>>>>>
>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From testeador01 at gmail.com  Thu Jan 28 13:51:31 2010
From: testeador01 at gmail.com (Milena)
Date: Thu, 28 Jan 2010 16:51:31 -0500
Subject: [Freeswitch-users] ERR root tag missing
Message-ID: <c29b930b1001281351r3bebc2f2ta0279a9bbd4b6c0e@mail.gmail.com>

Hello :)
First, do not hijack threads, if you want to post about a different problem,
do not click reply and then change the subject, please create a NEW MESSAGE.

About your question, are you using xml_curl? or any other dialplan seekers?

-Milena

2010/1/28 Troy Anderson <troy at tlainvestments.com>

> I'm seeing this error quite often on my systems:
>        2010-01-28 12:35:46.703112 [ERR] switch_xml.c:1571 Error[[error near
> line 1]: root tag missing]
>
> I've looked at freeswitch.xml.fsxml to see if I could find some kind of
> malformed XML, but with no luck.  Which Is line 1is it referring to?
>
> Line 1 of freeswitch.xml.fsxml is a comment, with the first actual XML line
> like:
>        <document type="freeswitch/xml">
>
> This error always happens right after after a mod_dialplan_xml.c:408 log
> message, so I'm led to believe my dialplan XML is messed up, but I cannot
> see where.
>
> In freeswitch.xml.fsxml near the dialplan section, this is what I have:
>
> ...
> <section name="configuration" description="Various Configuration">
>        <!-- CUT a bunch of stuff ... CUT -->
> </section>
> <section name="dialplan" description="Regex/XML Dialplan">
>        <context name="my_office">
>                <extension name="unloop">
>                        <condition field="${unroll_loops}"
> expression="^true$"/>
>                        <condition field="${sip_looped_call}"
> expression="^true$">
>                                <action application="deflect"
> data="${destination_number}"/>
>                        </condition>
>                </extension>
>                <extension name="set_domain" continue="true">
>                        <condition field="${domain_name}" expression="^$"/>
>                        <condition field="source" expression="mod_sofia"/>
>                        <condition field="${sip_auth_realm}"
> expression="^$">
>                                <action application="set"
> data="domain_name=10.0.0.120"/>
>                                <anti-action application="set"
> data="domain_name=${sip_auth_realm}"/>
>                        </condition>
>                </extension>
> ...
> </section>
> ...
>
> Thanks for any ideas!
>
> -Troy
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Prometheus001 at gmx.net  Thu Jan 28 13:54:19 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 28 Jan 2010 22:54:19 +0100
Subject: [Freeswitch-users] Set an gateway parameter for external gateway
	(skype)
Message-ID: <4B62078B.5060406@gmx.net>

Hello,

I have a main freeswitch server and a separate Freeswitch/Skype server
with mod_skypiax.

I want main freeswitch server to tell the Freeswitch/Skype server to use
a dedicated Skype interface(interface1, interface2 etc). What is the
best way to pass this variable?
Some ideas from my side

    * set caller_id_number or caller_id_number_name, as these are
      overwritten by Skype anyway. But this is not a clean solution
    * use another UDP port for each gateway, but this is a huge effort
      if the number of gateways becomes larger
    * set a dedicated sip header for this. Reading sip-header is easy
      (documented) but how to set my own sip header entry in FS?
    * any other idea?


Best regards
Peter




From anthony.minessale at gmail.com  Thu Jan 28 14:42:20 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 Jan 2010 16:42:20 -0600
Subject: [Freeswitch-users] Voicemail via web interface
In-Reply-To: <4B61EAE4.2070607@acsol.net>
References: <4B61ACB3.50903@acsol.net>
	<090DDB5F675E4FC48512CA3FC52FA096@greyhawk.tonecommander.com>
	<4B61EAE4.2070607@acsol.net>
Message-ID: <191c3a031001281442m2b74ebb5xffa19113f54f0948@mail.gmail.com>

yes sadly mp3 up sampled to 11khz is the only thing that works with that
flash player.


On Thu, Jan 28, 2010 at 1:52 PM, John <john at acsol.net> wrote:

> Thanks Robert. I believe the issue is probably because our files are in
> WAV format and not MP3.
>
> On 1/28/2010 12:28 PM, Robert Hadley wrote:
> > Using Firefox I was asked to install the latest Flash plugin and then I
> > could play the messages from the webpage directly.  IE8 never asked to
> add
> > the plugin that I noticed.
> > -RobertH
> >
> >
> > -----Original Message-----
> > From: John [mailto:john at acsol.net]
> > Sent: Thursday, January 28, 2010 7:27 AM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: [Freeswitch-users] Voicemail via web interface
> >
> > Hello,
> > Can you point me to any additional information about the voice mail via
> > web interface? I have it up and running; however if you click the play
> > button there is no playback, if you click download it will play in MS
> > media player. Thanks John
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Thu Jan 28 14:43:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 Jan 2010 16:43:50 -0600
Subject: [Freeswitch-users] prefix on exten
In-Reply-To: <968062.52038.qm@web34307.mail.mud.yahoo.com>
References: <968062.52038.qm@web34307.mail.mud.yahoo.com>
Message-ID: <191c3a031001281443y3e03ca6n5c8880b159ae57fc@mail.gmail.com>

1) capture it in the regex and put the () around the part without the
prefix.
2) do the same thing as asterisk with ${destination_number:4}



On Thu, Jan 28, 2010 at 3:14 PM, FERNANDO VILLARROEL
<fvillarroel at yahoo.com>wrote:

> Dear.
>
> If i receive a call from a customer with some prefx like 1234
>
> How i can do in order to forward this call with out prefix like Asterisk
>
> {ENTEN:4}
>
> Regards.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mustafa.pk at gmail.com  Thu Jan 28 15:21:33 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Fri, 29 Jan 2010 04:21:33 +0500
Subject: [Freeswitch-users] Set an gateway parameter for external
	gateway (skype)
In-Reply-To: <4B62078B.5060406@gmx.net>
References: <4B62078B.5060406@gmx.net>
Message-ID: <8213d6071001281521u27347956nf2c89c3cd2e72745@mail.gmail.com>

i don't know if it's a wise solution, but you can make a db insert
before bridge on freeswitch server, kypiax server can query db after
answer to get an idea!

On Fri, Jan 29, 2010 at 2:54 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello,
>
> I have a main freeswitch server and a separate Freeswitch/Skype server
> with mod_skypiax.
>
> I want main freeswitch server to tell the Freeswitch/Skype server to use
> a dedicated Skype interface(interface1, interface2 etc). What is the
> best way to pass this variable?
> Some ideas from my side
>
> ? ?* set caller_id_number or caller_id_number_name, as these are
> ? ? ?overwritten by Skype anyway. But this is not a clean solution
> ? ?* use another UDP port for each gateway, but this is a huge effort
> ? ? ?if the number of gateways becomes larger
> ? ?* set a dedicated sip header for this. Reading sip-header is easy
> ? ? ?(documented) but how to set my own sip header entry in FS?
> ? ?* any other idea?
>
>
> Best regards
> Peter
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com



From wasim at convergence.pk  Thu Jan 28 15:29:54 2010
From: wasim at convergence.pk (Wasim Baig)
Date: Fri, 29 Jan 2010 04:29:54 +0500
Subject: [Freeswitch-users] Set an gateway parameter for external
	gateway (skype)
In-Reply-To: <8213d6071001281521u27347956nf2c89c3cd2e72745@mail.gmail.com>
References: <4B62078B.5060406@gmx.net>
	<8213d6071001281521u27347956nf2c89c3cd2e72745@mail.gmail.com>
Message-ID: <b8ad2a5b1001281529n4f5ac2acy266ad324b0da9318@mail.gmail.com>

Setting a sip header is a more elegant way ...

http://wiki.freeswitch.org/wiki/Sofia#Adding_Request_Headers

-wasim

On Fri, Jan 29, 2010 at 4:21 AM, Ghulam Mustafa <mustafa.pk at gmail.com>wrote:

> i don't know if it's a wise solution, but you can make a db insert
> before bridge on freeswitch server, kypiax server can query db after
> answer to get an idea!
>
> On Fri, Jan 29, 2010 at 2:54 AM, Peter P GMX <Prometheus001 at gmx.net>
> wrote:
> > Hello,
> >
> > I have a main freeswitch server and a separate Freeswitch/Skype server
> > with mod_skypiax.
> >
> > I want main freeswitch server to tell the Freeswitch/Skype server to use
> > a dedicated Skype interface(interface1, interface2 etc). What is the
> > best way to pass this variable?
> > Some ideas from my side
> >
> >    * set caller_id_number or caller_id_number_name, as these are
> >      overwritten by Skype anyway. But this is not a clean solution
> >    * use another UDP port for each gateway, but this is a huge effort
> >      if the number of gateways becomes larger
> >    * set a dedicated sip header for this. Reading sip-header is easy
> >      (documented) but how to set my own sip header entry in FS?
> >    * any other idea?
> >
> >
> > Best regards
> > Peter
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
peace be upon you ...
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From emptysands at gmail.com  Wed Jan 27 21:54:03 2010
From: emptysands at gmail.com (Nicholas Lee)
Date: Thu, 28 Jan 2010 18:54:03 +1300
Subject: [Freeswitch-users] Hybrid Encryption?
Message-ID: <2b6116b31001272154l3c0bbe80y8bf3db94961e8e1d@mail.gmail.com>

The TLS wiki page talks about [1] Freeswitch being able to act as a past
though proxy for a SIP phone. Turning an unencrypted SIP+RTP session into a
SIPS+SRTP session.

Is there howto guide for the above?

I'm also wondering if Freeswitch could do a drop-in encryption.  ie. Phone
<-> Asterisk replaced with Phone <-> FS  <==> FS <-> Asterisk, where <==>
is encrypted.


Nicholas

[1] http://wiki.freeswitch.org/wiki/Tls#Hybrid_Encryption
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From bobc at panztel.biz  Thu Jan 28 14:28:30 2010
From: bobc at panztel.biz (Bob Coleman)
Date: Fri, 29 Jan 2010 11:28:30 +1300
Subject: [Freeswitch-users] prefix on exten
In-Reply-To: <968062.52038.qm@web34307.mail.mud.yahoo.com>
References: <968062.52038.qm@web34307.mail.mud.yahoo.com>
Message-ID: <8543d2b11001281428m737fb7bev23c7bfd802d2dfee@mail.gmail.com>

In your dialplan do something like this:

<include>
  <extension name="remotefs">
    <condition field="destination_number" expression="^1234(\d+)$">
     <action application="bridge" data="
sofia/internal/$1 at 192.168.50.195:5080"/<sofia/internal/$1 at 192.168.50.195:5080%22/>
>
    </condition>
  </extension>
</include>

The $1 variable has the number without the prefix of 1234.

I am sending the call to another FS box in this example

Bob

On Fri, Jan 29, 2010 at 10:14 AM, FERNANDO VILLARROEL <fvillarroel at yahoo.com
> wrote:

> Dear.
>
> If i receive a call from a customer with some prefx like 1234
>
> How i can do in order to forward this call with out prefix like Asterisk
>
> {ENTEN:4}
>
> Regards.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Thu Jan 28 16:19:14 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 28 Jan 2010 18:19:14 -0600
Subject: [Freeswitch-users] Hybrid Encryption?
In-Reply-To: <2b6116b31001272154l3c0bbe80y8bf3db94961e8e1d@mail.gmail.com>
References: <2b6116b31001272154l3c0bbe80y8bf3db94961e8e1d@mail.gmail.com>
Message-ID: <8A9EDC4E-C49B-488D-9DBF-169A185462AB@freeswitch.org>

Or you could do Phone <-> FS <=====> FS <-> Phone...

;)  Less complex.

/b

On Jan 27, 2010, at 11:54 PM, Nicholas Lee wrote:

> The TLS wiki page talks about [1] Freeswitch being able to act as a past though proxy for a SIP phone. Turning an unencrypted SIP+RTP session into a SIPS+SRTP session.
> 
> Is there howto guide for the above?
> 
> I'm also wondering if Freeswitch could do a drop-in encryption.  ie. Phone <-> Asterisk replaced with Phone <-> FS  <==> FS <-> Asterisk, where <==> is encrypted.
> 
> 
> Nicholas




From emptysands at gmail.com  Thu Jan 28 16:31:02 2010
From: emptysands at gmail.com (Nicholas Lee)
Date: Fri, 29 Jan 2010 08:31:02 +0800
Subject: [Freeswitch-users] Hybrid Encryption?
In-Reply-To: <8A9EDC4E-C49B-488D-9DBF-169A185462AB@freeswitch.org>
References: <2b6116b31001272154l3c0bbe80y8bf3db94961e8e1d@mail.gmail.com>
	<8A9EDC4E-C49B-488D-9DBF-169A185462AB@freeswitch.org>
Message-ID: <2b6116b31001281631u7ada7876wc419bb7afadd7ef7@mail.gmail.com>

Unfortunately it's not going to cover every situation.


Nicholas

On Fri, Jan 29, 2010 at 8:19 AM, Brian West <brian at freeswitch.org> wrote:

> Or you could do Phone <-> FS <=====> FS <-> Phone...
>
> ;)  Less complex.
>
> /b
>
> On Jan 27, 2010, at 11:54 PM, Nicholas Lee wrote:
>
> > The TLS wiki page talks about [1] Freeswitch being able to act as a past
> though proxy for a SIP phone. Turning an unencrypted SIP+RTP session into a
> SIPS+SRTP session.
> >
> > Is there howto guide for the above?
> >
> > I'm also wondering if Freeswitch could do a drop-in encryption.  ie.
> Phone <-> Asterisk replaced with Phone <-> FS  <==> FS <-> Asterisk, where
> <==> is encrypted.
> >
> >
> > Nicholas
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Thu Jan 28 16:52:39 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 28 Jan 2010 18:52:39 -0600
Subject: [Freeswitch-users] Hybrid Encryption?
In-Reply-To: <2b6116b31001281631u7ada7876wc419bb7afadd7ef7@mail.gmail.com>
References: <2b6116b31001272154l3c0bbe80y8bf3db94961e8e1d@mail.gmail.com>
	<8A9EDC4E-C49B-488D-9DBF-169A185462AB@freeswitch.org>
	<2b6116b31001281631u7ada7876wc419bb7afadd7ef7@mail.gmail.com>
Message-ID: <46C06209-9515-4B1B-B449-F55A51FF548B@freeswitch.org>

Then yes you could use FreeSWITCH to augment your Asterisk install and enable encryption from site to site.

/b


> Unfortunately it's not going to cover every situation.
> 
> 
> Nicholas




From dujinfang at gmail.com  Thu Jan 28 17:29:36 2010
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 29 Jan 2010 09:29:36 +0800
Subject: [Freeswitch-users] Set an gateway parameter for external
	gateway (skype)
In-Reply-To: <b8ad2a5b1001281529n4f5ac2acy266ad324b0da9318@mail.gmail.com>
References: <4B62078B.5060406@gmx.net>
	<8213d6071001281521u27347956nf2c89c3cd2e72745@mail.gmail.com>
	<b8ad2a5b1001281529n4f5ac2acy266ad324b0da9318@mail.gmail.com>
Message-ID: <23f91031001281729s403a9954nc7400d73fa69f6ea@mail.gmail.com>

why not just use dialplan matching?

assume FS1 has gateway named skype,

originate sofia/gateway/skype/<your_dedicated_skype_name>+<other_skype_name_you_want_to_dial


on your FS/skype server, set dialplan:


$destination_number to match    (.*)\+(.*), then you can

bridge skypiax/$1/$2

2010/1/29 Wasim Baig <wasim at convergence.pk>:
> Setting a sip header is a more elegant way ...
>
> http://wiki.freeswitch.org/wiki/Sofia#Adding_Request_Headers
>
> -wasim
>
> On Fri, Jan 29, 2010 at 4:21 AM, Ghulam Mustafa <mustafa.pk at gmail.com>
> wrote:
>>
>> i don't know if it's a wise solution, but you can make a db insert
>> before bridge on freeswitch server, kypiax server can query db after
>> answer to get an idea!
>>
>> On Fri, Jan 29, 2010 at 2:54 AM, Peter P GMX <Prometheus001 at gmx.net>
>> wrote:
>> > Hello,
>> >
>> > I have a main freeswitch server and a separate Freeswitch/Skype server
>> > with mod_skypiax.
>> >
>> > I want main freeswitch server to tell the Freeswitch/Skype server to use
>> > a dedicated Skype interface(interface1, interface2 etc). What is the
>> > best way to pass this variable?
>> > Some ideas from my side
>> >
>> > ? ?* set caller_id_number or caller_id_number_name, as these are
>> > ? ? ?overwritten by Skype anyway. But this is not a clean solution
>> > ? ?* use another UDP port for each gateway, but this is a huge effort
>> > ? ? ?if the number of gateways becomes larger
>> > ? ?* set a dedicated sip header for this. Reading sip-header is easy
>> > ? ? ?(documented) but how to set my own sip header entry in FS?
>> > ? ?* any other idea?
>> >
>> >
>> > Best regards
>> > Peter
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Ghulam Mustafa
>> cell: +92 333.611.7681
>> sip: cyrenity at ekiga.net
>> mail: mustafa.pk at gmail.com
>> web: cyrenity.wordpress.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
> peace be upon you ...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From emptysands at gmail.com  Thu Jan 28 18:08:29 2010
From: emptysands at gmail.com (Nicholas Lee)
Date: Fri, 29 Jan 2010 15:08:29 +1300
Subject: [Freeswitch-users] Hybrid Encryption?
In-Reply-To: <46C06209-9515-4B1B-B449-F55A51FF548B@freeswitch.org>
References: <2b6116b31001272154l3c0bbe80y8bf3db94961e8e1d@mail.gmail.com>
	<8A9EDC4E-C49B-488D-9DBF-169A185462AB@freeswitch.org>
	<2b6116b31001281631u7ada7876wc419bb7afadd7ef7@mail.gmail.com>
	<46C06209-9515-4B1B-B449-F55A51FF548B@freeswitch.org>
Message-ID: <2b6116b31001281808x1a004cd2ne7e8dcb9f16fec3e@mail.gmail.com>

Is there a way to do it transparently?  The FS proxies will past though the
extension creds.

On Fri, Jan 29, 2010 at 1:52 PM, Brian West <brian at freeswitch.org> wrote:

> Then yes you could use FreeSWITCH to augment your Asterisk install and
> enable encryption from site to site.
>
> /b
>
>
> > Unfortunately it's not going to cover every situation.
> >
> >
> > Nicholas
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From troy at tlainvestments.com  Thu Jan 28 19:18:03 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 28 Jan 2010 20:18:03 -0700
Subject: [Freeswitch-users] ERR root tag missing
In-Reply-To: <c29b930b1001281351r3bebc2f2ta0279a9bbd4b6c0e@mail.gmail.com>
References: <c29b930b1001281351r3bebc2f2ta0279a9bbd4b6c0e@mail.gmail.com>
Message-ID: <5B9401A1-DF37-4ED1-827A-2B95DAE7AEF2@tlainvestments.com>

Sorry about hijacking this thread!  I now know better.

I am using xml_curl, and I think that's the heads up I needed. Lemme check into what I'm returning from that...

Thanks!

On Jan 28, 2010, at 2:51 PM, Milena wrote:

> Hello :)
> First, do not hijack threads, if you want to post about a different problem, do not click reply and then change the subject, please create a NEW MESSAGE.
> 
> About your question, are you using xml_curl? or any other dialplan seekers?
> 
> -Milena
> 
> 2010/1/28 Troy Anderson <troy at tlainvestments.com>
> I'm seeing this error quite often on my systems:
>        2010-01-28 12:35:46.703112 [ERR] switch_xml.c:1571 Error[[error near line 1]: root tag missing]
> 
> I've looked at freeswitch.xml.fsxml to see if I could find some kind of malformed XML, but with no luck.  Which Is line 1is it referring to?
> 
> Line 1 of freeswitch.xml.fsxml is a comment, with the first actual XML line like:
>        <document type="freeswitch/xml">
> 
> This error always happens right after after a mod_dialplan_xml.c:408 log message, so I'm led to believe my dialplan XML is messed up, but I cannot see where.
> 
> In freeswitch.xml.fsxml near the dialplan section, this is what I have:
> 
> ...
> <section name="configuration" description="Various Configuration">
>        <!-- CUT a bunch of stuff ... CUT -->
> </section>
> <section name="dialplan" description="Regex/XML Dialplan">
>        <context name="my_office">
>                <extension name="unloop">
>                        <condition field="${unroll_loops}" expression="^true$"/>
>                        <condition field="${sip_looped_call}" expression="^true$">
>                                <action application="deflect" data="${destination_number}"/>
>                        </condition>
>                </extension>
>                <extension name="set_domain" continue="true">
>                        <condition field="${domain_name}" expression="^$"/>
>                        <condition field="source" expression="mod_sofia"/>
>                        <condition field="${sip_auth_realm}" expression="^$">
>                                <action application="set" data="domain_name=10.0.0.120"/>
>                                <anti-action application="set" data="domain_name=${sip_auth_realm}"/>
>                        </condition>
>                </extension>
> ...
> </section>
> ...
> 
> Thanks for any ideas!
> 
> -Troy
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mcampbellsmith at gmail.com  Thu Jan 28 19:32:26 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Fri, 29 Jan 2010 14:32:26 +1100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <191c3a031001272125t2b19d134lbb65dfebbb7a52b9@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
	<33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
	<191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>
	<33c87fa31001271939i668e9676y1f687dc182b4ca89@mail.gmail.com>
	<CF0D5382-EE44-48CB-BF14-27A443014336@freeswitch.org>
	<33c87fa31001272035t7c46e054t7ce2dd598fa40ebb@mail.gmail.com>
	<191c3a031001272125t2b19d134lbb65dfebbb7a52b9@mail.gmail.com>
Message-ID: <33c87fa31001281932h73466f87yf4715058f4632558@mail.gmail.com>

Hi !

I confirmed yesterday that if the SPA is not NAT'd, then the event is
sent.  I just removed NAT from the extension that I was having
problems with.

Looking at the db tables, it appears there are two - the
sofia_reg_internal.db and sofia_reg_internal_nat.db

Could it be that the sendevent command is only looking in the
sofia_reg_internal.db database and not sofia_reg_internal_nat.db ?



On Thu, Jan 28, 2010 at 4:25 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> You have to look in the sql db and compare the specified vals with the ones
> looked up from the event again the user and host need to match the db
>
> On Jan 27, 2010 10:41 PM, "Mark Campbell-Smith" <mcampbellsmith at gmail.com>
> wrote:
>
> Hi Brian,
>
> ?I've previously enabled siptrace for internal profile, but I see
> nothing sent and nothing received.
>
> On Thu, Jan 28, 2010 at 2:54 PM, Brian West <brian at freeswitch.org> wrote:
>> I'm suspecting the code...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From ranjtech at gmail.com  Thu Jan 28 21:57:03 2010
From: ranjtech at gmail.com (RR)
Date: Fri, 29 Jan 2010 00:57:03 -0500
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
Message-ID: <025701caa0a7$e1ca6200$a55f2600$@com>

Hi Brian,

 

Ok here's the sip trace captured at the softswitch. 

 

BTW, I noticed that during startup, I see FS printing out this message:

2010-01-29 01:30:05.634220 [NOTICE] sofia_reg.c:2267 Added gateway
'Test-Inbound' to profile 'external'

2010-01-29 01:30:05.634547 [ERR] sofia.c:1663 ERROR: username param is
REQUIRED!

2010-01-29 01:30:05.634657 [ERR] sofia.c:1663 ERROR: username param is
REQUIRED!

 

Which is weird, because as you can see from the config, the username is
infact present. Weird! Anyway, here's the trace

 

REGISTER sip:myswitch.net.au SIP/2.0

Via: SIP/2.0/UDP 173.xxx.xxx.xxx:5080;rport;branch=z9hG4bK1U32gpr9vj5eF

Max-Forwards: 70

From: <sip:user1 at myswitch.net.au;transport=udp>;tag=11Qey4tcrUH9g

To: <sip:user1 at myswitch.net.au;transport=udp>

Call-ID: cd589b73-031e-445f-a4d8-2fe334d81bbc

CSeq: 126255215 REGISTER

Contact:
<sip:gw+Test-Inbound at 173.xxx.xxx.xxx:5080;transport=udp;gw=Test-Inbound>

Expires: 60

User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15980

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY

Supported: timer, precondition, path, replaces

Content-Length: 0

SIP/2.0 409 Conflict

Via: SIP/2.0/UDP 173.xxx.xxx.xxx:5080;rport;branch=z9hG4bK1U32gpr9vj5eF

From: <sip:user1 at 10.1.3.20;transport=udp>;tag=11Qey4tcrUH9g

To: <sip:user1 at 10.1.3.20;transport=udp>

Call-ID: cd589b73-031e-445f-a4d8-2fe334d81bbc

CSeq: 126255215 REGISTER

Content-Length: 0

 

.and then this message just repeats again and again with every REGISTER
request.

 

Thanks for your help

\RR

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, January 28, 2010 2:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 409 Conflict When registering FS with
Softswitch

 

Any sip trace would work.. text.. or what not.. 

 

 

 

/b

 

On Jan 28, 2010, at 12:46 PM, RR wrote:





Hi brian,

 

Do you need the sip trace from within FS or just any packet capture like
from snoop/tcpdump etc will do? I don't know how to do a sip trace from
within FS

 

\R

 



__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4815 (20100128) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From magesh.freeswitch at gmail.com  Thu Jan 28 22:32:08 2010
From: magesh.freeswitch at gmail.com (Magesh R)
Date: Fri, 29 Jan 2010 01:32:08 -0500
Subject: [Freeswitch-users] min_dtmf_duration has not changed
Message-ID: <369c72d81001282232r6f7ef2f2m745e71fcdc6b73e2@mail.gmail.com>

Dear All,
   I am trying to change the min_dtmf_duration value by using fsctl. But it
didn't changed.

freeswitch at debian> fsctl min_dtmf_duration 800

+OK min dtmf duration: 400

     Is there any thing need to be set before changing this?

Thanks,
Mag.
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From christian.loeschenkohl at xpirio.com  Fri Jan 29 00:32:05 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Fri, 29 Jan 2010 09:32:05 +0100
Subject: [Freeswitch-users] wiki password recovery - no mail is send
Message-ID: <4B629D05.1060908@xpirio.com>

hello

the password recovery for the fs wiki doesn't seem to work.
no e-mail is send when entering the username and press "e-mail new password".

may i assist here, we do maintain a few wikis for ourself.

br


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From Prometheus001 at gmx.net  Fri Jan 29 03:32:05 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 29 Jan 2010 12:32:05 +0100
Subject: [Freeswitch-users] Set an gateway parameter for
 external	gateway (skype)
In-Reply-To: <23f91031001281729s403a9954nc7400d73fa69f6ea@mail.gmail.com>
References: <4B62078B.5060406@gmx.net>	<8213d6071001281521u27347956nf2c89c3cd2e72745@mail.gmail.com>	<b8ad2a5b1001281529n4f5ac2acy266ad324b0da9318@mail.gmail.com>
	<23f91031001281729s403a9954nc7400d73fa69f6ea@mail.gmail.com>
Message-ID: <4B62C735.4010309@gmx.net>

Hello Sven

I think that's a good idea. I just tried: "+" is not valid in a Skype
username. So I may use it.

I will try that and give some feedback.

Best regards
Peter


Seven Du schrieb:
> why not just use dialplan matching?
>
> assume FS1 has gateway named skype,
>
> originate sofia/gateway/skype/<your_dedicated_skype_name>+<other_skype_name_you_want_to_dial
>
>
> on your FS/skype server, set dialplan:
>
>
> $destination_number to match    (.*)\+(.*), then you can
>
> bridge skypiax/$1/$2
>
> 2010/1/29 Wasim Baig <wasim at convergence.pk>:
>   
>> Setting a sip header is a more elegant way ...
>>
>> http://wiki.freeswitch.org/wiki/Sofia#Adding_Request_Headers
>>
>> -wasim
>>
>> On Fri, Jan 29, 2010 at 4:21 AM, Ghulam Mustafa <mustafa.pk at gmail.com>
>> wrote:
>>     
>>> i don't know if it's a wise solution, but you can make a db insert
>>> before bridge on freeswitch server, kypiax server can query db after
>>> answer to get an idea!
>>>
>>> On Fri, Jan 29, 2010 at 2:54 AM, Peter P GMX <Prometheus001 at gmx.net>
>>> wrote:
>>>       
>>>> Hello,
>>>>
>>>> I have a main freeswitch server and a separate Freeswitch/Skype server
>>>> with mod_skypiax.
>>>>
>>>> I want main freeswitch server to tell the Freeswitch/Skype server to use
>>>> a dedicated Skype interface(interface1, interface2 etc). What is the
>>>> best way to pass this variable?
>>>> Some ideas from my side
>>>>
>>>>    * set caller_id_number or caller_id_number_name, as these are
>>>>      overwritten by Skype anyway. But this is not a clean solution
>>>>    * use another UDP port for each gateway, but this is a huge effort
>>>>      if the number of gateways becomes larger
>>>>    * set a dedicated sip header for this. Reading sip-header is easy
>>>>      (documented) but how to set my own sip header entry in FS?
>>>>    * any other idea?
>>>>
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>         
>>>
>>> --
>>> Ghulam Mustafa
>>> cell: +92 333.611.7681
>>> sip: cyrenity at ekiga.net
>>> mail: mustafa.pk at gmail.com
>>> web: cyrenity.wordpress.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>       
>>
>> --
>> wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
>> peace be upon you ...
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From Prometheus001 at gmx.net  Fri Jan 29 03:41:22 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 29 Jan 2010 12:41:22 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with
 modified	snd-dummy (one way audio)
In-Reply-To: <7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>
References: <4B60555B.2020004@gmx.net>	<7b197bef1001270823t15f2b5e3vf9b5f20081ed9a2b@mail.gmail.com>
	<4B6070A8.6050607@gmx.net>	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com>
	<4B607944.4040700@gmx.net>
	<4B608014.4030902@gmx.net>	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>
	<4B61ECE0.10409@gmx.net>	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>
	<4B61FCAB.5040707@gmx.net>
	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>
Message-ID: <4B62C962.7000601@gmx.net>

I now reinstalled the original sound drivers
Unfortunaltely the sound problems remain, not that worse but they are there:
Audio is still (almost) one way. Almost means:

    * SIP -> Skype ok
    * Skype=> SIP I hear only some scratching on very loud audio

Could it be a volume problem? But snd-dummy should have no volume
properties, right?

Best regards
Peter


Giovanni Maruzzelli schrieb:
> with three instances you will assign the hw:0 device to skype client
> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
> Must work. Pay attention to assign the same device name to all devices
> needed by a skype instance (sound devices window): playback, capture
> AND ring.
>
> Or maybe is a bug of ALSA on Debian...
>
> -giovanni
>
> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>> #2 to the Skype accounts. Still no sound.
>> On the frist call there is one way audio, on the following calls there
>> is no audio at all.
>> This is weird.
>>
>> Best regards
>> Peter
>>
>> Giovanni Maruzzelli schrieb:
>>     
>>> Ciao Peter,
>>>
>>> Never tested on Debian 5.
>>>
>>> When you write "same problem" you are referring to the audio going one
>>> way only (btw, which way?) with the custom audio driver?
>>>
>>> Have you tried with multiple instances of the regular Debian
>>> snd-dummy, as I wrote in a mail before?
>>>
>>> -gm
>>>
>>>
>>>
>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>       
>>>> Hello Giovanni,
>>>>
>>>> I did so but the same problem again.
>>>>
>>>> Did you ever test in on Debian 5.0?
>>>>
>>>> Best reards
>>>> Peter
>>>>
>>>> Giovanni Maruzzelli schrieb:
>>>>
>>>>         
>>>>> good, so you have only one sound device, the right one.
>>>>>
>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>
>>>>> -gm
>>>>>
>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>
>>>>>
>>>>>           
>>>>>> I installed alsa-utile,
>>>>>>
>>>>>> now I get:
>>>>>>
>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>  Subdevices: 127/128
>>>>>>  Subdevice #0: subdevice #0
>>>>>>  Subdevice #1: subdevice #1
>>>>>>  Subdevice #2: subdevice #2
>>>>>>  Subdevice #3: subdevice #3
>>>>>>  Subdevice #4: subdevice #4
>>>>>>  Subdevice #5: subdevice #5
>>>>>>  Subdevice #6: subdevice #6
>>>>>>  Subdevice #7: subdevice #7
>>>>>>  Subdevice #8: subdevice #8
>>>>>>  Subdevice #9: subdevice #9
>>>>>>  Subdevice #10: subdevice #10
>>>>>>  Subdevice #11: subdevice #11
>>>>>>  Subdevice #12: subdevice #12
>>>>>>  Subdevice #13: subdevice #13
>>>>>>  Subdevice #14: subdevice #14
>>>>>>  Subdevice #15: subdevice #15
>>>>>>  Subdevice #16: subdevice #16
>>>>>>  Subdevice #17: subdevice #17
>>>>>>  Subdevice #18: subdevice #18
>>>>>>  Subdevice #19: subdevice #19
>>>>>>  Subdevice #20: subdevice #20
>>>>>>  Subdevice #21: subdevice #21
>>>>>>  Subdevice #22: subdevice #22
>>>>>>  Subdevice #23: subdevice #23
>>>>>>  Subdevice #24: subdevice #24
>>>>>>  Subdevice #25: subdevice #25
>>>>>>  Subdevice #26: subdevice #26
>>>>>>  Subdevice #27: subdevice #27
>>>>>>  Subdevice #28: subdevice #28
>>>>>>  Subdevice #29: subdevice #29
>>>>>>  Subdevice #30: subdevice #30
>>>>>>  Subdevice #31: subdevice #31
>>>>>>  Subdevice #32: subdevice #32
>>>>>>  Subdevice #33: subdevice #33
>>>>>>  Subdevice #34: subdevice #34
>>>>>>  Subdevice #35: subdevice #35
>>>>>>  Subdevice #36: subdevice #36
>>>>>>  Subdevice #37: subdevice #37
>>>>>>  Subdevice #38: subdevice #38
>>>>>>  Subdevice #39: subdevice #39
>>>>>>  Subdevice #40: subdevice #40
>>>>>>  Subdevice #41: subdevice #41
>>>>>>  Subdevice #42: subdevice #42
>>>>>>  Subdevice #43: subdevice #43
>>>>>>  Subdevice #44: subdevice #44
>>>>>>  Subdevice #45: subdevice #45
>>>>>>  Subdevice #46: subdevice #46
>>>>>>  Subdevice #47: subdevice #47
>>>>>>  Subdevice #48: subdevice #48
>>>>>>  Subdevice #49: subdevice #49
>>>>>>  Subdevice #50: subdevice #50
>>>>>>  Subdevice #51: subdevice #51
>>>>>>  Subdevice #52: subdevice #52
>>>>>>  Subdevice #53: subdevice #53
>>>>>>  Subdevice #54: subdevice #54
>>>>>>  Subdevice #55: subdevice #55
>>>>>>  Subdevice #56: subdevice #56
>>>>>>  Subdevice #57: subdevice #57
>>>>>>  Subdevice #58: subdevice #58
>>>>>>  Subdevice #59: subdevice #59
>>>>>>  Subdevice #60: subdevice #60
>>>>>>  Subdevice #61: subdevice #61
>>>>>>  Subdevice #62: subdevice #62
>>>>>>  Subdevice #63: subdevice #63
>>>>>>  Subdevice #64: subdevice #64
>>>>>>  Subdevice #65: subdevice #65
>>>>>>  Subdevice #66: subdevice #66
>>>>>>  Subdevice #67: subdevice #67
>>>>>>  Subdevice #68: subdevice #68
>>>>>>  Subdevice #69: subdevice #69
>>>>>>  Subdevice #70: subdevice #70
>>>>>>  Subdevice #71: subdevice #71
>>>>>>  Subdevice #72: subdevice #72
>>>>>>  Subdevice #73: subdevice #73
>>>>>>  Subdevice #74: subdevice #74
>>>>>>  Subdevice #75: subdevice #75
>>>>>>  Subdevice #76: subdevice #76
>>>>>>  Subdevice #77: subdevice #77
>>>>>>  Subdevice #78: subdevice #78
>>>>>>  Subdevice #79: subdevice #79
>>>>>>  Subdevice #80: subdevice #80
>>>>>>  Subdevice #81: subdevice #81
>>>>>>  Subdevice #82: subdevice #82
>>>>>>  Subdevice #83: subdevice #83
>>>>>>  Subdevice #84: subdevice #84
>>>>>>  Subdevice #85: subdevice #85
>>>>>>  Subdevice #86: subdevice #86
>>>>>>  Subdevice #87: subdevice #87
>>>>>>  Subdevice #88: subdevice #88
>>>>>>  Subdevice #89: subdevice #89
>>>>>>  Subdevice #90: subdevice #90
>>>>>>  Subdevice #91: subdevice #91
>>>>>>  Subdevice #92: subdevice #92
>>>>>>  Subdevice #93: subdevice #93
>>>>>>  Subdevice #94: subdevice #94
>>>>>>  Subdevice #95: subdevice #95
>>>>>>  Subdevice #96: subdevice #96
>>>>>>  Subdevice #97: subdevice #97
>>>>>>  Subdevice #98: subdevice #98
>>>>>>  Subdevice #99: subdevice #99
>>>>>>  Subdevice #100: subdevice #100
>>>>>>  Subdevice #101: subdevice #101
>>>>>>  Subdevice #102: subdevice #102
>>>>>>  Subdevice #103: subdevice #103
>>>>>>  Subdevice #104: subdevice #104
>>>>>>  Subdevice #105: subdevice #105
>>>>>>  Subdevice #106: subdevice #106
>>>>>>  Subdevice #107: subdevice #107
>>>>>>  Subdevice #108: subdevice #108
>>>>>>  Subdevice #109: subdevice #109
>>>>>>  Subdevice #110: subdevice #110
>>>>>>  Subdevice #111: subdevice #111
>>>>>>  Subdevice #112: subdevice #112
>>>>>>  Subdevice #113: subdevice #113
>>>>>>  Subdevice #114: subdevice #114
>>>>>>  Subdevice #115: subdevice #115
>>>>>>  Subdevice #116: subdevice #116
>>>>>>  Subdevice #117: subdevice #117
>>>>>>  Subdevice #118: subdevice #118
>>>>>>  Subdevice #119: subdevice #119
>>>>>>  Subdevice #120: subdevice #120
>>>>>>  Subdevice #121: subdevice #121
>>>>>>  Subdevice #122: subdevice #122
>>>>>>  Subdevice #123: subdevice #123
>>>>>>  Subdevice #124: subdevice #124
>>>>>>  Subdevice #125: subdevice #125
>>>>>>  Subdevice #126: subdevice #126
>>>>>>  Subdevice #127: subdevice #127
>>>>>>
>>>>>>
>>>>>> Peter P GMX schrieb:
>>>>>>
>>>>>>
>>>>>>             
>>>>>>> Her's the output:
>>>>>>>
>>>>>>> skype:~# aplay -l
>>>>>>> bash: aplay: command not found
>>>>>>>
>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>> what's the output of:
>>>>>>>>
>>>>>>>> aplay -l
>>>>>>>>
>>>>>>>> ?
>>>>>>>>
>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>
>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>
>>>>>>>>
>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>>>> Thanks Giovanni,
>>>>>>>>>
>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>
>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>
>>>>>>>>> Best regards
>>>>>>>>> Peter
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>>>>>> This warning is harmless:
>>>>>>>>>>
>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>
>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>                     
>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>
>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>
>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>> 8.04).
>>>>>>>>>>>
>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>
>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>
>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>
>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>> soundcard.
>>>>>>>>>>>
>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>
>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>
>>>>>>>>>>> ciao for now,
>>>>>>>>>>>
>>>>>>>>>>> -giovanni
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>
>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>> instructions in
>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>
>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>> suppressed
>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>> suppressed
>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>
>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>
>>>>>>>>>>>> Second question:
>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>
>>>>>>>>>>>> Best regards
>>>>>>>>>>>> Peter
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>                         
>>>>>>>>>>> --
>>>>>>>>>>> Sincerely,
>>>>>>>>>>>
>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>                       
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>                   
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>>             
>>>>>
>>>>>           
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   



From codecomplete at free.fr  Fri Jan 29 03:52:26 2010
From: codecomplete at free.fr (Fred-145)
Date: Fri, 29 Jan 2010 12:52:26 +0100
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
References: <27338355.post@talk.nabble.com>
	<27338355.post-WJuSqJV8a7jJsTRiRinrng@public.gmane.org>
	<201001270750.16772.sos@sokhapkin.dyndns.org>
Message-ID: <eui5m5hg6tqemri4r5u250hu72mjddmjnt@4ax.com>

On Wed, 27 Jan 2010 07:50:16 -0500, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
>set bypass_media=true

Thanks everyone for the feedback.

Are there drawbacks to having RTP pakets flow directly between the SIP
end-points?




From mustafa.pk at gmail.com  Fri Jan 29 04:03:39 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Fri, 29 Jan 2010 17:03:39 +0500
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <4B62C962.7000601@gmx.net>
References: <4B60555B.2020004@gmx.net>
	<7b197bef1001270907x21e5cf6s487e2b4e30b9eae8@mail.gmail.com>
	<4B607944.4040700@gmx.net> <4B608014.4030902@gmx.net>
	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com>
	<4B61ECE0.10409@gmx.net>
	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com>
	<4B61FCAB.5040707@gmx.net>
	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com>
	<4B62C962.7000601@gmx.net>
Message-ID: <8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>

did you enable debug mode while compiling custom snd-dummy? if  yes
try re-compiling with debug mode disabled.

-m

On Fri, Jan 29, 2010 at 4:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> I now reinstalled the original sound drivers
> Unfortunaltely the sound problems remain, not that worse but they are there:
> Audio is still (almost) one way. Almost means:
>
> ? ?* SIP -> Skype ok
> ? ?* Skype=> SIP I hear only some scratching on very loud audio
>
> Could it be a volume problem? But snd-dummy should have no volume
> properties, right?
>
> Best regards
> Peter
>
>
> Giovanni Maruzzelli schrieb:
>> with three instances you will assign the hw:0 device to skype client
>> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
>> Must work. Pay attention to assign the same device name to all devices
>> needed by a skype instance (sound devices window): playback, capture
>> AND ring.
>>
>> Or maybe is a bug of ALSA on Debian...
>>
>> -giovanni
>>
>> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>
>>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>>> #2 to the Skype accounts. Still no sound.
>>> On the frist call there is one way audio, on the following calls there
>>> is no audio at all.
>>> This is weird.
>>>
>>> Best regards
>>> Peter
>>>
>>> Giovanni Maruzzelli schrieb:
>>>
>>>> Ciao Peter,
>>>>
>>>> Never tested on Debian 5.
>>>>
>>>> When you write "same problem" you are referring to the audio going one
>>>> way only (btw, which way?) with the custom audio driver?
>>>>
>>>> Have you tried with multiple instances of the regular Debian
>>>> snd-dummy, as I wrote in a mail before?
>>>>
>>>> -gm
>>>>
>>>>
>>>>
>>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>
>>>>
>>>>> Hello Giovanni,
>>>>>
>>>>> I did so but the same problem again.
>>>>>
>>>>> Did you ever test in on Debian 5.0?
>>>>>
>>>>> Best reards
>>>>> Peter
>>>>>
>>>>> Giovanni Maruzzelli schrieb:
>>>>>
>>>>>
>>>>>> good, so you have only one sound device, the right one.
>>>>>>
>>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>>
>>>>>> -gm
>>>>>>
>>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>
>>>>>>
>>>>>>
>>>>>>> I installed alsa-utile,
>>>>>>>
>>>>>>> now I get:
>>>>>>>
>>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>> ?Subdevices: 127/128
>>>>>>> ?Subdevice #0: subdevice #0
>>>>>>> ?Subdevice #1: subdevice #1
>>>>>>> ?Subdevice #2: subdevice #2
>>>>>>> ?Subdevice #3: subdevice #3
>>>>>>> ?Subdevice #4: subdevice #4
>>>>>>> ?Subdevice #5: subdevice #5
>>>>>>> ?Subdevice #6: subdevice #6
>>>>>>> ?Subdevice #7: subdevice #7
>>>>>>> ?Subdevice #8: subdevice #8
>>>>>>> ?Subdevice #9: subdevice #9
>>>>>>> ?Subdevice #10: subdevice #10
>>>>>>> ?Subdevice #11: subdevice #11
>>>>>>> ?Subdevice #12: subdevice #12
>>>>>>> ?Subdevice #13: subdevice #13
>>>>>>> ?Subdevice #14: subdevice #14
>>>>>>> ?Subdevice #15: subdevice #15
>>>>>>> ?Subdevice #16: subdevice #16
>>>>>>> ?Subdevice #17: subdevice #17
>>>>>>> ?Subdevice #18: subdevice #18
>>>>>>> ?Subdevice #19: subdevice #19
>>>>>>> ?Subdevice #20: subdevice #20
>>>>>>> ?Subdevice #21: subdevice #21
>>>>>>> ?Subdevice #22: subdevice #22
>>>>>>> ?Subdevice #23: subdevice #23
>>>>>>> ?Subdevice #24: subdevice #24
>>>>>>> ?Subdevice #25: subdevice #25
>>>>>>> ?Subdevice #26: subdevice #26
>>>>>>> ?Subdevice #27: subdevice #27
>>>>>>> ?Subdevice #28: subdevice #28
>>>>>>> ?Subdevice #29: subdevice #29
>>>>>>> ?Subdevice #30: subdevice #30
>>>>>>> ?Subdevice #31: subdevice #31
>>>>>>> ?Subdevice #32: subdevice #32
>>>>>>> ?Subdevice #33: subdevice #33
>>>>>>> ?Subdevice #34: subdevice #34
>>>>>>> ?Subdevice #35: subdevice #35
>>>>>>> ?Subdevice #36: subdevice #36
>>>>>>> ?Subdevice #37: subdevice #37
>>>>>>> ?Subdevice #38: subdevice #38
>>>>>>> ?Subdevice #39: subdevice #39
>>>>>>> ?Subdevice #40: subdevice #40
>>>>>>> ?Subdevice #41: subdevice #41
>>>>>>> ?Subdevice #42: subdevice #42
>>>>>>> ?Subdevice #43: subdevice #43
>>>>>>> ?Subdevice #44: subdevice #44
>>>>>>> ?Subdevice #45: subdevice #45
>>>>>>> ?Subdevice #46: subdevice #46
>>>>>>> ?Subdevice #47: subdevice #47
>>>>>>> ?Subdevice #48: subdevice #48
>>>>>>> ?Subdevice #49: subdevice #49
>>>>>>> ?Subdevice #50: subdevice #50
>>>>>>> ?Subdevice #51: subdevice #51
>>>>>>> ?Subdevice #52: subdevice #52
>>>>>>> ?Subdevice #53: subdevice #53
>>>>>>> ?Subdevice #54: subdevice #54
>>>>>>> ?Subdevice #55: subdevice #55
>>>>>>> ?Subdevice #56: subdevice #56
>>>>>>> ?Subdevice #57: subdevice #57
>>>>>>> ?Subdevice #58: subdevice #58
>>>>>>> ?Subdevice #59: subdevice #59
>>>>>>> ?Subdevice #60: subdevice #60
>>>>>>> ?Subdevice #61: subdevice #61
>>>>>>> ?Subdevice #62: subdevice #62
>>>>>>> ?Subdevice #63: subdevice #63
>>>>>>> ?Subdevice #64: subdevice #64
>>>>>>> ?Subdevice #65: subdevice #65
>>>>>>> ?Subdevice #66: subdevice #66
>>>>>>> ?Subdevice #67: subdevice #67
>>>>>>> ?Subdevice #68: subdevice #68
>>>>>>> ?Subdevice #69: subdevice #69
>>>>>>> ?Subdevice #70: subdevice #70
>>>>>>> ?Subdevice #71: subdevice #71
>>>>>>> ?Subdevice #72: subdevice #72
>>>>>>> ?Subdevice #73: subdevice #73
>>>>>>> ?Subdevice #74: subdevice #74
>>>>>>> ?Subdevice #75: subdevice #75
>>>>>>> ?Subdevice #76: subdevice #76
>>>>>>> ?Subdevice #77: subdevice #77
>>>>>>> ?Subdevice #78: subdevice #78
>>>>>>> ?Subdevice #79: subdevice #79
>>>>>>> ?Subdevice #80: subdevice #80
>>>>>>> ?Subdevice #81: subdevice #81
>>>>>>> ?Subdevice #82: subdevice #82
>>>>>>> ?Subdevice #83: subdevice #83
>>>>>>> ?Subdevice #84: subdevice #84
>>>>>>> ?Subdevice #85: subdevice #85
>>>>>>> ?Subdevice #86: subdevice #86
>>>>>>> ?Subdevice #87: subdevice #87
>>>>>>> ?Subdevice #88: subdevice #88
>>>>>>> ?Subdevice #89: subdevice #89
>>>>>>> ?Subdevice #90: subdevice #90
>>>>>>> ?Subdevice #91: subdevice #91
>>>>>>> ?Subdevice #92: subdevice #92
>>>>>>> ?Subdevice #93: subdevice #93
>>>>>>> ?Subdevice #94: subdevice #94
>>>>>>> ?Subdevice #95: subdevice #95
>>>>>>> ?Subdevice #96: subdevice #96
>>>>>>> ?Subdevice #97: subdevice #97
>>>>>>> ?Subdevice #98: subdevice #98
>>>>>>> ?Subdevice #99: subdevice #99
>>>>>>> ?Subdevice #100: subdevice #100
>>>>>>> ?Subdevice #101: subdevice #101
>>>>>>> ?Subdevice #102: subdevice #102
>>>>>>> ?Subdevice #103: subdevice #103
>>>>>>> ?Subdevice #104: subdevice #104
>>>>>>> ?Subdevice #105: subdevice #105
>>>>>>> ?Subdevice #106: subdevice #106
>>>>>>> ?Subdevice #107: subdevice #107
>>>>>>> ?Subdevice #108: subdevice #108
>>>>>>> ?Subdevice #109: subdevice #109
>>>>>>> ?Subdevice #110: subdevice #110
>>>>>>> ?Subdevice #111: subdevice #111
>>>>>>> ?Subdevice #112: subdevice #112
>>>>>>> ?Subdevice #113: subdevice #113
>>>>>>> ?Subdevice #114: subdevice #114
>>>>>>> ?Subdevice #115: subdevice #115
>>>>>>> ?Subdevice #116: subdevice #116
>>>>>>> ?Subdevice #117: subdevice #117
>>>>>>> ?Subdevice #118: subdevice #118
>>>>>>> ?Subdevice #119: subdevice #119
>>>>>>> ?Subdevice #120: subdevice #120
>>>>>>> ?Subdevice #121: subdevice #121
>>>>>>> ?Subdevice #122: subdevice #122
>>>>>>> ?Subdevice #123: subdevice #123
>>>>>>> ?Subdevice #124: subdevice #124
>>>>>>> ?Subdevice #125: subdevice #125
>>>>>>> ?Subdevice #126: subdevice #126
>>>>>>> ?Subdevice #127: subdevice #127
>>>>>>>
>>>>>>>
>>>>>>> Peter P GMX schrieb:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> Her's the output:
>>>>>>>>
>>>>>>>> skype:~# aplay -l
>>>>>>>> bash: aplay: command not found
>>>>>>>>
>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>>> what's the output of:
>>>>>>>>>
>>>>>>>>> aplay -l
>>>>>>>>>
>>>>>>>>> ?
>>>>>>>>>
>>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>>
>>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>> Thanks Giovanni,
>>>>>>>>>>
>>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>>
>>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>>
>>>>>>>>>> Best regards
>>>>>>>>>> Peter
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> This warning is harmless:
>>>>>>>>>>>
>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>
>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>>
>>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>>
>>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>>> 8.04).
>>>>>>>>>>>>
>>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>>
>>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>>
>>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>>
>>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>>> soundcard.
>>>>>>>>>>>>
>>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>>
>>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>>
>>>>>>>>>>>> ciao for now,
>>>>>>>>>>>>
>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>>
>>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>>> instructions in
>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>>
>>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>>
>>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>>
>>>>>>>>>>>>> Second question:
>>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>>
>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>> --
>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>
>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com



From lakindia89 at gmail.com  Fri Jan 29 04:36:12 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 29 Jan 2010 18:06:12 +0530
Subject: [Freeswitch-users] Multiple DTMF on group_confirm_key
In-Reply-To: <191c3a031001280914l56a035s74fd6d4b00dd9b3d@mail.gmail.com>
References: <7d79b3931001280411u6262f627xca723c64de5e118d@mail.gmail.com>
	<191c3a031001280914l56a035s74fd6d4b00dd9b3d@mail.gmail.com>
Message-ID: <7d79b3931001290436j2061a8dcg5cf7c5144103eb00@mail.gmail.com>

I tested by executing a script. It works great. But a small doubt.
Assume that I made a parallel dial using bridge application.
Normally, when one party answer the call, other party end will be hanged up.

But if I use group_confirm_key=exec and group_confirm_file=perl script.pl,
both the end can answer, and call bridged with the person who finished the
script first.

What I've to do if I need to execute the script only for the person who
answer's the call first?


On Thu, Jan 28, 2010 at 10:44 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you have to use a script (See the wiki for executing a script)
> then you can read in as many digits as you want and do what you need.
>
>
> On Thu, Jan 28, 2010 at 6:11 AM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Hi all,
>>
>> I've experimented with group confirm key and group confirm file. It works
>> great. However, I was unable to give multiple DTMF digits to get the
>> confirmation.
>>
>> I've set group_confirm_key=1234, I thought it will ask the 4 digits from
>> the user. But it simply taken 1 and when the user presses 1, the call got
>> bridged.
>>
>> Is there any way to specify multiple dtmf to be confirmed??
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at celliax.org  Fri Jan 29 05:13:00 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 29 Jan 2010 14:13:00 +0100
Subject: [Freeswitch-users] Set an gateway parameter for external
	gateway (skype)
In-Reply-To: <4B62C735.4010309@gmx.net>
References: <4B62078B.5060406@gmx.net>
	<8213d6071001281521u27347956nf2c89c3cd2e72745@mail.gmail.com> 
	<b8ad2a5b1001281529n4f5ac2acy266ad324b0da9318@mail.gmail.com> 
	<23f91031001281729s403a9954nc7400d73fa69f6ea@mail.gmail.com> 
	<4B62C735.4010309@gmx.net>
Message-ID: <7b197bef1001290513i14d3530avfb7298f5f97582f7@mail.gmail.com>

I would use SIP for intra-FS communications (as anthm rightly told me
when I was implementing a remote-protocol-kludge to have the skype
clients running in a machine different from the primary FS server -
that's probably what you want to obtain - not having skype clients
running in the primary FS server)

so FS1 <-> SIP <-> FS2 <-> skype

you can make good use of an URI like "skype/skypeusername" as detailed
in the wiki page
http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#Dialplan

So, you can have something like FS1 calling sip:skype/skypeusername at
FS2 , not sure about syntax tough.

-gm

On Fri, Jan 29, 2010 at 12:32 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello Sven
>
> I think that's a good idea. I just tried: "+" is not valid in a Skype
> username. So I may use it.
>
> I will try that and give some feedback.
>
> Best regards
> Peter
>
>
> Seven Du schrieb:
>> why not just use dialplan matching?
>>
>> assume FS1 has gateway named skype,
>>
>> originate sofia/gateway/skype/<your_dedicated_skype_name>+<other_skype_name_you_want_to_dial
>>
>>
>> on your FS/skype server, set dialplan:
>>
>>
>> $destination_number to match ? ?(.*)\+(.*), then you can
>>
>> bridge skypiax/$1/$2
>>
>> 2010/1/29 Wasim Baig <wasim at convergence.pk>:
>>
>>> Setting a sip header is a more elegant way ...
>>>
>>> http://wiki.freeswitch.org/wiki/Sofia#Adding_Request_Headers
>>>
>>> -wasim
>>>
>>> On Fri, Jan 29, 2010 at 4:21 AM, Ghulam Mustafa <mustafa.pk at gmail.com>
>>> wrote:
>>>
>>>> i don't know if it's a wise solution, but you can make a db insert
>>>> before bridge on freeswitch server, kypiax server can query db after
>>>> answer to get an idea!
>>>>
>>>> On Fri, Jan 29, 2010 at 2:54 AM, Peter P GMX <Prometheus001 at gmx.net>
>>>> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> I have a main freeswitch server and a separate Freeswitch/Skype server
>>>>> with mod_skypiax.
>>>>>
>>>>> I want main freeswitch server to tell the Freeswitch/Skype server to use
>>>>> a dedicated Skype interface(interface1, interface2 etc). What is the
>>>>> best way to pass this variable?
>>>>> Some ideas from my side
>>>>>
>>>>> ? ?* set caller_id_number or caller_id_number_name, as these are
>>>>> ? ? ?overwritten by Skype anyway. But this is not a clean solution
>>>>> ? ?* use another UDP port for each gateway, but this is a huge effort
>>>>> ? ? ?if the number of gateways becomes larger
>>>>> ? ?* set a dedicated sip header for this. Reading sip-header is easy
>>>>> ? ? ?(documented) but how to set my own sip header entry in FS?
>>>>> ? ?* any other idea?
>>>>>
>>>>>
>>>>> Best regards
>>>>> Peter
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> --
>>>> Ghulam Mustafa
>>>> cell: +92 333.611.7681
>>>> sip: cyrenity at ekiga.net
>>>> mail: mustafa.pk at gmail.com
>>>> web: cyrenity.wordpress.com
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> --
>>> wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
>>> peace be upon you ...
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From gmaruzz at celliax.org  Fri Jan 29 05:20:38 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 29 Jan 2010 14:20:38 +0100
Subject: [Freeswitch-users] mod_skypiax - help needed with modified
	snd-dummy (one way audio)
In-Reply-To: <8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>
References: <4B60555B.2020004@gmx.net> <4B607944.4040700@gmx.net> 
	<4B608014.4030902@gmx.net>
	<7b197bef1001271215o64918f56s3d16e51528fc5f66@mail.gmail.com> 
	<4B61ECE0.10409@gmx.net>
	<7b197bef1001281210u43a907edi13afe794cf0e1a2e@mail.gmail.com> 
	<4B61FCAB.5040707@gmx.net>
	<7b197bef1001281341h7ee2c58fj79b3886630901f29@mail.gmail.com> 
	<4B62C962.7000601@gmx.net>
	<8213d6071001290403g2c08fe39gd92eb0446bb6485a@mail.gmail.com>
Message-ID: <7b197bef1001290520x37179cb1o63273410e9ef5bc@mail.gmail.com>

Peter,

Can you connect on IRC (irc.freenode.net #freeswitch)? I'm gmaruzz there.

Can you restate your problems? I've lost connection :)

with snd-dummy custom you can create *one only* snd-dummy instance, so
*one only* fake soundcard. If you create more, will not work. But with
that one fake soundcard you can use 64 skype client instances, all
with the same soundcard hardware device (hw:n).

with original snd-dummy you can create a max of 8 instances, so 8 fake
soundcards, and with each fake soundcard you can  use a max of 8 skype
client instances.

use the hardware devices, not the default devices (use the "hw:n")

-giovanni

On Fri, Jan 29, 2010 at 1:03 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:
> did you enable debug mode while compiling custom snd-dummy? if ?yes
> try re-compiling with debug mode disabled.
>
> -m
>
> On Fri, Jan 29, 2010 at 4:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>> I now reinstalled the original sound drivers
>> Unfortunaltely the sound problems remain, not that worse but they are there:
>> Audio is still (almost) one way. Almost means:
>>
>> ? ?* SIP -> Skype ok
>> ? ?* Skype=> SIP I hear only some scratching on very loud audio
>>
>> Could it be a volume problem? But snd-dummy should have no volume
>> properties, right?
>>
>> Best regards
>> Peter
>>
>>
>> Giovanni Maruzzelli schrieb:
>>> with three instances you will assign the hw:0 device to skype client
>>> 0...7, hw:1 to skype client 8...15 and hw:2 to skype client 16...23.
>>> Must work. Pay attention to assign the same device name to all devices
>>> needed by a skype instance (sound devices window): playback, capture
>>> AND ring.
>>>
>>> Or maybe is a bug of ALSA on Debian...
>>>
>>> -giovanni
>>>
>>> On Thu, Jan 28, 2010 at 10:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>> I crated 3 instances of snd-dummy, this worked. I assigned then Instance
>>>> #2 to the Skype accounts. Still no sound.
>>>> On the frist call there is one way audio, on the following calls there
>>>> is no audio at all.
>>>> This is weird.
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>> Giovanni Maruzzelli schrieb:
>>>>
>>>>> Ciao Peter,
>>>>>
>>>>> Never tested on Debian 5.
>>>>>
>>>>> When you write "same problem" you are referring to the audio going one
>>>>> way only (btw, which way?) with the custom audio driver?
>>>>>
>>>>> Have you tried with multiple instances of the regular Debian
>>>>> snd-dummy, as I wrote in a mail before?
>>>>>
>>>>> -gm
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Jan 28, 2010 at 9:00 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>
>>>>>
>>>>>> Hello Giovanni,
>>>>>>
>>>>>> I did so but the same problem again.
>>>>>>
>>>>>> Did you ever test in on Debian 5.0?
>>>>>>
>>>>>> Best reards
>>>>>> Peter
>>>>>>
>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>
>>>>>>
>>>>>>> good, so you have only one sound device, the right one.
>>>>>>>
>>>>>>> Use the one with hw:0 in the window that skype gives you to set sound devices
>>>>>>>
>>>>>>> -gm
>>>>>>>
>>>>>>> On Wed, Jan 27, 2010 at 7:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> I installed alsa-utile,
>>>>>>>>
>>>>>>>> now I get:
>>>>>>>>
>>>>>>>> skype:/var/cache/apt/archives# aplay -l
>>>>>>>> **** List of PLAYBACK Hardware Devices ****
>>>>>>>> card 0: Dummy [Dummy], device 0: Dummy PCM [Dummy PCM]
>>>>>>>> ?Subdevices: 127/128
>>>>>>>> ?Subdevice #0: subdevice #0
>>>>>>>> ?Subdevice #1: subdevice #1
>>>>>>>> ?Subdevice #2: subdevice #2
>>>>>>>> ?Subdevice #3: subdevice #3
>>>>>>>> ?Subdevice #4: subdevice #4
>>>>>>>> ?Subdevice #5: subdevice #5
>>>>>>>> ?Subdevice #6: subdevice #6
>>>>>>>> ?Subdevice #7: subdevice #7
>>>>>>>> ?Subdevice #8: subdevice #8
>>>>>>>> ?Subdevice #9: subdevice #9
>>>>>>>> ?Subdevice #10: subdevice #10
>>>>>>>> ?Subdevice #11: subdevice #11
>>>>>>>> ?Subdevice #12: subdevice #12
>>>>>>>> ?Subdevice #13: subdevice #13
>>>>>>>> ?Subdevice #14: subdevice #14
>>>>>>>> ?Subdevice #15: subdevice #15
>>>>>>>> ?Subdevice #16: subdevice #16
>>>>>>>> ?Subdevice #17: subdevice #17
>>>>>>>> ?Subdevice #18: subdevice #18
>>>>>>>> ?Subdevice #19: subdevice #19
>>>>>>>> ?Subdevice #20: subdevice #20
>>>>>>>> ?Subdevice #21: subdevice #21
>>>>>>>> ?Subdevice #22: subdevice #22
>>>>>>>> ?Subdevice #23: subdevice #23
>>>>>>>> ?Subdevice #24: subdevice #24
>>>>>>>> ?Subdevice #25: subdevice #25
>>>>>>>> ?Subdevice #26: subdevice #26
>>>>>>>> ?Subdevice #27: subdevice #27
>>>>>>>> ?Subdevice #28: subdevice #28
>>>>>>>> ?Subdevice #29: subdevice #29
>>>>>>>> ?Subdevice #30: subdevice #30
>>>>>>>> ?Subdevice #31: subdevice #31
>>>>>>>> ?Subdevice #32: subdevice #32
>>>>>>>> ?Subdevice #33: subdevice #33
>>>>>>>> ?Subdevice #34: subdevice #34
>>>>>>>> ?Subdevice #35: subdevice #35
>>>>>>>> ?Subdevice #36: subdevice #36
>>>>>>>> ?Subdevice #37: subdevice #37
>>>>>>>> ?Subdevice #38: subdevice #38
>>>>>>>> ?Subdevice #39: subdevice #39
>>>>>>>> ?Subdevice #40: subdevice #40
>>>>>>>> ?Subdevice #41: subdevice #41
>>>>>>>> ?Subdevice #42: subdevice #42
>>>>>>>> ?Subdevice #43: subdevice #43
>>>>>>>> ?Subdevice #44: subdevice #44
>>>>>>>> ?Subdevice #45: subdevice #45
>>>>>>>> ?Subdevice #46: subdevice #46
>>>>>>>> ?Subdevice #47: subdevice #47
>>>>>>>> ?Subdevice #48: subdevice #48
>>>>>>>> ?Subdevice #49: subdevice #49
>>>>>>>> ?Subdevice #50: subdevice #50
>>>>>>>> ?Subdevice #51: subdevice #51
>>>>>>>> ?Subdevice #52: subdevice #52
>>>>>>>> ?Subdevice #53: subdevice #53
>>>>>>>> ?Subdevice #54: subdevice #54
>>>>>>>> ?Subdevice #55: subdevice #55
>>>>>>>> ?Subdevice #56: subdevice #56
>>>>>>>> ?Subdevice #57: subdevice #57
>>>>>>>> ?Subdevice #58: subdevice #58
>>>>>>>> ?Subdevice #59: subdevice #59
>>>>>>>> ?Subdevice #60: subdevice #60
>>>>>>>> ?Subdevice #61: subdevice #61
>>>>>>>> ?Subdevice #62: subdevice #62
>>>>>>>> ?Subdevice #63: subdevice #63
>>>>>>>> ?Subdevice #64: subdevice #64
>>>>>>>> ?Subdevice #65: subdevice #65
>>>>>>>> ?Subdevice #66: subdevice #66
>>>>>>>> ?Subdevice #67: subdevice #67
>>>>>>>> ?Subdevice #68: subdevice #68
>>>>>>>> ?Subdevice #69: subdevice #69
>>>>>>>> ?Subdevice #70: subdevice #70
>>>>>>>> ?Subdevice #71: subdevice #71
>>>>>>>> ?Subdevice #72: subdevice #72
>>>>>>>> ?Subdevice #73: subdevice #73
>>>>>>>> ?Subdevice #74: subdevice #74
>>>>>>>> ?Subdevice #75: subdevice #75
>>>>>>>> ?Subdevice #76: subdevice #76
>>>>>>>> ?Subdevice #77: subdevice #77
>>>>>>>> ?Subdevice #78: subdevice #78
>>>>>>>> ?Subdevice #79: subdevice #79
>>>>>>>> ?Subdevice #80: subdevice #80
>>>>>>>> ?Subdevice #81: subdevice #81
>>>>>>>> ?Subdevice #82: subdevice #82
>>>>>>>> ?Subdevice #83: subdevice #83
>>>>>>>> ?Subdevice #84: subdevice #84
>>>>>>>> ?Subdevice #85: subdevice #85
>>>>>>>> ?Subdevice #86: subdevice #86
>>>>>>>> ?Subdevice #87: subdevice #87
>>>>>>>> ?Subdevice #88: subdevice #88
>>>>>>>> ?Subdevice #89: subdevice #89
>>>>>>>> ?Subdevice #90: subdevice #90
>>>>>>>> ?Subdevice #91: subdevice #91
>>>>>>>> ?Subdevice #92: subdevice #92
>>>>>>>> ?Subdevice #93: subdevice #93
>>>>>>>> ?Subdevice #94: subdevice #94
>>>>>>>> ?Subdevice #95: subdevice #95
>>>>>>>> ?Subdevice #96: subdevice #96
>>>>>>>> ?Subdevice #97: subdevice #97
>>>>>>>> ?Subdevice #98: subdevice #98
>>>>>>>> ?Subdevice #99: subdevice #99
>>>>>>>> ?Subdevice #100: subdevice #100
>>>>>>>> ?Subdevice #101: subdevice #101
>>>>>>>> ?Subdevice #102: subdevice #102
>>>>>>>> ?Subdevice #103: subdevice #103
>>>>>>>> ?Subdevice #104: subdevice #104
>>>>>>>> ?Subdevice #105: subdevice #105
>>>>>>>> ?Subdevice #106: subdevice #106
>>>>>>>> ?Subdevice #107: subdevice #107
>>>>>>>> ?Subdevice #108: subdevice #108
>>>>>>>> ?Subdevice #109: subdevice #109
>>>>>>>> ?Subdevice #110: subdevice #110
>>>>>>>> ?Subdevice #111: subdevice #111
>>>>>>>> ?Subdevice #112: subdevice #112
>>>>>>>> ?Subdevice #113: subdevice #113
>>>>>>>> ?Subdevice #114: subdevice #114
>>>>>>>> ?Subdevice #115: subdevice #115
>>>>>>>> ?Subdevice #116: subdevice #116
>>>>>>>> ?Subdevice #117: subdevice #117
>>>>>>>> ?Subdevice #118: subdevice #118
>>>>>>>> ?Subdevice #119: subdevice #119
>>>>>>>> ?Subdevice #120: subdevice #120
>>>>>>>> ?Subdevice #121: subdevice #121
>>>>>>>> ?Subdevice #122: subdevice #122
>>>>>>>> ?Subdevice #123: subdevice #123
>>>>>>>> ?Subdevice #124: subdevice #124
>>>>>>>> ?Subdevice #125: subdevice #125
>>>>>>>> ?Subdevice #126: subdevice #126
>>>>>>>> ?Subdevice #127: subdevice #127
>>>>>>>>
>>>>>>>>
>>>>>>>> Peter P GMX schrieb:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>> Her's the output:
>>>>>>>>>
>>>>>>>>> skype:~# aplay -l
>>>>>>>>> bash: aplay: command not found
>>>>>>>>>
>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>> I don't think you got two snd-dummy loaded (but maybe yes)
>>>>>>>>>> what's the output of:
>>>>>>>>>>
>>>>>>>>>> aplay -l
>>>>>>>>>>
>>>>>>>>>> ?
>>>>>>>>>>
>>>>>>>>>> If instead you are referring to the choices that skype clients offers
>>>>>>>>>> you in the "set audio devices" window, choose Dummy PCM (hw0:0)
>>>>>>>>>>
>>>>>>>>>> Eg: not the "default", but the "hardware" one
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Wed, Jan 27, 2010 at 5:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> Thanks Giovanni,
>>>>>>>>>>>
>>>>>>>>>>> I think there may be the problem, that I have 2 sound devices now:
>>>>>>>>>>> - Dummy PCM (hw0:0) (this is from debian install)
>>>>>>>>>>> - Dummy PCM Default Audio device (defauzlt: CARD=Dummy) (this is new
>>>>>>>>>>> since I compiled alsa newly)
>>>>>>>>>>>
>>>>>>>>>>> I tried both, but both do not work. How do I get rid of the old alsa device?
>>>>>>>>>>> By the way: I uninstalled Alsa before I installed the new driver
>>>>>>>>>>> (apt-get remove alsa-utils alsa-base).
>>>>>>>>>>>
>>>>>>>>>>> Best regards
>>>>>>>>>>> Peter
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Giovanni Maruzzelli schrieb:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>> This warning is harmless:
>>>>>>>>>>>>
>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>
>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:26 PM, Giovanni Maruzzelli
>>>>>>>>>>>> <gmaruzz at celliax.org> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>> Ciao Peter
>>>>>>>>>>>>>
>>>>>>>>>>>>> one instance of snd-dummy "customized" is enough for 64 instances of
>>>>>>>>>>>>> skype clients, no need (and do not works) with more instances of
>>>>>>>>>>>>> snd-dummy-customized.
>>>>>>>>>>>>>
>>>>>>>>>>>>> Maybe you got the one-way problem because of kernel at 250HZ (don't
>>>>>>>>>>>>> know). It uses to works well on a tickless kernel at 100HZ (eg: ubuntu
>>>>>>>>>>>>> 8.04).
>>>>>>>>>>>>>
>>>>>>>>>>>>> Or maybe you have to check and modify which sound devices the skype
>>>>>>>>>>>>> clients are using (try to check that with snd-summy-custom loaded,
>>>>>>>>>>>>> maybe with the ssh -X trick (as in the wiki page).
>>>>>>>>>>>>>
>>>>>>>>>>>>> To load more than one snd-dummy-original (the non modified one), you
>>>>>>>>>>>>> do this with the modprobe command, as in:
>>>>>>>>>>>>>
>>>>>>>>>>>>> rmmod snd-dummy
>>>>>>>>>>>>> modprobe snd-dummy enable=1,1,1
>>>>>>>>>>>>>
>>>>>>>>>>>>> this command will enable three instances of snd-dummy original, so
>>>>>>>>>>>>> you'll have three fake soundcards, and you'll have to setup each group
>>>>>>>>>>>>> of 8 skype instances to use sound devices from one fake soundcard, RG:
>>>>>>>>>>>>> no more than 8 skype client instances can use one instance of fake
>>>>>>>>>>>>> soundcard.
>>>>>>>>>>>>>
>>>>>>>>>>>>> Also, please update the mod_skypiax code (svn up in its directory) I
>>>>>>>>>>>>> just committed some improvements.
>>>>>>>>>>>>>
>>>>>>>>>>>>> If you have any other doubts, or need more info, don't hesitate to
>>>>>>>>>>>>> write the mailing list again,
>>>>>>>>>>>>>
>>>>>>>>>>>>> ciao for now,
>>>>>>>>>>>>>
>>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Wed, Jan 27, 2010 at 4:01 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>> I have mod_skypiax working nicely so far with 2 Skype channels. Thanks
>>>>>>>>>>>>>> to all contributors, excellent work!
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> In order to have more than 8 channels working, I have followed the
>>>>>>>>>>>>>> instructions in
>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>>>>>>>>>>> and compiled alsa-driver-1.0.20 with the modified dummy.c file. (System
>>>>>>>>>>>>>> ist Debian 5.0R3)
>>>>>>>>>>>>>> It compiled well however when I start snd-dummy I only have
>>>>>>>>>>>>>> one-way-audio and my logs show
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Jan 27 15:28:41 skype kernel: [ 3984.318403] snd-dummy skypiax driver,
>>>>>>>>>>>>>> /usr/src/alsa-driver-1.0.20/drivers/../alsa-kernel/drivers/dummy.c:920
>>>>>>>>>>>>>> working on a machine with 250HZ kernel
>>>>>>>>>>>>>> Jan 27 15:28:50 skype kernel: [ 3994.795786] process `skype' is using
>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>> Jan 27 15:28:56 skype kernel: [ 4005.289907] __ratelimit: 490 messages
>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4012.458310] process `skype' is using
>>>>>>>>>>>>>> obsolete setsockopt SO_BSDCOMPAT
>>>>>>>>>>>>>> Jan 27 15:29:01 skype kernel: [ 4013.326290] __ratelimit: 499 messages
>>>>>>>>>>>>>> suppressed
>>>>>>>>>>>>>> If I reinstall alsa from deb everything sworks fine again (of course
>>>>>>>>>>>>>> with the current limitations).
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> First question: Has anybody had this issue before? How can I solve this?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Second question:
>>>>>>>>>>>>>> As I do not need 64 channels or more: how do I manage, that Skype
>>>>>>>>>>>>>> instances 9..15 use a second instance of snd-dummy as addressed in the wiki?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Best regards
>>>>>>>>>>>>>> Peter
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>> --
>>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>>
>>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From q.edward at gmail.com  Fri Jan 29 06:04:23 2010
From: q.edward at gmail.com (Edward Q.)
Date: Fri, 29 Jan 2010 09:04:23 -0500
Subject: [Freeswitch-users] Blind dialed number
Message-ID: <89313a91001290604q500c114cs31ce39f5a4d65922@mail.gmail.com>

Hi guys.
Sorry for been a newbie on all this. And thank you for your help in
advanced.
Here is the scenario I am trying to accomplish.
I would like to have the phone number inside a database not in the
extension.xml file.
Here is what I am trying to do.
Create a let's say 1001.xml file ( if needed )  but that file instead of
having the phone number to be dialed,  go to MySQL and look for the phone
number there. Or when freeswitch is told to call that extension instead of
looking for the 1001.xml file go directly to mysql to look for the phone
number associated to that extension to dial it out.
That way the phone numbers are always kept private at all times.
What we are doing is.
>From the webserver customer clicks on a link -> link dials the phone number
in the xml file automatically but I would like to keep the phone number on
the DB server not in the xml file.
Can anyone point me please in the right direction?
Really appreciated.
Ed
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From brian at freeswitch.org  Fri Jan 29 06:26:53 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 29 Jan 2010 08:26:53 -0600
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
In-Reply-To: <eui5m5hg6tqemri4r5u250hu72mjddmjnt@4ax.com>
References: <27338355.post@talk.nabble.com>
	<27338355.post-WJuSqJV8a7jJsTRiRinrng@public.gmane.org>
	<201001270750.16772.sos@sokhapkin.dyndns.org>
	<eui5m5hg6tqemri4r5u250hu72mjddmjnt@4ax.com>
Message-ID: <57E0B3FB-166A-4CF2-82E8-267D2FADB9DC@freeswitch.org>

Well if nat is involved you might have issues.

/b

On Jan 29, 2010, at 5:52 AM, Fred-145 wrote:

> Are there drawbacks to having RTP pakets flow directly between the SIP
> end-points?




From jmesquita at freeswitch.org  Fri Jan 29 06:33:54 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 29 Jan 2010 12:33:54 -0200
Subject: [Freeswitch-users] Blind dialed number
In-Reply-To: <89313a91001290604q500c114cs31ce39f5a4d65922@mail.gmail.com>
References: <89313a91001290604q500c114cs31ce39f5a4d65922@mail.gmail.com>
Message-ID: <ebbc2d551001290633t57fae340v80a9a82153f28c64@mail.gmail.com>

A couple of options there.

ESL, mod_xml_curl or even mod_python, mod_spidermonkey, lua, etc...

Pick your flavor.

Jo?o Mesquita


On Fri, Jan 29, 2010 at 12:04 PM, Edward Q. <q.edward at gmail.com> wrote:

> Hi guys.
> Sorry for been a newbie on all this. And thank you for your help in
> advanced.
> Here is the scenario I am trying to accomplish.
> I would like to have the phone number inside a database not in the
> extension.xml file.
> Here is what I am trying to do.
> Create a let's say 1001.xml file ( if needed )  but that file instead of
> having the phone number to be dialed,  go to MySQL and look for the phone
> number there. Or when freeswitch is told to call that extension instead of
> looking for the 1001.xml file go directly to mysql to look for the phone
> number associated to that extension to dial it out.
> That way the phone numbers are always kept private at all times.
> What we are doing is.
> From the webserver customer clicks on a link -> link dials the phone number
> in the xml file automatically but I would like to keep the phone number on
> the DB server not in the xml file.
> Can anyone point me please in the right direction?
> Really appreciated.
> Ed
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rob4manhere at gmail.com  Fri Jan 29 06:48:00 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 29 Jan 2010 08:48:00 -0600
Subject: [Freeswitch-users] Blind dialed number
In-Reply-To: <ebbc2d551001290633t57fae340v80a9a82153f28c64@mail.gmail.com>
References: <89313a91001290604q500c114cs31ce39f5a4d65922@mail.gmail.com>
	<ebbc2d551001290633t57fae340v80a9a82153f28c64@mail.gmail.com>
Message-ID: <81B863DC-CF42-4ADE-8F82-EC62B51B03F3@gmail.com>

Agreed- you have lots of options.

A side note though- if your requirement is that the number is kept  
private from freeswitch, you're going to run into issues.  Freeswitch  
will eventually have to know the number in order to call it, and it  
will log its activity... so if there are people who aren't supposed to  
see the number but can see 1001.xml, they could also see the  
freeswitch logs.  They could also open the fs_cli and see activity  
there.  I guess you could lock everything like that down but just  
things to think through if that kind of compartmentalization is a big  
requirement.

Rob

On Jan 29, 2010, at 8:33 AM, Jo?o Mesquita wrote:

> A couple of options there.
>
> ESL, mod_xml_curl or even mod_python, mod_spidermonkey, lua, etc...
>
> Pick your flavor.
>
> Jo?o Mesquita
>
>
> On Fri, Jan 29, 2010 at 12:04 PM, Edward Q. <q.edward at gmail.com>  
> wrote:
> Hi guys.
> Sorry for been a newbie on all this. And thank you for your help in  
> advanced.
> Here is the scenario I am trying to accomplish.
> I would like to have the phone number inside a database not in the  
> extension.xml file.
> Here is what I am trying to do.
> Create a let's say 1001.xml file ( if needed )  but that file  
> instead of having the phone number to be dialed,  go to MySQL and  
> look for the phone number there. Or when freeswitch is told to call  
> that extension instead of looking for the 1001.xml file go directly  
> to mysql to look for the phone number associated to that extension  
> to dial it out.
> That way the phone numbers are always kept private at all times.
> What we are doing is.
> From the webserver customer clicks on a link -> link dials the phone  
> number in the xml file automatically but I would like to keep the  
> phone number on the DB server not in the xml file.
> Can anyone point me please in the right direction?
> Really appreciated.
> Ed
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Fri Jan 29 06:53:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jan 2010 06:53:50 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Agenda
Message-ID: <87f2f3b91001290653x417398aaj8900a4da9b1aa83b@mail.gmail.com>

Greetings,

This week's conference call agenda is posted here:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_29

Please add your items as the agenda is very light this week. We do have a
few things to discuss, though, so please hop on and bring a friend!

Talk to you all soon,
Michael
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From tim at communicatefreely.net  Fri Jan 29 06:59:53 2010
From: tim at communicatefreely.net (Tim St. Pierre)
Date: Fri, 29 Jan 2010 09:59:53 -0500
Subject: [Freeswitch-users] Blind dialed number
In-Reply-To: <89313a91001290604q500c114cs31ce39f5a4d65922@mail.gmail.com>
References: <89313a91001290604q500c114cs31ce39f5a4d65922@mail.gmail.com>
Message-ID: <4B62F7E9.6000202@communicatefreely.net>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Edward,

I am building a dialplan that runs almost everything out of a database, and what seemed easiest was
to use mod_xml_curl and a php script (substitute your favorite language here).

When a call comes in, FS makes a query to the web server.  A PHP script looks at the posted values,
and if it matches certain criteria (I usually match on context first, before I do anything.), a
small piece of XML dialplan is returned.

Usually what I do is return an extension with a pattern match that will always match what is dialed,
along with the actions I want FS to take.  In this way, the PHP script is handling all the decision
making - FS only posts the call data, and the exact action to take is returned.  I'm not using the
XML files for dialplan at all - it is rendered on the fly by PHP.

The other nice thing about this is that it scales very well.  The web server, database, and FS
components can all be on separate boxes if it's a very high volume environment.  mod_xml_curl can be
set up with multiple failover options.

If you want to mix and match text vs. curl XML dialplan, just have some logic in your script that
returns only the root tags if the call doesn't match one of the database driven extensions.  FS will
include the returned result in with the XML dialplan, so the call can match either.

Good luck!

- -Tim

Edward Q. wrote:
> Hi guys.
> Sorry for been a newbie on all this. And thank you for your help in
> advanced.
> Here is the scenario I am trying to accomplish.
> I would like to have the phone number inside a database not in the
> extension.xml file.
> Here is what I am trying to do.
> Create a let's say 1001.xml file ( if needed )  but that file instead of
> having the phone number to be dialed,  go to MySQL and look for the
> phone number there. Or when freeswitch is told to call that extension
> instead of looking for the 1001.xml file go directly to mysql to look
> for the phone number associated to that extension to dial it out.
> That way the phone numbers are always kept private at all times.
> What we are doing is.
> From the webserver customer clicks on a link -> link dials the phone
> number in the xml file automatically but I would like to keep the phone
> number on the DB server not in the xml file.
> Can anyone point me please in the right direction?
> Really appreciated.
> Ed
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

- --
Tim St. Pierre
IP Voice technician
Communicate Freely
1-877-291-8647 x5101
sip:5101 at communicatefreely.net
tim at communicatefreely.net
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From msc at freeswitch.org  Fri Jan 29 07:11:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jan 2010 07:11:07 -0800
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E90BF91C@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E90BD7F4@prato.activenetwerx.local>
	<20100127210836.5ED5D2E3B2D@cuneorg-email.cune.pri>
	<CA5A491E9DEFBE4CB777DE97E21575E90BF91C@prato.activenetwerx.local>
Message-ID: <87f2f3b91001290711t3ee71079wa0674941d8dbc741@mail.gmail.com>

Are you really dialing that phone number or is that just redacted? Try
dialing 919-386-9900 for testing.
-MC

On Wed, Jan 27, 2010 at 6:09 PM, Joseph L. Casale <jcasale at activenetwerx.com
> wrote:

> >Debian 5.0.3
>
> Well, given the time I had tonight, I tried on my CentOS 5.3 box.
> The incoming log is the first block, and an outgoing log is the
> second block at http://pastebin.freeswitch.org/11965
>
> When I call in, I can hear it get answered, as I play the wav file
> I hear the tone go very low, but no sound.
>
> When I try to call out, nothing happens?
>
> Is there anything in the log that might standout from your perspective?
>
> Thanks everyone!
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.villasmil.work at gmail.com  Fri Jan 29 07:22:37 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 29 Jan 2010 16:22:37 +0100
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <5d2828f1001281205u2eccc999re6f25a55b07156c7@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
	<5d2828f1001281205u2eccc999re6f25a55b07156c7@mail.gmail.com>
Message-ID: <9853f4ff1001290722m4007e620sf6db3c61c1495a4b@mail.gmail.com>

Oh yes, here's an example:

function dbConnect()
     -- connect to db
     require "luasql.mysql"
     env = assert(luasql.mysql())
     conn = assert(env:connect("freeswitch","user","userpass","localhost"))
end

function getpin()
     session:streamFile(card_greeting_audio_file)

     card_pin = session:getDigits(4, "#", 3000);

     if card_pin > "" then
          freeswitch.consoleLog("info", "CARD INFO: PIN...........:
".. card_pin .."\n");
          cur = assert(
               conn:execute( "select * from cards_table where pin ="..
card_pin ..";" )
               )

          -- print all rows, the rows will be indexed by field names
          row = cur:fetch ({}, "a")
          fsLog("ROWS: ".. cur:numrows() )
          if cur:numrows() > 0 then pinok=true end
          while row do

               fsLog("CARD INFO: Batch.........: ".. row.batch        )
               fsLog("CARD INFO: Card Name.....: ".. row.card_name    )
               fsLog("CARD INFO: Ratetable.....: ".. row.ratetable    )
               fsLog("CARD INFO: Initial Bal...: ".. row.init_bal     )
               fsLog("CARD INFO: Curr Balance..: ".. row.balance      )

               batch, ratetable, init_bal, balance = row.batch,
row.ratetable, row.init_bal, row.balance

               SetVar("card_pin",card_pin)
               SetVar("card_batch", batch)
               SetVar("card_ratetable", ratetable)
               SetVar("card_init_bal", init_bal)
               SetVar("card_balance", balance)

            -- reusing the table of results
            row = cur:fetch (row, "a")
          end
     else
          pinok=false
          session:streamFile(card_invalid_pin_audio_file)
     end
end


On Thu, Jan 28, 2010 at 9:05 PM, Mike van Lammeren
<mike at van.lammeren.net> wrote:
> And you can make queries against your MySQL database, and get results, etc.?
>
> On Thu, Jan 28, 2010 at 2:43 AM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> Hello,
>>
>> That works fine:
>>
>> box:~# lua testdb.lua
>> box:~#
>>
>>
>> David
>>
>> On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
>> <mike at van.lammeren.net> wrote:
>> > Have you tried running a Lua script that includes the library from
>> > outside
>> > of FreeSWITCH? What does that do?
>> >
>> > On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com>
>> > wrote:
>> >>
>> >> I tried running ldconfig on the directory containing mysql.so, but it
>> >> did
>> >> not help.
>> >> So it sounds like there could be a bug in the latter versions?
>> >>
>> >> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
>> >> <david.villasmil.work at gmail.com> wrote:
>> >>>
>> >>> I got the same error, my script was working with no problems before an
>> >>> update to trunk.
>> >>>
>> >>> David
>> >>>
>> >>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com>
>> >>> wrote:
>> >>> > Hi, I followed the instructions in the Lua documentation for setting
>> >>> > up
>> >>> > luasql, but when I try to run my script I get:
>> >>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading
>> >>> > module
>> >>> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
>> >>> > ?? ? ? ?/usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore
>> >>> > segment
>> >>> > prot
>> >>> > after reloc: Permission denied
>> >>> > stack traceback:
>> >>> > ?? ? ? ?[C]: ?
>> >>> > ?? ? ? ?[C]: in function 'require'
>> >>> > ?? ? ? ?/usr/local/freeswitch/scripts/l.lua:2: in main chunk
>> >>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
>> >>> > I changed the permissions for mysql.so and for my script to 777, so
>> >>> > I'm
>> >>> > not
>> >>> > sure where the permission problem could be.
>> >>> > I'd appreciate any suggestions.
>> >>> > Thanks,
>> >>> > Adam
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> >
>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Fri Jan 29 07:25:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jan 2010 07:25:53 -0800
Subject: [Freeswitch-users] wiki password recovery - no mail is send
In-Reply-To: <4B629D05.1060908@xpirio.com>
References: <4B629D05.1060908@xpirio.com>
Message-ID: <87f2f3b91001290725o13dde05fq9426d8ef45e46035@mail.gmail.com>

Email me off list and I'll help you with this. I've never been able to
reproduce this symptom but some others have. In any case I will assist you
with getting your wiki account updated.
-MC

2010/1/29 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> hello
>
> the password recovery for the fs wiki doesn't seem to work.
> no e-mail is send when entering the username and press "e-mail new
> password".
>
> may i assist here, we do maintain a few wikis for ourself.
>
> br
>
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Jan 29 07:30:55 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jan 2010 07:30:55 -0800
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <9853f4ff1001290722m4007e620sf6db3c61c1495a4b@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
	<5d2828f1001281205u2eccc999re6f25a55b07156c7@mail.gmail.com>
	<9853f4ff1001290722m4007e620sf6db3c61c1495a4b@mail.gmail.com>
Message-ID: <87f2f3b91001290730y652086far4008036c47beed89@mail.gmail.com>

David,

Are you using Lua and lusql for some exotic call handling scenarios? If so,
would you mind posting some examples to the wiki and then linking here?
Also, if you can join the community conference call today that would be
great!

Thanks,
MC

On Fri, Jan 29, 2010 at 7:22 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Oh yes, here's an example:
>
> function dbConnect()
>     -- connect to db
>     require "luasql.mysql"
>     env = assert(luasql.mysql())
>     conn = assert(env:connect("freeswitch","user","userpass","localhost"))
> end
>
> function getpin()
>     session:streamFile(card_greeting_audio_file)
>
>     card_pin = session:getDigits(4, "#", 3000);
>
>     if card_pin > "" then
>          freeswitch.consoleLog("info", "CARD INFO: PIN...........:
> ".. card_pin .."\n");
>          cur = assert(
>               conn:execute( "select * from cards_table where pin ="..
> card_pin ..";" )
>               )
>
>          -- print all rows, the rows will be indexed by field names
>          row = cur:fetch ({}, "a")
>          fsLog("ROWS: ".. cur:numrows() )
>          if cur:numrows() > 0 then pinok=true end
>          while row do
>
>               fsLog("CARD INFO: Batch.........: ".. row.batch        )
>               fsLog("CARD INFO: Card Name.....: ".. row.card_name    )
>               fsLog("CARD INFO: Ratetable.....: ".. row.ratetable    )
>               fsLog("CARD INFO: Initial Bal...: ".. row.init_bal     )
>               fsLog("CARD INFO: Curr Balance..: ".. row.balance      )
>
>               batch, ratetable, init_bal, balance = row.batch,
> row.ratetable, row.init_bal, row.balance
>
>               SetVar("card_pin",card_pin)
>               SetVar("card_batch", batch)
>               SetVar("card_ratetable", ratetable)
>               SetVar("card_init_bal", init_bal)
>               SetVar("card_balance", balance)
>
>            -- reusing the table of results
>            row = cur:fetch (row, "a")
>          end
>     else
>          pinok=false
>          session:streamFile(card_invalid_pin_audio_file)
>     end
> end
>
>
> On Thu, Jan 28, 2010 at 9:05 PM, Mike van Lammeren
> <mike at van.lammeren.net> wrote:
> > And you can make queries against your MySQL database, and get results,
> etc.?
> >
> > On Thu, Jan 28, 2010 at 2:43 AM, David Villasmil
> > <david.villasmil.work at gmail.com> wrote:
> >>
> >> Hello,
> >>
> >> That works fine:
> >>
> >> box:~# lua testdb.lua
> >> box:~#
> >>
> >>
> >> David
> >>
> >> On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
> >> <mike at van.lammeren.net> wrote:
> >> > Have you tried running a Lua script that includes the library from
> >> > outside
> >> > of FreeSWITCH? What does that do?
> >> >
> >> > On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com>
> >> > wrote:
> >> >>
> >> >> I tried running ldconfig on the directory containing mysql.so, but it
> >> >> did
> >> >> not help.
> >> >> So it sounds like there could be a bug in the latter versions?
> >> >>
> >> >> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
> >> >> <david.villasmil.work at gmail.com> wrote:
> >> >>>
> >> >>> I got the same error, my script was working with no problems before
> an
> >> >>> update to trunk.
> >> >>>
> >> >>> David
> >> >>>
> >> >>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com>
> >> >>> wrote:
> >> >>> > Hi, I followed the instructions in the Lua documentation for
> setting
> >> >>> > up
> >> >>> > luasql, but when I try to run my script I get:
> >> >>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading
> >> >>> > module
> >> >>> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
> >> >>> >         /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore
> >> >>> > segment
> >> >>> > prot
> >> >>> > after reloc: Permission denied
> >> >>> > stack traceback:
> >> >>> >         [C]: ?
> >> >>> >         [C]: in function 'require'
> >> >>> >         /usr/local/freeswitch/scripts/l.lua:2: in main chunk
> >> >>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
> >> >>> > I changed the permissions for mysql.so and for my script to 777,
> so
> >> >>> > I'm
> >> >>> > not
> >> >>> > sure where the permission problem could be.
> >> >>> > I'd appreciate any suggestions.
> >> >>> > Thanks,
> >> >>> > Adam
> >> >>> >
> >> >>> > _______________________________________________
> >> >>> > FreeSWITCH-users mailing list
> >> >>> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >
> >> >>> >
> >> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> > http://www.freeswitch.org
> >> >>> >
> >> >>> >
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.villasmil.work at gmail.com  Fri Jan 29 07:51:36 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 29 Jan 2010 16:51:36 +0100
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <87f2f3b91001290730y652086far4008036c47beed89@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
	<5d2828f1001281205u2eccc999re6f25a55b07156c7@mail.gmail.com>
	<9853f4ff1001290722m4007e620sf6db3c61c1495a4b@mail.gmail.com>
	<87f2f3b91001290730y652086far4008036c47beed89@mail.gmail.com>
Message-ID: <9853f4ff1001290751q29c12e58oc550bd37fbf225c9@mail.gmail.com>

Today what time/timezone?

On Fri, Jan 29, 2010 at 4:30 PM, Michael Collins <msc at freeswitch.org> wrote:
> David,
>
> Are you using Lua and lusql for some exotic call handling scenarios? If so,
> would you mind posting some examples to the wiki and then linking here?
> Also, if you can join the community conference call today that would be
> great!
>
> Thanks,
> MC
>
> On Fri, Jan 29, 2010 at 7:22 AM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> Oh yes, here's an example:
>>
>> function dbConnect()
>> ? ? -- connect to db
>> ? ? require "luasql.mysql"
>> ? ? env = assert(luasql.mysql())
>> ? ? conn = assert(env:connect("freeswitch","user","userpass","localhost"))
>> end
>>
>> function getpin()
>> ? ? session:streamFile(card_greeting_audio_file)
>>
>> ? ? card_pin = session:getDigits(4, "#", 3000);
>>
>> ? ? if card_pin > "" then
>> ? ? ? ? ?freeswitch.consoleLog("info", "CARD INFO: PIN...........:
>> ".. card_pin .."\n");
>> ? ? ? ? ?cur = assert(
>> ? ? ? ? ? ? ? conn:execute( "select * from cards_table where pin ="..
>> card_pin ..";" )
>> ? ? ? ? ? ? ? )
>>
>> ? ? ? ? ?-- print all rows, the rows will be indexed by field names
>> ? ? ? ? ?row = cur:fetch ({}, "a")
>> ? ? ? ? ?fsLog("ROWS: ".. cur:numrows() )
>> ? ? ? ? ?if cur:numrows() > 0 then pinok=true end
>> ? ? ? ? ?while row do
>>
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Batch.........: ".. row.batch ? ? ? ?)
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Card Name.....: ".. row.card_name ? ?)
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Ratetable.....: ".. row.ratetable ? ?)
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Initial Bal...: ".. row.init_bal ? ? )
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Curr Balance..: ".. row.balance ? ? ?)
>>
>> ? ? ? ? ? ? ? batch, ratetable, init_bal, balance = row.batch,
>> row.ratetable, row.init_bal, row.balance
>>
>> ? ? ? ? ? ? ? SetVar("card_pin",card_pin)
>> ? ? ? ? ? ? ? SetVar("card_batch", batch)
>> ? ? ? ? ? ? ? SetVar("card_ratetable", ratetable)
>> ? ? ? ? ? ? ? SetVar("card_init_bal", init_bal)
>> ? ? ? ? ? ? ? SetVar("card_balance", balance)
>>
>> ? ? ? ? ? ?-- reusing the table of results
>> ? ? ? ? ? ?row = cur:fetch (row, "a")
>> ? ? ? ? ?end
>> ? ? else
>> ? ? ? ? ?pinok=false
>> ? ? ? ? ?session:streamFile(card_invalid_pin_audio_file)
>> ? ? end
>> end
>>
>>
>> On Thu, Jan 28, 2010 at 9:05 PM, Mike van Lammeren
>> <mike at van.lammeren.net> wrote:
>> > And you can make queries against your MySQL database, and get results,
>> > etc.?
>> >
>> > On Thu, Jan 28, 2010 at 2:43 AM, David Villasmil
>> > <david.villasmil.work at gmail.com> wrote:
>> >>
>> >> Hello,
>> >>
>> >> That works fine:
>> >>
>> >> box:~# lua testdb.lua
>> >> box:~#
>> >>
>> >>
>> >> David
>> >>
>> >> On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
>> >> <mike at van.lammeren.net> wrote:
>> >> > Have you tried running a Lua script that includes the library from
>> >> > outside
>> >> > of FreeSWITCH? What does that do?
>> >> >
>> >> > On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> I tried running ldconfig on the directory containing mysql.so, but
>> >> >> it
>> >> >> did
>> >> >> not help.
>> >> >> So it sounds like there could be a bug in the latter versions?
>> >> >>
>> >> >> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
>> >> >> <david.villasmil.work at gmail.com> wrote:
>> >> >>>
>> >> >>> I got the same error, my script was working with no problems before
>> >> >>> an
>> >> >>> update to trunk.
>> >> >>>
>> >> >>> David
>> >> >>>
>> >> >>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com>
>> >> >>> wrote:
>> >> >>> > Hi, I followed the instructions in the Lua documentation for
>> >> >>> > setting
>> >> >>> > up
>> >> >>> > luasql, but when I try to run my script I get:
>> >> >>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading
>> >> >>> > module
>> >> >>> > 'luasql.mysql' from file
>> >> >>> > '/usr/local/lib/lua/5.1/luasql/mysql.so':
>> >> >>> > ?? ? ? ?/usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore
>> >> >>> > segment
>> >> >>> > prot
>> >> >>> > after reloc: Permission denied
>> >> >>> > stack traceback:
>> >> >>> > ?? ? ? ?[C]: ?
>> >> >>> > ?? ? ? ?[C]: in function 'require'
>> >> >>> > ?? ? ? ?/usr/local/freeswitch/scripts/l.lua:2: in main chunk
>> >> >>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
>> >> >>> > I changed the permissions for mysql.so and for my script to 777,
>> >> >>> > so
>> >> >>> > I'm
>> >> >>> > not
>> >> >>> > sure where the permission problem could be.
>> >> >>> > I'd appreciate any suggestions.
>> >> >>> > Thanks,
>> >> >>> > Adam
>> >> >>> >
>> >> >>> > _______________________________________________
>> >> >>> > FreeSWITCH-users mailing list
>> >> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>> >
>> >> >>> >
>> >> >>> >
>> >> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> > http://www.freeswitch.org
>> >> >>> >
>> >> >>> >
>> >> >>>
>> >> >>> _______________________________________________
>> >> >>> FreeSWITCH-users mailing list
>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>
>> >> >>>
>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> http://www.freeswitch.org
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From freeswitch-list at puzzled.xs4all.nl  Fri Jan 29 07:56:45 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick)
Date: Fri, 29 Jan 2010 16:56:45 +0100
Subject: [Freeswitch-users] wiki password recovery - no mail is send
In-Reply-To: <87f2f3b91001290725o13dde05fq9426d8ef45e46035@mail.gmail.com>
References: <4B629D05.1060908@xpirio.com>
	<87f2f3b91001290725o13dde05fq9426d8ef45e46035@mail.gmail.com>
Message-ID: <4B63053D.20805@puzzled.xs4all.nl>

On 01/29/2010 04:25 PM, Michael Collins wrote:
> Email me off list and I'll help you with this. I've never been able to
> reproduce this symptom but some others have. In any case I will assist
> you with getting your wiki account updated.
> -MC

I have seen this happening if the receiving mailserver uses greylisting
and the email is sent directly by the Wiki app. If all the Wiki does is
fire email and forget then it will not try to resend after being
greylisted. The solution would be to have the Wiki send the email to a
local MTA which takes care of delivering it.

Regards,
Patrick



From rob4manhere at gmail.com  Fri Jan 29 07:59:08 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 29 Jan 2010 09:59:08 -0600
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <9853f4ff1001290751q29c12e58oc550bd37fbf225c9@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
	<5d2828f1001281205u2eccc999re6f25a55b07156c7@mail.gmail.com>
	<9853f4ff1001290722m4007e620sf6db3c61c1495a4b@mail.gmail.com>
	<87f2f3b91001290730y652086far4008036c47beed89@mail.gmail.com>
	<9853f4ff1001290751q29c12e58oc550bd37fbf225c9@mail.gmail.com>
Message-ID: <EDB39B7D-87D2-40F4-B18F-EDBFA77ADFBB@gmail.com>

http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_29


Calling Instructions

Friday January 29 at 1700 UTC (1100 CST)


sip:888 at conference.freeswitch.org or via the good old PSTN at  
+1-919-386-9900
Or click on this link
Or call Skype the skype user "skypiax5", then press "1" on the Skype  
dialpad (max 20 concurrent users).
Codecs: PCMU/PCMA, G.722, CELT, Speex, Skype, among others




On Jan 29, 2010, at 9:51 AM, David Villasmil wrote:

> Today what time/timezone?
>
> On Fri, Jan 29, 2010 at 4:30 PM, Michael Collins  
> <msc at freeswitch.org> wrote:
>> David,
>>
>> Are you using Lua and lusql for some exotic call handling  
>> scenarios? If so,
>> would you mind posting some examples to the wiki and then linking  
>> here?
>> Also, if you can join the community conference call today that  
>> would be
>> great!
>>
>> Thanks,
>> MC
>>
>> On Fri, Jan 29, 2010 at 7:22 AM, David Villasmil
>> <david.villasmil.work at gmail.com> wrote:
>>>
>>> Oh yes, here's an example:
>>>
>>> function dbConnect()
>>>     -- connect to db
>>>     require "luasql.mysql"
>>>     env = assert(luasql.mysql())
>>>     conn =  
>>> assert(env:connect("freeswitch","user","userpass","localhost"))
>>> end
>>>
>>> function getpin()
>>>     session:streamFile(card_greeting_audio_file)
>>>
>>>     card_pin = session:getDigits(4, "#", 3000);
>>>
>>>     if card_pin > "" then
>>>          freeswitch.consoleLog("info", "CARD INFO: PIN...........:
>>> ".. card_pin .."\n");
>>>          cur = assert(
>>>               conn:execute( "select * from cards_table where pin  
>>> ="..
>>> card_pin ..";" )
>>>               )
>>>
>>>          -- print all rows, the rows will be indexed by field names
>>>          row = cur:fetch ({}, "a")
>>>          fsLog("ROWS: ".. cur:numrows() )
>>>          if cur:numrows() > 0 then pinok=true end
>>>          while row do
>>>
>>>               fsLog("CARD INFO: Batch.........: "..  
>>> row.batch        )
>>>               fsLog("CARD INFO: Card Name.....: "..  
>>> row.card_name    )
>>>               fsLog("CARD INFO: Ratetable.....: "..  
>>> row.ratetable    )
>>>               fsLog("CARD INFO: Initial Bal...: "..  
>>> row.init_bal     )
>>>               fsLog("CARD INFO: Curr Balance..: "..  
>>> row.balance      )
>>>
>>>               batch, ratetable, init_bal, balance = row.batch,
>>> row.ratetable, row.init_bal, row.balance
>>>
>>>               SetVar("card_pin",card_pin)
>>>               SetVar("card_batch", batch)
>>>               SetVar("card_ratetable", ratetable)
>>>               SetVar("card_init_bal", init_bal)
>>>               SetVar("card_balance", balance)
>>>
>>>            -- reusing the table of results
>>>            row = cur:fetch (row, "a")
>>>          end
>>>     else
>>>          pinok=false
>>>          session:streamFile(card_invalid_pin_audio_file)
>>>     end
>>> end
>>>
>>>
>>> On Thu, Jan 28, 2010 at 9:05 PM, Mike van Lammeren
>>> <mike at van.lammeren.net> wrote:
>>>> And you can make queries against your MySQL database, and get  
>>>> results,
>>>> etc.?
>>>>
>>>> On Thu, Jan 28, 2010 at 2:43 AM, David Villasmil
>>>> <david.villasmil.work at gmail.com> wrote:
>>>>>
>>>>> Hello,
>>>>>
>>>>> That works fine:
>>>>>
>>>>> box:~# lua testdb.lua
>>>>> box:~#
>>>>>
>>>>>
>>>>> David
>>>>>
>>>>> On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
>>>>> <mike at van.lammeren.net> wrote:
>>>>>> Have you tried running a Lua script that includes the library  
>>>>>> from
>>>>>> outside
>>>>>> of FreeSWITCH? What does that do?
>>>>>>
>>>>>> On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com 
>>>>>> >
>>>>>> wrote:
>>>>>>>
>>>>>>> I tried running ldconfig on the directory containing mysql.so,  
>>>>>>> but
>>>>>>> it
>>>>>>> did
>>>>>>> not help.
>>>>>>> So it sounds like there could be a bug in the latter versions?
>>>>>>>
>>>>>>> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
>>>>>>> <david.villasmil.work at gmail.com> wrote:
>>>>>>>>
>>>>>>>> I got the same error, my script was working with no problems  
>>>>>>>> before
>>>>>>>> an
>>>>>>>> update to trunk.
>>>>>>>>
>>>>>>>> David
>>>>>>>>
>>>>>>>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com 
>>>>>>>> >
>>>>>>>> wrote:
>>>>>>>>> Hi, I followed the instructions in the Lua documentation for
>>>>>>>>> setting
>>>>>>>>> up
>>>>>>>>> luasql, but when I try to run my script I get:
>>>>>>>>> 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading
>>>>>>>>> module
>>>>>>>>> 'luasql.mysql' from file
>>>>>>>>> '/usr/local/lib/lua/5.1/luasql/mysql.so':
>>>>>>>>>         /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore
>>>>>>>>> segment
>>>>>>>>> prot
>>>>>>>>> after reloc: Permission denied
>>>>>>>>> stack traceback:
>>>>>>>>>         [C]: ?
>>>>>>>>>         [C]: in function 'require'
>>>>>>>>>         /usr/local/freeswitch/scripts/l.lua:2: in main chunk
>>>>>>>>> I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
>>>>>>>>> I changed the permissions for mysql.so and for my script to  
>>>>>>>>> 777,
>>>>>>>>> so
>>>>>>>>> I'm
>>>>>>>>> not
>>>>>>>>> sure where the permission problem could be.
>>>>>>>>> I'd appreciate any suggestions.
>>>>>>>>> Thanks,
>>>>>>>>> Adam
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>
>>>>>>>>
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>>
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From william.suffill at gmail.com  Fri Jan 29 08:02:06 2010
From: william.suffill at gmail.com (William Suffill)
Date: Fri, 29 Jan 2010 11:02:06 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <9853f4ff1001290751q29c12e58oc550bd37fbf225c9@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com> 
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com> 
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com> 
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com> 
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com> 
	<5d2828f1001281205u2eccc999re6f25a55b07156c7@mail.gmail.com> 
	<9853f4ff1001290722m4007e620sf6db3c61c1495a4b@mail.gmail.com> 
	<87f2f3b91001290730y652086far4008036c47beed89@mail.gmail.com> 
	<9853f4ff1001290751q29c12e58oc550bd37fbf225c9@mail.gmail.com>
Message-ID: <6b65470d1001290802q4f92b1e8wc60e8b61615be812@mail.gmail.com>

Friday January 29 at 1700 UTC (1100 CST)
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org> or
via the good old PSTN at +1-919-386-9900
Or click on this link<http://www.phonefromhere.com/freeswitch/freeswitch.xsql>
Or call Skype the skype user "skypiax5", then press "1" on the Skype dialpad
(max 20 concurrent users).
Codecs: PCMU/PCMA, G.722, CELT, Speex, Skype, among others


More info here http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_29
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From jcasale at activenetwerx.com  Fri Jan 29 08:21:34 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 29 Jan 2010 16:21:34 +0000
Subject: [Freeswitch-users] Openzap w/ DAHDi Linux 2.2.1
In-Reply-To: <87f2f3b91001290711t3ee71079wa0674941d8dbc741@mail.gmail.com>
References: <CA5A491E9DEFBE4CB777DE97E21575E90BD7F4@prato.activenetwerx.local>
	<20100127210836.5ED5D2E3B2D@cuneorg-email.cune.pri>
	<CA5A491E9DEFBE4CB777DE97E21575E90BF91C@prato.activenetwerx.local>
	<87f2f3b91001290711t3ee71079wa0674941d8dbc741@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E90C2546@prato.activenetwerx.local>

>Are you really dialing that phone number or is that just redacted? Try dialing 919-386-9900 for testing.

Hi Michael,
It was redacted. I can get back in at night to continue testing, but
given incoming doesn't work I doubt I will have any luck. The dialplan
to bypass the sip provider requires a 9, then the 10 digit #.

On the incoming call, I see the two errors:
[ERR] zap_io.c:1599 I/O backend does not support command 2[4|5]!

Any idea what that pertains to? From there on, everything looks
positive as it find the dialplan and plays the wave, I just can't
hear anything at all.

Even though this card doesn't have a hw ec, w/ mg2 or hpec the quality
was flawless. I'd love to have it running again.

For the record, using the 4630 rev of zaptel as per
http://wiki.freeswitch.org/wiki/Zaptel_Tutorial
I was able to get some success, but it was very intermittent.

Thanks for any insight,
jlc



From codecomplete at free.fr  Fri Jan 29 08:21:52 2010
From: codecomplete at free.fr (Fred-145)
Date: Fri, 29 Jan 2010 17:21:52 +0100
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
References: <27338355.post@talk.nabble.com>
	<27338355.post-WJuSqJV8a7jJsTRiRinrng@public.gmane.org>
	<201001270750.16772.sos@sokhapkin.dyndns.org>
	<eui5m5hg6tqemri4r5u250hu72mjddmjnt@4ax.com>
	<eui5m5hg6tqemri4r5u250hu72mjddmjnt-e09XROE/p8c@public.gmane.org>
	<57E0B3FB-166A-4CF2-82E8-267D2FADB9DC@freeswitch.org>
Message-ID: <gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b@4ax.com>

On Fri, 29 Jan 2010 08:26:53 -0600, Brian West
<brian at freeswitch.org> wrote:
>> Are there drawbacks to having RTP pakets flow directly between the SIP
>> end-points?
>
>Well if nat is involved you might have issues.

Since Freeswitch has handled the initial connection and the NAT box
has opened the range of UDP ports for RTP/RTCP, in what case would the
two end-points have a problem with NAT? Is it possible for end-points
to send IP/port information in the midst of a conversation?




From david.villasmil.work at gmail.com  Fri Jan 29 08:23:45 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 29 Jan 2010 17:23:45 +0100
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <87f2f3b91001290730y652086far4008036c47beed89@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<5d2828f1001272227l2a3a409dvd74b4fc5d4880a1@mail.gmail.com>
	<9853f4ff1001272343i5dc95e86t7cd7d76e368eb83c@mail.gmail.com>
	<5d2828f1001281205u2eccc999re6f25a55b07156c7@mail.gmail.com>
	<9853f4ff1001290722m4007e620sf6db3c61c1495a4b@mail.gmail.com>
	<87f2f3b91001290730y652086far4008036c47beed89@mail.gmail.com>
Message-ID: <9853f4ff1001290823l5ef053e2j2feaf1190a23d834@mail.gmail.com>

Where in the wiki?

On Fri, Jan 29, 2010 at 4:30 PM, Michael Collins <msc at freeswitch.org> wrote:
> David,
>
> Are you using Lua and lusql for some exotic call handling scenarios? If so,
> would you mind posting some examples to the wiki and then linking here?
> Also, if you can join the community conference call today that would be
> great!
>
> Thanks,
> MC
>
> On Fri, Jan 29, 2010 at 7:22 AM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>>
>> Oh yes, here's an example:
>>
>> function dbConnect()
>> ? ? -- connect to db
>> ? ? require "luasql.mysql"
>> ? ? env = assert(luasql.mysql())
>> ? ? conn = assert(env:connect("freeswitch","user","userpass","localhost"))
>> end
>>
>> function getpin()
>> ? ? session:streamFile(card_greeting_audio_file)
>>
>> ? ? card_pin = session:getDigits(4, "#", 3000);
>>
>> ? ? if card_pin > "" then
>> ? ? ? ? ?freeswitch.consoleLog("info", "CARD INFO: PIN...........:
>> ".. card_pin .."\n");
>> ? ? ? ? ?cur = assert(
>> ? ? ? ? ? ? ? conn:execute( "select * from cards_table where pin ="..
>> card_pin ..";" )
>> ? ? ? ? ? ? ? )
>>
>> ? ? ? ? ?-- print all rows, the rows will be indexed by field names
>> ? ? ? ? ?row = cur:fetch ({}, "a")
>> ? ? ? ? ?fsLog("ROWS: ".. cur:numrows() )
>> ? ? ? ? ?if cur:numrows() > 0 then pinok=true end
>> ? ? ? ? ?while row do
>>
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Batch.........: ".. row.batch ? ? ? ?)
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Card Name.....: ".. row.card_name ? ?)
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Ratetable.....: ".. row.ratetable ? ?)
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Initial Bal...: ".. row.init_bal ? ? )
>> ? ? ? ? ? ? ? fsLog("CARD INFO: Curr Balance..: ".. row.balance ? ? ?)
>>
>> ? ? ? ? ? ? ? batch, ratetable, init_bal, balance = row.batch,
>> row.ratetable, row.init_bal, row.balance
>>
>> ? ? ? ? ? ? ? SetVar("card_pin",card_pin)
>> ? ? ? ? ? ? ? SetVar("card_batch", batch)
>> ? ? ? ? ? ? ? SetVar("card_ratetable", ratetable)
>> ? ? ? ? ? ? ? SetVar("card_init_bal", init_bal)
>> ? ? ? ? ? ? ? SetVar("card_balance", balance)
>>
>> ? ? ? ? ? ?-- reusing the table of results
>> ? ? ? ? ? ?row = cur:fetch (row, "a")
>> ? ? ? ? ?end
>> ? ? else
>> ? ? ? ? ?pinok=false
>> ? ? ? ? ?session:streamFile(card_invalid_pin_audio_file)
>> ? ? end
>> end
>>
>>
>> On Thu, Jan 28, 2010 at 9:05 PM, Mike van Lammeren
>> <mike at van.lammeren.net> wrote:
>> > And you can make queries against your MySQL database, and get results,
>> > etc.?
>> >
>> > On Thu, Jan 28, 2010 at 2:43 AM, David Villasmil
>> > <david.villasmil.work at gmail.com> wrote:
>> >>
>> >> Hello,
>> >>
>> >> That works fine:
>> >>
>> >> box:~# lua testdb.lua
>> >> box:~#
>> >>
>> >>
>> >> David
>> >>
>> >> On Thu, Jan 28, 2010 at 7:27 AM, Mike van Lammeren
>> >> <mike at van.lammeren.net> wrote:
>> >> > Have you tried running a Lua script that includes the library from
>> >> > outside
>> >> > of FreeSWITCH? What does that do?
>> >> >
>> >> > On Wed, Jan 27, 2010 at 11:47 PM, Adam Wilt <wiltingtree at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> I tried running ldconfig on the directory containing mysql.so, but
>> >> >> it
>> >> >> did
>> >> >> not help.
>> >> >> So it sounds like there could be a bug in the latter versions?
>> >> >>
>> >> >> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil
>> >> >> <david.villasmil.work at gmail.com> wrote:
>> >> >>>
>> >> >>> I got the same error, my script was working with no problems before
>> >> >>> an
>> >> >>> update to trunk.
>> >> >>>
>> >> >>> David
>> >> >>>
>> >> >>> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com>
>> >> >>> wrote:
>> >> >>> > Hi, I followed the instructions in the Lua documentation for
>> >> >>> > setting
>> >> >>> > up
>> >> >>> > luasql, but when I try to run my script I get:
>> >> >>> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading
>> >> >>> > module
>> >> >>> > 'luasql.mysql' from file
>> >> >>> > '/usr/local/lib/lua/5.1/luasql/mysql.so':
>> >> >>> > ?? ? ? ?/usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore
>> >> >>> > segment
>> >> >>> > prot
>> >> >>> > after reloc: Permission denied
>> >> >>> > stack traceback:
>> >> >>> > ?? ? ? ?[C]: ?
>> >> >>> > ?? ? ? ?[C]: in function 'require'
>> >> >>> > ?? ? ? ?/usr/local/freeswitch/scripts/l.lua:2: in main chunk
>> >> >>> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
>> >> >>> > I changed the permissions for mysql.so and for my script to 777,
>> >> >>> > so
>> >> >>> > I'm
>> >> >>> > not
>> >> >>> > sure where the permission problem could be.
>> >> >>> > I'd appreciate any suggestions.
>> >> >>> > Thanks,
>> >> >>> > Adam
>> >> >>> >
>> >> >>> > _______________________________________________
>> >> >>> > FreeSWITCH-users mailing list
>> >> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>> >
>> >> >>> >
>> >> >>> >
>> >> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> > http://www.freeswitch.org
>> >> >>> >
>> >> >>> >
>> >> >>>
>> >> >>> _______________________________________________
>> >> >>> FreeSWITCH-users mailing list
>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>
>> >> >>>
>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> http://www.freeswitch.org
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Fri Jan 29 08:38:01 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jan 2010 08:38:01 -0800
Subject: [Freeswitch-users] wiki password recovery - no mail is send
In-Reply-To: <4B63053D.20805@puzzled.xs4all.nl>
References: <4B629D05.1060908@xpirio.com>
	<87f2f3b91001290725o13dde05fq9426d8ef45e46035@mail.gmail.com>
	<4B63053D.20805@puzzled.xs4all.nl>
Message-ID: <87f2f3b91001290838m337ea44dwe7e3ea7e9c3f1ac2@mail.gmail.com>

On Fri, Jan 29, 2010 at 7:56 AM, Patrick
<freeswitch-list at puzzled.xs4all.nl>wrote:

> On 01/29/2010 04:25 PM, Michael Collins wrote:
> > Email me off list and I'll help you with this. I've never been able to
> > reproduce this symptom but some others have. In any case I will assist
> > you with getting your wiki account updated.
> > -MC
>
> I have seen this happening if the receiving mailserver uses greylisting
> and the email is sent directly by the Wiki app. If all the Wiki does is
> fire email and forget then it will not try to resend after being
> greylisted. The solution would be to have the Wiki send the email to a
> local MTA which takes care of delivering it.
>
> Sounds like a job for ... Super Raymond! :D
-MC
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From Russell.Mosemann at cune.org  Fri Jan 29 08:43:59 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 29 Jan 2010 16:43:59 -0000
Subject: [Freeswitch-users] Equivalent to
	Asterisk's	"directrtpsetup=yes"?
In-Reply-To: <gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b@4ax.com>
Message-ID: <20100129164359.9B3C22B10D@cuneorg-email.cune.pri>

Fred-145 <codecomplete at free.fr> said:

> Since Freeswitch has handled the initial connection and the NAT box
> has opened the range of UDP ports for RTP/RTCP, in what case would the
> two end-points have a problem with NAT? 

The ports are open between the endpoint and Freeswitch. The ports are not
open between the two endpoints themselves. If each endpoint is behind its
own NAT, neither endpoint will be able to contact the other endpoint
unless some kind of forwarding is set up on the firewall to map the
external IP address and port to an internal IP address and port.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From brian at freeswitch.org  Fri Jan 29 09:04:07 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 29 Jan 2010 11:04:07 -0600
Subject: [Freeswitch-users] wiki password recovery - no mail is send
In-Reply-To: <87f2f3b91001290838m337ea44dwe7e3ea7e9c3f1ac2@mail.gmail.com>
References: <4B629D05.1060908@xpirio.com>
	<87f2f3b91001290725o13dde05fq9426d8ef45e46035@mail.gmail.com>
	<4B63053D.20805@puzzled.xs4all.nl>
	<87f2f3b91001290838m337ea44dwe7e3ea7e9c3f1ac2@mail.gmail.com>
Message-ID: <9F5F7247-3ED1-497E-8DDB-0ED43C91158B@freeswitch.org>

aka Mud Puddle!

/b

On Jan 29, 2010, at 10:38 AM, Michael Collins wrote:

> Sounds like a job for ... Super Raymond! :D
> -MC




From ranjtech at gmail.com  Fri Jan 29 09:57:46 2010
From: ranjtech at gmail.com (RR)
Date: Fri, 29 Jan 2010 12:57:46 -0500
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <025701caa0a7$e1ca6200$a55f2600$@com>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
Message-ID: <02b001caa10c$913b9100$b3b2b300$@com>

Any ideas? Anyone? All I can tell is that the Contact Header looks a bit
dodgey but how that's causing a 409 is what I don't get. 

 

Help please?

\RR

 

From: RR [mailto:ranjtech at gmail.com] 
Sent: Friday, January 29, 2010 12:57 AM
To: freeswitch-users at lists.freeswitch.org
Subject: RE: [Freeswitch-users] 409 Conflict When registering FS with
Softswitch

 

Hi Brian,

 

Ok here's the sip trace captured at the softswitch. 

 

BTW, I noticed that during startup, I see FS printing out this message:

2010-01-29 01:30:05.634220 [NOTICE] sofia_reg.c:2267 Added gateway
'Test-Inbound' to profile 'external'

2010-01-29 01:30:05.634547 [ERR] sofia.c:1663 ERROR: username param is
REQUIRED!

2010-01-29 01:30:05.634657 [ERR] sofia.c:1663 ERROR: username param is
REQUIRED!

 

Which is weird, because as you can see from the config, the username is
infact present. Weird! Anyway, here's the trace

 

REGISTER sip:myswitch.net.au SIP/2.0

Via: SIP/2.0/UDP 173.xxx.xxx.xxx:5080;rport;branch=z9hG4bK1U32gpr9vj5eF

Max-Forwards: 70

From: <sip:user1 at myswitch.net.au;transport=udp>;tag=11Qey4tcrUH9g

To: <sip:user1 at myswitch.net.au;transport=udp>

Call-ID: cd589b73-031e-445f-a4d8-2fe334d81bbc

CSeq: 126255215 REGISTER

Contact:
<sip:gw+Test-Inbound at 173.xxx.xxx.xxx:5080;transport=udp;gw=Test-Inbound>

Expires: 60

User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15980

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY

Supported: timer, precondition, path, replaces

Content-Length: 0

 

SIP/2.0 409 Conflict

Via: SIP/2.0/UDP 173.xxx.xxx.xxx:5080;rport;branch=z9hG4bK1U32gpr9vj5eF

From: <sip:user1 at 10.1.3.20;transport=udp>;tag=11Qey4tcrUH9g

To: <sip:user1 at 10.1.3.20;transport=udp>

Call-ID: cd589b73-031e-445f-a4d8-2fe334d81bbc

CSeq: 126255215 REGISTER

Content-Length: 0

 

.and then this message just repeats again and again with every REGISTER
request.

 

Thanks for your help

\RR

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, January 28, 2010 2:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 409 Conflict When registering FS with
Softswitch

 

Any sip trace would work.. text.. or what not.. 

 

 

 

/b

 

On Jan 28, 2010, at 12:46 PM, RR wrote:

 

Hi brian,

 

Do you need the sip trace from within FS or just any packet capture like
from snoop/tcpdump etc will do? I don't know how to do a sip trace from
within FS

 

\R

 



__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4815 (20100128) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com



__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4816 (20100128) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4818 (20100129) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From brian at freeswitch.org  Fri Jan 29 10:04:53 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 29 Jan 2010 12:04:53 -0600
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <02b001caa10c$913b9100$b3b2b300$@com>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
Message-ID: <F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>

contact header is fine... but if your switch requires the username in the contact then you can set extension-in-contact on the gateway to force that.

/b

On Jan 29, 2010, at 11:57 AM, RR wrote:

> Any ideas? Anyone? All I can tell is that the Contact Header looks a bit dodgey but how that?s causing a 409 is what I don?t get.
>  
> Help please?
> \RR
>  

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From ranjtech at gmail.com  Fri Jan 29 10:44:17 2010
From: ranjtech at gmail.com (RR)
Date: Fri, 29 Jan 2010 13:44:17 -0500
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
	<F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
Message-ID: <02c501caa113$105532b0$30ff9810$@com>

Thanks mate! Specifying the extension same as the username and then using
extension-in-contact fixed the problem. It now registers successfully with
the switch. 

 

I do though still get the message:

2010-01-29 01:30:05.634220 [NOTICE] sofia_reg.c:2267 Added gateway
'Test-Inbound' to profile 'external'

2010-01-29 01:30:05.634547 [ERR] sofia.c:1663 ERROR: username param is
REQUIRED!

2010-01-29 01:30:05.634657 [ERR] sofia.c:1663 ERROR: username param is
REQUIRED!

 

What is that about? I have the username param stated in the gateway
profile!!

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Friday, January 29, 2010 1:05 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 409 Conflict When registering FS with
Softswitch

 

contact header is fine... but if your switch requires the username in the
contact then you can set extension-in-contact on the gateway to force that.

 

/b

 

On Jan 29, 2010, at 11:57 AM, RR wrote:





Any ideas? Anyone? All I can tell is that the Contact Header looks a bit
dodgey but how that's causing a 409 is what I don't get.

 

Help please?

\RR

 

 



__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4818 (20100129) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From jerry.richards at teotech.com  Fri Jan 29 11:04:09 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 29 Jan 2010 11:04:09 -0800
Subject: [Freeswitch-users] Posting a Bounty
In-Reply-To: <mailman.2031.1251316570.2787.freeswitch-users@lists.freeswitch.org>
References: <mailman.2031.1251316570.2787.freeswitch-users@lists.freeswitch.org>
Message-ID: <F5D23A7C72D34626AC8BA8EFC1331AF1@greyhawk.tonecommander.com>

 
How do I post a new Bounty request?

Thanks,
Jerry




From brian at freeswitch.org  Fri Jan 29 11:08:17 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 29 Jan 2010 13:08:17 -0600
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <02c501caa113$105532b0$30ff9810$@com>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
	<F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
	<02c501caa113$105532b0$30ff9810$@com>
Message-ID: <E91AAF1B-2BCB-4D9E-9A49-221480CD8BF5@freeswitch.org>

You'll require a username param .. are you on svn trunk?

/b

On Jan 29, 2010, at 12:44 PM, RR wrote:

> Thanks mate! Specifying the extension same as the username and then using extension-in-contact fixed the problem. It now registers successfully with the switch.
>  
> I do though still get the message:
> 2010-01-29 01:30:05.634220 [NOTICE] sofia_reg.c:2267 Added gateway 'Test-Inbound' to profile 'external'
> 2010-01-29 01:30:05.634547 [ERR] sofia.c:1663 ERROR: username param is REQUIRED!
> 2010-01-29 01:30:05.634657 [ERR] sofia.c:1663 ERROR: username param is REQUIRED!

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From msc at freeswitch.org  Fri Jan 29 11:10:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jan 2010 11:10:49 -0800
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <02c501caa113$105532b0$30ff9810$@com>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
	<F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
	<02c501caa113$105532b0$30ff9810$@com>
Message-ID: <87f2f3b91001291110l226ce6d8u6e806d0a1b782c5f@mail.gmail.com>

On Fri, Jan 29, 2010 at 10:44 AM, RR <ranjtech at gmail.com> wrote:

>  Thanks mate! Specifying the extension same as the username and then using
> extension-in-contact fixed the problem. It now registers successfully with
> the switch.
>
>
>
> I do though still get the message:
>
> *2010-01-29 01:30:05.634220 [NOTICE] sofia_reg.c:2267 Added gateway
> 'Test-Inbound' to profile 'external'*
>
> *2010-01-29 01:30:05.634547 [ERR] sofia.c:1663 ERROR: username param is
> REQUIRED!*
>
> *2010-01-29 01:30:05.634657 [ERR] sofia.c:1663 ERROR: username param is
> REQUIRED!*
>
>
>
> What is that about? I have the username param stated in the gateway
> profile!!
>

Do you possibly have some other XML files floating around that don't have a
username param? It's curious that it said this error twice. It makes me
think that possibly a different file or files is causing that...

-MC
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From msc at freeswitch.org  Fri Jan 29 11:14:56 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jan 2010 11:14:56 -0800
Subject: [Freeswitch-users] Posting a Bounty
In-Reply-To: <F5D23A7C72D34626AC8BA8EFC1331AF1@greyhawk.tonecommander.com>
References: <mailman.2031.1251316570.2787.freeswitch-users@lists.freeswitch.org>
	<F5D23A7C72D34626AC8BA8EFC1331AF1@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91001291114s91dc125n324f215a71a80946@mail.gmail.com>

On Fri, Jan 29, 2010 at 11:04 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

>
> How do I post a new Bounty request?
>
> In JIRA you can do this:
http://jira.freeswitch.org/browse/BOUNTY

Once you post the bounty you can link to it here in the mailing list to let
everyone know about it.
-MC
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From ranjtech at gmail.com  Fri Jan 29 13:08:50 2010
From: ranjtech at gmail.com (RR)
Date: Fri, 29 Jan 2010 16:08:50 -0500
Subject: [Freeswitch-users] 409 Conflict When registering FS with
	Softswitch
In-Reply-To: <87f2f3b91001291110l226ce6d8u6e806d0a1b782c5f@mail.gmail.com>
References: <020c01ca9fe9$1d5952f0$580bf8d0$@com>
	<1254B3C5-D7D3-413D-BA82-54FE4789B360@freeswitch.org>
	<022701caa04a$44f60b80$cee22280$@com>
	<697C7F93-3737-4ABD-8934-6A48DC09C088@freeswitch.org>
	<025701caa0a7$e1ca6200$a55f2600$@com>
	<02b001caa10c$913b9100$b3b2b300$@com>
	<F03237B7-89EE-4CE4-A67F-FCB8E8D26700@freeswitch.org>
	<02c501caa113$105532b0$30ff9810$@com>
	<87f2f3b91001291110l226ce6d8u6e806d0a1b782c5f@mail.gmail.com>
Message-ID: <02dd01caa127$428aa760$c79ff620$@com>

I don't think so..this is the first time I've started to configure FS, and
this was the first xml file I have touched. 

 

Anyway, I saved this gateway xml file, rm -rf'ed the entire conf directory,
did a make current, did a make samples, restored the gateway xml config and
now I don't see the error about username param.

 

Thanks for the help. Now let's see what I can do next in FS, I know it's a
lot to learn. Wish I'd jumped right into it 2 years ago when it was being
developed!

 

I guess the first thing to learn is Dialplan because now I want to make an
inbound call from another endpoint registered to my softswitch to be able to
call an extension registered to FS by calling user1

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Friday, January 29, 2010 2:11 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 409 Conflict When registering FS with
Softswitch

 

 

On Fri, Jan 29, 2010 at 10:44 AM, RR <ranjtech at gmail.com> wrote:

Thanks mate! Specifying the extension same as the username and then using
extension-in-contact fixed the problem. It now registers successfully with
the switch. 

 

I do though still get the message:

2010-01-29 01:30:05.634220 [NOTICE] sofia_reg.c:2267 Added gateway
'Test-Inbound' to profile 'external'

2010-01-29 01:30:05.634547 [ERR] sofia.c:1663 ERROR: username param is
REQUIRED!

2010-01-29 01:30:05.634657 [ERR] sofia.c:1663 ERROR: username param is
REQUIRED!

 

What is that about? I have the username param stated in the gateway
profile!!


Do you possibly have some other XML files floating around that don't have a
username param? It's curious that it said this error twice. It makes me
think that possibly a different file or files is causing that...

-MC

 

 

__________ Information from ESET NOD32 Antivirus, version of virus signature
database 4818 (20100129) __________

 

The message was checked by ESET NOD32 Antivirus.

 

http://www.eset.com

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From jerry.richards at teotech.com  Fri Jan 29 13:14:45 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 29 Jan 2010 13:14:45 -0800
Subject: [Freeswitch-users] PRI Simulator
In-Reply-To: <mailman.5734.1264787871.2495.freeswitch-users@lists.freeswitch.org>
References: <mailman.5734.1264787871.2495.freeswitch-users@lists.freeswitch.org>
Message-ID: <AF84ED7F907F493DAC0A29C8FBC54E44@greyhawk.tonecommander.com>


Can anyone recommend a good PRI simulator?  Sorry this is off topic a bit.

Thanks,
Jerry




From mouncifbb at gmail.com  Fri Jan 29 14:43:38 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Fri, 29 Jan 2010 17:43:38 -0500
Subject: [Freeswitch-users] mod_lcr problem
Message-ID: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>

i can't make use of mod_lcr using Intra/Interstate rating, I am using
svn: FreeSWITCH Version 1.0.trunk (16517)


lcr mysql table structure:

CREATE TABLE `lcr` (
  `id` INT(11) NOT NULL AUTO_INCREMENT,
  `digits` VARCHAR(15) DEFAULT NULL,
  `rate` FLOAT(11,5) DEFAULT NULL,
  `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
  `intralata_rate` FLOAT(11,5) DEFAULT NULL,
  `carrier_id` INT(11) NOT NULL,
  `lead_strip` INT(11) NOT NULL,
  `trail_strip` INT(11) NOT NULL,
  `prefix` VARCHAR(16) NOT NULL,
  `suffix` VARCHAR(16) NOT NULL,
  `lcr_profile` VARCHAR(32) DEFAULT NULL,
  `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
  `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
  `quality` FLOAT(10,6) NOT NULL,
  `reliability` FLOAT(10,6) NOT NULL,
  `cid` VARCHAR(32) NOT NULL DEFAULT '',
  `enabled` TINYINT(1) NOT NULL DEFAULT '1',
  PRIMARY KEY  (`id`),
  KEY `carrier_id` (`carrier_id`),
  KEY `digits` (`digits`),
  KEY `lcr_profile` (`lcr_profile`),
  KEY `digits_profile_cid_rate` USING BTREE (`digits`),
  CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES `carriers`
(`id`) ON DELETE CASCADE ON UPDATE CASCADE
) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1



*lcr_admin show profiles*

Name:           default
 custom sql:    SELECT l.digits, c.carrier_name, l.${lcr_rate_field},
cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip, l.trail_strip,
l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
(${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN date_start AND
date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality DESC,
 reliability DESC, rand();
 has %:         false
 has vars:      true
 has intrastate:        true
 has intralata: true
 has npanxx:    true
 Reorder rate:  enabled
 Info in headers:       disabled
 Quote IN() List:       disabled




*lc**r 617642  default*  returns rate from the rate field table and not
intra/inter state fields rates.


Any ideas? thanks!
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From freeswitch at aastral.net  Fri Jan 29 15:52:21 2010
From: freeswitch at aastral.net (Bill W)
Date: Fri, 29 Jan 2010 18:52:21 -0500
Subject: [Freeswitch-users] Posting a Bounty
In-Reply-To: <F5D23A7C72D34626AC8BA8EFC1331AF1@greyhawk.tonecommander.com>
References: <mailman.2031.1251316570.2787.freeswitch-users@lists.freeswitch.org>
	<F5D23A7C72D34626AC8BA8EFC1331AF1@greyhawk.tonecommander.com>
Message-ID: <4B6374B5.1050500@aastral.net>

Hey Jerry,

Just go to jira.freeswitch.org, log in, create new issue with project=bounty

Hope this helps,
Bill



Jerry Richards wrote:
>  
> How do I post a new Bounty request?
> 
> Thanks,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Fri Jan 29 16:21:39 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 29 Jan 2010 16:21:39 -0800
Subject: [Freeswitch-users] PRI Simulator
In-Reply-To: <AF84ED7F907F493DAC0A29C8FBC54E44@greyhawk.tonecommander.com>
References: <mailman.5734.1264787871.2495.freeswitch-users@lists.freeswitch.org>
	<AF84ED7F907F493DAC0A29C8FBC54E44@greyhawk.tonecommander.com>
Message-ID: <1AF7532D-2B17-4D6E-97A0-421C595FF674@freeswitch.org>

What's your budget?

Sent from my iPhone

On Jan 29, 2010, at 1:14 PM, "Jerry Richards" <jerry.richards at teotech.com 
 > wrote:

>
> Can anyone recommend a good PRI simulator?  Sorry this is off topic  
> a bit.
>
> Thanks,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From rupa at rupa.com  Fri Jan 29 16:37:07 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 29 Jan 2010 18:37:07 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
Message-ID: <b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>

turn console logging up to debug and redo the lcr lookup.  The sql
statements along with status info will show up.  This should give
enough information to debug what is happening.

I'm assuming the npanxx table is actually populated and not just existing?

When doing the lookup from the cli you have to tell lcr what CID to
use (remember, it is relative to the src/dest number).  I'm pretty
sure you get something on the console log when you don't specify a CID
when using the commandline.  Anyway:

lcr 617642 ?default 6176421212

should give you intralata.

Note that the definition of intralata doesn't mean "local" for some
providers.  Some providers define local to "same ratecenter" which is
even more restrictive.

On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <mouncifbb at gmail.com> wrote:
> i can't make use of mod_lcr using Intra/Interstate rating, I am using
> svn:?FreeSWITCH Version 1.0.trunk (16517)
>
> lcr mysql table structure:
> CREATE TABLE `lcr` (
> ??`id` INT(11) NOT NULL AUTO_INCREMENT,
> ??`digits` VARCHAR(15) DEFAULT NULL,
> ??`rate` FLOAT(11,5) DEFAULT NULL,
> ??`intrastate_rate` FLOAT(11,5) DEFAULT NULL,
> ??`intralata_rate` FLOAT(11,5) DEFAULT NULL,
> ??`carrier_id` INT(11) NOT NULL,
> ??`lead_strip` INT(11) NOT NULL,
> ??`trail_strip` INT(11) NOT NULL,
> ??`prefix` VARCHAR(16) NOT NULL,
> ??`suffix` VARCHAR(16) NOT NULL,
> ??`lcr_profile` VARCHAR(32) DEFAULT NULL,
> ??`date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
> ??`date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
> ??`quality` FLOAT(10,6) NOT NULL,
> ??`reliability` FLOAT(10,6) NOT NULL,
> ??`cid` VARCHAR(32) NOT NULL DEFAULT '',
> ??`enabled` TINYINT(1) NOT NULL DEFAULT '1',
> ??PRIMARY KEY ?(`id`),
> ??KEY `carrier_id` (`carrier_id`),
> ??KEY `digits` (`digits`),
> ??KEY `lcr_profile` (`lcr_profile`),
> ??KEY `digits_profile_cid_rate` USING BTREE (`digits`),
> ??CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES `carriers`
> (`id`) ON DELETE CASCADE ON UPDATE CASCADE
> ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>
>
> lcr_admin show profiles
> Name: ? ? ? ? ? default
> ?custom sql: ? ?SELECT l.digits, c.carrier_name, l.${lcr_rate_field},
> cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip, l.trail_strip,
> l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
> l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
> c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
> (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN date_start AND
> date_end ORDER BY digits DESC, ?${lcr_rate_field}, ?quality DESC,
> ?reliability DESC, rand();
> ?has %: ? ? ? ? false
> ?has vars: ? ? ?true
> ?has intrastate: ? ? ? ?true
> ?has intralata: true
> ?has npanxx: ? ?true
> ?Reorder rate: ?enabled
> ?Info in headers: ? ? ? disabled
> ?Quote IN() List: ? ? ? disabled
>
>
>
> lcr 617642 ?default ?returns rate from the rate field table and not
> intra/inter state fields rates.
>
> Any ideas? thanks!
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa



From mouncifbb at gmail.com  Fri Jan 29 20:30:36 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Fri, 29 Jan 2010 23:30:36 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
Message-ID: <ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>

Tried it and it's not giving me intralata  instead I get interstate, does
the npa_nxx_company_ocn table needs to be used in this case?, also do I have
to have the rate field in lcr table?


lcr 617642  default 6176421212

 | Digit Match | Carrier  | Rate     | Codec | CID Regexp | Dialstring
                                                     |
 | 617642      | carrier1 | 0.00500  |       |            |
[lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |


2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr is
[617642  default 6176421212]
2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
[6176421212]
2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing [state:0
lata:0] so rate field is [rate]
2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
c.id=cg.carrier_id
WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
(617642, 61764, 6176, 617, 61, 6) AND CURRENT_TIMESTAMP BETWEEN date_start
AND date_end ORDER BY digits DESC,  rate, rand();
2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
[lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to head of
list
2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
[lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642


Thank you Rupa!

On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> turn console logging up to debug and redo the lcr lookup.  The sql
> statements along with status info will show up.  This should give
> enough information to debug what is happening.
>
> I'm assuming the npanxx table is actually populated and not just existing?
>
> When doing the lookup from the cli you have to tell lcr what CID to
> use (remember, it is relative to the src/dest number).  I'm pretty
> sure you get something on the console log when you don't specify a CID
> when using the commandline.  Anyway:
>
> lcr 617642  default 6176421212
>
> should give you intralata.
>
> Note that the definition of intralata doesn't mean "local" for some
> providers.  Some providers define local to "same ratecenter" which is
> even more restrictive.
>
> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <mouncifbb at gmail.com>
> wrote:
> > i can't make use of mod_lcr using Intra/Interstate rating, I am using
> > svn: FreeSWITCH Version 1.0.trunk (16517)
> >
> > lcr mysql table structure:
> > CREATE TABLE `lcr` (
> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
> >   `digits` VARCHAR(15) DEFAULT NULL,
> >   `rate` FLOAT(11,5) DEFAULT NULL,
> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
> >   `carrier_id` INT(11) NOT NULL,
> >   `lead_strip` INT(11) NOT NULL,
> >   `trail_strip` INT(11) NOT NULL,
> >   `prefix` VARCHAR(16) NOT NULL,
> >   `suffix` VARCHAR(16) NOT NULL,
> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
> >   `quality` FLOAT(10,6) NOT NULL,
> >   `reliability` FLOAT(10,6) NOT NULL,
> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
> >   PRIMARY KEY  (`id`),
> >   KEY `carrier_id` (`carrier_id`),
> >   KEY `digits` (`digits`),
> >   KEY `lcr_profile` (`lcr_profile`),
> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
> `carriers`
> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
> >
> >
> > lcr_admin show profiles
> > Name:           default
> >  custom sql:    SELECT l.digits, c.carrier_name, l.${lcr_rate_field},
> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
> l.trail_strip,
> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN date_start
> AND
> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality DESC,
> >  reliability DESC, rand();
> >  has %:         false
> >  has vars:      true
> >  has intrastate:        true
> >  has intralata: true
> >  has npanxx:    true
> >  Reorder rate:  enabled
> >  Info in headers:       disabled
> >  Quote IN() List:       disabled
> >
> >
> >
> > lcr 617642  default  returns rate from the rate field table and not
> > intra/inter state fields rates.
> >
> > Any ideas? thanks!
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mouncifbb at gmail.com  Fri Jan 29 20:42:04 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Fri, 29 Jan 2010 23:42:04 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
Message-ID: <ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>

Also the Provider has presented the rates in this format?

NPANXXLATA OCN INTER INTRA


On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> Tried it and it's not giving me intralata  instead I get interstate, does
> the npa_nxx_company_ocn table needs to be used in this case?, also do I have
> to have the rate field in lcr table?
>
>
> lcr 617642  default 6176421212
>
>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp | Dialstring
>                                                        |
>  | 617642      | carrier1 | 0.00500  |       |            |
> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>
>
> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr is
> [617642  default 6176421212]
> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
> [6176421212]
> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing [state:0
> lata:0] so rate field is [rate]
> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
> (617642, 61764, 6176, 617, 61, 6) AND CURRENT_TIMESTAMP BETWEEN date_start
> AND date_end ORDER BY digits DESC,  rate, rand();
> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to head of
> list
> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>
>
> Thank you Rupa!
>
>
> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> turn console logging up to debug and redo the lcr lookup.  The sql
>> statements along with status info will show up.  This should give
>> enough information to debug what is happening.
>>
>> I'm assuming the npanxx table is actually populated and not just existing?
>>
>> When doing the lookup from the cli you have to tell lcr what CID to
>> use (remember, it is relative to the src/dest number).  I'm pretty
>> sure you get something on the console log when you don't specify a CID
>> when using the commandline.  Anyway:
>>
>> lcr 617642  default 6176421212
>>
>> should give you intralata.
>>
>> Note that the definition of intralata doesn't mean "local" for some
>> providers.  Some providers define local to "same ratecenter" which is
>> even more restrictive.
>>
>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <mouncifbb at gmail.com>
>> wrote:
>> > i can't make use of mod_lcr using Intra/Interstate rating, I am using
>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>> >
>> > lcr mysql table structure:
>> > CREATE TABLE `lcr` (
>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>> >   `digits` VARCHAR(15) DEFAULT NULL,
>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>> >   `carrier_id` INT(11) NOT NULL,
>> >   `lead_strip` INT(11) NOT NULL,
>> >   `trail_strip` INT(11) NOT NULL,
>> >   `prefix` VARCHAR(16) NOT NULL,
>> >   `suffix` VARCHAR(16) NOT NULL,
>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>> >   `quality` FLOAT(10,6) NOT NULL,
>> >   `reliability` FLOAT(10,6) NOT NULL,
>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>> >   PRIMARY KEY  (`id`),
>> >   KEY `carrier_id` (`carrier_id`),
>> >   KEY `digits` (`digits`),
>> >   KEY `lcr_profile` (`lcr_profile`),
>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>> `carriers`
>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>> >
>> >
>> > lcr_admin show profiles
>> > Name:           default
>> >  custom sql:    SELECT l.digits, c.carrier_name, l.${lcr_rate_field},
>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>> l.trail_strip,
>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN date_start
>> AND
>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality DESC,
>> >  reliability DESC, rand();
>> >  has %:         false
>> >  has vars:      true
>> >  has intrastate:        true
>> >  has intralata: true
>> >  has npanxx:    true
>> >  Reorder rate:  enabled
>> >  Info in headers:       disabled
>> >  Quote IN() List:       disabled
>> >
>> >
>> >
>> > lcr 617642  default  returns rate from the rate field table and not
>> > intra/inter state fields rates.
>> >
>> > Any ideas? thanks!
>> >
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From marketing at cluecon.com  Fri Jan 29 22:47:45 2010
From: marketing at cluecon.com (Michael Collins)
Date: Fri, 29 Jan 2010 22:47:45 -0800
Subject: [Freeswitch-users] ClueCon MMX - Save the Date!
Message-ID: <87f2f3b91001292247u5f058054yca8590bb9c39ae65@mail.gmail.com>

ClueCon MMX (2010) will be here before you know it! Please mark your
calendars: August 3-5, 2010. Start talking up ClueCon with your peers,
coworkers, business owners, CEOs, potential sponsors, and anyone else you
can think of. It's coming fast, so start getting ready now. Lots more
information will be coming soon.

Looking forward to seeing everyone this August!
-ClueCon Team
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From robin at swip.net  Fri Jan 29 07:31:29 2010
From: robin at swip.net (Robin Vleij)
Date: Fri, 29 Jan 2010 16:31:29 +0100
Subject: [Freeswitch-users] CPS per mod_sofia profile limited?
Message-ID: <4B62FF51.8070608@swip.net>

Hi guys,

Doing a bit of testing / benchmarking with FS 1.04 (and 1.05 SVN). Found
out some interesting things, I think.

The setup is like this:

SIPP Client -> FS -> SIPP Server

The dialplan is as simple as it gets:

<extension name="test-out">
  <condition field="network_addr" expression="^130\.244\.190\.150$"/>
  <condition field="destination_number" expression="^(\d+)$">
    <action application="set" data="proxy_media=true"/>
    <action application="set" data="hangup_after_bridge=true"/>
    <action application="bridge"
data="{sip_contact_user=transit}sofia/gateway/${distributor(gwg1)}/$1|sofia/gateway/${distributor(gwg2)}/$1"/>
  </condition>
</extension>

For the rest it's running CSV cdr's, commented out all modules I'm not
using, etc etc all that I could find on the wiki and the Interwebs.
Hardware / OS: from the shelve quadcore Xeon, debian 64-bit, 12GB
memory. SIPP is running a 500ms RTP pcap and the other side echos back.

I had a few test setups then:

1: FS SVN, 1 sofia profile where the gateways were configured and the
server_IP:5060 was used.

2. FS SVN, 2 sofia profiles where the gateways where in an seperate
profile (server_IP:5070) and the "customer facing" side was the original
profile.

3. FS 1.04, same as above

4. FS 1.04, 4 sofia profiles, distributor to spread load over 2 incoming
and 2 outgoing profiles.

Now the interesting thing was that under 1 I could go up, almost without
any CPU load, to 50cps. As soon as I went over this, calls where handled
 slower and "ongoing" calls would pile up untill it became really slow.
CPU load went to 100% on the FS process (both user and system time).
Lots of interupts and context switches. No throughput anymore untill I
lower and wait till the "buffer" is empty and FS is keeping up again.

Under 2, I was able to increase the CPS to about 100 with the same effect.

3 then went much better, I was able to increase CPS to about 200 cps and
response times in SIPP went up slighty untill it just hits some kind of
limit and calls are handled slower.

4 is pretty cool. Here I can run 2 sipp clients both doing 150cps to the
gateways that are spread on the distribution module, so I spread traffic
over 2 profiles. With 300 cps in total, FS is keeping up and I have 30%
idle CPU. However, increasing to over 300cps gives problems again, even
though I have idle CPU left!

All in all, I have a feeling that a single sip profile can't run more
than a certain limit untill it gets into some problem. Depending on if
I'm running SVN or 1.04 that limit seems to be 50cps or a bit higher.
After that limit it starts piling up "ongoing" calls, by taking time to
handle them and when that limit gets too high it's too late. All in all
really fine, I just set the system wide limit to a little under that
"threshold". But when I'm running just UNDER the threshold it's not CPU
that's a problem. Theoretically I should be able to run (based on the
CPU usage at 300cps) about 400cps.

When running at 300 I get SOME failed calls and I see

"switch_core_state_machine.c:525 a9a60636-0cea-11df-85a1-09c991f2afc5
Timeout waiting for next instruction in CS_NEW!"

in the console.

I didn't find much on how people do high cps setups and it feels a bit
like a "friday afternoon solution" to run multiple sofia profiles on the
same machine in order to max out the system.

Maybe I'm missing something and I know it's not an exact science this,
but I'm not sure "all is OK" because I'm not slowly getting to a 100%
cpu (or disk / network) usage, I hit some kind of limit after which
stuff goes wrong.

Anyone any input!

/Robin



From paul.gore.j at gmail.com  Fri Jan 29 20:20:46 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Fri, 29 Jan 2010 23:20:46 -0500
Subject: [Freeswitch-users] Logging question
Message-ID: <da4ab26e1001292020o280f330ft62a278b93c7fb1fd@mail.gmail.com>

Hi there,
I am running FS 1.0.trunk (14501) (I know it's old but we serve a small
community and don't have time to upgrade/test the latest/greatest). I am
having troubles understanding how to switch SIP trace in log files, I tried

fsctl loglevel debug
sofia tracelevel debug

but it seem to have no effect, I only get sofia debug messages but no
detailed SIP info.
What also puzzling me is if I do

console loglevel 0

I still get debug information on console.
What am I doing wrong?
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From mike at jerris.com  Fri Jan 29 23:50:45 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 Jan 2010 02:50:45 -0500
Subject: [Freeswitch-users] Hybrid Encryption?
In-Reply-To: <2b6116b31001281808x1a004cd2ne7e8dcb9f16fec3e@mail.gmail.com>
References: <2b6116b31001272154l3c0bbe80y8bf3db94961e8e1d@mail.gmail.com>
	<8A9EDC4E-C49B-488D-9DBF-169A185462AB@freeswitch.org>
	<2b6116b31001281631u7ada7876wc419bb7afadd7ef7@mail.gmail.com>
	<46C06209-9515-4B1B-B449-F55A51FF548B@freeswitch.org>
	<2b6116b31001281808x1a004cd2ne7e8dcb9f16fec3e@mail.gmail.com>
Message-ID: <B2BE2748-ADA4-4D1C-8004-99AD72FB7806@jerris.com>

Freeswitch isn't a proxy, and no, we don't provide support for passthrough auth like this.  A proxy would, but not sure of any proxy based solution that would do the srtp work for you.

Mike

On Jan 28, 2010, at 9:08 PM, Nicholas Lee wrote:

> Is there a way to do it transparently?  The FS proxies will past though the extension creds.
> 
> On Fri, Jan 29, 2010 at 1:52 PM, Brian West <brian at freeswitch.org> wrote:
> Then yes you could use FreeSWITCH to augment your Asterisk install and enable encryption from site to site.
> 
> /b
> 
> 
> > Unfortunately it's not going to cover every situation.
> >
> >
> > Nicholas
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Fri Jan 29 23:56:30 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 Jan 2010 02:56:30 -0500
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001281932h73466f87yf4715058f4632558@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<191c3a031001271709i43f104c1md628818aa61b062@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
	<33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
	<191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>
	<33c87fa31001271939i668e9676y1f687dc182b4ca89@mail.gmail.com>
	<CF0D5382-EE44-48CB-BF14-27A443014336@freeswitch.org>
	<33c87fa31001272035t7c46e054t7ce2dd598fa40ebb@mail.gmail.com>
	<191c3a031001272125t2b19d134lbb65dfebbb7a52b9@mail.gmail.com>
	<33c87fa31001281932h73466f87yf4715058f4632558@mail.gmail.com>
Message-ID: <69FCAB67-B24A-4B10-B4E9-00A0FC55324E@jerris.com>

So they are in 2 different profiles then.  If you are doing them exactly the same as you said, the issue is your telling it profile internal, when it is really nat, so it does not find it.

Mike

On Jan 28, 2010, at 10:32 PM, Mark Campbell-Smith wrote:

> Hi !
> 
> I confirmed yesterday that if the SPA is not NAT'd, then the event is
> sent.  I just removed NAT from the extension that I was having
> problems with.
> 
> Looking at the db tables, it appears there are two - the
> sofia_reg_internal.db and sofia_reg_internal_nat.db
> 
> Could it be that the sendevent command is only looking in the
> sofia_reg_internal.db database and not sofia_reg_internal_nat.db ?
> 




From oseslija at gmail.com  Sat Jan 30 01:14:19 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sat, 30 Jan 2010 10:14:19 +0100
Subject: [Freeswitch-users] Custom NOTIFY message in FS
In-Reply-To: <33c87fa31001281932h73466f87yf4715058f4632558@mail.gmail.com>
References: <33c87fa31001240247s783c2e07ndaa54cc78bd94b2d@mail.gmail.com>
	<33c87fa31001271726l1d4e66bdi802cf273d1fd3498@mail.gmail.com>
	<191c3a031001271741h384c9399qc1028602b94d19ed@mail.gmail.com>
	<33c87fa31001271834r58e6c61vc9ae94dac0a72c87@mail.gmail.com>
	<191c3a031001271925y48866edbv272b0e961dbaf518@mail.gmail.com>
	<33c87fa31001271939i668e9676y1f687dc182b4ca89@mail.gmail.com>
	<CF0D5382-EE44-48CB-BF14-27A443014336@freeswitch.org>
	<33c87fa31001272035t7c46e054t7ce2dd598fa40ebb@mail.gmail.com>
	<191c3a031001272125t2b19d134lbb65dfebbb7a52b9@mail.gmail.com>
	<33c87fa31001281932h73466f87yf4715058f4632558@mail.gmail.com>
Message-ID: <4468a6771001300114m68a18e4fk74627314ad7182fd@mail.gmail.com>

I have reboot working from fs_cli with the NATed SPA.

Regards,
Ognjen

On Fri, Jan 29, 2010 at 4:32 AM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Hi !
>
> I confirmed yesterday that if the SPA is not NAT'd, then the event is
> sent.  I just removed NAT from the extension that I was having
> problems with.
>
> Looking at the db tables, it appears there are two - the
> sofia_reg_internal.db and sofia_reg_internal_nat.db
>
> Could it be that the sendevent command is only looking in the
> sofia_reg_internal.db database and not sofia_reg_internal_nat.db ?
>
>
>
> On Thu, Jan 28, 2010 at 4:25 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > You have to look in the sql db and compare the specified vals with the
> ones
> > looked up from the event again the user and host need to match the db
> >
> > On Jan 27, 2010 10:41 PM, "Mark Campbell-Smith" <
> mcampbellsmith at gmail.com>
> > wrote:
> >
> > Hi Brian,
> >
> >  I've previously enabled siptrace for internal profile, but I see
> > nothing sent and nothing received.
> >
> > On Thu, Jan 28, 2010 at 2:54 PM, Brian West <brian at freeswitch.org>
> wrote:
> >> I'm suspecting the code...
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From errotan at gmail.com  Sat Jan 30 03:33:21 2010
From: errotan at gmail.com (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Sat, 30 Jan 2010 12:33:21 +0100
Subject: [Freeswitch-users] CPS per mod_sofia profile limited?
In-Reply-To: <4B62FF51.8070608@swip.net>
References: <4B62FF51.8070608@swip.net>
Message-ID: <201001301233.21516.errotan@gmail.com>

2010. janu?r 29. 16.31.29 Robin Vleij d?tummal ezt ?rta:
> Hi guys,
> 
> Doing a bit of testing / benchmarking with FS 1.04 (and 1.05 SVN). Found
> out some interesting things, I think.
> 
> The setup is like this:
> 
> SIPP Client -> FS -> SIPP Server
> 
> The dialplan is as simple as it gets:
> 
> <extension name="test-out">
>   <condition field="network_addr" expression="^130\.244\.190\.150$"/>
>   <condition field="destination_number" expression="^(\d+)$">
>     <action application="set" data="proxy_media=true"/>
>     <action application="set" data="hangup_after_bridge=true"/>
>     <action application="bridge"
> data="{sip_contact_user=transit}sofia/gateway/${distributor(gwg1)}/$1|sofia
> /gateway/${distributor(gwg2)}/$1"/> </condition>
> </extension>
> 
> For the rest it's running CSV cdr's, commented out all modules I'm not
> using, etc etc all that I could find on the wiki and the Interwebs.
> Hardware / OS: from the shelve quadcore Xeon, debian 64-bit, 12GB
> memory. SIPP is running a 500ms RTP pcap and the other side echos back.
> 
> I had a few test setups then:
> 
> 1: FS SVN, 1 sofia profile where the gateways were configured and the
> server_IP:5060 was used.
> 
> 2. FS SVN, 2 sofia profiles where the gateways where in an seperate
> profile (server_IP:5070) and the "customer facing" side was the original
> profile.
> 
> 3. FS 1.04, same as above
> 
> 4. FS 1.04, 4 sofia profiles, distributor to spread load over 2 incoming
> and 2 outgoing profiles.
> 
> Now the interesting thing was that under 1 I could go up, almost without
> any CPU load, to 50cps. As soon as I went over this, calls where handled
>  slower and "ongoing" calls would pile up untill it became really slow.
> CPU load went to 100% on the FS process (both user and system time).
> Lots of interupts and context switches. No throughput anymore untill I
> lower and wait till the "buffer" is empty and FS is keeping up again.
> 
> Under 2, I was able to increase the CPS to about 100 with the same effect.
> 
> 3 then went much better, I was able to increase CPS to about 200 cps and
> response times in SIPP went up slighty untill it just hits some kind of
> limit and calls are handled slower.
> 
> 4 is pretty cool. Here I can run 2 sipp clients both doing 150cps to the
> gateways that are spread on the distribution module, so I spread traffic
> over 2 profiles. With 300 cps in total, FS is keeping up and I have 30%
> idle CPU. However, increasing to over 300cps gives problems again, even
> though I have idle CPU left!
> 
> All in all, I have a feeling that a single sip profile can't run more
> than a certain limit untill it gets into some problem. Depending on if
> I'm running SVN or 1.04 that limit seems to be 50cps or a bit higher.
> After that limit it starts piling up "ongoing" calls, by taking time to
> handle them and when that limit gets too high it's too late. All in all
> really fine, I just set the system wide limit to a little under that
> "threshold". But when I'm running just UNDER the threshold it's not CPU
> that's a problem. Theoretically I should be able to run (based on the
> CPU usage at 300cps) about 400cps.
> 
> When running at 300 I get SOME failed calls and I see
> 
> "switch_core_state_machine.c:525 a9a60636-0cea-11df-85a1-09c991f2afc5
> Timeout waiting for next instruction in CS_NEW!"
> 
> in the console.
> 
> I didn't find much on how people do high cps setups and it feels a bit
> like a "friday afternoon solution" to run multiple sofia profiles on the
> same machine in order to max out the system.
> 
> Maybe I'm missing something and I know it's not an exact science this,
> but I'm not sure "all is OK" because I'm not slowly getting to a 100%
> cpu (or disk / network) usage, I hit some kind of limit after which
> stuff goes wrong.
> 
> Anyone any input!
> 
> /Robin
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

CPU usage is not the only thing that limit your calls. Have you set the 
recommended ulimit settings and / or started fs with the -waste option ?

http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations



From codecomplete at free.fr  Sat Jan 30 05:53:25 2010
From: codecomplete at free.fr (Fred-145)
Date: Sat, 30 Jan 2010 14:53:25 +0100
Subject: [Freeswitch-users] Equivalent to
	Asterisk's	"directrtpsetup=yes"?
References: <gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b@4ax.com>
	<gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b-e09XROE/p8c@public.gmane.org>
	<20100129164359.9B3C22B10D@cuneorg-email.cune.pri>
Message-ID: <j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8@4ax.com>

On Fri, 29 Jan 2010 16:43:59 -0000,
<Russell.Mosemann at cune.org> wrote:
>The ports are open between the endpoint and Freeswitch. The ports are not
>open between the two endpoints themselves. If each endpoint is behind its
>own NAT, neither endpoint will be able to contact the other endpoint
>unless some kind of forwarding is set up on the firewall to map the
>external IP address and port to an internal IP address and port.

Thanks but the context I was refering to is...
1. Freeswitch is configured in BypassMedia mode
2. The firewall and the local end-points are configured so that a
series of UDP ranges are mapped to their respective end-point (eg.
UDP100-1003 for extension #1, 1004-1007 for #2, etc.)
... so that RTP packets flow directly between the two end-points

Brian says above that there might be cases where NAT could be a
problem. When could this happen?

I'd like to get to the bottom of this so that in case a server is a
bit short on CPU/network power, I know that there's the alternative of
RTP packets by-passing the server... but I also need to know what
issues this setup can cause.

Thank you.




From yehavi.bourvine at gmail.com  Sat Jan 30 06:01:22 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 30 Jan 2010 16:01:22 +0200
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001251206x21242589he3014081a7e110d1@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<D56D65EB-B150-4A59-8B66-68BF4C58108B@jerris.com>
	<d361ca7b1001241004kcb73f11hc400bdb37c9d372b@mail.gmail.com>
	<3A27F063-E0C0-4178-A3AF-068956B55846@jerris.com>
	<d361ca7b1001241117q7f1379bewe1bc85279ef67c83@mail.gmail.com>
	<alpine.GSO.2.00.1001250716120.17161@h1.pp.uhwv.np.vy>
	<d361ca7b1001242150t29b36ebcy6c2e40ca3d360c00@mail.gmail.com>
	<d361ca7b1001251143p7f396dd8h11d372731a26f4d7@mail.gmail.com>
	<224C684A-B357-42E4-98AA-0EE238A27A49@jerris.com>
	<d361ca7b1001251206x21242589he3014081a7e110d1@mail.gmail.com>
Message-ID: <d361ca7b1001300601r29530dcfm65c67e9121d3ddde@mail.gmail.com>

It works ok now (fixed on r16534).

                Thanks! __Yehavi:

2010/1/25 Yehavi Bourvine <yehavi.bourvine at gmail.com>

> OK. The sources are under /home/freeswitch. Note that it has Hebrew modules
> which we are now developing, but the problem is seen with the vanilla
> version as well.
>
> The execs and all other stuff is at /freeswitch.
>
>                       regards, __Yehavi:
>
> 2010/1/25 Michael Jerris <mike at jerris.com>
>
> I have not had a chance to actually try it yet.  I will let you know.
>>
>> Mike
>>
>> On Jan 25, 2010, at 2:43 PM, Yehavi Bourvine wrote:
>>
>> > Hello Mike,
>> >
>> >   I see you are logged-in into our machine. Before I go to sleep: Do you
>> need any help?
>> >
>> >                     Thanks, __yehavi:
>> >
>>
>>
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From dftoro at yahoo.com  Sat Jan 30 06:27:44 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Sat, 30 Jan 2010 06:27:44 -0800 (PST)
Subject: [Freeswitch-users] mutiple playback files (unescape_char)
	Windows
In-Reply-To: <191c3a031001261426h7fd87e1fpf95824788d639557@mail.gmail.com>
Message-ID: <406990.22243.qm@web33501.mail.mud.yahoo.com>

Hi, 

With 4 slashes (\\\\) works fine

Thank you

Diego Toro
http://lacarretade.blogspot.com/


--- On Tue, 1/26/10, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] mutiple playback files (unescape_char) Windows
> To: freeswitch-users at lists.freeswitch.org
> Date: Tuesday, January 26, 2010, 5:26 PM
> please update again and try 4 slashes
> 
> you need 4 because the expand vars on the data=""
> will eat the 4 down to 2
> then the splitter on ! will turn \\s into \s
> 
> 
> 
> 
> 
> 
> On Tue, Jan 26, 2010 at 3:02 PM, Diego Toro <dftoro at yahoo.com>
> wrote:
> 
> Hi, sorry, I explain better. Using \\\\ is
> also changed when path matches a character such as
> \s,\n... My alternative on Windows is to use
> '/' like path separator.
> 
> 
> 
> 
> 
> Diego Toro
> 
> http://lacarretade.blogspot.com/
> 
> 
> 
> 
> 
> --- On Tue, 1/26/10, Anthony
> Minessale <anthony.minessale at gmail.com>
> wrote:
> 
> 
> 
> > From: Anthony Minessale <anthony.minessale at gmail.com>
> 
> > Subject: Re: [Freeswitch-users] mutiple playback files
> (unescape_char) Windows
> 
> > To: freeswitch-users at lists.freeswitch.org
> 
> > Date: Tuesday, January 26, 2010, 11:27 AM
> 
> > I didn't
> understand that
> 
> >
> 
> > On Tue, Jan 26, 2010 at 9:58 AM,
> 
> > Diego Toro <dftoro at yahoo.com>
> 
> > wrote:
> 
> >
> 
> > Hi, using \\\\ the is changed also
> when
> 
> > there is a match with an escape character
> 
> > (\s,\n...)
> 
> >
> 
> >
> 
> >
> 
> > Thank you
> 
> >
> 
> >
> 
> >
> 
> > Diego Toro
> 
> >
> 
> > http://lacarretade.blogspot.com/
> 
> >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> > --- On Mon, 1/25/10, Anthony Minessale <anthony.minessale at gmail.com>
> 
> > wrote:
> 
> >
> 
> >
> 
> >
> 
> > > From: Anthony Minessale <anthony.minessale at gmail.com>
> 
> >
> 
> > > Subject: Re: [Freeswitch-users]
> 
> > mutiple playback files (unescape_char) Windows
> 
> >
> 
> > > To: freeswitch-users at lists.freeswitch.org
> 
> >
> 
> > > Date: Monday, January 25, 2010, 12:20 PM
> 
> >
> 
> > > its possible your
> 
> > string hits the parser
> 
> >
> 
> > > more than once.
> 
> >
> 
> > > try using 4 \
> 
> >
> 
> > >
> 
> >
> 
> > > \\\\sound
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > > On Sun, Jan 24, 2010 at 4:03 AM,
> 
> >
> 
> > > Michael Jerris <mike at jerris.com>
> 
> >
> 
> > > wrote:
> 
> >
> 
> > >
> 
> >
> 
> > > As noted on that bug, you should be
> 
> >
> 
> > > able to either use \\ or / for the path
> 
> > separator
> 
> >
> 
> > > there and it should work.
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > > Mike
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > > On Jan 22, 2010, at 9:18 AM, Diego Toro wrote:
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > > > Hi, with svn r16440 the problem persists, I
> 
> > creted a
> 
> >
> 
> > > jira report http://jira.freeswitch.org/browse/LBSNDF-8
> 
> >
> 
> > > this is a minor issue, but activing playback
> delimiter
> 
> > no
> 
> >
> 
> > > audio file can be played. On FS the audio files
> are
> 
> > placed
> 
> >
> 
> > > in the \sound\ directory, building the
> path
> 
> > on
> 
> >
> 
> > > Windows would be \sound '\s'
> which is
> 
> >
> 
> > > replaced by 'ound'.
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > > >
> 
> >
> 
> > >
> 
> >
> 
> > > > Thank you
> 
> >
> 
> > >
> 
> >
> 
> > > >
> 
> >
> 
> > >
> 
> >
> 
> > > > Diego Toro
> 
> >
> 
> > >
> 
> >
> 
> > > > http://lacarretade.blogspot.com/
> 
> >
> 
> > >
> 
> >
> 
> > > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > > _______________________________________________
> 
> >
> 
> > >
> 
> >
> 
> > > FreeSWITCH-users mailing list
> 
> >
> 
> > >
> 
> >
> 
> > > FreeSWITCH-users at lists.freeswitch.org
> 
> >
> 
> > >
> 
> >
> 
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> >
> 
> > >
> 
> >
> 
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> >
> 
> > >
> 
> >
> 
> > > http://www.freeswitch.org
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > > --
> 
> >
> 
> > > Anthony Minessale II
> 
> >
> 
> > >
> 
> >
> 
> > > FreeSWITCH http://www.freeswitch.org/
> 
> >
> 
> > > ClueCon http://www.cluecon.com/
> 
> >
> 
> > >
> 
> >
> 
> > > Twitter: http://twitter.com/FreeSWITCH_wire
> 
> >
> 
> > >
> 
> >
> 
> > > AIM: anthm
> 
> >
> 
> > > MSN:anthony_minessale at hotmail.com
> 
> >
> 
> > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> 
> >
> 
> > >
> 
> >
> 
> > > IRC: irc.freenode.net
> 
> >
> 
> > > #freeswitch
> 
> >
> 
> > >
> 
> >
> 
> > > FreeSWITCH Developer Conference
> 
> >
> 
> > > sip:888 at conference.freeswitch.org
> 
> >
> 
> > > iax:guest at conference.freeswitch.org/888
> 
> >
> 
> > >
> 
> >
> 
> > > googletalk:conf+888 at conference.freeswitch.org
> 
> >
> 
> > > pstn:+19193869900
> 
> >
> 
> > >
> 
> >
> 
> > >
> 
> >
> 
> > > -----Inline Attachment Follows-----
> 
> >
> 
> > >
> 
> >
> 
> > > _______________________________________________
> 
> >
> 
> > > FreeSWITCH-users mailing list
> 
> >
> 
> > > FreeSWITCH-users at lists.freeswitch.org
> 
> >
> 
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> >
> 
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> >
> 
> > > http://www.freeswitch.org
> 
> >
> 
> > >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> >
> 
> > _______________________________________________
> 
> >
> 
> > FreeSWITCH-users mailing list
> 
> >
> 
> > FreeSWITCH-users at lists.freeswitch.org
> 
> >
> 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> >
> 
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> >
> 
> > http://www.freeswitch.org
> 
> >
> 
> >
> 
> >
> 
> >
> 
> > --
> 
> > Anthony Minessale II
> 
> >
> 
> > FreeSWITCH http://www.freeswitch.org/
> 
> > ClueCon http://www.cluecon.com/
> 
> >
> 
> > Twitter: http://twitter.com/FreeSWITCH_wire
> 
> >
> 
> > AIM: anthm
> 
> > MSN:anthony_minessale at hotmail.com
> 
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> 
> >
> 
> > IRC: irc.freenode.net
> 
> > #freeswitch
> 
> >
> 
> > FreeSWITCH Developer Conference
> 
> > sip:888 at conference.freeswitch.org
> 
> > iax:guest at conference.freeswitch.org/888
> 
> >
> 
> > googletalk:conf+888 at conference.freeswitch.org
> 
> > pstn:+19193869900
> 
> >
> 
> >
> 
> > -----Inline Attachment Follows-----
> 
> >
> 
> > _______________________________________________
> 
> > FreeSWITCH-users mailing list
> 
> > FreeSWITCH-users at lists.freeswitch.org
> 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> > http://www.freeswitch.org
> 
> >
> 
> 
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> 
> FreeSWITCH-users mailing list
> 
> FreeSWITCH-users at lists.freeswitch.org
> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> 
> IRC: irc.freenode.net
> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> 
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From sharad at coraltele.com  Sat Jan 30 06:43:10 2010
From: sharad at coraltele.com (Sharad)
Date: Sat, 30 Jan 2010 06:43:10 -0800 (PST)
Subject: [Freeswitch-users] freeswitch with T.38
Message-ID: <1264862590045-4485495.post@n2.nabble.com>


Is paid freeswitch available with T.38 support. ?

regards
Sharad
-- 
View this message in context: http://n2.nabble.com/freeswitch-with-T-38-tp4485495p4485495.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From lakindia89 at gmail.com  Sat Jan 30 06:48:53 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Sat, 30 Jan 2010 20:18:53 +0530
Subject: [Freeswitch-users] nixevent behavior
Message-ID: <7d79b3931001300648j6aa55258yfc496d9cea5c4b8b@mail.gmail.com>

Dear all

I've done the following sample script to experiment the nixevent. I found
some difference in behavior because of nixevent. Let me explain my question
down the script.

require ESL;
use IO::Socket::INET;

my $ip = "192.168.1.222";
my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
Proto => 'tcp',  Listen => 1,  Reuse => 1 );
die "Could not create socket: $!\n" unless $sock;
for(;;) {
    my $new_sock = $sock->accept();
    next if (not defined ($new_sock));
    my $pid = fork();
    if ($pid) {
        close($new_sock);
        next;
    }
    print "CHILD PID: $$\n";
    my $host = $new_sock->sockhost();
    my $fd = fileno($new_sock);

    my $con = new ESL::ESLconnection($fd);
    my $info = $con->getInfo();

    my $uuid = $info->getHeader("unique-id");

    printf "Connected call %s, from %s\n", $uuid,
$info->getHeader("caller-caller-id-number");
    my $r=$con->execute("answer");
    $con->events("plain","all");
##########################
    $con->send("nixevent DTMF");
    my $val=$con->api("create_uuid");
    $val = $val->getBody();            # LINE 1
    chomp($val);
    print "UUID is $val\n";
    my $e = $con->recvEvent();
    $val = $e->getBody();               # LINE 2
    chomp($val);
    print "UUID is $val\n";
    close($new_sock);
}

# If the line ($con->send("nixevent DTMF");) is commented, then the result
of create_uuid is obtained in LINE 1.
# else, the result isn't obtained in the LINE 1 and it has nothing. The
result is obtained only when I do a recvEvent,
# followed by a getBody (LINE 2)

Just want to know why the behavior differs when nixevent is present???
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From rupa at rupa.com  Sat Jan 30 07:02:31 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 30 Jan 2010 09:02:31 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
Message-ID: <b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>

Something is still missing from the logs. Note the query of the npanxx
table, the flags being set, and the rate field being chosen.  Umm..
oh, what version of fs are you running?

Yes, the npa_nxx_ocn table needs to be loaded up as described in:
http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
(there is a link to that from mod_lcr's wiki page).

An example from my own setup:

2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
is [12148267711 default 12148267712]
2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to [12148267712]
2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR (npa=214
AND nxx=826)
2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count: 1
2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
[state:1 lata:1] so rate field is [intralata_rate]
2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT l.digits
AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
l.intralata_rate as lcr_rate_field,        cg.prefix AS lcr_gw_prefix,
cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
 WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
date_start AND date_end   ORDER BY digits DESC, intralata_rate,
random();
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
[lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
head of list
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
[lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of list
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
[lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of list
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
[lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to end of list
2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
[lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
[...]

On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <mouncifbb at gmail.com> wrote:
> Also the Provider has presented the rates in this format?
> NPANXXLATA OCN INTER INTRA
>
>
> On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <mouncifbb at gmail.com>
> wrote:
>>
>> Tried it and it's not giving me?intralata??instead I get interstate, does
>> the?npa_nxx_company_ocn table needs to be used in this case?, also do I have
>> to have the rate field in lcr table?
>>
>> lcr 617642 ?default 6176421212
>> ?| Digit Match | Carrier ?| Rate ? ? | Codec | CID Regexp | Dialstring
>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?|
>> ?| 617642 ? ? ?| carrier1 | 0.00500 ?| ? ? ? | ? ? ? ? ? ?|
>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>> [617642 ?default 6176421212]
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>> [6176421212]
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing [state:0
>> lata:0] so rate field is [rate]
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
>> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled
>> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND CURRENT_TIMESTAMP
>> BETWEEN date_start AND date_end ORDER BY digits DESC, ?rate, rand();
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>> of list
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>
>> Thank you Rupa!
>>
>> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>> turn console logging up to debug and redo the lcr lookup. ?The sql
>>> statements along with status info will show up. ?This should give
>>> enough information to debug what is happening.
>>>
>>> I'm assuming the npanxx table is actually populated and not just
>>> existing?
>>>
>>> When doing the lookup from the cli you have to tell lcr what CID to
>>> use (remember, it is relative to the src/dest number). ?I'm pretty
>>> sure you get something on the console log when you don't specify a CID
>>> when using the commandline. ?Anyway:
>>>
>>> lcr 617642 ?default 6176421212
>>>
>>> should give you intralata.
>>>
>>> Note that the definition of intralata doesn't mean "local" for some
>>> providers. ?Some providers define local to "same ratecenter" which is
>>> even more restrictive.
>>>
>>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <mouncifbb at gmail.com>
>>> wrote:
>>> > i can't make use of mod_lcr using Intra/Interstate rating, I am using
>>> > svn:?FreeSWITCH Version 1.0.trunk (16517)
>>> >
>>> > lcr mysql table structure:
>>> > CREATE TABLE `lcr` (
>>> > ??`id` INT(11) NOT NULL AUTO_INCREMENT,
>>> > ??`digits` VARCHAR(15) DEFAULT NULL,
>>> > ??`rate` FLOAT(11,5) DEFAULT NULL,
>>> > ??`intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>> > ??`intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>> > ??`carrier_id` INT(11) NOT NULL,
>>> > ??`lead_strip` INT(11) NOT NULL,
>>> > ??`trail_strip` INT(11) NOT NULL,
>>> > ??`prefix` VARCHAR(16) NOT NULL,
>>> > ??`suffix` VARCHAR(16) NOT NULL,
>>> > ??`lcr_profile` VARCHAR(32) DEFAULT NULL,
>>> > ??`date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>>> > ??`date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>> > ??`quality` FLOAT(10,6) NOT NULL,
>>> > ??`reliability` FLOAT(10,6) NOT NULL,
>>> > ??`cid` VARCHAR(32) NOT NULL DEFAULT '',
>>> > ??`enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>> > ??PRIMARY KEY ?(`id`),
>>> > ??KEY `carrier_id` (`carrier_id`),
>>> > ??KEY `digits` (`digits`),
>>> > ??KEY `lcr_profile` (`lcr_profile`),
>>> > ??KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>> > ??CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>>> > `carriers`
>>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>> >
>>> >
>>> > lcr_admin show profiles
>>> > Name: ? ? ? ? ? default
>>> > ?custom sql: ? ?SELECT l.digits, c.carrier_name, l.${lcr_rate_field},
>>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>> > l.trail_strip,
>>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
>>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
>>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN date_start
>>> > AND
>>> > date_end ORDER BY digits DESC, ?${lcr_rate_field}, ?quality DESC,
>>> > ?reliability DESC, rand();
>>> > ?has %: ? ? ? ? false
>>> > ?has vars: ? ? ?true
>>> > ?has intrastate: ? ? ? ?true
>>> > ?has intralata: true
>>> > ?has npanxx: ? ?true
>>> > ?Reorder rate: ?enabled
>>> > ?Info in headers: ? ? ? disabled
>>> > ?Quote IN() List: ? ? ? disabled
>>> >
>>> >
>>> >
>>> > lcr 617642 ?default ?returns rate from the rate field table and not
>>> > intra/inter state fields rates.
>>> >
>>> > Any ideas? thanks!
>>> >
>>> >
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa



From rupa at rupa.com  Sat Jan 30 07:03:48 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 30 Jan 2010 09:03:48 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
Message-ID: <b39273531001300703oe2d8289j6279eec3bdd3dc0d@mail.gmail.com>

Can you give me the first few lines of their rate table?

Is it:

NPANXXLATA = prefix
OCN = rate for same ocn
INTER = rate for interlata
INTRA = rate for intralata

or something else?

On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <mouncifbb at gmail.com> wrote:
> Also the Provider has presented the rates in this format?
> NPANXXLATA OCN INTER INTRA
>
>
> On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <mouncifbb at gmail.com>
> wrote:
>>
>> Tried it and it's not giving me?intralata??instead I get interstate, does
>> the?npa_nxx_company_ocn table needs to be used in this case?, also do I have
>> to have the rate field in lcr table?
>>
>> lcr 617642 ?default 6176421212
>> ?| Digit Match | Carrier ?| Rate ? ? | Codec | CID Regexp | Dialstring
>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?|
>> ?| 617642 ? ? ?| carrier1 | 0.00500 ?| ? ? ? | ? ? ? ? ? ?|
>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>> [617642 ?default 6176421212]
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>> [6176421212]
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing [state:0
>> lata:0] so rate field is [rate]
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
>> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled
>> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND CURRENT_TIMESTAMP
>> BETWEEN date_start AND date_end ORDER BY digits DESC, ?rate, rand();
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>> of list
>> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>
>> Thank you Rupa!
>>
>> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>> turn console logging up to debug and redo the lcr lookup. ?The sql
>>> statements along with status info will show up. ?This should give
>>> enough information to debug what is happening.
>>>
>>> I'm assuming the npanxx table is actually populated and not just
>>> existing?
>>>
>>> When doing the lookup from the cli you have to tell lcr what CID to
>>> use (remember, it is relative to the src/dest number). ?I'm pretty
>>> sure you get something on the console log when you don't specify a CID
>>> when using the commandline. ?Anyway:
>>>
>>> lcr 617642 ?default 6176421212
>>>
>>> should give you intralata.
>>>
>>> Note that the definition of intralata doesn't mean "local" for some
>>> providers. ?Some providers define local to "same ratecenter" which is
>>> even more restrictive.
>>>
>>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <mouncifbb at gmail.com>
>>> wrote:
>>> > i can't make use of mod_lcr using Intra/Interstate rating, I am using
>>> > svn:?FreeSWITCH Version 1.0.trunk (16517)
>>> >
>>> > lcr mysql table structure:
>>> > CREATE TABLE `lcr` (
>>> > ??`id` INT(11) NOT NULL AUTO_INCREMENT,
>>> > ??`digits` VARCHAR(15) DEFAULT NULL,
>>> > ??`rate` FLOAT(11,5) DEFAULT NULL,
>>> > ??`intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>> > ??`intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>> > ??`carrier_id` INT(11) NOT NULL,
>>> > ??`lead_strip` INT(11) NOT NULL,
>>> > ??`trail_strip` INT(11) NOT NULL,
>>> > ??`prefix` VARCHAR(16) NOT NULL,
>>> > ??`suffix` VARCHAR(16) NOT NULL,
>>> > ??`lcr_profile` VARCHAR(32) DEFAULT NULL,
>>> > ??`date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>>> > ??`date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>> > ??`quality` FLOAT(10,6) NOT NULL,
>>> > ??`reliability` FLOAT(10,6) NOT NULL,
>>> > ??`cid` VARCHAR(32) NOT NULL DEFAULT '',
>>> > ??`enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>> > ??PRIMARY KEY ?(`id`),
>>> > ??KEY `carrier_id` (`carrier_id`),
>>> > ??KEY `digits` (`digits`),
>>> > ??KEY `lcr_profile` (`lcr_profile`),
>>> > ??KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>> > ??CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>>> > `carriers`
>>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>> >
>>> >
>>> > lcr_admin show profiles
>>> > Name: ? ? ? ? ? default
>>> > ?custom sql: ? ?SELECT l.digits, c.carrier_name, l.${lcr_rate_field},
>>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>> > l.trail_strip,
>>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
>>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE
>>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN date_start
>>> > AND
>>> > date_end ORDER BY digits DESC, ?${lcr_rate_field}, ?quality DESC,
>>> > ?reliability DESC, rand();
>>> > ?has %: ? ? ? ? false
>>> > ?has vars: ? ? ?true
>>> > ?has intrastate: ? ? ? ?true
>>> > ?has intralata: true
>>> > ?has npanxx: ? ?true
>>> > ?Reorder rate: ?enabled
>>> > ?Info in headers: ? ? ? disabled
>>> > ?Quote IN() List: ? ? ? disabled
>>> >
>>> >
>>> >
>>> > lcr 617642 ?default ?returns rate from the rate field table and not
>>> > intra/inter state fields rates.
>>> >
>>> > Any ideas? thanks!
>>> >
>>> >
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa



From abid_freeswitch at live.com  Fri Jan 29 23:48:38 2010
From: abid_freeswitch at live.com (Abid Saleem)
Date: Sat, 30 Jan 2010 12:48:38 +0500
Subject: [Freeswitch-users] SS7 & MGCP support
Message-ID: <SNT106-W1023694FA10CC0F668ADCDF45A0@phx.gbl>


Hi,

 

I am new to FreeSwitch. Please help me answer my following
questions I could not find on wiki documentation.

 

?        
Since it is a softswitch also, does it support
SS7, MGCP and Megaco protocols to control media gateways?

?        
Does it support call shops business model?

?        
How to add new SIP user accounts into it that
can be used to register to it. I know one way is to copy and paste          1000.xml
file and edit it in the conf/directory folder. What is the optimal way to do
this task?o   Is there any GUI available. If yes how can I make it work and private label it.? 	

 

Thanks for your great help.
Regards-----------Abid SaleemProduct ManagerComcerto Bahrain W.L.L 		 	   		  
_________________________________________________________________
Hotmail: Powerful Free email with security by Microsoft.
https://signup.live.com/signup.aspx?id=60969
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From mike at jerris.com  Sat Jan 30 08:12:35 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 Jan 2010 11:12:35 -0500
Subject: [Freeswitch-users] little hangup problem - prepaid application
In-Reply-To: <7abab2411001252034kaf90e64g97b5d5c8f7d65e20@mail.gmail.com>
References: <4B5E0608.3070001@xpirio.com>
	<201001251620.35308.sos@sokhapkin.dyndns.org>
	<7abab2411001251713p31e542b2xf6204ff13c556d03@mail.gmail.com>
	<201001252234.08162.sos@sokhapkin.dyndns.org>
	<7abab2411001252034kaf90e64g97b5d5c8f7d65e20@mail.gmail.com>
Message-ID: <11212980-94E0-4C75-BACC-D2069644D663@jerris.com>

http://wiki.freeswitch.org/wiki/Lua is a good place to start.

On Jan 25, 2010, at 11:34 PM, Frank Church wrote:

> That is new to me, does that mean that all the languages linked in
> with Freeswitch have access to the events and variables in FS at all
> times?
> 
> Can you link me to the documenation that describes this part in more
> detail and some examples?
> 
> 2010/1/26 Sergey Okhapkin <sos at sokhapkin.dyndns.org>:
>> No record is sent to the script, but the script has access to all channel
>> variables.
>> 
>> On Monday 25 January 2010, Frank Church wrote:
>>> I am new to Freeswitch and I am interested in how it works. When the
>>> record is sent to the lua program what format is it sent in?
>>> 
> Frank Church

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From mike at jerris.com  Sat Jan 30 08:19:59 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 Jan 2010 11:19:59 -0500
Subject: [Freeswitch-users] External Profile Problem
In-Reply-To: <BAY143-W267C2F235ED9FCFA619FA1A15D0@phx.gbl>
References: <BAY143-W20A3102CC2FF36EF1C170FA15D0@phx.gbl>,
	<443888.41110.qm@web33505.mail.mud.yahoo.com>
	<BAY143-W267C2F235ED9FCFA619FA1A15D0@phx.gbl>
Message-ID: <74639FD5-11D5-4828-9D14-1E659CC30F52@jerris.com>

Contexts are sets of dialplan rules.  This allows you to have different rules for different "contexts"  such as, people dialing from the outside world are in one context, your internal users are in another.

Mike

On Jan 27, 2010, at 10:48 AM, juan camilo ospina quintero wrote:

> hi thanks
> 
> sorry but i dont really understand what a context is.
> 
> so, when i put <action application="transfer" data="$0 XML default"/>
> what does it really does, what it means that transfer to new context,
> 
> 
> bye
> 
> > Date: Wed, 27 Jan 2010 05:27:38 -0800
> > From: dftoro at yahoo.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] External Profile Problem
> > 
> > Hi, 
> > 
> > <action application="transfer" data="$0 XML default"/>
> > 
> > You must take into account that transfer application not "transfer" a call to destination, only transfer it to a new context
> > 
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer
> > 
> > 
> > Diego Toro
> > http://lacarretade.blogspot.com/
> > 
> > 
> > --- On Tue, 1/26/10, juan camilo ospina quintero <camilin2212 at hotmail.com> wrote:
> > 
> > > From: juan camilo ospina quintero <camilin2212 at hotmail.com>
> > > Subject: Re: [Freeswitch-users] External Profile Problem
> > > To: "freeswitch" <freeswitch-users at lists.freeswitch.org>
> > > Date: Tuesday, January 26, 2010, 11:06 PM
> > > 
> > > 
> > > 
> > > 
> > > 
> > > Hi
> > > 
> > > This works fine
> > > 
> > > <extension name="redirect">
> > >  <condition field="network_addr"
> > > expression="^192\.168\.2\.9$"/> 
> > > 
> > >  <condition field="destination_number"
> > > expression="^1(\d+)$">
> > >  <action application="bridge"
> > > data="sofia/external/$0 at 127.0.0.1:5070"/>
> > > </extension>
> > > 
> > > this redirects from freeswitch to sailfin (127.0.0.1:5070),
> > > and is in default.xml, in the dialplan.
> > > 
> > > the problem is this
> > > 
> > > <extension name="receive">
> > > <condition field="network_addr"
> > > expression="^127\.0\.0\.1$"/>
> > > <condition field="destination_number"
> > > expression="^1(\d+)$">
> > > <action application="transfer"
> > > data="$0 XML default"/>
> > > </condition>
> > > </extension>
> > > this doesnt work, this configuration can be found in
> > > public.xml in the dialplan, the idea of
> > > this is that when a sip invite comes from sailfin
> > > (127.0.0.1) transfer the invite to the destination number
> > > 
> > > the both configurations above are the only configuration i
> > > have change from  the default instalation of 
> > > freeswitch.
> > > 
> > > i would like to have some hep with this thanks
> > > 
> > > here is the trace log again
> > > 
> > > 2010-01-26 20:14:29.512927 [NOTICE]
> > > switch_channel.c:602 New Channel sofia/external/1000
> > > at 192.168.2.9 [5177e93a-0ae1-11df-afc9-db39c681a2f1]
> > > 2010-01-26 20:14:29.512927 [NOTICE] sofia.c:3527 Hangup
> > > sofia/external/1000
> > > at 192.168.2.9 [CS_NEW] [MANDATORY_IE_MISSING]
> > > send 632 bytes to udp/[192.168.2.9]:5070 at
> > > 01:14:29.517927:
> > > 
> > > ------------------------------------------------------------------------
> > > SIP/2.0 480 Temporarily Unavailable
> > > Via: SIP/2.0/UDP
> > > 192.168.153.1:5070;branch=z9hG4bKdaacdd64d693615c451ab9db43f9c71c2626;received=192.168.2.9
> > > From: <sip:1000
> > > at 192.168.2.9>;tag=g4xfbi12-3
> > > To: <sip:1001
> > > at 192.168.2.9:5080>;tag=4r91165pvcycB
> > > Call-Id: 192.168.153.1_3_3990383226484831353
> > > Cseq: 1 INVITE
> > > User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
> > > Accept: application/sdp
> > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
> > > MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
> > > Supported: timer, precondition, path, replaces
> > > Allow-Events: talk, refer
> > > Reason:
> > > Q.850;cause=96;text="MANDATORY_IE_MISSING"
> > > Content-Length: 0
> > > 
> > > 
> > > ------------------------------------------------------------------------
> > > 2010-01-26 20:14:29.525646 [NOTICE]
> > > switch_core_session.c:1086 Session 9 (sofia/external/1000
> > > at 192.168.2.9) Ended
> > > 2010-01-26 20:14:29.525646 [NOTICE]
> > > switch_core_session.c:1088 Close Channel sofia/external/1000
> > > at 192.168.2.9 [CS_DESTROY]
> > > 
> > > 
> > > 
> > > > Date: Tue, 26 Jan 2010 22:09:29 -0500
> > > > From: frank at carmickle.com
> > > > To: freeswitch-users at lists.freeswitch.org
> > > > Subject: Re: [Freeswitch-users] External Profile
> > > Problem
> > > > 
> > > > On Tue, Jan 26, juan camilo ospina quintero wrote:
> > > > > 
> > > > > Hi, 
> > > > > 
> > > > > im trying to establish a simple conference using
> > > freeswitch and sailfin, sailfin is
> > > > > and application server that works with
> > > SipSevlets.
> > > > > the all thing works as follow.
> > > > > 
> > > > > two softphone register with freeswitch, extension
> > > 1000 and 1001
> > > > > 1000 sends and invite to 1001, this invite goes
> > > to sailfin, i use this
> > > > > 
> > > > > <extension name="redirect">
> > > > > <condition
> > > field="network_addr"
> > > expression="^192\.168\.2\.9$"/> 
> > > 
> > > > > <condition
> > > field="destination_number"
> > > expression="^1(\d+)$">
> > > > > <action application="bridge"
> > > data="sofia/external/$0 at 127.0.0.1:5070"/>
> > > > 
> > > > And what is the external profile listening on? 
> > > Probably not the loopback address. Set up another profile
> > > listening on 127.0.0.1 and bridge to that.
> > > > 
> > > > I could be off base here because you haven't given
> > > us very much info about your freeswitch configurations.
> > > > 
> > > > --FC
> > > > 
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> > > 
> > > Windows Live: Friends
> > > get your Flickr, Yelp, and Digg updates when they e-mail
> > > you. 
> > > 
> > > -----Inline Attachment Follows-----
> > > 
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > > 
> > 
> > 
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> Hotmail: Trusted email with powerful SPAM protection. Sign up now. _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Sat Jan 30 08:48:18 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 30 Jan 2010 10:48:18 -0600
Subject: [Freeswitch-users] nixevent behavior
In-Reply-To: <7d79b3931001300648j6aa55258yfc496d9cea5c4b8b@mail.gmail.com>
References: <7d79b3931001300648j6aa55258yfc496d9cea5c4b8b@mail.gmail.com>
Message-ID: <191c3a031001300848h65d65c9cg9b355cd07e922@mail.gmail.com>

use $e = $con->sendRecv("command");

every time
for each send you do you must do a recv so this does both.


On Sat, Jan 30, 2010 at 8:48 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Dear all
>
> I've done the following sample script to experiment the nixevent. I found
> some difference in behavior because of nixevent. Let me explain my question
> down the script.
>
> require ESL;
> use IO::Socket::INET;
>
> my $ip = "192.168.1.222";
> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8447',
> Proto => 'tcp',  Listen => 1,  Reuse => 1 );
> die "Could not create socket: $!\n" unless $sock;
> for(;;) {
>     my $new_sock = $sock->accept();
>     next if (not defined ($new_sock));
>     my $pid = fork();
>     if ($pid) {
>         close($new_sock);
>         next;
>     }
>     print "CHILD PID: $$\n";
>     my $host = $new_sock->sockhost();
>     my $fd = fileno($new_sock);
>
>     my $con = new ESL::ESLconnection($fd);
>     my $info = $con->getInfo();
>
>     my $uuid = $info->getHeader("unique-id");
>
>     printf "Connected call %s, from %s\n", $uuid,
> $info->getHeader("caller-caller-id-number");
>     my $r=$con->execute("answer");
>     $con->events("plain","all");
> ##########################
>     $con->send("nixevent DTMF");
>     my $val=$con->api("create_uuid");
>     $val = $val->getBody();            # LINE 1
>     chomp($val);
>     print "UUID is $val\n";
>     my $e = $con->recvEvent();
>     $val = $e->getBody();               # LINE 2
>     chomp($val);
>     print "UUID is $val\n";
>     close($new_sock);
> }
>
> # If the line ($con->send("nixevent DTMF");) is commented, then the result
> of create_uuid is obtained in LINE 1.
> # else, the result isn't obtained in the LINE 1 and it has nothing. The
> result is obtained only when I do a recvEvent,
> # followed by a getBody (LINE 2)
>
> Just want to know why the behavior differs when nixevent is present???
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Sat Jan 30 08:54:02 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 30 Jan 2010 10:54:02 -0600
Subject: [Freeswitch-users] freeswitch with T.38
In-Reply-To: <1264862590045-4485495.post@n2.nabble.com>
References: <1264862590045-4485495.post@n2.nabble.com>
Message-ID: <191c3a031001300854r2dbc1cbeje06393183271b629@mail.gmail.com>

You can pay for commercial support contract to get priority bug fixes.
As for t38 you will have to wait for that feature a bit longer.


On Sat, Jan 30, 2010 at 8:43 AM, Sharad <sharad at coraltele.com> wrote:

>
> Is paid freeswitch available with T.38 support. ?
>
> regards
> Sharad
> --
> View this message in context:
> http://n2.nabble.com/freeswitch-with-T-38-tp4485495p4485495.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Sat Jan 30 08:57:02 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 30 Jan 2010 10:57:02 -0600
Subject: [Freeswitch-users] CPS per mod_sofia profile limited?
In-Reply-To: <201001301233.21516.errotan@gmail.com>
References: <4B62FF51.8070608@swip.net> <201001301233.21516.errotan@gmail.com>
Message-ID: <191c3a031001300857w36920b94r183f2e861db6baae@mail.gmail.com>

Also keep in mind that the industry standard is 50 which is the capacity to
take over for the real standard of 25 in a fail-over scenario.  So you
should be happy you even get 300cps for free.

The sofia stack can be improved but we are not the creators of this sip
stack.  There is little to no work being done on that project right now and
we are happy with what we have until we can get the lead dev to work on
improving it with us when he has the time.



On Sat, Jan 30, 2010 at 5:33 AM, Pusk?s Zsolt <errotan at gmail.com> wrote:

> 2010. janu?r 29. 16.31.29 Robin Vleij d?tummal ezt ?rta:
> > Hi guys,
> >
> > Doing a bit of testing / benchmarking with FS 1.04 (and 1.05 SVN). Found
> > out some interesting things, I think.
> >
> > The setup is like this:
> >
> > SIPP Client -> FS -> SIPP Server
> >
> > The dialplan is as simple as it gets:
> >
> > <extension name="test-out">
> >   <condition field="network_addr" expression="^130\.244\.190\.150$"/>
> >   <condition field="destination_number" expression="^(\d+)$">
> >     <action application="set" data="proxy_media=true"/>
> >     <action application="set" data="hangup_after_bridge=true"/>
> >     <action application="bridge"
> >
> data="{sip_contact_user=transit}sofia/gateway/${distributor(gwg1)}/$1|sofia
> > /gateway/${distributor(gwg2)}/$1"/> </condition>
> > </extension>
> >
> > For the rest it's running CSV cdr's, commented out all modules I'm not
> > using, etc etc all that I could find on the wiki and the Interwebs.
> > Hardware / OS: from the shelve quadcore Xeon, debian 64-bit, 12GB
> > memory. SIPP is running a 500ms RTP pcap and the other side echos back.
> >
> > I had a few test setups then:
> >
> > 1: FS SVN, 1 sofia profile where the gateways were configured and the
> > server_IP:5060 was used.
> >
> > 2. FS SVN, 2 sofia profiles where the gateways where in an seperate
> > profile (server_IP:5070) and the "customer facing" side was the original
> > profile.
> >
> > 3. FS 1.04, same as above
> >
> > 4. FS 1.04, 4 sofia profiles, distributor to spread load over 2 incoming
> > and 2 outgoing profiles.
> >
> > Now the interesting thing was that under 1 I could go up, almost without
> > any CPU load, to 50cps. As soon as I went over this, calls where handled
> >  slower and "ongoing" calls would pile up untill it became really slow.
> > CPU load went to 100% on the FS process (both user and system time).
> > Lots of interupts and context switches. No throughput anymore untill I
> > lower and wait till the "buffer" is empty and FS is keeping up again.
> >
> > Under 2, I was able to increase the CPS to about 100 with the same
> effect.
> >
> > 3 then went much better, I was able to increase CPS to about 200 cps and
> > response times in SIPP went up slighty untill it just hits some kind of
> > limit and calls are handled slower.
> >
> > 4 is pretty cool. Here I can run 2 sipp clients both doing 150cps to the
> > gateways that are spread on the distribution module, so I spread traffic
> > over 2 profiles. With 300 cps in total, FS is keeping up and I have 30%
> > idle CPU. However, increasing to over 300cps gives problems again, even
> > though I have idle CPU left!
> >
> > All in all, I have a feeling that a single sip profile can't run more
> > than a certain limit untill it gets into some problem. Depending on if
> > I'm running SVN or 1.04 that limit seems to be 50cps or a bit higher.
> > After that limit it starts piling up "ongoing" calls, by taking time to
> > handle them and when that limit gets too high it's too late. All in all
> > really fine, I just set the system wide limit to a little under that
> > "threshold". But when I'm running just UNDER the threshold it's not CPU
> > that's a problem. Theoretically I should be able to run (based on the
> > CPU usage at 300cps) about 400cps.
> >
> > When running at 300 I get SOME failed calls and I see
> >
> > "switch_core_state_machine.c:525 a9a60636-0cea-11df-85a1-09c991f2afc5
> > Timeout waiting for next instruction in CS_NEW!"
> >
> > in the console.
> >
> > I didn't find much on how people do high cps setups and it feels a bit
> > like a "friday afternoon solution" to run multiple sofia profiles on the
> > same machine in order to max out the system.
> >
> > Maybe I'm missing something and I know it's not an exact science this,
> > but I'm not sure "all is OK" because I'm not slowly getting to a 100%
> > cpu (or disk / network) usage, I hit some kind of limit after which
> > stuff goes wrong.
> >
> > Anyone any input!
> >
> > /Robin
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> CPU usage is not the only thing that limit your calls. Have you set the
> recommended ulimit settings and / or started fs with the -waste option ?
>
> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From mouncifbb at gmail.com  Sat Jan 30 13:38:22 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Sat, 30 Jan 2010 16:38:22 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
Message-ID: <ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>

  NPANXX,"LATA","OCN","NTER","INTRA"  201007,"224","7229","0.0059","0.0127"
201040,"224","9206","0.0036","0.0036"
FreeSWITCH Version 1.0.trunk (16540)


Also I noticed the *npa_nxx_ocn* table never get consulted.

I also see this now when making a real call instead of running thorugh CLI

EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890 NANPA_STD)
2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate channel var is
[undef]*
2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based on
interstate rates
2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on 16179470893
using profile NANPA_STD
2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing [state:0
lata:0] so rate field is [rate]

called number 6179470890  caller ID: 6179472456

any ideas??




On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> Something is still missing from the logs. Note the query of the npanxx
> table, the flags being set, and the rate field being chosen.  Umm..
> oh, what version of fs are you running?
>
> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>
> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
> (there is a link to that from mod_lcr's wiki page).
>
> An example from my own setup:
>
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
> is [12148267711 default 12148267712]
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
> [12148267712]
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR (npa=214
> AND nxx=826)
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count: 1
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
> [state:1 lata:1] so rate field is [intralata_rate]
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT l.digits
> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
> l.intralata_rate as lcr_rate_field,        cg.prefix AS lcr_gw_prefix,
> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
> random();
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>
> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
> head of list
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of
> list
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of
> list
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to end of
> list
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>
> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
> [...]
>
> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <mouncifbb at gmail.com>
> wrote:
> > Also the Provider has presented the rates in this format?
> > NPANXXLATA OCN INTER INTRA
> >
> >
> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <mouncifbb at gmail.com>
> > wrote:
> >>
> >> Tried it and it's not giving me intralata  instead I get interstate,
> does
> >> the npa_nxx_company_ocn table needs to be used in this case?, also do I
> have
> >> to have the rate field in lcr table?
> >>
> >> lcr 617642  default 6176421212
> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp | Dialstring
> >>                                                        |
> >>  | 617642      | carrier1 | 0.00500  |       |            |
> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
> >>
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr is
> >> [617642  default 6176421212]
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
> >> [6176421212]
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing [state:0
> >> lata:0] so rate field is [rate]
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM
> lcr
> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
> l.enabled
> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
> CURRENT_TIMESTAMP
> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate, rand();
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
> >> of list
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
> >>
> >> Thank you Rupa!
> >>
> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> >>>
> >>> turn console logging up to debug and redo the lcr lookup.  The sql
> >>> statements along with status info will show up.  This should give
> >>> enough information to debug what is happening.
> >>>
> >>> I'm assuming the npanxx table is actually populated and not just
> >>> existing?
> >>>
> >>> When doing the lookup from the cli you have to tell lcr what CID to
> >>> use (remember, it is relative to the src/dest number).  I'm pretty
> >>> sure you get something on the console log when you don't specify a CID
> >>> when using the commandline.  Anyway:
> >>>
> >>> lcr 617642  default 6176421212
> >>>
> >>> should give you intralata.
> >>>
> >>> Note that the definition of intralata doesn't mean "local" for some
> >>> providers.  Some providers define local to "same ratecenter" which is
> >>> even more restrictive.
> >>>
> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <mouncifbb at gmail.com
> >
> >>> wrote:
> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I am using
> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
> >>> >
> >>> > lcr mysql table structure:
> >>> > CREATE TABLE `lcr` (
> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
> >>> >   `carrier_id` INT(11) NOT NULL,
> >>> >   `lead_strip` INT(11) NOT NULL,
> >>> >   `trail_strip` INT(11) NOT NULL,
> >>> >   `prefix` VARCHAR(16) NOT NULL,
> >>> >   `suffix` VARCHAR(16) NOT NULL,
> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
> >>> >   `quality` FLOAT(10,6) NOT NULL,
> >>> >   `reliability` FLOAT(10,6) NOT NULL,
> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
> >>> >   PRIMARY KEY  (`id`),
> >>> >   KEY `carrier_id` (`carrier_id`),
> >>> >   KEY `digits` (`digits`),
> >>> >   KEY `lcr_profile` (`lcr_profile`),
> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
> >>> > `carriers`
> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
> >>> >
> >>> >
> >>> > lcr_admin show profiles
> >>> > Name:           default
> >>> >  custom sql:    SELECT l.digits, c.carrier_name, l.${lcr_rate_field},
> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
> >>> > l.trail_strip,
> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
> WHERE
> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits
> IN
> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
> date_start
> >>> > AND
> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality DESC,
> >>> >  reliability DESC, rand();
> >>> >  has %:         false
> >>> >  has vars:      true
> >>> >  has intrastate:        true
> >>> >  has intralata: true
> >>> >  has npanxx:    true
> >>> >  Reorder rate:  enabled
> >>> >  Info in headers:       disabled
> >>> >  Quote IN() List:       disabled
> >>> >
> >>> >
> >>> >
> >>> > lcr 617642  default  returns rate from the rate field table and not
> >>> > intra/inter state fields rates.
> >>> >
> >>> > Any ideas? thanks!
> >>> >
> >>> >
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>>
> >>>
> >>> --
> >>> -Rupa
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rupa at rupa.com  Sat Jan 30 15:59:27 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 30 Jan 2010 17:59:27 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
Message-ID: <b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>

Stuff inline.

On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

>  NPANXX,"LATA","OCN","NTER","INTRA"  201007,"224","7229","0.0059","0.0127"
> 201040,"224","9206","0.0036","0.0036"
>

Looks like they give you the LATA and OCN values with the prefix.  We
(should) look that up ourselves.


> FreeSWITCH Version 1.0.trunk (16540)
>
>
> Also I noticed the *npa_nxx_ocn* table never get consulted.
>
> I also see this now when making a real call instead of running thorugh CLI
>
> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890 NANPA_STD)
> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate channel var
> is [undef]*


This is fine.  it is a leftover from when you would tell mod_lcr via a
channel var that it should do intrastate.  I later had mod_lcr do the lookup
itself, but we still honor the old var.  There are no channel vars
associated with the cli, so you wouldn't see that msg.


>
> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based on
> interstate rates
> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on 16179470893
> using profile NANPA_STD
> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing [state:0
> lata:0] so rate field is [rate]
>
> called number 6179470890  caller ID: 6179472456
>
> any ideas??
>
>
Only thing that jumps out at me.

The output from lcr_admin show profiles showed only the default one.  On the
dialplan you use the NANPA_STD profile.  Can you check lcr_admin list and
see if that profile is defined and if so if it says it is using the npanxx
table?




>
>
>
>
> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> Something is still missing from the logs. Note the query of the npanxx
>> table, the flags being set, and the rate field being chosen.  Umm..
>> oh, what version of fs are you running?
>>
>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>
>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>> (there is a link to that from mod_lcr's wiki page).
>>
>> An example from my own setup:
>>
>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>> is [12148267711 default 12148267712]
>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>> [12148267712]
>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR (npa=214
>> AND nxx=826)
>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count: 1
>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>> [state:1 lata:1] so rate field is [intralata_rate]
>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT l.digits
>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>> l.intralata_rate as lcr_rate_field,        cg.prefix AS lcr_gw_prefix,
>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>> random();
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>
>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
>> head of list
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of
>> list
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of
>> list
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to end of
>> list
>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>
>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>> [...]
>>
>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <mouncifbb at gmail.com>
>> wrote:
>> > Also the Provider has presented the rates in this format?
>> > NPANXXLATA OCN INTER INTRA
>> >
>> >
>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <mouncifbb at gmail.com
>> >
>> > wrote:
>> >>
>> >> Tried it and it's not giving me intralata  instead I get interstate,
>> does
>> >> the npa_nxx_company_ocn table needs to be used in this case?, also do I
>> have
>> >> to have the rate field in lcr table?
>> >>
>> >> lcr 617642  default 6176421212
>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp | Dialstring
>> >>                                                        |
>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>> >>
>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>> >> [617642  default 6176421212]
>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>> >> [6176421212]
>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing [state:0
>> >> lata:0] so rate field is [rate]
>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM
>> lcr
>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
>> l.enabled
>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>> CURRENT_TIMESTAMP
>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate, rand();
>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to
>> head
>> >> of list
>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>> >>
>> >> Thank you Rupa!
>> >>
>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>> >>>
>> >>> turn console logging up to debug and redo the lcr lookup.  The sql
>> >>> statements along with status info will show up.  This should give
>> >>> enough information to debug what is happening.
>> >>>
>> >>> I'm assuming the npanxx table is actually populated and not just
>> >>> existing?
>> >>>
>> >>> When doing the lookup from the cli you have to tell lcr what CID to
>> >>> use (remember, it is relative to the src/dest number).  I'm pretty
>> >>> sure you get something on the console log when you don't specify a CID
>> >>> when using the commandline.  Anyway:
>> >>>
>> >>> lcr 617642  default 6176421212
>> >>>
>> >>> should give you intralata.
>> >>>
>> >>> Note that the definition of intralata doesn't mean "local" for some
>> >>> providers.  Some providers define local to "same ratecenter" which is
>> >>> even more restrictive.
>> >>>
>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <
>> mouncifbb at gmail.com>
>> >>> wrote:
>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I am
>> using
>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>> >>> >
>> >>> > lcr mysql table structure:
>> >>> > CREATE TABLE `lcr` (
>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>> >>> >   `carrier_id` INT(11) NOT NULL,
>> >>> >   `lead_strip` INT(11) NOT NULL,
>> >>> >   `trail_strip` INT(11) NOT NULL,
>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>> >>> >   PRIMARY KEY  (`id`),
>> >>> >   KEY `carrier_id` (`carrier_id`),
>> >>> >   KEY `digits` (`digits`),
>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>> >>> > `carriers`
>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>> >>> >
>> >>> >
>> >>> > lcr_admin show profiles
>> >>> > Name:           default
>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>> l.${lcr_rate_field},
>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>> >>> > l.trail_strip,
>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>> WHERE
>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits
>> IN
>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>> date_start
>> >>> > AND
>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality DESC,
>> >>> >  reliability DESC, rand();
>> >>> >  has %:         false
>> >>> >  has vars:      true
>> >>> >  has intrastate:        true
>> >>> >  has intralata: true
>> >>> >  has npanxx:    true
>> >>> >  Reorder rate:  enabled
>> >>> >  Info in headers:       disabled
>> >>> >  Quote IN() List:       disabled
>> >>> >
>> >>> >
>> >>> >
>> >>> > lcr 617642  default  returns rate from the rate field table and not
>> >>> > intra/inter state fields rates.
>> >>> >
>> >>> > Any ideas? thanks!
>> >>> >
>> >>> >
>> >>> >
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> -Rupa
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From mouncifbb at gmail.com  Sat Jan 30 16:23:33 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Sat, 30 Jan 2010 19:23:33 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
Message-ID: <ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>

yes I use NANPA_STD profile instead of default cause I thought the custom
profile was causing issues, but looks like it's returning same results.

There is this line in thw wiki:
intra lata/state selection is done manually by setting the channel variables
*intrastate* or *intralata* to the value *true*.

do I have to set these ? if yes how?

Thanks

On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Stuff inline.
>
> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>
>>  NPANXX,"LATA","OCN","NTER","INTRA"
>> 201007,"224","7229","0.0059","0.0127"
>> 201040,"224","9206","0.0036","0.0036"
>>
>
> Looks like they give you the LATA and OCN values with the prefix.  We
> (should) look that up ourselves.
>
>
>> FreeSWITCH Version 1.0.trunk (16540)
>>
>>
>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>
>> I also see this now when making a real call instead of running thorugh CLI
>>
>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890 NANPA_STD)
>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate channel var
>> is [undef]*
>
>
> This is fine.  it is a leftover from when you would tell mod_lcr via a
> channel var that it should do intrastate.  I later had mod_lcr do the lookup
> itself, but we still honor the old var.  There are no channel vars
> associated with the cli, so you wouldn't see that msg.
>
>
>>
>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based on
>> interstate rates
>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>> 16179470893 using profile NANPA_STD
>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing [state:0
>> lata:0] so rate field is [rate]
>>
>> called number 6179470890  caller ID: 6179472456
>>
>> any ideas??
>>
>>
> Only thing that jumps out at me.
>
> The output from lcr_admin show profiles showed only the default one.  On
> the dialplan you use the NANPA_STD profile.  Can you check lcr_admin list
> and see if that profile is defined and if so if it says it is using the
> npanxx table?
>
>
>
>
>>
>>
>>
>>
>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>> Something is still missing from the logs. Note the query of the npanxx
>>> table, the flags being set, and the rate field being chosen.  Umm..
>>> oh, what version of fs are you running?
>>>
>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>> (there is a link to that from mod_lcr's wiki page).
>>>
>>> An example from my own setup:
>>>
>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>>> is [12148267711 default 12148267712]
>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>> [12148267712]
>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR (npa=214
>>> AND nxx=826)
>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count: 1
>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>> [state:1 lata:1] so rate field is [intralata_rate]
>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT l.digits
>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS lcr_gw_prefix,
>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>> random();
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>
>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
>>> head of list
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of
>>> list
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of
>>> list
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to end
>>> of list
>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>
>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>> [...]
>>>
>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <mouncifbb at gmail.com>
>>> wrote:
>>> > Also the Provider has presented the rates in this format?
>>> > NPANXXLATA OCN INTER INTRA
>>> >
>>> >
>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <
>>> mouncifbb at gmail.com>
>>> > wrote:
>>> >>
>>> >> Tried it and it's not giving me intralata  instead I get interstate,
>>> does
>>> >> the npa_nxx_company_ocn table needs to be used in this case?, also do
>>> I have
>>> >> to have the rate field in lcr table?
>>> >>
>>> >> lcr 617642  default 6176421212
>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp | Dialstring
>>> >>                                                        |
>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>> |
>>> >>
>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr
>>> is
>>> >> [617642  default 6176421212]
>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>> >> [6176421212]
>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>> [state:0
>>> >> lata:0] so rate field is [rate]
>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>> gw_suffix,
>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid
>>> FROM lcr
>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
>>> l.enabled
>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>> CURRENT_TIMESTAMP
>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate, rand();
>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to
>>> head
>>> >> of list
>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>> >>
>>> >> Thank you Rupa!
>>> >>
>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com>
>>> wrote:
>>> >>>
>>> >>> turn console logging up to debug and redo the lcr lookup.  The sql
>>> >>> statements along with status info will show up.  This should give
>>> >>> enough information to debug what is happening.
>>> >>>
>>> >>> I'm assuming the npanxx table is actually populated and not just
>>> >>> existing?
>>> >>>
>>> >>> When doing the lookup from the cli you have to tell lcr what CID to
>>> >>> use (remember, it is relative to the src/dest number).  I'm pretty
>>> >>> sure you get something on the console log when you don't specify a
>>> CID
>>> >>> when using the commandline.  Anyway:
>>> >>>
>>> >>> lcr 617642  default 6176421212
>>> >>>
>>> >>> should give you intralata.
>>> >>>
>>> >>> Note that the definition of intralata doesn't mean "local" for some
>>> >>> providers.  Some providers define local to "same ratecenter" which is
>>> >>> even more restrictive.
>>> >>>
>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <
>>> mouncifbb at gmail.com>
>>> >>> wrote:
>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I am
>>> using
>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>> >>> >
>>> >>> > lcr mysql table structure:
>>> >>> > CREATE TABLE `lcr` (
>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>> >>> >   PRIMARY KEY  (`id`),
>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>> >>> >   KEY `digits` (`digits`),
>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>>> >>> > `carriers`
>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>> >>> >
>>> >>> >
>>> >>> > lcr_admin show profiles
>>> >>> > Name:           default
>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>> l.${lcr_rate_field},
>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>> >>> > l.trail_strip,
>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON
>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>> WHERE
>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits
>>> IN
>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>> date_start
>>> >>> > AND
>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality DESC,
>>> >>> >  reliability DESC, rand();
>>> >>> >  has %:         false
>>> >>> >  has vars:      true
>>> >>> >  has intrastate:        true
>>> >>> >  has intralata: true
>>> >>> >  has npanxx:    true
>>> >>> >  Reorder rate:  enabled
>>> >>> >  Info in headers:       disabled
>>> >>> >  Quote IN() List:       disabled
>>> >>> >
>>> >>> >
>>> >>> >
>>> >>> > lcr 617642  default  returns rate from the rate field table and not
>>> >>> > intra/inter state fields rates.
>>> >>> >
>>> >>> > Any ideas? thanks!
>>> >>> >
>>> >>> >
>>> >>> >
>>> >>> >
>>> >>> > _______________________________________________
>>> >>> > FreeSWITCH-users mailing list
>>> >>> > FreeSWITCH-users at lists.freeswitch.org
>>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> >
>>> >>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> > http://www.freeswitch.org
>>> >>> >
>>> >>> >
>>> >>>
>>> >>>
>>> >>>
>>> >>> --
>>> >>> -Rupa
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rupa at rupa.com  Sat Jan 30 16:45:35 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 30 Jan 2010 18:45:35 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
Message-ID: <b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>

turn up logging to debug again, and then reload mod_lcr.  It'll spit out a
bunch of crap when it tests out each profile you have defined.  Give me the
full log (here or in pastebin.freeswitch.org).  That may show more useful
info as to why things are mucked up?

On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> yes I use NANPA_STD profile instead of default cause I thought the custom
> profile was causing issues, but looks like it's returning same results.
>
> There is this line in thw wiki:
> intra lata/state selection is done manually by setting the channel
> variables *intrastate* or *intralata* to the value *true*.
>
> do I have to set these ? if yes how?
>
> Thanks
>
>
> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> Stuff inline.
>>
>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>>
>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>> 201007,"224","7229","0.0059","0.0127"
>>> 201040,"224","9206","0.0036","0.0036"
>>>
>>
>> Looks like they give you the LATA and OCN values with the prefix.  We
>> (should) look that up ourselves.
>>
>>
>>> FreeSWITCH Version 1.0.trunk (16540)
>>>
>>>
>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>
>>> I also see this now when making a real call instead of running thorugh
>>> CLI
>>>
>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890 NANPA_STD)
>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate channel
>>> var is [undef]*
>>
>>
>> This is fine.  it is a leftover from when you would tell mod_lcr via a
>> channel var that it should do intrastate.  I later had mod_lcr do the lookup
>> itself, but we still honor the old var.  There are no channel vars
>> associated with the cli, so you wouldn't see that msg.
>>
>>
>>>
>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based on
>>> interstate rates
>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>> 16179470893 using profile NANPA_STD
>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>> lata:0] so rate field is [rate]
>>>
>>> called number 6179470890  caller ID: 6179472456
>>>
>>> any ideas??
>>>
>>>
>> Only thing that jumps out at me.
>>
>> The output from lcr_admin show profiles showed only the default one.  On
>> the dialplan you use the NANPA_STD profile.  Can you check lcr_admin list
>> and see if that profile is defined and if so if it says it is using the
>> npanxx table?
>>
>>
>>
>>
>>>
>>>
>>>
>>>
>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>>> Something is still missing from the logs. Note the query of the npanxx
>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>> oh, what version of fs are you running?
>>>>
>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>
>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>> (there is a link to that from mod_lcr's wiki page).
>>>>
>>>> An example from my own setup:
>>>>
>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>>>> is [12148267711 default 12148267712]
>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>> [12148267712]
>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR (npa=214
>>>> AND nxx=826)
>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count: 1
>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT l.digits
>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS lcr_gw_prefix,
>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>> random();
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>
>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
>>>> head of list
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>
>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of
>>>> list
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end of
>>>> list
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>
>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to end
>>>> of list
>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>
>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>> [...]
>>>>
>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <mouncifbb at gmail.com>
>>>> wrote:
>>>> > Also the Provider has presented the rates in this format?
>>>> > NPANXXLATA OCN INTER INTRA
>>>> >
>>>> >
>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <
>>>> mouncifbb at gmail.com>
>>>> > wrote:
>>>> >>
>>>> >> Tried it and it's not giving me intralata  instead I get interstate,
>>>> does
>>>> >> the npa_nxx_company_ocn table needs to be used in this case?, also do
>>>> I have
>>>> >> to have the rate field in lcr table?
>>>> >>
>>>> >> lcr 617642  default 6176421212
>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>> Dialstring
>>>> >>                                                        |
>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>> |
>>>> >>
>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr
>>>> is
>>>> >> [617642  default 6176421212]
>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>> >> [6176421212]
>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>> [state:0
>>>> >> lata:0] so rate field is [rate]
>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>> l.digits,
>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>> gw_suffix,
>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid
>>>> FROM lcr
>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
>>>> l.enabled
>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>> CURRENT_TIMESTAMP
>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate, rand();
>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to
>>>> head
>>>> >> of list
>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>>> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>> >>
>>>> >> Thank you Rupa!
>>>> >>
>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com>
>>>> wrote:
>>>> >>>
>>>> >>> turn console logging up to debug and redo the lcr lookup.  The sql
>>>> >>> statements along with status info will show up.  This should give
>>>> >>> enough information to debug what is happening.
>>>> >>>
>>>> >>> I'm assuming the npanxx table is actually populated and not just
>>>> >>> existing?
>>>> >>>
>>>> >>> When doing the lookup from the cli you have to tell lcr what CID to
>>>> >>> use (remember, it is relative to the src/dest number).  I'm pretty
>>>> >>> sure you get something on the console log when you don't specify a
>>>> CID
>>>> >>> when using the commandline.  Anyway:
>>>> >>>
>>>> >>> lcr 617642  default 6176421212
>>>> >>>
>>>> >>> should give you intralata.
>>>> >>>
>>>> >>> Note that the definition of intralata doesn't mean "local" for some
>>>> >>> providers.  Some providers define local to "same ratecenter" which
>>>> is
>>>> >>> even more restrictive.
>>>> >>>
>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <
>>>> mouncifbb at gmail.com>
>>>> >>> wrote:
>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I am
>>>> using
>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>> >>> >
>>>> >>> > lcr mysql table structure:
>>>> >>> > CREATE TABLE `lcr` (
>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>> >>> >   PRIMARY KEY  (`id`),
>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>> >>> >   KEY `digits` (`digits`),
>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>>>> >>> > `carriers`
>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>> >>> >
>>>> >>> >
>>>> >>> > lcr_admin show profiles
>>>> >>> > Name:           default
>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>> l.${lcr_rate_field},
>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>> >>> > l.trail_strip,
>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c
>>>> ON
>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>> WHERE
>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>> digits IN
>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>> date_start
>>>> >>> > AND
>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality DESC,
>>>> >>> >  reliability DESC, rand();
>>>> >>> >  has %:         false
>>>> >>> >  has vars:      true
>>>> >>> >  has intrastate:        true
>>>> >>> >  has intralata: true
>>>> >>> >  has npanxx:    true
>>>> >>> >  Reorder rate:  enabled
>>>> >>> >  Info in headers:       disabled
>>>> >>> >  Quote IN() List:       disabled
>>>> >>> >
>>>> >>> >
>>>> >>> >
>>>> >>> > lcr 617642  default  returns rate from the rate field table and
>>>> not
>>>> >>> > intra/inter state fields rates.
>>>> >>> >
>>>> >>> > Any ideas? thanks!
>>>> >>> >
>>>> >>> >
>>>> >>> >
>>>> >>> >
>>>> >>> > _______________________________________________
>>>> >>> > FreeSWITCH-users mailing list
>>>> >>> > FreeSWITCH-users at lists.freeswitch.org
>>>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>> >
>>>> >>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>> > http://www.freeswitch.org
>>>> >>> >
>>>> >>> >
>>>> >>>
>>>> >>>
>>>> >>>
>>>> >>> --
>>>> >>> -Rupa
>>>> >>>
>>>> >>> _______________________________________________
>>>> >>> FreeSWITCH-users mailing list
>>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>> http://www.freeswitch.org
>>>> >>
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From jwssr at charter.net  Sat Jan 30 10:11:42 2010
From: jwssr at charter.net (jwssr)
Date: Sat, 30 Jan 2010 10:11:42 -0800 (PST)
Subject: [Freeswitch-users]  problem with hard phone
Message-ID: <27386032.post@talk.nabble.com>


Iam having probles with my hard phone...van access ip0020...which is not
completing the 'B' leg on incoming calls to it
and 
on outgoing calls....immedialtely issuing a 488...
and 
fs log states ...
switch_core_codec.c:537 Codec G722 Exists but not at the desired
implementation. 16000hz 20ms
2010-01-30 11:49:07.736157 [ERR] sofia_glue.c:2126 Can't load codec?
...
I have multiple soft phones (sflphones  extensions 1000-1019), a dlink wifi
dph-540 (extension 4001 ip=192.168.1.190), and this van access ip0020 hard
phone (extension 2001 ip=192.168.1.180).....fs running on 192.168.1.102.

all phones sans '2001"  work perfectly..to/fro vitelity and local on lan.

Im almost certain that the problem lies in the port number assigned to the
uri.  wireshark shows that call was cancel because port could not be reached 
and 
port number is missing on ua shown by fs cli.

Call-ID:        af4cfd49b52bf8b3b3ec8db3e8a309e5 at 192.168.1.190
User:           4001 at 192.168.1.102
Contact:        4001 <sip:4001 at 192.168.1.190:5060>
...
...

Call-ID:        4c1ed9ae64405367 at 192.168.1.180
User:           2001 at 192.168.1.102
Contact:        "user" <sip:2001 at 192.168.1.180>
...                                   notice missing port no. here

Call-ID:        9be86b04-720f-4f7c-af52-b4f67dccf76b
User:           1000 at 192.168.1.102
Contact:        "jon" <sip:1000 at 192.168.1.101:5060>  
....

wireshark.....
292 14.903431   192.168.1.102         192.168.1.180         ICMP    
Destination unreachable (Port unreachable)
295 14.927845   192.168.1.102         192.168.1.180         SIP     
Request: BYE sip:2001 at 192.168.1.180

I have output from siptrace...but did not want to clutter up forum with too
much detail but can provide.

I also have screenshots of html config of phone...

I would appreciate some help please

thanks
-- 
View this message in context: http://old.nabble.com/problem-with-hard-phone-tp27386032p27386032.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From mike at jerris.com  Sat Jan 30 17:57:38 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 Jan 2010 20:57:38 -0500
Subject: [Freeswitch-users] CPS per mod_sofia profile limited?
In-Reply-To: <191c3a031001300857w36920b94r183f2e861db6baae@mail.gmail.com>
References: <4B62FF51.8070608@swip.net> <201001301233.21516.errotan@gmail.com>
	<191c3a031001300857w36920b94r183f2e861db6baae@mail.gmail.com>
Message-ID: <A1448B87-D14D-4181-892A-9ECCD5EB41F0@jerris.com>

For clarification, is it correct that your getting worse numbers for sustainable cps on SVN then on 1.0.4?  I would be interested in the numbers you would get with bypass_media=true instead of proxy_media=true and with neither setting set as well.  Also, make sure your logging level is low and try putting the db dir on a ram disk.  Thanks for the info.  As tony said, there are clearly some bottlenecks in the sofia library, but if you really need to pass media, your test is not very accurate for real life, and the results are likely not very useful to you.  You should use a more realistic length of call.  In bypass media I would suspect that length of call matters very little.  In proxy media or normal mode, the performance of the box is much more of a calculation on number of calls than cps as a result of the context switching from having to move the media and you will see that this plays a much more significant role on realistic call lengths (unless you really have lots of 1/2 second calls).  Some other tips.  While this extension may be trivial, what else is there in your dialplan context?  Anything above that extension could cause a significant impact.  Do you have any of the presence features enabled?  These do significantly impact call handling performance even if your sipp scenarios do not send any of those packets.

Mike


On Jan 30, 2010, at 11:57 AM, Anthony Minessale wrote:

> Also keep in mind that the industry standard is 50 which is the capacity to take over for the real standard of 25 in a fail-over scenario.  So you should be happy you even get 300cps for free.
> 
> The sofia stack can be improved but we are not the creators of this sip stack.  There is little to no work being done on that project right now and we are happy with what we have until we can get the lead dev to work on improving it with us when he has the time.
> 
> 
> 
> On Sat, Jan 30, 2010 at 5:33 AM, Pusk?s Zsolt <errotan at gmail.com> wrote:
> 2010. janu?r 29. 16.31.29 Robin Vleij d?tummal ezt ?rta:
> > Hi guys,
> >
> > Doing a bit of testing / benchmarking with FS 1.04 (and 1.05 SVN). Found
> > out some interesting things, I think.
> >
> > The setup is like this:
> >
> > SIPP Client -> FS -> SIPP Server
> >
> > The dialplan is as simple as it gets:
> >
> > <extension name="test-out">
> >   <condition field="network_addr" expression="^130\.244\.190\.150$"/>
> >   <condition field="destination_number" expression="^(\d+)$">
> >     <action application="set" data="proxy_media=true"/>
> >     <action application="set" data="hangup_after_bridge=true"/>
> >     <action application="bridge"
> > data="{sip_contact_user=transit}sofia/gateway/${distributor(gwg1)}/$1|sofia
> > /gateway/${distributor(gwg2)}/$1"/> </condition>
> > </extension>
> >
> > For the rest it's running CSV cdr's, commented out all modules I'm not
> > using, etc etc all that I could find on the wiki and the Interwebs.
> > Hardware / OS: from the shelve quadcore Xeon, debian 64-bit, 12GB
> > memory. SIPP is running a 500ms RTP pcap and the other side echos back.
> >
> > I had a few test setups then:
> >
> > 1: FS SVN, 1 sofia profile where the gateways were configured and the
> > server_IP:5060 was used.
> >
> > 2. FS SVN, 2 sofia profiles where the gateways where in an seperate
> > profile (server_IP:5070) and the "customer facing" side was the original
> > profile.
> >
> > 3. FS 1.04, same as above
> >
> > 4. FS 1.04, 4 sofia profiles, distributor to spread load over 2 incoming
> > and 2 outgoing profiles.
> >
> > Now the interesting thing was that under 1 I could go up, almost without
> > any CPU load, to 50cps. As soon as I went over this, calls where handled
> >  slower and "ongoing" calls would pile up untill it became really slow.
> > CPU load went to 100% on the FS process (both user and system time).
> > Lots of interupts and context switches. No throughput anymore untill I
> > lower and wait till the "buffer" is empty and FS is keeping up again.
> >
> > Under 2, I was able to increase the CPS to about 100 with the same effect.
> >
> > 3 then went much better, I was able to increase CPS to about 200 cps and
> > response times in SIPP went up slighty untill it just hits some kind of
> > limit and calls are handled slower.
> >
> > 4 is pretty cool. Here I can run 2 sipp clients both doing 150cps to the
> > gateways that are spread on the distribution module, so I spread traffic
> > over 2 profiles. With 300 cps in total, FS is keeping up and I have 30%
> > idle CPU. However, increasing to over 300cps gives problems again, even
> > though I have idle CPU left!
> >
> > All in all, I have a feeling that a single sip profile can't run more
> > than a certain limit untill it gets into some problem. Depending on if
> > I'm running SVN or 1.04 that limit seems to be 50cps or a bit higher.
> > After that limit it starts piling up "ongoing" calls, by taking time to
> > handle them and when that limit gets too high it's too late. All in all
> > really fine, I just set the system wide limit to a little under that
> > "threshold". But when I'm running just UNDER the threshold it's not CPU
> > that's a problem. Theoretically I should be able to run (based on the
> > CPU usage at 300cps) about 400cps.
> >
> > When running at 300 I get SOME failed calls and I see
> >
> > "switch_core_state_machine.c:525 a9a60636-0cea-11df-85a1-09c991f2afc5
> > Timeout waiting for next instruction in CS_NEW!"
> >
> > in the console.
> >
> > I didn't find much on how people do high cps setups and it feels a bit
> > like a "friday afternoon solution" to run multiple sofia profiles on the
> > same machine in order to max out the system.
> >
> > Maybe I'm missing something and I know it's not an exact science this,
> > but I'm not sure "all is OK" because I'm not slowly getting to a 100%
> > cpu (or disk / network) usage, I hit some kind of limit after which
> > stuff goes wrong.
> >
> > Anyone any input!
> >
> > /Robin
> >
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From mike at jerris.com  Sat Jan 30 18:01:47 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 Jan 2010 21:01:47 -0500
Subject: [Freeswitch-users] problem with hard phone
In-Reply-To: <27386032.post@talk.nabble.com>
References: <27386032.post@talk.nabble.com>
Message-ID: <FA59609C-96AB-41CB-8922-EA444A124221@jerris.com>

Your log below seems to indicate this is a codec negotiation issue, I doubt it has anything to do with the port thing you mention below.  The message below looks like your phone is requesting g722 at 16000hz.  This would seem logical as its a wideband codec.  There is a long ugly story behind this, but the rfc requires everyone to lie and say 8000hz on g722 instead.  Your phone seems to not be following this rule and I think that is the cause of your issues.  You should update your phone to some firmware that fixes this bug, correct it with some configuration setting on the phone, or disable g722 on the phone altogether.

Mike


On Jan 30, 2010, at 1:11 PM, jwssr wrote:

> 
> Iam having probles with my hard phone...van access ip0020...which is not
> completing the 'B' leg on incoming calls to it
> and 
> on outgoing calls....immedialtely issuing a 488...
> and 
> fs log states ...
> switch_core_codec.c:537 Codec G722 Exists but not at the desired
> implementation. 16000hz 20ms
> 2010-01-30 11:49:07.736157 [ERR] sofia_glue.c:2126 Can't load codec?
> ...
> I have multiple soft phones (sflphones  extensions 1000-1019), a dlink wifi
> dph-540 (extension 4001 ip=192.168.1.190), and this van access ip0020 hard
> phone (extension 2001 ip=192.168.1.180).....fs running on 192.168.1.102.
> 
> all phones sans '2001"  work perfectly..to/fro vitelity and local on lan.
> 
> Im almost certain that the problem lies in the port number assigned to the
> uri.  wireshark shows that call was cancel because port could not be reached 
> and 
> port number is missing on ua shown by fs cli.
> 
> Call-ID:        af4cfd49b52bf8b3b3ec8db3e8a309e5 at 192.168.1.190
> User:           4001 at 192.168.1.102
> Contact:        4001 <sip:4001 at 192.168.1.190:5060>
> ...
> ...
> 
> Call-ID:        4c1ed9ae64405367 at 192.168.1.180
> User:           2001 at 192.168.1.102
> Contact:        "user" <sip:2001 at 192.168.1.180>
> ...                                   notice missing port no. here
> 
> Call-ID:        9be86b04-720f-4f7c-af52-b4f67dccf76b
> User:           1000 at 192.168.1.102
> Contact:        "jon" <sip:1000 at 192.168.1.101:5060>  
> ....
> 
> wireshark.....
> 292 14.903431   192.168.1.102         192.168.1.180         ICMP    
> Destination unreachable (Port unreachable)
> 295 14.927845   192.168.1.102         192.168.1.180         SIP     
> Request: BYE sip:2001 at 192.168.1.180
> 
> I have output from siptrace...but did not want to clutter up forum with too
> much detail but can provide.
> 
> I also have screenshots of html config of phone...
> 
> I would appreciate some help please
> 
> thanks
> -- 
> View this message in context: http://old.nabble.com/problem-with-hard-phone-tp27386032p27386032.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Sat Jan 30 18:35:14 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 Jan 2010 21:35:14 -0500
Subject: [Freeswitch-users] Strategies for reliably detecting nat on
	B-leg?
In-Reply-To: <4B608987.9090606@aastral.net>
References: <4B60785F.6030505@aastral.net>	<FB139E6B-F685-491C-A672-16844878200E@freeswitch.org>
	<191c3a031001271005t62631e77sf6d9ca406054ba00@mail.gmail.com>
	<4B608987.9090606@aastral.net>
Message-ID: <2ADEAAFC-502E-474B-92BD-D6CBACCF885F@jerris.com>

There is no even remotely reliable way to tell.  The only thing you can tell is for some devices you can know they ARE behind nat, you can't ever know reliably that they are not.  That being said, I am not sure that is really the measure you are looking for.  It may be enough to know that the devices reliably tell you their external ip and port in their sdp.  For the a leg, you certainly know this before hand.  For the b leg you would not at dialplan time.  If you are looking to save bandwidth, the only real way of dealing with this would be to perform these actions at some point after the initial call is set up.  This will allow freeswitch to handle any initial audio indications that may be necessary, and for you to definitively know that it has 2 remote endpoints capable of at least handling nat for rtp.  You could do something from a monitoring application such as using uuid_media to trigger the freeswitch box out of the media path after the call has set up and you have confirmed the endpoints are well behaved.  The one thing you should be careful of in this case would be if freeswitch ever has to do an auto-adjust when initially setting up the media, this means the other end lied about either its ip or port.  This could be an indication of both software that can not report its remote IP in sdp or of a firewall that is changing the port.  The firewall changing port scenario you can not use bypass media unless the other device uses an auto-adjust method like freeswitch.  If it just lies about its IP, you could conceivably re-write the sdp with the correct ip and port, but freeswitch does not support this and I don't see it being likely that this would be added.  For you to accomplish this, you likely wold need to add some events to better monitor auto-adjust in the rtp and have some application (either as a freeswitch module, script, or some app attached to event socket) monitor these events and take action.

Mike

On Jan 27, 2010, at 1:44 PM, Bill W wrote:

> Thanks for the reply!
> 
> Just to make sure we're on the same page,  my FreeSWITCH sesrver has a 
> public IP, and I'm trying to bypass media whenever possible to reduce my 
> bandwidth usage.   My concern is trying to bypass media when one of the 
> remote endpoints (b-leg) is behind NAT (since I can reliably detect nat 
> with aggressive-nat on the A-leg).
> 
> Isn't local-network-acl and autonat:x.x.x.x for FS behind nat?
> 
> Does this new information change your responses?
> 
> Thanks again!
> Bill
> 
> 
> 
> Anthony Minessale wrote:
>> also you can set
>> sip_sticky_contact=true
>> channel var which will make that session turn on nat lock in the b leg 
>> so they can't change the contact to a nat addr
>> 
>> add it in {} to your dial string like
>> 
>> {sip_sticky_contact=true}sofia/internal/foo at bar.com <mailto:foo at bar.com>
>> 
>> 
>> 
>> 
>> On Wed, Jan 27, 2010 at 11:50 AM, Brian West <brian at freeswitch.org 
>> <mailto:brian at freeswitch.org>> wrote:
>> 
>>    update to trunk.  and don't use agressive-nat, set
>>    local-network-acl, set the ext-rtp-ip and ext-sip-ip to
>>    autonat:x.x.x.x or if you're behind a natpmp or upnp router set it
>>    to auto-nat.
>> 
>>    It should just work.  Again you have no real way to know if the far
>>    end client never lies to you.  Which it should never do anyway.
>>     Endpoints should know how to traverse their own nat and not leave
>>    it up to the registrar to figure it out.
>> 
>>    /b
>> 
>>    On Jan 27, 2010, at 11:31 AM, Bill W wrote:
>> 
>>> Thoughts?  Suggestions?
>> 
>> 
>>    _______________________________________________
>>    FreeSWITCH-users mailing list
>>    FreeSWITCH-users at lists.freeswitch.org
>>    <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>    http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com 
>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>> 
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org 
>> <mailto:sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888 
>> <http://iax:guest at conference.freeswitch.org/888>
>> googletalk:conf+888 at conference.freeswitch.org 
>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>> 
>> 
>> ------------------------------------------------------------------------
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Sat Jan 30 18:44:55 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 Jan 2010 21:44:55 -0500
Subject: [Freeswitch-users] Multiple DTMF on group_confirm_key
In-Reply-To: <7d79b3931001290436j2061a8dcg5cf7c5144103eb00@mail.gmail.com>
References: <7d79b3931001280411u6262f627xca723c64de5e118d@mail.gmail.com>
	<191c3a031001280914l56a035s74fd6d4b00dd9b3d@mail.gmail.com>
	<7d79b3931001290436j2061a8dcg5cf7c5144103eb00@mail.gmail.com>
Message-ID: <A8C99DF5-D9ED-481C-84E7-3DA5100C2576@jerris.com>

This does not make any sense.  If the one person answers the call but fails to enter the digits, what are you going to do with the caller just hang up on them?  


On Jan 29, 2010, at 7:36 AM, lakshmanan ganapathy wrote:

> I tested by executing a script. It works great. But a small doubt. 
> Assume that I made a parallel dial using bridge application.
> Normally, when one party answer the call, other party end will be hanged up.
> 
> But if I use group_confirm_key=exec and group_confirm_file=perl script.pl, both the end can answer, and call bridged with the person who finished the script first.
> 
> What I've to do if I need to execute the script only for the person who answer's the call first?
> 
> 
> On Thu, Jan 28, 2010 at 10:44 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> you have to use a script (See the wiki for executing a script)
> then you can read in as many digits as you want and do what you need.
> 

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From anthony.minessale at gmail.com  Sat Jan 30 20:19:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 30 Jan 2010 22:19:07 -0600
Subject: [Freeswitch-users] Multiple DTMF on group_confirm_key
In-Reply-To: <191c3a031001302017r7dee2c09vb4ef934335bc2f87@mail.gmail.com>
References: <7d79b3931001280411u6262f627xca723c64de5e118d@mail.gmail.com>
	<191c3a031001280914l56a035s74fd6d4b00dd9b3d@mail.gmail.com>
	<7d79b3931001290436j2061a8dcg5cf7c5144103eb00@mail.gmail.com>
	<191c3a031001302017r7dee2c09vb4ef934335bc2f87@mail.gmail.com>
Message-ID: <191c3a031001302019keac00f7j2deef44daa810304@mail.gmail.com>

If the both answer you can hangup on the one who entered the wrong or no
info.  The winner is whichever one exits the script first and is not hungup.

On Jan 29, 2010 6:43 AM, "lakshmanan ganapathy" <lakindia89 at gmail.com>
wrote:

I tested by executing a script. It works great. But a small doubt.
Assume that I made a parallel dial using bridge application.
Normally, when one party answer the call, other party end will be hanged up.

But if I use group_confirm_key=exec and group_confirm_file=perl script.pl,
both the end can answer, and call bridged with the person who finished the
script first.

What I've to do if I need to execute the script only for the person who
answer's the call first?




On Thu, Jan 28, 2010 at 10:44 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:
>
> you ...

_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From mouncifbb at gmail.com  Sat Jan 30 20:57:08 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Sat, 30 Jan 2010 23:57:08 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
Message-ID: <ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>

OK going back to use default profile to keep things simple below 2 results

Using:

lcr 16179470890  default  19785223241  ( this one consult
npa_nxx_company_ocn)

lcr 6179470890  default  9785223241 ( this one don't!! )



freeswitch> lcr 16179470890  default  19785223241
2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
[16179470890  default  19785223241]
2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
[19785223241]
2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
lata:1] so rate field is [intralata_rate]
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway
cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
l.enabled = '1' AND digits IN (16179470890, 1617947089, 161794708, 16179470,
1617947, 161794, 16179, 1617, 161, 16, 1) AND CURRENT_TIMESTAMP BETWEEN
date_start AND date_end ORDER BY digits DESC,  intralata_rate, rand();
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
[lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to head of
list
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
[lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
06179470890 at proxy.carrier2.net:5060
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end of
list after carrier1

 | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
Dialstring
|
 | 1           | carrier1 | 0.00000  |       |            |
[lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
|
 | 1           | carrier2 | 0.00000  |       |            |
[lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
06179470890 at proxy.carrier2.net:5060 |

2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
[lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
[lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
06179470890 at proxy.carrier2.net:5060





freeswitch> lcr 6179470890  default  9785223241
2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
[6179470890  default  9785223241]
2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
[9785223241]
2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
lata:0] so rate field is [rate]
2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
c.id=cg.carrier_id
WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
(6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
rate, rand();
2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
[lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to head of
list
2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
[lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890

 | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
Dialstring                                                                |
 | 617947      | carrier1 | 0.09000  |       |            |
[lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |










On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> turn up logging to debug again, and then reload mod_lcr.  It'll spit out a
> bunch of crap when it tests out each profile you have defined.  Give me the
> full log (here or in pastebin.freeswitch.org).  That may show more useful
> info as to why things are mucked up?
>
>
> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>
>> yes I use NANPA_STD profile instead of default cause I thought the custom
>> profile was causing issues, but looks like it's returning same results.
>>
>> There is this line in thw wiki:
>> intra lata/state selection is done manually by setting the channel
>> variables *intrastate* or *intralata* to the value *true*.
>>
>> do I have to set these ? if yes how?
>>
>> Thanks
>>
>>
>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>> Stuff inline.
>>>
>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>>>
>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>> 201007,"224","7229","0.0059","0.0127"
>>>> 201040,"224","9206","0.0036","0.0036"
>>>>
>>>
>>> Looks like they give you the LATA and OCN values with the prefix.  We
>>> (should) look that up ourselves.
>>>
>>>
>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>
>>>>
>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>
>>>> I also see this now when making a real call instead of running thorugh
>>>> CLI
>>>>
>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890 NANPA_STD)
>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate channel
>>>> var is [undef]*
>>>
>>>
>>> This is fine.  it is a leftover from when you would tell mod_lcr via a
>>> channel var that it should do intrastate.  I later had mod_lcr do the lookup
>>> itself, but we still honor the old var.  There are no channel vars
>>> associated with the cli, so you wouldn't see that msg.
>>>
>>>
>>>>
>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based on
>>>> interstate rates
>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>> 16179470893 using profile NANPA_STD
>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>>> lata:0] so rate field is [rate]
>>>>
>>>> called number 6179470890  caller ID: 6179472456
>>>>
>>>> any ideas??
>>>>
>>>>
>>> Only thing that jumps out at me.
>>>
>>> The output from lcr_admin show profiles showed only the default one.  On
>>> the dialplan you use the NANPA_STD profile.  Can you check lcr_admin list
>>> and see if that profile is defined and if so if it says it is using the
>>> npanxx table?
>>>
>>>
>>>
>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>>
>>>>> Something is still missing from the logs. Note the query of the npanxx
>>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>>> oh, what version of fs are you running?
>>>>>
>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>
>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>
>>>>> An example from my own setup:
>>>>>
>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>>>>> is [12148267711 default 12148267712]
>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>> [12148267712]
>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR (npa=214
>>>>> AND nxx=826)
>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count: 1
>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT l.digits
>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS lcr_gw_prefix,
>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>> random();
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>
>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
>>>>> head of list
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>
>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end
>>>>> of list
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end
>>>>> of list
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>
>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to end
>>>>> of list
>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>
>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>> [...]
>>>>>
>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <
>>>>> mouncifbb at gmail.com> wrote:
>>>>> > Also the Provider has presented the rates in this format?
>>>>> > NPANXXLATA OCN INTER INTRA
>>>>> >
>>>>> >
>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <
>>>>> mouncifbb at gmail.com>
>>>>> > wrote:
>>>>> >>
>>>>> >> Tried it and it's not giving me intralata  instead I get interstate,
>>>>> does
>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?, also
>>>>> do I have
>>>>> >> to have the rate field in lcr table?
>>>>> >>
>>>>> >> lcr 617642  default 6176421212
>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>> Dialstring
>>>>> >>                                                        |
>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>> >>
>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>> >>
>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to lcr
>>>>> is
>>>>> >> [617642  default 6176421212]
>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>>> >> [6176421212]
>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>> [state:0
>>>>> >> lata:0] so rate field is [rate]
>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>> l.digits,
>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>> gw_suffix,
>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid
>>>>> FROM lcr
>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
>>>>> l.enabled
>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>> CURRENT_TIMESTAMP
>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate, rand();
>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>> Dialstring
>>>>> >>
>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to
>>>>> head
>>>>> >> of list
>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>> Dialstring
>>>>> >>
>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>> >>
>>>>> >> Thank you Rupa!
>>>>> >>
>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com>
>>>>> wrote:
>>>>> >>>
>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The sql
>>>>> >>> statements along with status info will show up.  This should give
>>>>> >>> enough information to debug what is happening.
>>>>> >>>
>>>>> >>> I'm assuming the npanxx table is actually populated and not just
>>>>> >>> existing?
>>>>> >>>
>>>>> >>> When doing the lookup from the cli you have to tell lcr what CID to
>>>>> >>> use (remember, it is relative to the src/dest number).  I'm pretty
>>>>> >>> sure you get something on the console log when you don't specify a
>>>>> CID
>>>>> >>> when using the commandline.  Anyway:
>>>>> >>>
>>>>> >>> lcr 617642  default 6176421212
>>>>> >>>
>>>>> >>> should give you intralata.
>>>>> >>>
>>>>> >>> Note that the definition of intralata doesn't mean "local" for some
>>>>> >>> providers.  Some providers define local to "same ratecenter" which
>>>>> is
>>>>> >>> even more restrictive.
>>>>> >>>
>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <
>>>>> mouncifbb at gmail.com>
>>>>> >>> wrote:
>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I am
>>>>> using
>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>> >>> >
>>>>> >>> > lcr mysql table structure:
>>>>> >>> > CREATE TABLE `lcr` (
>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>> >>> >   KEY `digits` (`digits`),
>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>>>>> >>> > `carriers`
>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>> >>> >
>>>>> >>> >
>>>>> >>> > lcr_admin show profiles
>>>>> >>> > Name:           default
>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>> l.${lcr_rate_field},
>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>>> >>> > l.trail_strip,
>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c
>>>>> ON
>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>> WHERE
>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>> digits IN
>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>> date_start
>>>>> >>> > AND
>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality DESC,
>>>>> >>> >  reliability DESC, rand();
>>>>> >>> >  has %:         false
>>>>> >>> >  has vars:      true
>>>>> >>> >  has intrastate:        true
>>>>> >>> >  has intralata: true
>>>>> >>> >  has npanxx:    true
>>>>> >>> >  Reorder rate:  enabled
>>>>> >>> >  Info in headers:       disabled
>>>>> >>> >  Quote IN() List:       disabled
>>>>> >>> >
>>>>> >>> >
>>>>> >>> >
>>>>> >>> > lcr 617642  default  returns rate from the rate field table and
>>>>> not
>>>>> >>> > intra/inter state fields rates.
>>>>> >>> >
>>>>> >>> > Any ideas? thanks!
>>>>> >>> >
>>>>> >>> >
>>>>> >>> >
>>>>> >>> >
>>>>> >>> > _______________________________________________
>>>>> >>> > FreeSWITCH-users mailing list
>>>>> >>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>> >
>>>>> >>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >>> > http://www.freeswitch.org
>>>>> >>> >
>>>>> >>> >
>>>>> >>>
>>>>> >>>
>>>>> >>>
>>>>> >>> --
>>>>> >>> -Rupa
>>>>> >>>
>>>>> >>> _______________________________________________
>>>>> >>> FreeSWITCH-users mailing list
>>>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >>> http://www.freeswitch.org
>>>>> >>
>>>>> >
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sat Jan 30 21:38:07 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 31 Jan 2010 00:38:07 -0500
Subject: [Freeswitch-users] Inbound sip invite from external gateway
In-Reply-To: <BAY143-W25E4C7D5F62FD42D7BA37BA15E0@phx.gbl>
References: <BAY143-W25E4C7D5F62FD42D7BA37BA15E0@phx.gbl>
Message-ID: <AB6382BE-DE8E-4239-8325-54A5A3538297@jerris.com>

If you look at the debug you can see all the condition matching and  
variable expansion.  Is it not matching or is $0 not expanding?  Maybe  
try to use $1 instead.  Regardless, the debug of the dialplan should  
point to exactly what is not matching.

On Jan 26, 2010, at 4:10 PM, juan camilo ospina quintero <camilin2212 at hotmail.com 
 > wrote:

> hi to all
>
> im already do the integration with. Freeswitch sends invite messages  
> to sailfin, in sailfin there is a sip
> servlet that acts as a proxy, this means it receives the invite from  
> extension1000 and send the invite back
> to freeswitch at extension 1001, but i get the freeswitch messages  
> go to sailfin, but i dont get freeswitch
> to understand sailfin messages.
>
> there is my configuration for sending messages and for receiving  
> messages
>
> In /freeswitch/conf/dialplan/default.xml
>
>     <extension name="redirect">
>       <condition field="network_addr" expression="^192\.168\.2\.9$"/>
>       <condition field="destination_number" expression="^1(\d+)$">
>           <action application="bridge" data="sofia/external/ 
> $0 at 127.0.0.1:5070"/>
>       </condition>
>     </extension>
>
> this works fine, it redirects the messages to sailfin in 127.0.0.1
>
>
> In /freeswitch/conf/dialplan/public.xml
>
>     <extension name="receive">
>       <condition field="network_addr" expression="^127\.0\.0\.1$"/>
>       <condition field="destination_number" expression="^1(\d+)$">
>           <action application="bridge" data="sofia/internal/ 
> $0 at 192.168.2.9:5060"/>
>       </condition>
>     </extension>
>
> this doesnt work, i also use <action application="transfer"  
> data="$1001 XML default/> instead <action application="bridge"  
> data="sofia/internal/$0 at 192.168.2.9:5060"/>
>
> but still doesnt work, the invite that sailfin sends appears in the  
> freeswitch console, but the 1001 extension doesnt get it
>
> Keep your friends updated? even when you?re not signed in.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From mike at jerris.com  Sat Jan 30 21:46:39 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 31 Jan 2010 00:46:39 -0500
Subject: [Freeswitch-users] NAT keep alive
In-Reply-To: <8657153B18DB4AB3B17DB7A4BAAFF862@Terminal>
References: <8657153B18DB4AB3B17DB7A4BAAFF862@Terminal>
Message-ID: <28B8210E-BF86-4478-8140-21F0C9A135E8@jerris.com>

http://wiki.freeswitch.org/wiki/Sofia.conf.xml#nat-options-ping

It is a Sofia profile param

On Jan 26, 2010, at 3:57 AM, "Airsignal" <airsignal at wavecable.com>  
wrote:

> Good Evening:
>
> I am trying to get my switch to send keep alives to the ata's in the  
> field.
>
> <param name="nat-options-ping" value="true"/>   seems to be meant  
> for this.
>
> where should it go?  I can little documentation discussing this...
>
>
> Thanks!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From mike at jerris.com  Sat Jan 30 21:58:07 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 31 Jan 2010 00:58:07 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
Message-ID: <536A30FA-3494-40C2-9B9D-D8F63CA6BCC6@jerris.com>

http://www.google.com/search?q=cannot+restore+segment+prot+after+reloc:+Permission+denied

Google says this is selinux configured to enforcing without setting it  
up properly to allow what your trying to do.  Try disabling selinux,  
and if that works and you want selinux enabled, you will need to come  
up with the propper config.

Mike

On Jan 27, 2010, at 11:47 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> I tried running ldconfig on the directory containing mysql.so, but  
> it did not help.
> So it sounds like there could be a bug in the latter versions?
>
>
> On Wed, Jan 27, 2010 at 8:09 PM, David Villasmil <david.villasmil.work at gmail.com 
> > wrote:
> I got the same error, my script was working with no problems before an
> update to trunk.
>
> David
>
> On Thu, Jan 28, 2010 at 1:15 AM, Adam Wilt <wiltingtree at gmail.com>  
> wrote:
> > Hi, I followed the instructions in the Lua documentation for  
> setting up
> > luasql, but when I try to run my script I get:
> > 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading  
> module
> > 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
> >         /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore  
> segment prot
> > after reloc: Permission denied
> > stack traceback:
> >         [C]: ?
> >         [C]: in function 'require'
> >         /usr/local/freeswitch/scripts/l.lua:2: in main chunk
> > I'm running FreeSWITCH version 1.4 and luasql version 2.1.1.
> > I changed the permissions for mysql.so and for my script to 777,  
> so I'm not
> > sure where the permission problem could be.
> > I'd appreciate any suggestions.
> > Thanks,
> > Adam
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From rm at callrica.co.za  Sat Jan 30 22:05:39 2010
From: rm at callrica.co.za (Roly Maz)
Date: Sun, 31 Jan 2010 08:05:39 +0200
Subject: [Freeswitch-users] How to populate Voice recording metadata with
	custom CRM data?
Message-ID: <003c01caa23b$83ed0800$8bc71800$@co.za>

Hi All,  

 

I am using Freeswitch 1.0.4 (Latest Windows Installer version) on Windows 7
Ultimate for testing

 

Assuming an outbound call, what would be the most sensible approach to pass
custom CRM data into the voice-recording? 

 

I would like the voice recording of the call to include the customers social
security number in the file title, or even metadata.

 

In other words, is there a way to pass custom info, at time of call, to the
dialplan for use in the creating the voice recording. 

 

Any pointers would be much appreciated

 

Rgds

 

Roly

 

 

 

 

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From oseslija at gmail.com  Sun Jan 31 02:47:42 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sun, 31 Jan 2010 11:47:42 +0100
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
In-Reply-To: <j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8@4ax.com>
References: <gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b@4ax.com>
	<gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b-e09XROE/p8c@public.gmane.org>
	<20100129164359.9B3C22B10D@cuneorg-email.cune.pri>
	<j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8@4ax.com>
Message-ID: <4468a6771001310247n4bf5d1a4rf4aea9f2d6c35a1f@mail.gmail.com>

If FreeSWITCH is configured in bypass-media mode, and the endpoint behing
NAT cannot use any of the NAT avoiding techiques to send public IP in the
SDP (STUN etc.) then you'll have issues.

You can do what do I do, which is to make different sofia profiles for NATed
and non-NATED endpoints (FS has many server-side nat traversal mechanisms).

Regards,
Ognjen

On Sat, Jan 30, 2010 at 2:53 PM, Fred-145 <codecomplete at free.fr> wrote:

> On Fri, 29 Jan 2010 16:43:59 -0000,
> <Russell.Mosemann at cune.org> wrote:
> >The ports are open between the endpoint and Freeswitch. The ports are not
> >open between the two endpoints themselves. If each endpoint is behind its
> >own NAT, neither endpoint will be able to contact the other endpoint
> >unless some kind of forwarding is set up on the firewall to map the
> >external IP address and port to an internal IP address and port.
>
> Thanks but the context I was refering to is...
> 1. Freeswitch is configured in BypassMedia mode
> 2. The firewall and the local end-points are configured so that a
> series of UDP ranges are mapped to their respective end-point (eg.
> UDP100-1003 for extension #1, 1004-1007 for #2, etc.)
> ... so that RTP packets flow directly between the two end-points
>
> Brian says above that there might be cases where NAT could be a
> problem. When could this happen?
>
> I'd like to get to the bottom of this so that in case a server is a
> bit short on CPU/network power, I know that there's the alternative of
> RTP packets by-passing the server... but I also need to know what
> issues this setup can cause.
>
> Thank you.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch-list at puzzled.xs4all.nl  Sun Jan 31 05:04:28 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick)
Date: Sun, 31 Jan 2010 14:04:28 +0100
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <536A30FA-3494-40C2-9B9D-D8F63CA6BCC6@jerris.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<536A30FA-3494-40C2-9B9D-D8F63CA6BCC6@jerris.com>
Message-ID: <4B657FDC.5080109@puzzled.xs4all.nl>

On 01/31/2010 06:58 AM, Michael Jerris wrote:
> http://www.google.com/search?q=cannot+restore+segment+prot+after+reloc:+Permission+denied
> 
> Google says this is selinux configured to enforcing without setting it
> up properly to allow what your trying to do.  Try disabling selinux, and
> if that works and you want selinux enabled, you will need to come up
> with the propper config.

To fix a similar error message this is what I had in an old spec file:
/sbin/restorecon -v /usr/lib64/somelib.so

Iirc this is not the proper way to fix this and one should use the chcon
command (chcon -t ...) or create an selinux policy. man chcon and google
has more info.

Regards,
Patrick






From rupa at rupa.com  Sun Jan 31 05:32:04 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 31 Jan 2010 07:32:04 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
Message-ID: <b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>

On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:

> OK going back to use default profile to keep things simple below 2 results
>
> Using:
>
> lcr 16179470890  default  19785223241  ( this one consult
> npa_nxx_company_ocn)
>
> lcr 6179470890  default  9785223241 ( this one don't!! )
>
>
>
Oh, right!  mod_lcr really expects you to normalize your prefix to e164
format.  I thought there was discussion about this in the wiki, but maybe
not.  For simple prefix matching it doesn't matter, but for things that make
decisions based on the # (like the lata/state stuff) it does.

npanxx lookup only makes sense for NANPA numbers.  NANPA numbers have a
country code of "1" and a total length of 11 (including the 1).

This is the only rational way to do it when you have a rate table with both
domestic (NANPA) and international prefixes.


> freeswitch> lcr 16179470890  default  19785223241
> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
> [16179470890  default  19785223241]
> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
> [19785223241]
> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
> OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
> npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
> lata:1] so rate field is [intralata_rate]
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
> c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
> gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
> l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway
> cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
> l.enabled = '1' AND digits IN (16179470890, 1617947089, 161794708, 16179470,
> 1617947, 161794, 16179, 1617, 161, 16, 1) AND CURRENT_TIMESTAMP BETWEEN
> date_start AND date_end ORDER BY digits DESC,  intralata_rate, rand();
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to head of
> list
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
> 06179470890 at proxy.carrier2.net:5060
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end of
> list after carrier1
>
>
>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
> Dialstring
> |
>  | 1           | carrier1 | 0.00000  |       |            |
> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
> |
>  | 1           | carrier2 | 0.00000  |       |            |
> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
> 06179470890 at proxy.carrier2.net:5060 |
>
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
> 06179470890 at proxy.carrier2.net:5060
>
>
>
>
>
> freeswitch> lcr 6179470890  default  9785223241
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
> [6179470890  default  9785223241]
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
> [9785223241]
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
> lata:0] so rate field is [rate]
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
> (6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
> AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
> rate, rand();
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to head of
> list
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>
>
>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
> Dialstring                                                                |
>  | 617947      | carrier1 | 0.09000  |       |            |
> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |
>
>
>
>
>
>
>
>
>
>
>
> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> turn up logging to debug again, and then reload mod_lcr.  It'll spit out a
>> bunch of crap when it tests out each profile you have defined.  Give me the
>> full log (here or in pastebin.freeswitch.org).  That may show more useful
>> info as to why things are mucked up?
>>
>>
>> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>>
>>> yes I use NANPA_STD profile instead of default cause I thought the custom
>>> profile was causing issues, but looks like it's returning same results.
>>>
>>> There is this line in thw wiki:
>>> intra lata/state selection is done manually by setting the channel
>>> variables *intrastate* or *intralata* to the value *true*.
>>>
>>> do I have to set these ? if yes how?
>>>
>>> Thanks
>>>
>>>
>>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>>> Stuff inline.
>>>>
>>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>>>>
>>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>>> 201007,"224","7229","0.0059","0.0127"
>>>>> 201040,"224","9206","0.0036","0.0036"
>>>>>
>>>>
>>>> Looks like they give you the LATA and OCN values with the prefix.  We
>>>> (should) look that up ourselves.
>>>>
>>>>
>>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>>
>>>>>
>>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>>
>>>>> I also see this now when making a real call instead of running thorugh
>>>>> CLI
>>>>>
>>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890 NANPA_STD)
>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate channel
>>>>> var is [undef]*
>>>>
>>>>
>>>> This is fine.  it is a leftover from when you would tell mod_lcr via a
>>>> channel var that it should do intrastate.  I later had mod_lcr do the lookup
>>>> itself, but we still honor the old var.  There are no channel vars
>>>> associated with the cli, so you wouldn't see that msg.
>>>>
>>>>
>>>>>
>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based
>>>>> on interstate rates
>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>>> 16179470893 using profile NANPA_STD
>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>>>> lata:0] so rate field is [rate]
>>>>>
>>>>> called number 6179470890  caller ID: 6179472456
>>>>>
>>>>> any ideas??
>>>>>
>>>>>
>>>> Only thing that jumps out at me.
>>>>
>>>> The output from lcr_admin show profiles showed only the default one.  On
>>>> the dialplan you use the NANPA_STD profile.  Can you check lcr_admin list
>>>> and see if that profile is defined and if so if it says it is using the
>>>> npanxx table?
>>>>
>>>>
>>>>
>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>
>>>>>> Something is still missing from the logs. Note the query of the npanxx
>>>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>>>> oh, what version of fs are you running?
>>>>>>
>>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>>
>>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>>
>>>>>> An example from my own setup:
>>>>>>
>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>>>>>> is [12148267711 default 12148267712]
>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>>> [12148267712]
>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
>>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
>>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR (npa=214
>>>>>> AND nxx=826)
>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count: 1
>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT l.digits
>>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS lcr_gw_prefix,
>>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
>>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
>>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>>> random();
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>
>>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
>>>>>> head of list
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>
>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end
>>>>>> of list
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end
>>>>>> of list
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>
>>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to
>>>>>> end of list
>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>
>>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>>> [...]
>>>>>>
>>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <
>>>>>> mouncifbb at gmail.com> wrote:
>>>>>> > Also the Provider has presented the rates in this format?
>>>>>> > NPANXXLATA OCN INTER INTRA
>>>>>> >
>>>>>> >
>>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <
>>>>>> mouncifbb at gmail.com>
>>>>>> > wrote:
>>>>>> >>
>>>>>> >> Tried it and it's not giving me intralata  instead I get
>>>>>> interstate, does
>>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?, also
>>>>>> do I have
>>>>>> >> to have the rate field in lcr table?
>>>>>> >>
>>>>>> >> lcr 617642  default 6176421212
>>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>>> Dialstring
>>>>>> >>                                                        |
>>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>>> >>
>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>>> >>
>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to
>>>>>> lcr is
>>>>>> >> [617642  default 6176421212]
>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>>>> >> [6176421212]
>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>>> [state:0
>>>>>> >> lata:0] so rate field is [rate]
>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>>> l.digits,
>>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>>> gw_suffix,
>>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid
>>>>>> FROM lcr
>>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
>>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
>>>>>> l.enabled
>>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>>> CURRENT_TIMESTAMP
>>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate,
>>>>>> rand();
>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>> Dialstring
>>>>>> >>
>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1 to
>>>>>> head
>>>>>> >> of list
>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>> Dialstring
>>>>>> >>
>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>> >>
>>>>>> >> Thank you Rupa!
>>>>>> >>
>>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com>
>>>>>> wrote:
>>>>>> >>>
>>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The sql
>>>>>> >>> statements along with status info will show up.  This should give
>>>>>> >>> enough information to debug what is happening.
>>>>>> >>>
>>>>>> >>> I'm assuming the npanxx table is actually populated and not just
>>>>>> >>> existing?
>>>>>> >>>
>>>>>> >>> When doing the lookup from the cli you have to tell lcr what CID
>>>>>> to
>>>>>> >>> use (remember, it is relative to the src/dest number).  I'm pretty
>>>>>> >>> sure you get something on the console log when you don't specify a
>>>>>> CID
>>>>>> >>> when using the commandline.  Anyway:
>>>>>> >>>
>>>>>> >>> lcr 617642  default 6176421212
>>>>>> >>>
>>>>>> >>> should give you intralata.
>>>>>> >>>
>>>>>> >>> Note that the definition of intralata doesn't mean "local" for
>>>>>> some
>>>>>> >>> providers.  Some providers define local to "same ratecenter" which
>>>>>> is
>>>>>> >>> even more restrictive.
>>>>>> >>>
>>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <
>>>>>> mouncifbb at gmail.com>
>>>>>> >>> wrote:
>>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I am
>>>>>> using
>>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>>> >>> >
>>>>>> >>> > lcr mysql table structure:
>>>>>> >>> > CREATE TABLE `lcr` (
>>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>>> >>> >   KEY `digits` (`digits`),
>>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>>>>>> >>> > `carriers`
>>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>>> >>> >
>>>>>> >>> >
>>>>>> >>> > lcr_admin show profiles
>>>>>> >>> > Name:           default
>>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>>> l.${lcr_rate_field},
>>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>>>> >>> > l.trail_strip,
>>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c
>>>>>> ON
>>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>> WHERE
>>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>>> digits IN
>>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>>> date_start
>>>>>> >>> > AND
>>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality
>>>>>> DESC,
>>>>>> >>> >  reliability DESC, rand();
>>>>>> >>> >  has %:         false
>>>>>> >>> >  has vars:      true
>>>>>> >>> >  has intrastate:        true
>>>>>> >>> >  has intralata: true
>>>>>> >>> >  has npanxx:    true
>>>>>> >>> >  Reorder rate:  enabled
>>>>>> >>> >  Info in headers:       disabled
>>>>>> >>> >  Quote IN() List:       disabled
>>>>>> >>> >
>>>>>> >>> >
>>>>>> >>> >
>>>>>> >>> > lcr 617642  default  returns rate from the rate field table and
>>>>>> not
>>>>>> >>> > intra/inter state fields rates.
>>>>>> >>> >
>>>>>> >>> > Any ideas? thanks!
>>>>>> >>> >
>>>>>> >>> >
>>>>>> >>> >
>>>>>> >>> >
>>>>>> >>> > _______________________________________________
>>>>>> >>> > FreeSWITCH-users mailing list
>>>>>> >>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >>> >
>>>>>> >>> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >>> > http://www.freeswitch.org
>>>>>> >>> >
>>>>>> >>> >
>>>>>> >>>
>>>>>> >>>
>>>>>> >>>
>>>>>> >>> --
>>>>>> >>> -Rupa
>>>>>> >>>
>>>>>> >>> _______________________________________________
>>>>>> >>> FreeSWITCH-users mailing list
>>>>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> >>> http://www.freeswitch.org
>>>>>> >>
>>>>>> >
>>>>>> >
>>>>>> > _______________________________________________
>>>>>> > FreeSWITCH-users mailing list
>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> > http://www.freeswitch.org
>>>>>> >
>>>>>> >
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From scottferri09 at gmail.com  Sun Jan 31 05:45:04 2010
From: scottferri09 at gmail.com (Scott Fernandez)
Date: Sun, 31 Jan 2010 19:15:04 +0530
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <874941.17255.qm@web33502.mail.mud.yahoo.com>
References: <874941.17255.qm@web33502.mail.mud.yahoo.com>
Message-ID: <df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>

Hi,

Thx for the information. Can I have some detailed steps to configure
mod_managed class call control and how do we write the API commands in .Net
applications?

In addition, how do we get the current STATE of the call when I use webapi?.
Because it is required for me to route the call to the user upon it is
answered or disconnect  it.

Thanks,
Scott

On Wed, Jan 20, 2010 at 8:47 PM, Diego Toro <dftoro at yahoo.com> wrote:

> Hi, the answer is yes, you can to use mod_managed wich offer C# managed
> class to call control http://wiki.freeswitch.org/wiki/Mod_managed. Or
> using managed ESL (libs/esl/managed) which offer C# managed class to receive
> and send events and commands to FreeSwitch.
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
> --- On Wed, 1/20/10, Scott Fernandez <scottferri09 at gmail.com> wrote:
>
> > From: Scott Fernandez <scottferri09 at gmail.com>
> > Subject: Re: [Freeswitch-users] Establishing a Call from .Net based
> application
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Wednesday, January 20, 2010, 2:17 AM
>  > Thanks Dome. Will try it out and get back to
> > you if I come across any issues.
> >
> > Regards,
> > Scott.
> >
> > On Wed, Jan 20, 2010 at 11:02 AM,
> > Dome Charoenyost <dome at tel.co.th>
> > wrote:
> >
> > Please try http://wiki.freeswitch.org/wiki/Webapi
> >
> >
> > you can create class and map to webapi.
> >
> >
> >
> > Dome C.
> >
> >
> >
> > 2010/1/19 Scott Fernandez <scottferri09 at gmail.com>:
> >
> > > Hi,
> >
> > >
> >
> > > Is there any API modules available for me to initiate
> > a call from .Net based
> >
> > > application?.
> >
> > >
> >
> > > The idea is to include the API modules if any with the
> > .NET base classes so
> >
> > > that the API commands will be made available on it. I
> > know it is doable when
> >
> > > I use socket programming in .NET in which Telnet
> > session is created.
> >
> > > However, this would potentially hamper the performance
> > of the application
> >
> > > because of multiple sessions that will be created for
> > each call.
> >
> > >
> >
> > > Other than that, Is there any Freeswitch API modules
> > (like plug-ins)
> >
> > > available in order to include it into the .Net classes
> > and start building
> >
> > > the customized application?
> >
> > >
> >
> > > Any help from any one is highly appreciated.
> >
> > >
> >
> > > Thanks,
> >
> > > Scott
> >
> > >
> >
> > >
> > _______________________________________________
> >
> > > FreeSWITCH-users mailing list
> >
> > > FreeSWITCH-users at lists.freeswitch.org
> >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > http://www.freeswitch.org
> >
> > >
> >
> > >
> >
> >
> >
> > _______________________________________________
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > http://www.freeswitch.org
> >
> >
> >
> >
> > -----Inline Attachment Follows-----
>  >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Sun Jan 31 06:21:35 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 31 Jan 2010 08:21:35 -0600
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <4B657FDC.5080109@puzzled.xs4all.nl>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<9853f4ff1001271709w60445c0ar7d3f4cb5fee36d4@mail.gmail.com>
	<eb7797d71001272047l48614934v3a35567f9fd0aa25@mail.gmail.com>
	<536A30FA-3494-40C2-9B9D-D8F63CA6BCC6@jerris.com>
	<4B657FDC.5080109@puzzled.xs4all.nl>
Message-ID: <191c3a031001310621s20264d79u27afbf8ff0ba1a64@mail.gmail.com>

Be careful with lua and sql
I have heard countless reports of the luasql leaking memory like a fire
hydrant.....

We may need to make our own odbc obj so every embedded lang can share it.
But it takes time and resources.

On Jan 31, 2010 7:11 AM, "Patrick" <freeswitch-list at puzzled.xs4all.nl>
wrote:

On 01/31/2010 06:58 AM, Michael Jerris wrote:
> http://www.google.com/search?q=cannot+restore+segmen...
To fix a similar error message this is what I had in an old spec file:
/sbin/restorecon -v /usr/lib64/somelib.so

Iirc this is not the proper way to fix this and one should use the chcon
command (chcon -t ...) or create an selinux policy. man chcon and google
has more info.

Regards,
Patrick





_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at l...
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From anthony.minessale at gmail.com  Sun Jan 31 06:24:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 31 Jan 2010 08:24:34 -0600
Subject: [Freeswitch-users] How to populate Voice recording metadata
	with custom CRM data?
In-Reply-To: <003c01caa23b$83ed0800$8bc71800$@co.za>
References: <003c01caa23b$83ed0800$8bc71800$@co.za>
Message-ID: <191c3a031001310624u34d02ccbtfd8db2766f263827@mail.gmail.com>

There are channel vars that all begin record_
Check the wiki and default config example in the dp

On Jan 31, 2010 12:18 AM, "Roly Maz" <rm at callrica.co.za> wrote:

 Hi All,



I am using Freeswitch 1.0.4 (Latest Windows Installer version) on Windows 7
Ultimate for testing



Assuming an outbound call, what would be the most sensible approach to pass
custom CRM data into the voice-recording?



I would like the voice recording of the call to include the customers social
security number in the file title, or even metadata.



In other words, is there a way to pass custom info, at time of call, to the
dialplan for use in the creating the voice recording.



Any pointers would be much appreciated



Rgds



Roly









_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From stevendt at primrosebank.net  Sun Jan 31 06:30:11 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sun, 31 Jan 2010 14:30:11 -0000
Subject: [Freeswitch-users] Trunk Version Number
Message-ID: <2360B060EDF44D368707DD12064BD416@bp1.ad.bp.com>

Hi,

Running the latest SVN (16453) under Windows, the console "Version" command displays :-

"FreeSWITCH Version 1.0.trunk (UNKNOWN)"

Should the version number not include a meaningful build version in the brackets ?

regards
Dave
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From sos at sokhapkin.dyndns.org  Sun Jan 31 06:33:17 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 31 Jan 2010 09:33:17 -0500
Subject: [Freeswitch-users] error loading module 'luasql.mysql'
In-Reply-To: <191c3a031001310621s20264d79u27afbf8ff0ba1a64@mail.gmail.com>
References: <eb7797d71001271615n2b86c197wc943d533fc0ffbe1@mail.gmail.com>
	<4B657FDC.5080109@puzzled.xs4all.nl>
	<191c3a031001310621s20264d79u27afbf8ff0ba1a64@mail.gmail.com>
Message-ID: <201001310933.17301.sos@sokhapkin.dyndns.org>

There is memory leak with luasql mysql module, but everything is fine with 
luasql odbc module.

Is someone working on lua interface to freeswitch core odbc functions?

On Sunday 31 January 2010, Anthony Minessale wrote:
> Be careful with lua and sql
> I have heard countless reports of the luasql leaking memory like a fire
> hydrant.....
>
> We may need to make our own odbc obj so every embedded lang can share it.
> But it takes time and resources.
>
> On Jan 31, 2010 7:11 AM, "Patrick" <freeswitch-list at puzzled.xs4all.nl>
> wrote:
>
> On 01/31/2010 06:58 AM, Michael Jerris wrote:
> > http://www.google.com/search?q=cannot+restore+segmen...
>
> To fix a similar error message this is what I had in an old spec file:
> /sbin/restorecon -v /usr/lib64/somelib.so
>
> Iirc this is not the proper way to fix this and one should use the chcon
> command (chcon -t ...) or create an selinux policy. man chcon and google
> has more info.
>
> Regards,
> Patrick
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at l...





From robin at swip.net  Sun Jan 31 06:42:54 2010
From: robin at swip.net (Robin Vleij)
Date: Sun, 31 Jan 2010 15:42:54 +0100
Subject: [Freeswitch-users] CPS per mod_sofia profile limited?
In-Reply-To: <201001301233.21516.errotan@gmail.com>
References: <4B62FF51.8070608@swip.net> <201001301233.21516.errotan@gmail.com>
Message-ID: <4B6596EE.8080103@swip.net>

On 1/30/10 12:33 PM, Pusk?s Zsolt wrote:

Hi!

> CPU usage is not the only thing that limit your calls. Have you set the
> recommended ulimit settings and / or started fs with the -waste option ?

Yes, I did read the wiki & docs and do have the right setup for max 
performance. The -waste option didn't seem to do much in the 1.05 setup 
I was testing in, but I can try again on 1.04 tomorrow when I continue 
testing.

/Robin



From robin at swip.net  Sun Jan 31 06:45:39 2010
From: robin at swip.net (Robin Vleij)
Date: Sun, 31 Jan 2010 15:45:39 +0100
Subject: [Freeswitch-users] CPS per mod_sofia profile limited?
In-Reply-To: <191c3a031001300857w36920b94r183f2e861db6baae@mail.gmail.com>
References: <4B62FF51.8070608@swip.net> <201001301233.21516.errotan@gmail.com>
	<191c3a031001300857w36920b94r183f2e861db6baae@mail.gmail.com>
Message-ID: <4B659793.70408@swip.net>

On 1/30/10 5:57 PM, Anthony Minessale wrote:

Hi Anthony,

> Also keep in mind that the industry standard is 50 which is the capacity
> to take over for the real standard of 25 in a fail-over scenario.  So
> you should be happy you even get 300cps for free.

Yeah, no discussion there. I think 300 on standard hardware seems really 
good. Especially compared to some commercial product we've seen / read 
about. :) It was just that it doesn't feel like the hardware is fully 
used, but some invisible wall we hit, ie bug maybe. That's why I'm 
asking if anyone has seen this. Whatever limit we'll have, we'll set in 
the system wide limits and we're done, so it's no show-stopper in 
putting it in production. I've also done a long term test with a few 
thousand ongoing calls at 50cps over the weekend, see how it goes with 
memory usage and such.

> and we are happy with what we have until we can get the lead dev to work
> on improving it with us when he has the time.

OK.

/robin



From robin at swip.net  Sun Jan 31 06:51:11 2010
From: robin at swip.net (Robin Vleij)
Date: Sun, 31 Jan 2010 15:51:11 +0100
Subject: [Freeswitch-users] CPS per mod_sofia profile limited?
In-Reply-To: <A1448B87-D14D-4181-892A-9ECCD5EB41F0@jerris.com>
References: <4B62FF51.8070608@swip.net>
	<201001301233.21516.errotan@gmail.com>	<191c3a031001300857w36920b94r183f2e861db6baae@mail.gmail.com>
	<A1448B87-D14D-4181-892A-9ECCD5EB41F0@jerris.com>
Message-ID: <4B6598DF.3020407@swip.net>

On 1/31/10 2:57 AM, Michael Jerris wrote:

Hi Mike!

> For clarification, is it correct that your getting worse numbers for
> sustainable cps on SVN then on 1.0.4? I would be interested in the

Yes, 1.04 offered a lot better performance in my test scenario, using 
the same call scripts. I agree that the scenario isn't really realistic, 
I should test with at least 30 second calls.

> numbers you would get with bypass_media=true instead of proxy_media=true

Thought about the same, and funnily enough that didn't help. But in the 
long list of things I tried I don't remember if I tried it on 1.04, 
without media proxying. I expect a much higher performance just 
bypassing the media. I'll try again tomorrow on 1.04 and post the results.

> and with neither setting set as well. Also, make sure your logging level
> is low and try putting the db dir on a ram disk. Thanks for the info. As

I check disk performance using iostat and they're on 2% usage. It's a 
10k rpm sas disk we're writing to, with ram cache on the raid 
controller. Also -nosql didn't have any influence at all on performance.

> that length of call matters very little. In proxy media or normal mode,
> the performance of the box is much more of a calculation on number of
> calls than cps as a result of the context switching from having to move

Exactly. I'll adjust the lengt of the call to 30 or 60 seconds and then 
try to see how many calls we can have. That's in the end what matters 
really and not cps on it's own. Have to create a 60 second pcap first. :)

> calls). Some other tips. While this extension may be trivial, what else
> is there in your dialplan context? Anything above that extension could

Nothing, it's a single entry dialplan and that entry is thus on the top.

> cause a significant impact. Do you have any of the presence features

All presence features disabled.

> enabled? These do significantly impact call handling performance even if
> your sipp scenarios do not send any of those packets.

Yep. :)

Thanks for the input so far!

Maybe after I get some final results I put it on the wiki for future 
reference to load tests?

/robin



From anthony.minessale at gmail.com  Sun Jan 31 10:40:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 31 Jan 2010 12:40:51 -0600
Subject: [Freeswitch-users] CPS per mod_sofia profile limited?
In-Reply-To: <191c3a031001311023w526a2c4cpbd93e2bcea8fd07e@mail.gmail.com>
References: <4B62FF51.8070608@swip.net> <201001301233.21516.errotan@gmail.com>
	<191c3a031001300857w36920b94r183f2e861db6baae@mail.gmail.com>
	<A1448B87-D14D-4181-892A-9ECCD5EB41F0@jerris.com>
	<4B6598DF.3020407@swip.net>
	<191c3a031001311021n1f9c89ffq1a13de517ea98fb2@mail.gmail.com>
	<191c3a031001311022k217817aanbdd6830672cc063b@mail.gmail.com>
	<191c3a031001311023w526a2c4cpbd93e2bcea8fd07e@mail.gmail.com>
Message-ID: <191c3a031001311040r3932da76m77bd2ccabe1689c3@mail.gmail.com>

There is a tradeoff between cps and audio quality and timing accuracy.

Try -vm -nocal if you want to mimic 1.0.4 we don't discuss performance here.

On Jan 31, 2010 8:56 AM, "Robin Vleij" <robin at swip.net> wrote:

On 1/31/10 2:57 AM, Michael Jerris wrote:

Hi Mike!


> For clarification, is it correct that your getting worse numbers for
> sustainable cps on SVN the...
Yes, 1.04 offered a lot better performance in my test scenario, using
the same call scripts. I agree that the scenario isn't really realistic,
I should test with at least 30 second calls.


> numbers you would get with bypass_media=true instead of proxy_media=true
Thought about the same, and funnily enough that didn't help. But in the
long list of things I tried I don't remember if I tried it on 1.04,
without media proxying. I expect a much higher performance just
bypassing the media. I'll try again tomorrow on 1.04 and post the results.


> and with neither setting set as well. Also, make sure your logging level
> is low and try putting...
I check disk performance using iostat and they're on 2% usage. It's a
10k rpm sas disk we're writing to, with ram cache on the raid
controller. Also -nosql didn't have any influence at all on performance.


> that length of call matters very little. In proxy media or normal mode,
> the performance of the ...
Exactly. I'll adjust the lengt of the call to 30 or 60 seconds and then
try to see how many calls we can have. That's in the end what matters
really and not cps on it's own. Have to create a 60 second pcap first. :)


> calls). Some other tips. While this extension may be trivial, what else
> is there in your dialpl...
Nothing, it's a single entry dialplan and that entry is thus on the top.


> cause a significant impact. Do you have any of the presence features
All presence features disabled.


> enabled? These do significantly impact call handling performance even if
> your sipp scenarios do...
Yep. :)

Thanks for the input so far!

Maybe after I get some final results I put it on the wiki for future
reference to load tests?

/robin


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at list...
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From mbsip at gazeta.pl  Sun Jan 31 09:05:28 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sun, 31 Jan 2010 18:05:28 +0100
Subject: [Freeswitch-users] voicemail_greeting_number - question
Message-ID: <28f27f5d1001310905r41b16ca7r5ef1f236f76a070c@mail.gmail.com>

Hi ALL,

I am playing around with VM and want to play user recorded greeting
instead of default one.
I've scaned wiki Mod_Voicemail and found proper parameter
"voicemail_greeting_number".
Unfortunately there is a lack of example hence i dont know if it is
already working.

Aforementioned param was placed in /conf/directory/default/1000.xml
file (param name="voicemail_greeting_number", i tried many values)
The effect is that the default greeting is played.

Is this param embeeded into FS right now?
How to use it?
Is there any other place I should do the changes?

I am running  FreeSWITCH Version 1.0.trunk (16456).

Thx in advance.
Maciej



From tayeb.meftah at gmail.com  Sun Jan 31 12:21:29 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 31 Jan 2010 21:21:29 +0100
Subject: [Freeswitch-users] determining the source of receyved call in
	public context
Message-ID: <4B65E649.2040007@gmail.com>

hi,
how do i determine the gateway or the Ip of a receyved call from ITSP's?
i am calling my did from my mobile, but i see is processing the the 
mobile number no the ITSP User or did
thanks



From codecomplete at free.fr  Sun Jan 31 12:41:11 2010
From: codecomplete at free.fr (Fred-145)
Date: Sun, 31 Jan 2010 21:41:11 +0100
Subject: [Freeswitch-users] Equivalent to Asterisk's
	"directrtpsetup=yes"?
References: <gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b@4ax.com>
	<gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b-e09XROE/p8c@public.gmane.org>
	<20100129164359.9B3C22B10D@cuneorg-email.cune.pri>
	<j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8@4ax.com>
	<j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8-e09XROE/p8c@public.gmane.org>
	<4468a6771001310247n4bf5d1a4rf4aea9f2d6c35a1f@mail.gmail.com>
Message-ID: <qiqbm55lueuqup1m6r6d0mff36b4l4fb9l@4ax.com>

On Sun, 31 Jan 2010 11:47:42 +0100, Ognjen Seslija
<oseslija at gmail.com> wrote:
>If FreeSWITCH is configured in bypass-media mode, and the endpoint behing
>NAT cannot use any of the NAT avoiding techiques to send public IP in the
>SDP (STUN etc.) then you'll have issues.

In this scenario, Freeswitch and the SIP end-points are configured to
handle NAT. I'm just curious to know what issues/drawbacks I should
expect if I decide to lower the CPU/network load on the Freeswitch
server if I decide to configure it in bypass-media mode.

Are there features that aren't available when the two end-points speak
RTP directly instead of having RTP packets go through Freeswitch?

>You can do what do I do, which is to make different sofia profiles for NATed
>and non-NATED endpoints (FS has many server-side nat traversal mechanisms).

If you have time, I'm interested in knowing more about your setup.




From mbsip at gazeta.pl  Sun Jan 31 12:46:34 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sun, 31 Jan 2010 21:46:34 +0100
Subject: [Freeswitch-users]  vm-disk-quota
Message-ID: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>

Hi ALL,

Maybe this question will be piece of cake for most of you, but it
makes me think.

I would like to configure "vm-disk-quota" for all users i have.
I followed the wiki page and provided:
<param name="vm-disk-quota" value="10"/>
to /conf/directory/default/1000.xml

After reloadxml, incoming call give me "mod_voicemail.c:3057 Voicemail
disk quota is exceeded" feedback
No surprise for me because i had more less 10 voice mails already
recorded (before the vm-disk-quota was set up).
Strange is that increasing value even to 100 does not change anything.
The same thing with deleting recordings from user directory.
The only wayout is to set it to default value=0 (even FS shutdown
doesn't change anything)

I am wondering why vm-disk-quota produces "Voicemail disk quota is
exceeded" all the time
Where the module is looking for stored voicemail recordings.

Below is part of my configuration.
1) /conf/autoload_configs/voicemail.conf.xml
<param name="storage-dir" value="/usr/local/freeswitch/vm"/>
2) /conf/directory/default/1000.xml
<param name="vm-disk-quota" value="100"/>
3) /vm/FS_ip_address/1000 is empty


Thanks in advance.
Maciej



From mbsip at gazeta.pl  Sun Jan 31 13:25:28 2010
From: mbsip at gazeta.pl (mbsip)
Date: Sun, 31 Jan 2010 22:25:28 +0100
Subject: [Freeswitch-users] vm-disk-quota
In-Reply-To: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>
References: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>
Message-ID: <28f27f5d1001311325v5dc1fbeegc7ce27f21925a233@mail.gmail.com>

Small change after wiping out all db voicemail_msgs table in
voicemail_default.db - i am able to record just one voicemail.

Strange, isn't it?

Thx,
Maciej.



From mouncifbb at gmail.com  Sun Jan 31 14:18:26 2010
From: mouncifbb at gmail.com (Mouncifbb)
Date: Sun, 31 Jan 2010 17:18:26 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001291637s937e7ffn31b71e1110ce11ae@mail.gmail.com>
	<ade741d61001292030u38704cddhf5f5566f665095e3@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
	<b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
Message-ID: <9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>

So the CID must have 1 at front also? Usually people
Send only npa and nxx ex 6176427788 7817612233
Do I need to alter it?

Sent from my iPhone

On Jan 31, 2010, at 8:32 AM, Rupa Schomaker <rupa at rupa.com> wrote:

>
>
> On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane <mouncifbb at gmail.com 
> > wrote:
> OK going back to use default profile to keep things simple below 2  
> results
>
> Using:
>
> lcr 16179470890  default  19785223241  ( this one consult  
> npa_nxx_company_ocn)
>
> lcr 6179470890  default  9785223241 ( this one don't!! )
>
>
>
> Oh, right!  mod_lcr really expects you to normalize your prefix to  
> e164 format.  I thought there was discussion about this in the wiki,  
> but maybe not.  For simple prefix matching it doesn't matter, but  
> for things that make decisions based on the # (like the lata/state  
> stuff) it does.
>
> npanxx lookup only makes sense for NANPA numbers.  NANPA numbers  
> have a country code of "1" and a total length of 11 (including the 1).
>
> This is the only rational way to do it when you have a rate table  
> with both domestic (NANPA) and international prefixes.
>
>
> freeswitch> lcr 16179470890  default  19785223241
> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr  
> is [16179470890  default  19785223241]
> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to  
> [19785223241]
> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT  
> 'state', count(DISTINCT state) FROM npa_nxx_company_ocn WHERE  
> (npa=617 AND nxx=947) OR (npa=978 AND nxx=522) UNION SELECT 'lata',  
> count(DISTINCT lata) FROM npa_nxx_company_ocn WHERE (npa=617 AND  
> nxx=947) OR (npa=978 AND nxx=522)
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing  
> [state:1 lata:1] so rate field is [intralata_rate]
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT  
> l.digits, c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix,  
> cg.suffix AS gw_suffix, l.lead_strip, l.trail_strip, l.prefix,  
> l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON  
> l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id  
> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND  
> digits IN (16179470890, 1617947089, 161794708, 16179470, 1617947,  
> 161794, 16179, 1617, 161, 16, 1) AND CURRENT_TIMESTAMP BETWEEN  
> date_start AND date_end ORDER BY digits DESC,  intralata_rate, rand();
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning  
> Dialstring [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/ 
> carrier1/16179470890
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to  
> head of list
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning  
> Dialstring [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/ 
> 06179470890 at proxy.carrier2.net:5060
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to  
> end of list after carrier1
>
>
>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |  
> Dialstring  
>     
>     
>     
>     
>                                                                    |
>  | 1           | carrier1 | 0.00000  |       |            |  
> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/ 
> carrier1/16179470890                 |
>  | 1           | carrier2 | 0.00000  |       |            |  
> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/ 
> 06179470890 at proxy.carrier2.net:5060 |
>
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning  
> Dialstring [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/ 
> carrier1/16179470890
> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning  
> Dialstring [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/ 
> 06179470890 at proxy.carrier2.net:5060
>
>
>
>
>
> freeswitch> lcr 6179470890  default  9785223241
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr  
> is [6179470890  default  9785223241]
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to  
> [9785223241]
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing  
> [state:0 lata:0] so rate field is [rate]
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT  
> l.digits, c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix  
> AS gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix ,  
> cg.codec , l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id  
> JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1'  
> AND cg.enabled = '1' AND l.enabled = '1' AND digits IN (6179470890, 617947089 
> , 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6) AND  
> CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits  
> DESC,  rate, rand();
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning  
> Dialstring [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/ 
> carrier1/16179470890
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to  
> head of list
> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning  
> Dialstring [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/ 
> carrier1/16179470890
>
>
>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |  
> Dialstring  
>                                                                |
>  | 617947      | carrier1 | 0.09000  |       |            |  
> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/ 
> carrier1/16179470890 |
>
>
>
>
>
>
>
>
>
>
>
> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> turn up logging to debug again, and then reload mod_lcr.  It'll spit  
> out a bunch of crap when it tests out each profile you have  
> defined.  Give me the full log (here or in  
> pastebin.freeswitch.org).  That may show more useful info as to why  
> things are mucked up?
>
>
> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane  
> <mouncifbb at gmail.com> wrote:
> yes I use NANPA_STD profile instead of default cause I thought the  
> custom profile was causing issues, but looks like it's returning  
> same results.
>
> There is this line in thw wiki:
> intra lata/state selection is done manually by setting the channel  
> variables intrastate or intralata to the value true.
>
> do I have to set these ? if yes how?
>
> Thanks
>
>
> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> Stuff inline.
>
> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane  
> <mouncifbb at gmail.com> wrote:
> NPANXX,"LATA","OCN","NTER","INTRA"
> 201007,"224","7229","0.0059","0.0127"
> 201040,"224","9206","0.0036","0.0036"
>
>
> Looks like they give you the LATA and OCN values with the prefix.   
> We (should) look that up ourselves.
>
> FreeSWITCH Version 1.0.trunk (16540)
>
>
> Also I noticed the npa_nxx_ocn table never get consulted.
>
> I also see this now when making a real call instead of running  
> thorugh CLI
>
> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890 NANPA_STD)
> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 intrastate channel  
> var is [undef]
>
> This is fine.  it is a leftover from when you would tell mod_lcr via  
> a channel var that it should do intrastate.  I later had mod_lcr do  
> the lookup itself, but we still honor the old var.  There are no  
> channel vars associated with the cli, so you wouldn't see that msg.
>
>
> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes  
> based on interstate rates
> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on 16179470893 
>  using profile NANPA_STD
> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing  
> [state:0 lata:0] so rate field is [rate]
>
> called number 6179470890  caller ID: 6179472456
>
> any ideas??
>
>
> Only thing that jumps out at me.
>
> The output from lcr_admin show profiles showed only the default  
> one.  On the dialplan you use the NANPA_STD profile.  Can you check  
> lcr_admin list and see if that profile is defined and if so if it  
> says it is using the npanxx table?
>
>
>
>
>
>
>
> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker <rupa at rupa.com>  
> wrote:
> Something is still missing from the logs. Note the query of the npanxx
> table, the flags being set, and the rate field being chosen.  Umm..
> oh, what version of fs are you running?
>
> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
> (there is a link to that from mod_lcr's wiki page).
>
> An example from my own setup:
>
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
> is [12148267711 default 12148267712]
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to  
> [12148267712]
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR (npa=214
> AND nxx=826)
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count: 1
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
> [state:1 lata:1] so rate field is [intralata_rate]
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT l.digits
> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
> l.intralata_rate as lcr_rate_field,        cg.prefix AS lcr_gw_prefix,
> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
> random();
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/ 
> XXXX12148267711
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
> head of list
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/ 
> 12148267711
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to  
> end of list
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to  
> end of list
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/ 
> 12148267711
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to  
> end of list
> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/ 
> YYYY12148267711
> [...]
>
> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <mouncifbb at gmail.com 
> > wrote:
> > Also the Provider has presented the rates in this format?
> > NPANXXLATA OCN INTER INTRA
> >
> >
> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <mouncifbb at gmail.com 
> >
> > wrote:
> >>
> >> Tried it and it's not giving me intralata  instead I get  
> interstate, does
> >> the npa_nxx_company_ocn table needs to be used in this case?,  
> also do I have
> >> to have the rate field in lcr table?
> >>
> >> lcr 617642  default 6176421212
> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |  
> Dialstring
> >>                                                        |
> >>  | 617642      | carrier1 | 0.00500  |       |            |
> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/ 
> carrier1/1617642 |
> >>
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to  
> lcr is
> >> [617642  default 6176421212]
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
> >> [6176421212]
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing  
> [state:0
> >> lata:0] so rate field is [rate]
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT  
> l.digits,
> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS  
> gw_suffix,
> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,  
> l.cid FROM lcr
> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND  
> l.enabled
> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND  
> CURRENT_TIMESTAMP
> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate, rand 
> ();
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning  
> Dialstring
> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/ 
> carrier1/1617642
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1  
> to head
> >> of list
> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning  
> Dialstring
> >> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/ 
> carrier1/1617642
> >>
> >> Thank you Rupa!
> >>
> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <rupa at rupa.com>  
> wrote:
> >>>
> >>> turn console logging up to debug and redo the lcr lookup.  The sql
> >>> statements along with status info will show up.  This should give
> >>> enough information to debug what is happening.
> >>>
> >>> I'm assuming the npanxx table is actually populated and not just
> >>> existing?
> >>>
> >>> When doing the lookup from the cli you have to tell lcr what CID  
> to
> >>> use (remember, it is relative to the src/dest number).  I'm pretty
> >>> sure you get something on the console log when you don't specify  
> a CID
> >>> when using the commandline.  Anyway:
> >>>
> >>> lcr 617642  default 6176421212
> >>>
> >>> should give you intralata.
> >>>
> >>> Note that the definition of intralata doesn't mean "local" for  
> some
> >>> providers.  Some providers define local to "same ratecenter"  
> which is
> >>> even more restrictive.
> >>>
> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <mouncifbb at gmail.com 
> >
> >>> wrote:
> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I  
> am using
> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
> >>> >
> >>> > lcr mysql table structure:
> >>> > CREATE TABLE `lcr` (
> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
> >>> >   `carrier_id` INT(11) NOT NULL,
> >>> >   `lead_strip` INT(11) NOT NULL,
> >>> >   `trail_strip` INT(11) NOT NULL,
> >>> >   `prefix` VARCHAR(16) NOT NULL,
> >>> >   `suffix` VARCHAR(16) NOT NULL,
> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
> >>> >   `quality` FLOAT(10,6) NOT NULL,
> >>> >   `reliability` FLOAT(10,6) NOT NULL,
> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
> >>> >   PRIMARY KEY  (`id`),
> >>> >   KEY `carrier_id` (`carrier_id`),
> >>> >   KEY `digits` (`digits`),
> >>> >   KEY `lcr_profile` (`lcr_profile`),
> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
> >>> > `carriers`
> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
> >>> >
> >>> >
> >>> > lcr_admin show profiles
> >>> > Name:           default
> >>> >  custom sql:    SELECT l.digits, c.carrier_name, l.$ 
> {lcr_rate_field},
> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
> >>> > l.trail_strip,
> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers  
> c ON
> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON  
> c.id=cg.carrier_id WHERE
> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND  
> digits IN
> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN  
> date_start
> >>> > AND
> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality  
> DESC,
> >>> >  reliability DESC, rand();
> >>> >  has %:         false
> >>> >  has vars:      true
> >>> >  has intrastate:        true
> >>> >  has intralata: true
> >>> >  has npanxx:    true
> >>> >  Reorder rate:  enabled
> >>> >  Info in headers:       disabled
> >>> >  Quote IN() List:       disabled
> >>> >
> >>> >
> >>> >
> >>> > lcr 617642  default  returns rate from the rate field table  
> and not
> >>> > intra/inter state fields rates.
> >>> >
> >>> > Any ideas? thanks!
> >>> >
> >>> >
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
> freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>>
> >>>
> >>> --
> >>> -Rupa
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
> freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> -- 
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> -- 
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> -- 
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From Russell.Mosemann at cune.org  Sun Jan 31 14:35:21 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sun, 31 Jan 2010 16:35:21 -0600
Subject: [Freeswitch-users] Equivalent to Asterisk's"directrtpsetup=yes"?
In-Reply-To: <qiqbm55lueuqup1m6r6d0mff36b4l4fb9l@4ax.com>
References: <gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b@4ax.com><gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b-e09XROE/p8c@public.gmane.org><20100129164359.9B3C22B10D@cuneorg-email.cune.pri><j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8@4ax.com><j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8-e09XROE/p8c@public.gmane.org><4468a6771001310247n4bf5d1a4rf4aea9f2d6c35a1f@mail.gmail.com>
	<qiqbm55lueuqup1m6r6d0mff36b4l4fb9l@4ax.com>
Message-ID: <15D48404014D48D19F85CFFFC4BBC76F@cune.pri>

Fred-145 asked:

> Are there features that aren't available when the two end-points speak
> RTP directly instead of having RTP packets go through Freeswitch?

I wasn't paying close attention, but in a recent discussion, someone wanted to have moh in a bypass-media situation. I think there was a way to do that, but then there was an issue with going back to bypass-media after the call was taken off hold. If you look through the list archive, you should be able to find it.

--
Russell Mosemann




From rupa at rupa.com  Sun Jan 31 15:07:29 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 31 Jan 2010 17:07:29 -0600
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<ade741d61001292042i1912ca94ra012c94ece8d8350@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
	<b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
	<9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>
Message-ID: <b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>

Yes, you need to normalize the values passed to lcr.  Otherwise, how could
it work?

You can normalize the CID by matching and adding a 1 for 10 digit #s, or
removing the leading + or other things you might need then setting it back
to the profile using the set_profile_var app (
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var).
 (mod_cidlookup will set it after doing a #->name/area lookup - but for now
you can set it yourself)

You can normalize the DID by doing similar matching rules as above and then
transfering to that normalized DID for the rest of your call plan
processing.

I'm pretty sure mod_cidlookup has an example of normalizing... yeah:
http://wiki.freeswitch.org/wiki/Mod_cidlookup#Dialplan_Application

On Sun, Jan 31, 2010 at 4:18 PM, Mouncifbb <mouncifbb at gmail.com> wrote:

> So the CID must have 1 at front also? Usually people
> Send only npa and nxx ex 6176427788 7817612233
> Do I need to alter it?
>
> Sent from my iPhone
>
> On Jan 31, 2010, at 8:32 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>
>
> On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane < <mouncifbb at gmail.com>
> mouncifbb at gmail.com> wrote:
>
>> OK going back to use default profile to keep things simple below 2 results
>>
>> Using:
>>
>> lcr 16179470890  default  19785223241  ( this one consult
>> npa_nxx_company_ocn)
>>
>> lcr 6179470890  default  9785223241 ( this one don't!! )
>>
>>
>>
> Oh, right!  mod_lcr really expects you to normalize your prefix to e164
> format.  I thought there was discussion about this in the wiki, but maybe
> not.  For simple prefix matching it doesn't matter, but for things that make
> decisions based on the # (like the lata/state stuff) it does.
>
> npanxx lookup only makes sense for NANPA numbers.  NANPA numbers have a
> country code of "1" and a total length of 11 (including the 1).
>
> This is the only rational way to do it when you have a rate table with both
> domestic (NANPA) and international prefixes.
>
>
>> freeswitch> lcr 16179470890  default  19785223241
>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>> [16179470890  default  19785223241]
>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>> [19785223241]
>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
>> OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
>> npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
>> lata:1] so rate field is [intralata_rate]
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>> c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
>> gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>> l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN
>> carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND
>> cg.enabled = '1' AND l.enabled = '1' AND digits IN (16179470890, 1617947089,
>> 161794708, 16179470, 1617947, 161794, 16179, 1617, 161, 16, 1) AND
>> CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>> intralata_rate, rand();
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>> of list
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>> 06179470890 at proxy.carrier2.net:5060
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end of
>> list after carrier1
>>
>>
>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>> Dialstring
>> |
>>  | 1           | carrier1 | 0.00000  |       |            |
>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>> |
>>  | 1           | carrier2 | 0.00000  |       |            |
>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>> 06179470890 at proxy.carrier2.net:5060 |
>>
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>> 06179470890 at proxy.carrier2.net:5060
>>
>>
>>
>>
>>
>> freeswitch> lcr 6179470890  default  9785223241
>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>> [6179470890  default  9785223241]
>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>> [9785223241]
>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
>> lata:0] so rate field is [rate]
>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>> (6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
>> AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>> rate, rand();
>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>> of list
>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>
>>
>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>> Dialstring                                                                |
>>   | 617947      | carrier1 | 0.09000  |       |            |
>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker < <rupa at rupa.com>
>> rupa at rupa.com> wrote:
>>
>>> turn up logging to debug again, and then reload mod_lcr.  It'll spit out
>>> a bunch of crap when it tests out each profile you have defined.  Give me
>>> the full log (here or in <http://pastebin.freeswitch.org>
>>> pastebin.freeswitch.org).  That may show more useful info as to why
>>> things are mucked up?
>>>
>>>
>>> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>> mouncifbb at gmail.com> wrote:
>>>
>>>> yes I use NANPA_STD profile instead of default cause I thought the
>>>> custom profile was causing issues, but looks like it's returning same
>>>> results.
>>>>
>>>> There is this line in thw wiki:
>>>> intra lata/state selection is done manually by setting the channel
>>>> variables *intrastate* or *intralata* to the value *true*.
>>>>
>>>> do I have to set these ? if yes how?
>>>>
>>>> Thanks
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker < <rupa at rupa.com>
>>>> rupa at rupa.com> wrote:
>>>>
>>>>> Stuff inline.
>>>>>
>>>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>> mouncifbb at gmail.com> wrote:
>>>>>
>>>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>>>> 201007,"224","7229","0.0059","0.0127"
>>>>>> 201040,"224","9206","0.0036","0.0036"
>>>>>>
>>>>>
>>>>> Looks like they give you the LATA and OCN values with the prefix.  We
>>>>> (should) look that up ourselves.
>>>>>
>>>>>
>>>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>>>
>>>>>>
>>>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>>>
>>>>>> I also see this now when making a real call instead of running thorugh
>>>>>> CLI
>>>>>>
>>>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890
>>>>>> NANPA_STD)
>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate channel
>>>>>> var is [undef]*
>>>>>
>>>>>
>>>>> This is fine.  it is a leftover from when you would tell mod_lcr via a
>>>>> channel var that it should do intrastate.  I later had mod_lcr do the lookup
>>>>> itself, but we still honor the old var.  There are no channel vars
>>>>> associated with the cli, so you wouldn't see that msg.
>>>>>
>>>>>
>>>>>>
>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based
>>>>>> on interstate rates
>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>>>> 16179470893 using profile NANPA_STD
>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing
>>>>>> [state:0 lata:0] so rate field is [rate]
>>>>>>
>>>>>> called number 6179470890  caller ID: 6179472456
>>>>>>
>>>>>> any ideas??
>>>>>>
>>>>>>
>>>>> Only thing that jumps out at me.
>>>>>
>>>>> The output from lcr_admin show profiles showed only the default one.
>>>>>  On the dialplan you use the NANPA_STD profile.  Can you check lcr_admin
>>>>> list and see if that profile is defined and if so if it says it is using the
>>>>> npanxx table?
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker < <rupa at rupa.com>
>>>>>> rupa at rupa.com> wrote:
>>>>>>
>>>>>>> Something is still missing from the logs. Note the query of the
>>>>>>> npanxx
>>>>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>>>>> oh, what version of fs are you running?
>>>>>>>
>>>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>>> <http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name>
>>>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>>>
>>>>>>> An example from my own setup:
>>>>>>>
>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>>>>>>> is [12148267711 default 12148267712]
>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>>>> [12148267712]
>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT 'state',
>>>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata', count(DISTINCT
>>>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR
>>>>>>> (npa=214
>>>>>>> AND nxx=826)
>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count: 1
>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count:
>>>>>>> 1
>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT
>>>>>>> l.digits
>>>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS
>>>>>>> lcr_gw_prefix,
>>>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c ON
>>>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP BETWEEN
>>>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>>>> random();
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>>
>>>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us to
>>>>>>> head of list
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>>
>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end
>>>>>>> of list
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to end
>>>>>>> of list
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>>
>>>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to
>>>>>>> end of list
>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning Dialstring
>>>>>>>
>>>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>>>> [...]
>>>>>>>
>>>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>> > Also the Provider has presented the rates in this format?
>>>>>>> > NPANXXLATA OCN INTER INTRA
>>>>>>> >
>>>>>>> >
>>>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>> mouncifbb at gmail.com>
>>>>>>> > wrote:
>>>>>>> >>
>>>>>>> >> Tried it and it's not giving me intralata  instead I get
>>>>>>> interstate, does
>>>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?, also
>>>>>>> do I have
>>>>>>> >> to have the rate field in lcr table?
>>>>>>> >>
>>>>>>> >> lcr 617642  default 6176421212
>>>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>>>> Dialstring
>>>>>>> >>                                                        |
>>>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>>>> >>
>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>>>> >>
>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to
>>>>>>> lcr is
>>>>>>> >> [617642  default 6176421212]
>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>>>>>> >> [6176421212]
>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>> [state:0
>>>>>>> >> lata:0] so rate field is [rate]
>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>>>> l.digits,
>>>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>>>> gw_suffix,
>>>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid
>>>>>>> FROM lcr
>>>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON
>>>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND
>>>>>>> l.enabled
>>>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>>>> CURRENT_TIMESTAMP
>>>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate,
>>>>>>> rand();
>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>> Dialstring
>>>>>>> >>
>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1
>>>>>>> to head
>>>>>>> >> of list
>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>> Dialstring
>>>>>>> >>
>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>> >>
>>>>>>> >> Thank you Rupa!
>>>>>>> >>
>>>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker < <rupa at rupa.com>
>>>>>>> rupa at rupa.com> wrote:
>>>>>>> >>>
>>>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The
>>>>>>> sql
>>>>>>> >>> statements along with status info will show up.  This should give
>>>>>>> >>> enough information to debug what is happening.
>>>>>>> >>>
>>>>>>> >>> I'm assuming the npanxx table is actually populated and not just
>>>>>>> >>> existing?
>>>>>>> >>>
>>>>>>> >>> When doing the lookup from the cli you have to tell lcr what CID
>>>>>>> to
>>>>>>> >>> use (remember, it is relative to the src/dest number).  I'm
>>>>>>> pretty
>>>>>>> >>> sure you get something on the console log when you don't specify
>>>>>>> a CID
>>>>>>> >>> when using the commandline.  Anyway:
>>>>>>> >>>
>>>>>>> >>> lcr 617642  default 6176421212
>>>>>>> >>>
>>>>>>> >>> should give you intralata.
>>>>>>> >>>
>>>>>>> >>> Note that the definition of intralata doesn't mean "local" for
>>>>>>> some
>>>>>>> >>> providers.  Some providers define local to "same ratecenter"
>>>>>>> which is
>>>>>>> >>> even more restrictive.
>>>>>>> >>>
>>>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>> mouncifbb at gmail.com>
>>>>>>> >>> wrote:
>>>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I am
>>>>>>> using
>>>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>>>> >>> >
>>>>>>> >>> > lcr mysql table structure:
>>>>>>> >>> > CREATE TABLE `lcr` (
>>>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00',
>>>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>>>> >>> >   KEY `digits` (`digits`),
>>>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES
>>>>>>> >>> > `carriers`
>>>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>>>> >>> >
>>>>>>> >>> >
>>>>>>> >>> > lcr_admin show profiles
>>>>>>> >>> > Name:           default
>>>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>>>> l.${lcr_rate_field},
>>>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>>>>> >>> > l.trail_strip,
>>>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers
>>>>>>> c ON
>>>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>> WHERE
>>>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>>>> digits IN
>>>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>>>> date_start
>>>>>>> >>> > AND
>>>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality
>>>>>>> DESC,
>>>>>>> >>> >  reliability DESC, rand();
>>>>>>> >>> >  has %:         false
>>>>>>> >>> >  has vars:      true
>>>>>>> >>> >  has intrastate:        true
>>>>>>> >>> >  has intralata: true
>>>>>>> >>> >  has npanxx:    true
>>>>>>> >>> >  Reorder rate:  enabled
>>>>>>> >>> >  Info in headers:       disabled
>>>>>>> >>> >  Quote IN() List:       disabled
>>>>>>> >>> >
>>>>>>> >>> >
>>>>>>> >>> >
>>>>>>> >>> > lcr 617642  default  returns rate from the rate field table and
>>>>>>> not
>>>>>>> >>> > intra/inter state fields rates.
>>>>>>> >>> >
>>>>>>> >>> > Any ideas? thanks!
>>>>>>> >>> >
>>>>>>> >>> >
>>>>>>> >>> >
>>>>>>> >>> >
>>>>>>> >>> > _______________________________________________
>>>>>>> >>> > FreeSWITCH-users mailing list
>>>>>>> >>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >>> >
>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >>> >
>>>>>>> >>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>> >>> >
>>>>>>> >>> >
>>>>>>> >>>
>>>>>>> >>>
>>>>>>> >>>
>>>>>>> >>> --
>>>>>>> >>> -Rupa
>>>>>>> >>>
>>>>>>> >>> _______________________________________________
>>>>>>> >>> FreeSWITCH-users mailing list
>>>>>>> >>> <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>> >>
>>>>>>> >
>>>>>>> >
>>>>>>> > _______________________________________________
>>>>>>> > FreeSWITCH-users mailing list
>>>>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>> >
>>>>>>> >
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> -Rupa
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From mailinglist at fribert.dk  Sun Jan 31 15:38:04 2010
From: mailinglist at fribert.dk (mailinglist)
Date: Mon, 01 Feb 2010 00:38:04 +0100
Subject: [Freeswitch-users] Somebody help me understand the 'features' set
	up please :-)
Message-ID: <4B66226C020000E10000043C@mail.fribert.dk>

Ok, I've gotten the Freeswitch to register to my VoIP provider.
I've gotten my phones to register to Freeswitch, and I can receive and make calls, all very nice.
 
I've gotten a Sipura SPA901 and a Siemens Gigaset S68IP registered to the Freeswitch.
 
When I receive a call, I would like to be able to transfer the call to another phone, or change the call to a conference call with two local phones.
 
So I've been looking at the examples in the wiki, and I can't make them work, not as I understand them anyways. Especially the att_xfer seems to be able to do what I need.
 
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
 
As I understand Example1, I should answer the call, and then press *3 during the call, and either transfer it or change it to a threeway call.
 
I get the first part, create an extension in the dialplan called att_xfer.
But what is meant by the second par 'then bind this feature to DTMF 3', how do I enter that, and where?
 
I hope somebody can help me with this (again)?
 
 
 
Best regards
Fribse
 
/Running Freeswitch on pfSense running on a VMWare ESXi 4.0
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From rupa at rupa.com  Sun Jan 31 16:45:15 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 31 Jan 2010 18:45:15 -0600
Subject: [Freeswitch-users] Somebody help me understand the 'features'
	set up please :-)
In-Reply-To: <4B66226C020000E10000043C@mail.fribert.dk>
References: <4B66226C020000E10000043C@mail.fribert.dk>
Message-ID: <b39273531001311645u618ea87am8a3d84ee6b1bcfcb@mail.gmail.com>

Look at bind_meta_app in the default dialplan.  It binds the dtmf to
the features context.

On Sun, Jan 31, 2010 at 5:38 PM, mailinglist <mailinglist at fribert.dk> wrote:
> Ok, I've gotten the Freeswitch to register to my VoIP provider.
> I've gotten my phones to register to Freeswitch, and I can receive and make
> calls, all very nice.
>
> I've gotten a Sipura SPA901 and a Siemens Gigaset S68IP registered to the
> Freeswitch.
>
> When I receive a call, I would like to be able to transfer the call to
> another phone, or change the call to a conference call with two local
> phones.
>
> So I've been looking at the examples in the wiki, and I can't make them
> work, not as I understand them anyways. Especially the att_xfer seems to be
> able to do what I need.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
>
> As I understand Example1, I should answer the call, and then press *3 during
> the call, and either transfer it or change it to a threeway call.
>
> I get the first part, create an extension in the dialplan called att_xfer.
> But what is meant by the second par 'then bind this feature to DTMF 3', how
> do I enter that, and where?
>
> I hope somebody can help me with this (again)?
>
>
>
> Best regards
> Fribse
>
> /Running Freeswitch on pfSense running on a VMWare ESXi 4.0
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa



From mouncifbb at gmail.com  Sun Jan 31 19:13:10 2010
From: mouncifbb at gmail.com (Mouncif Benniane)
Date: Sun, 31 Jan 2010 22:13:10 -0500
Subject: [Freeswitch-users] mod_lcr problem
In-Reply-To: <b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
References: <ade741d61001291443l6134a3achf28be64fc7746c8e@mail.gmail.com>
	<b39273531001300702r1a81f094pac0bd20aeb5ceed6@mail.gmail.com>
	<ade741d61001301338x8014e86t9c9094403f3361f@mail.gmail.com>
	<b39273531001301559h60668746kd5c63ada240513f1@mail.gmail.com>
	<ade741d61001301623w70d7e449i44a9f1e9fb7beae4@mail.gmail.com>
	<b39273531001301645w10446308o3578479dcbf2da2@mail.gmail.com>
	<ade741d61001302057w6ce72355iee6e8b4a2872fc81@mail.gmail.com>
	<b39273531001310532y1de53609rf474755cec04f218@mail.gmail.com>
	<9AB8B620-5A69-4F08-B62C-FFEB03FC6762@gmail.com>
	<b39273531001311507hb1bfe02m8d370443b882ef81@mail.gmail.com>
Message-ID: <ade741d61001311913mf2916e6g20796f8c457b28f6@mail.gmail.com>

Got it! I appreciate your help very much!

On Sun, Jan 31, 2010 at 6:07 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Yes, you need to normalize the values passed to lcr.  Otherwise, how could
> it work?
>
> You can normalize the CID by matching and adding a 1 for 10 digit #s, or
> removing the leading + or other things you might need then setting it back
> to the profile using the set_profile_var app (
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_profile_var).
>  (mod_cidlookup will set it after doing a #->name/area lookup - but for now
> you can set it yourself)
>
> You can normalize the DID by doing similar matching rules as above and then
> transfering to that normalized DID for the rest of your call plan
> processing.
>
> I'm pretty sure mod_cidlookup has an example of normalizing... yeah:
> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Dialplan_Application
>
> On Sun, Jan 31, 2010 at 4:18 PM, Mouncifbb <mouncifbb at gmail.com> wrote:
>
>> So the CID must have 1 at front also? Usually people
>> Send only npa and nxx ex 6176427788 7817612233
>> Do I need to alter it?
>>
>> Sent from my iPhone
>>
>> On Jan 31, 2010, at 8:32 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>
>>
>> On Sat, Jan 30, 2010 at 10:57 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>> mouncifbb at gmail.com> wrote:
>>
>>> OK going back to use default profile to keep things simple below 2
>>> results
>>>
>>> Using:
>>>
>>> lcr 16179470890  default  19785223241  ( this one consult
>>> npa_nxx_company_ocn)
>>>
>>> lcr 6179470890  default  9785223241 ( this one don't!! )
>>>
>>>
>>>
>> Oh, right!  mod_lcr really expects you to normalize your prefix to e164
>> format.  I thought there was discussion about this in the wiki, but maybe
>> not.  For simple prefix matching it doesn't matter, but for things that make
>> decisions based on the # (like the lata/state stuff) it does.
>>
>> npanxx lookup only makes sense for NANPA numbers.  NANPA numbers have a
>> country code of "1" and a total length of 11 (including the 1).
>>
>> This is the only rational way to do it when you have a rate table with
>> both domestic (NANPA) and international prefixes.
>>
>>
>>> freeswitch> lcr 16179470890  default  19785223241
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>> [16179470890  default  19785223241]
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>> [19785223241]
>>> 2010-01-30 23:53:34.681842 [DEBUG] mod_lcr.c:736 SQL: SELECT 'state',
>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=617 AND nxx=947)
>>> OR (npa=978 AND nxx=522) UNION SELECT 'lata', count(DISTINCT lata) FROM
>>> npa_nxx_company_ocn WHERE (npa=617 AND nxx=947) OR (npa=978 AND nxx=522)
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: state, Count: 1
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:696 Type: lata, Count: 1
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:786 intra routing [state:1
>>> lata:1] so rate field is [intralata_rate]
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:802 we have an event
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>> c.carrier_name, l.intralata_rate, cg.prefix AS gw_prefix, cg.suffix AS
>>> gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>> l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN
>>> carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND
>>> cg.enabled = '1' AND l.enabled = '1' AND digits IN (16179470890, 1617947089,
>>> 161794708, 16179470, 1617947, 161794, 16179, 1617, 161, 16, 1) AND
>>> CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>> intralata_rate, rand();
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>> of list
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:667 adding carrier2 to end
>>> of list after carrier1
>>>
>>>
>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>> Dialstring
>>> |
>>>  | 1           | carrier1 | 0.00000  |       |            |
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> |
>>>  | 1           | carrier2 | 0.00000  |       |            |
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060 |
>>>
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.00000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:53:34.910842 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier2,lcr_rate=0.00000]sofia/external/
>>> 06179470890 at proxy.carrier2.net:5060
>>>
>>>
>>>
>>>
>>>
>>> freeswitch> lcr 6179470890  default  9785223241
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1329 data passed to lcr is
>>> [6179470890  default  9785223241]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:1365 Set Caller ID to
>>> [9785223241]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:786 intra routing [state:0
>>> lata:0] so rate field is [rate]
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:802 we have an event
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:826 SQL: SELECT l.digits,
>>> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix,
>>> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr
>>> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>> WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN
>>> (6179470890, 617947089, 61794708, 6179470, 617947, 61794, 6179, 617, 61, 6)
>>> AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC,
>>> rate, rand();
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:615 Adding carrier1 to head
>>> of list
>>> 2010-01-30 23:52:58.782633 [DEBUG] mod_lcr.c:307 Returning Dialstring
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890
>>>
>>>
>>>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>> Dialstring                                                                |
>>>   | 617947      | carrier1 | 0.09000  |       |            |
>>> [lcr_carrier=carrier1,lcr_rate=0.09000]sofia/gateway/carrier1/16179470890 |
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Sat, Jan 30, 2010 at 7:45 PM, Rupa Schomaker < <rupa at rupa.com>
>>> rupa at rupa.com> wrote:
>>>
>>>> turn up logging to debug again, and then reload mod_lcr.  It'll spit out
>>>> a bunch of crap when it tests out each profile you have defined.  Give me
>>>> the full log (here or in <http://pastebin.freeswitch.org>
>>>> pastebin.freeswitch.org).  That may show more useful info as to why
>>>> things are mucked up?
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 6:23 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>> mouncifbb at gmail.com> wrote:
>>>>
>>>>> yes I use NANPA_STD profile instead of default cause I thought the
>>>>> custom profile was causing issues, but looks like it's returning same
>>>>> results.
>>>>>
>>>>> There is this line in thw wiki:
>>>>> intra lata/state selection is done manually by setting the channel
>>>>> variables *intrastate* or *intralata* to the value *true*.
>>>>>
>>>>> do I have to set these ? if yes how?
>>>>>
>>>>> Thanks
>>>>>
>>>>>
>>>>> On Sat, Jan 30, 2010 at 6:59 PM, Rupa Schomaker < <rupa at rupa.com>
>>>>> rupa at rupa.com> wrote:
>>>>>
>>>>>> Stuff inline.
>>>>>>
>>>>>> On Sat, Jan 30, 2010 at 3:38 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>
>>>>>>>  NPANXX,"LATA","OCN","NTER","INTRA"
>>>>>>> 201007,"224","7229","0.0059","0.0127"
>>>>>>> 201040,"224","9206","0.0036","0.0036"
>>>>>>>
>>>>>>
>>>>>> Looks like they give you the LATA and OCN values with the prefix.  We
>>>>>> (should) look that up ourselves.
>>>>>>
>>>>>>
>>>>>>> FreeSWITCH Version 1.0.trunk (16540)
>>>>>>>
>>>>>>>
>>>>>>> Also I noticed the *npa_nxx_ocn* table never get consulted.
>>>>>>>
>>>>>>> I also see this now when making a real call instead of running
>>>>>>> thorugh CLI
>>>>>>>
>>>>>>> EXECUTE sofia/external/6179472456 at 174.x.x.x lcr(16179470890
>>>>>>> NANPA_STD)
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1230 *intrastate
>>>>>>> channel var is [undef]*
>>>>>>
>>>>>>
>>>>>> This is fine.  it is a leftover from when you would tell mod_lcr via a
>>>>>> channel var that it should do intrastate.  I later had mod_lcr do the lookup
>>>>>> itself, but we still honor the old var.  There are no channel vars
>>>>>> associated with the cli, so you wouldn't see that msg.
>>>>>>
>>>>>>
>>>>>>>
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1233 Select routes based
>>>>>>> on interstate rates
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:1252 LCR Lookup on
>>>>>>> 16179470893 using profile NANPA_STD
>>>>>>> 2010-01-30 16:28:56.685457 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>> [state:0 lata:0] so rate field is [rate]
>>>>>>>
>>>>>>> called number 6179470890  caller ID: 6179472456
>>>>>>>
>>>>>>> any ideas??
>>>>>>>
>>>>>>>
>>>>>> Only thing that jumps out at me.
>>>>>>
>>>>>> The output from lcr_admin show profiles showed only the default one.
>>>>>>  On the dialplan you use the NANPA_STD profile.  Can you check lcr_admin
>>>>>> list and see if that profile is defined and if so if it says it is using the
>>>>>> npanxx table?
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Sat, Jan 30, 2010 at 10:02 AM, Rupa Schomaker < <rupa at rupa.com>
>>>>>>> rupa at rupa.com> wrote:
>>>>>>>
>>>>>>>> Something is still missing from the logs. Note the query of the
>>>>>>>> npanxx
>>>>>>>> table, the flags being set, and the rate field being chosen.  Umm..
>>>>>>>> oh, what version of fs are you running?
>>>>>>>>
>>>>>>>> Yes, the npa_nxx_ocn table needs to be loaded up as described in:
>>>>>>>> <http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name>
>>>>>>>> http://wiki.freeswitch.org/wiki/Mod_cidlookup#Falling_back_to_.22City_State.22_in_the_absense_of_a_name
>>>>>>>> (there is a link to that from mod_lcr's wiki page).
>>>>>>>>
>>>>>>>> An example from my own setup:
>>>>>>>>
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1384 data passed to lcr
>>>>>>>> is [12148267711 default 12148267712]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:1420 Set Caller ID to
>>>>>>>> [12148267712]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:759 SQL: SELECT
>>>>>>>> 'state',
>>>>>>>> count(DISTINCT state) FROM npa_nxx_company_ocn WHERE (npa=214 AND
>>>>>>>> nxx=826) OR (npa=214 AND nxx=826) UNION SELECT 'lata',
>>>>>>>> count(DISTINCT
>>>>>>>> lata) FROM npa_nxx_company_ocn WHERE (npa=214 AND nxx=826) OR
>>>>>>>> (npa=214
>>>>>>>> AND nxx=826)
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: lata, Count:
>>>>>>>> 1
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:719 Type: state, Count:
>>>>>>>> 1
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:809 intra routing
>>>>>>>> [state:1 lata:1] so rate field is [intralata_rate]
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:825 we have an event
>>>>>>>> 2010-01-30 08:55:10.633951 [DEBUG] mod_lcr.c:849 SQL:  SELECT
>>>>>>>> l.digits
>>>>>>>> AS lcr_digits,         c.carrier_name AS lcr_carrier_name,
>>>>>>>> l.intralata_rate as lcr_rate_field,        cg.prefix AS
>>>>>>>> lcr_gw_prefix,
>>>>>>>> cg.suffix AS lcr_gw_suffix,        l.lead_strip AS lcr_lead_strip,
>>>>>>>> l.trail_strip AS lcr_trail_strip,        l.prefix AS lcr_prefix,
>>>>>>>> l.suffix AS lcr_suffix,        cg.codec AS lcr_codec, l.cid AS
>>>>>>>> lcr_cid,        'carriers' AS lcr_limit_realm, c.carrier_name AS
>>>>>>>> lcr_limit_id, 5 AS lcr_limit_max   FROM lcr l      JOIN carriers c
>>>>>>>> ON
>>>>>>>> l.carrier_id=c.id      JOIN carrier_gateway cg ON c.id
>>>>>>>> =cg.carrier_id
>>>>>>>>  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
>>>>>>>> AND digits_prefix @> '12148267711'      AND CURRENT_TIMESTAMP
>>>>>>>> BETWEEN
>>>>>>>> date_start AND date_end   ORDER BY digits DESC, intralata_rate,
>>>>>>>> random();
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=grnvoip_us,lcr_rate=0.00591]sofia/gateway/grnvoip/XXXX12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:638 Adding grnvoip_us
>>>>>>>> to
>>>>>>>> head of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax_atl/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=teliax,lcr_rate=0.01000]sofia/gateway/teliax/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding teliax to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=vitelity,lcr_rate=0.01440]sofia/gateway/vitelity/12148267711
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:660 Adding vitelity to
>>>>>>>> end of list
>>>>>>>> 2010-01-30 08:55:10.644013 [DEBUG] mod_lcr.c:314 Returning
>>>>>>>> Dialstring
>>>>>>>>
>>>>>>>> [lcr_carrier=grnvoip_std,lcr_rate=0.01500]sofia/gateway/grnvoip/YYYY12148267711
>>>>>>>> [...]
>>>>>>>>
>>>>>>>> On Fri, Jan 29, 2010 at 10:42 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com> wrote:
>>>>>>>> > Also the Provider has presented the rates in this format?
>>>>>>>> > NPANXXLATA OCN INTER INTRA
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > On Fri, Jan 29, 2010 at 11:30 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com>
>>>>>>>> > wrote:
>>>>>>>> >>
>>>>>>>> >> Tried it and it's not giving me intralata  instead I get
>>>>>>>> interstate, does
>>>>>>>> >> the npa_nxx_company_ocn table needs to be used in this case?,
>>>>>>>> also do I have
>>>>>>>> >> to have the rate field in lcr table?
>>>>>>>> >>
>>>>>>>> >> lcr 617642  default 6176421212
>>>>>>>> >>  | Digit Match | Carrier  | Rate     | Codec | CID Regexp |
>>>>>>>> Dialstring
>>>>>>>> >>                                                        |
>>>>>>>> >>  | 617642      | carrier1 | 0.00500  |       |            |
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642 |
>>>>>>>> >>
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1329 data passed to
>>>>>>>> lcr is
>>>>>>>> >> [617642  default 6176421212]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:1365 Set Caller ID
>>>>>>>> to
>>>>>>>> >> [6176421212]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:786 intra routing
>>>>>>>> [state:0
>>>>>>>> >> lata:0] so rate field is [rate]
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:802 we have an event
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:826 SQL: SELECT
>>>>>>>> l.digits,
>>>>>>>> >> c.carrier_name, l.rate, cg.prefix AS gw_prefix, cg.suffix AS
>>>>>>>> gw_suffix,
>>>>>>>> >> l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec ,
>>>>>>>> l.cid FROM lcr
>>>>>>>> >> l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg
>>>>>>>> ON
>>>>>>>> >> c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1'
>>>>>>>> AND l.enabled
>>>>>>>> >> = '1' AND digits IN (617642, 61764, 6176, 617, 61, 6) AND
>>>>>>>> CURRENT_TIMESTAMP
>>>>>>>> >> BETWEEN date_start AND date_end ORDER BY digits DESC,  rate,
>>>>>>>> rand();
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>> Dialstring
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:615 Adding carrier1
>>>>>>>> to head
>>>>>>>> >> of list
>>>>>>>> >> 2010-01-29 23:29:45.003307 [DEBUG] mod_lcr.c:307 Returning
>>>>>>>> Dialstring
>>>>>>>> >>
>>>>>>>> [lcr_carrier=carrier1,lcr_rate=0.00500]sofia/gateway/carrier1/1617642
>>>>>>>> >>
>>>>>>>> >> Thank you Rupa!
>>>>>>>> >>
>>>>>>>> >> On Fri, Jan 29, 2010 at 7:37 PM, Rupa Schomaker <<rupa at rupa.com>
>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>> >>>
>>>>>>>> >>> turn console logging up to debug and redo the lcr lookup.  The
>>>>>>>> sql
>>>>>>>> >>> statements along with status info will show up.  This should
>>>>>>>> give
>>>>>>>> >>> enough information to debug what is happening.
>>>>>>>> >>>
>>>>>>>> >>> I'm assuming the npanxx table is actually populated and not just
>>>>>>>> >>> existing?
>>>>>>>> >>>
>>>>>>>> >>> When doing the lookup from the cli you have to tell lcr what CID
>>>>>>>> to
>>>>>>>> >>> use (remember, it is relative to the src/dest number).  I'm
>>>>>>>> pretty
>>>>>>>> >>> sure you get something on the console log when you don't specify
>>>>>>>> a CID
>>>>>>>> >>> when using the commandline.  Anyway:
>>>>>>>> >>>
>>>>>>>> >>> lcr 617642  default 6176421212
>>>>>>>> >>>
>>>>>>>> >>> should give you intralata.
>>>>>>>> >>>
>>>>>>>> >>> Note that the definition of intralata doesn't mean "local" for
>>>>>>>> some
>>>>>>>> >>> providers.  Some providers define local to "same ratecenter"
>>>>>>>> which is
>>>>>>>> >>> even more restrictive.
>>>>>>>> >>>
>>>>>>>> >>> On Fri, Jan 29, 2010 at 4:43 PM, Mouncif Benniane <<mouncifbb at gmail.com>
>>>>>>>> mouncifbb at gmail.com>
>>>>>>>> >>> wrote:
>>>>>>>> >>> > i can't make use of mod_lcr using Intra/Interstate rating, I
>>>>>>>> am using
>>>>>>>> >>> > svn: FreeSWITCH Version 1.0.trunk (16517)
>>>>>>>> >>> >
>>>>>>>> >>> > lcr mysql table structure:
>>>>>>>> >>> > CREATE TABLE `lcr` (
>>>>>>>> >>> >   `id` INT(11) NOT NULL AUTO_INCREMENT,
>>>>>>>> >>> >   `digits` VARCHAR(15) DEFAULT NULL,
>>>>>>>> >>> >   `rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `intrastate_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `intralata_rate` FLOAT(11,5) DEFAULT NULL,
>>>>>>>> >>> >   `carrier_id` INT(11) NOT NULL,
>>>>>>>> >>> >   `lead_strip` INT(11) NOT NULL,
>>>>>>>> >>> >   `trail_strip` INT(11) NOT NULL,
>>>>>>>> >>> >   `prefix` VARCHAR(16) NOT NULL,
>>>>>>>> >>> >   `suffix` VARCHAR(16) NOT NULL,
>>>>>>>> >>> >   `lcr_profile` VARCHAR(32) DEFAULT NULL,
>>>>>>>> >>> >   `date_start` DATETIME NOT NULL DEFAULT '1970-01-01
>>>>>>>> 00:00:00',
>>>>>>>> >>> >   `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00',
>>>>>>>> >>> >   `quality` FLOAT(10,6) NOT NULL,
>>>>>>>> >>> >   `reliability` FLOAT(10,6) NOT NULL,
>>>>>>>> >>> >   `cid` VARCHAR(32) NOT NULL DEFAULT '',
>>>>>>>> >>> >   `enabled` TINYINT(1) NOT NULL DEFAULT '1',
>>>>>>>> >>> >   PRIMARY KEY  (`id`),
>>>>>>>> >>> >   KEY `carrier_id` (`carrier_id`),
>>>>>>>> >>> >   KEY `digits` (`digits`),
>>>>>>>> >>> >   KEY `lcr_profile` (`lcr_profile`),
>>>>>>>> >>> >   KEY `digits_profile_cid_rate` USING BTREE (`digits`),
>>>>>>>> >>> >   CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`)
>>>>>>>> REFERENCES
>>>>>>>> >>> > `carriers`
>>>>>>>> >>> > (`id`) ON DELETE CASCADE ON UPDATE CASCADE
>>>>>>>> >>> > ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > lcr_admin show profiles
>>>>>>>> >>> > Name:           default
>>>>>>>> >>> >  custom sql:    SELECT l.digits, c.carrier_name,
>>>>>>>> l.${lcr_rate_field},
>>>>>>>> >>> > cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip,
>>>>>>>> >>> > l.trail_strip,
>>>>>>>> >>> > l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers
>>>>>>>> c ON
>>>>>>>> >>> > l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id
>>>>>>>> WHERE
>>>>>>>> >>> > c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND
>>>>>>>> digits IN
>>>>>>>> >>> > (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN
>>>>>>>> date_start
>>>>>>>> >>> > AND
>>>>>>>> >>> > date_end ORDER BY digits DESC,  ${lcr_rate_field},  quality
>>>>>>>> DESC,
>>>>>>>> >>> >  reliability DESC, rand();
>>>>>>>> >>> >  has %:         false
>>>>>>>> >>> >  has vars:      true
>>>>>>>> >>> >  has intrastate:        true
>>>>>>>> >>> >  has intralata: true
>>>>>>>> >>> >  has npanxx:    true
>>>>>>>> >>> >  Reorder rate:  enabled
>>>>>>>> >>> >  Info in headers:       disabled
>>>>>>>> >>> >  Quote IN() List:       disabled
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > lcr 617642  default  returns rate from the rate field table
>>>>>>>> and not
>>>>>>>> >>> > intra/inter state fields rates.
>>>>>>>> >>> >
>>>>>>>> >>> > Any ideas? thanks!
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>> > _______________________________________________
>>>>>>>> >>> > FreeSWITCH-users mailing list
>>>>>>>> >>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >>> >
>>>>>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >>> >
>>>>>>>> >>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >>> >
>>>>>>>> >>> >
>>>>>>>> >>>
>>>>>>>> >>>
>>>>>>>> >>>
>>>>>>>> >>> --
>>>>>>>> >>> -Rupa
>>>>>>>> >>>
>>>>>>>> >>> _______________________________________________
>>>>>>>> >>> FreeSWITCH-users mailing list
>>>>>>>> >>> <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >>
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > _______________________________________________
>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>> >
>>>>>>>> >
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> -Rupa
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Sun Jan 31 20:36:25 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 31 Jan 2010 22:36:25 -0600
Subject: [Freeswitch-users] Equivalent to Asterisk's"directrtpsetup=yes"?
In-Reply-To: <15D48404014D48D19F85CFFFC4BBC76F@cune.pri>
References: <gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b@4ax.com><gk26m5t9sfgt6kan4qt41p22ivjfb0ek1b-e09XROE/p8c@public.gmane.org><20100129164359.9B3C22B10D@cuneorg-email.cune.pri><j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8@4ax.com><j4e8m5tqfl3fo8ug7d3sdgdiu7gofidbj8-e09XROE/p8c@public.gmane.org><4468a6771001310247n4bf5d1a4rf4aea9f2d6c35a1f@mail.gmail.com>
	<qiqbm55lueuqup1m6r6d0mff36b4l4fb9l@4ax.com>
	<15D48404014D48D19F85CFFFC4BBC76F@cune.pri>
Message-ID: <7FA9CDA6-EAB3-4C1D-9CD3-8D4B06B0F335@freeswitch.org>

Going back no_media after hold isn't supported yet.. Anthony said he would add it if someone really really wanted it and posted a bounty of $500 to cover his time to implement it.

/b

On Jan 31, 2010, at 4:35 PM, Russell Mosemann wrote:

> I wasn't paying close attention, but in a recent discussion, someone wanted to have moh in a bypass-media situation. I think there was a way to do that, but then there was an issue with going back to bypass-media after the call was taken off hold. If you look through the list archive, you should be able to find it.




From wasim at convergence.pk  Sun Jan 31 21:01:26 2010
From: wasim at convergence.pk (Wasim Baig)
Date: Mon, 1 Feb 2010 10:01:26 +0500
Subject: [Freeswitch-users] SS7 & MGCP support
In-Reply-To: <SNT106-W1023694FA10CC0F668ADCDF45A0@phx.gbl>
References: <SNT106-W1023694FA10CC0F668ADCDF45A0@phx.gbl>
Message-ID: <b8ad2a5b1001312101u68c2f5a9sa66fb45f114d6bac@mail.gmail.com>

On Sat, Jan 30, 2010 at 12:48 PM, Abid Saleem <abid_freeswitch at live.com>wrote:

> ?         Since it is a softswitch also, does it support SS7, MGCP and
> Megaco protocols to control media gateways?
>
I remember back in the day when FS was a glimmer in tony's eye and we had a
mostly working (but not stable) MGCP UA mode with it ... but then I don't
think any more work was done it. My requirement for it has phased away also,
although if anyone is really interested I'm sure we could come up with a
bounty for this, as MGCP is fairly basic.

The only option you have for SS7 is a commercial implementation from
Sangoma. No open source implementation for SS7 or SIGTRAN exist as of now,
although there are a couple of rumors in the air about diverse efforts.

> ?         Does it support call shops business model?
>
Technically yes, but the logic and billing etc is all up to you, so out of
the box, is it a preconfigured call shop system, no. Can it be used for one,
certainly. Will you have to work a bit for it, most cetainly, yes.

> ?         How to add new SIP user accounts into it that can be used to
> register to it. I know one way is to copy and paste          1000.xml file
> and edit it in the conf/directory folder. What is the optimal way to do this
> task?
>
You can use a DB for this as well and have the XML generated or used inline.
http://wiki.freeswitch.org/wiki/Mod_xml_curl

> o   Is there any GUI available. If yes how can I make it work and private
> label it.?
>
There are a couple of efforts. Read the wiki. You could use ASTPP with
reseller option to white label it.

-- 
wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
peace be upon you ...
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From mike at jerris.com  Sun Jan 31 21:53:07 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 00:53:07 -0500
Subject: [Freeswitch-users] Replace Internal IP with External IP in From
	Header
In-Reply-To: <4571ff701001261738w4f51b33dqee1d19b8d0e2236d@mail.gmail.com>
References: <4571ff701001261738w4f51b33dqee1d19b8d0e2236d@mail.gmail.com>
Message-ID: <27B2F566-0C42-4A55-8D62-D5E0D5D5EDE7@jerris.com>

Try current trunk, I think this is fixed now, if not, please open a bug on jira.FreeSWITCH.org


On Jan 26, 2010, at 8:38 PM, Code Ghar wrote:

> I followed the example in Freeswitch behind NAT (http://wiki.freeswitch.org/wiki/NAT_Traversal#Freeswitch_behind_NAT). In the Contact header of invite sent to an external gateway, I see sip:extension at ExternalIP:port but in the From header I see sip:extension at InternalIP. How can I change the From header of SIP message so that it displays the external IP instead of internal IP? The reason for doing this is that the external gateway authenticates and authorizes call based on the IP in From header. They expect an external IP and not an internal IP. 
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From mike at jerris.com  Sun Jan 31 21:58:21 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 00:58:21 -0500
Subject: [Freeswitch-users] Question about javascript
In-Reply-To: <20100127123533.602850e1.matsubara_tomoya@intec.co.jp>
References: <20100127123533.602850e1.matsubara_tomoya@intec.co.jp>
Message-ID: <7C99C342-AB78-4959-9508-9BB8CA5FE29F@jerris.com>

You can see if you have a memory leak by running in valgrind.  In the mean time, why would you do this in a lua script, all of this is easier and more clear to do right in dialplan.
also, your running ring ready, then answering, then setting ringback?  it seems like you have everything mixed up here.

This probably should just be a 5 line dialplan.  
set ringback
set continue on fail true
set hangup after bridge true
bridge (with api on answer)
playback test02.wav

Mike

On Jan 26, 2010, at 10:35 PM, Tomoya Matsubara wrote:

> Hello,
> 
> When the following scripts were tested, it seems to do the memory leak. 
> Please teach when there is a problem in this script. 
> 
> -- test script --
> session.execute("ring_ready");
> session.answer();
> session.setVariable("ringback", "%(1000, 2000, 440, 460)");
> 
> var bleg = new Session();
> var sound_wav = "sounds/test01.wav";
> var sound_leg = "both";
> var op = "api_on_answer=uuid_broadcast "+session.uuid+" "+sound_wav+" "+sound_leg;
> var ret = bleg.originate(session, "{"+op+"}" + "sofia/gateway/profile0_gateway1/1000");
> if(!session.ready()){
>  return;
> }
> 
> if(!ret){ // bleg not answered.
>  var sound_wav = "sounds/test02.wav";
>  session.streamFile(sound_wav);
>  if(session.ready()){
>    session.hangup();
>  }
>  return;
> }
> 
> if(bleg.ready()){
>  bridge(session, bleg);
> }
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Sun Jan 31 21:59:15 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 00:59:15 -0500
Subject: [Freeswitch-users] Wrong RTP port submitted?
In-Reply-To: <4B6011EF.6090706@gmx.net>
References: <4B6011EF.6090706@gmx.net>
Message-ID: <48056347-BFCC-4B38-8C29-9FB592033827@jerris.com>

please report this bug to jira.FreeSWITCH.org.

On Jan 27, 2010, at 5:14 AM, Peter P GMX wrote:

> I have defined the rtp port range for 12000-12100 in switch.conf.xml.
> However Freeswitch is offering a port 48320 in the invite message. The
> result is, that the incoming RTP stream is blocked by the firewall (I
> can see a reject for UDP 48320).
> Any hint how to solve this?




From mike at jerris.com  Sun Jan 31 22:04:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 01:04:20 -0500
Subject: [Freeswitch-users] Logging question
In-Reply-To: <da4ab26e1001292020o280f330ft62a278b93c7fb1fd@mail.gmail.com>
References: <da4ab26e1001292020o280f330ft62a278b93c7fb1fd@mail.gmail.com>
Message-ID: <8BEB200D-32AC-4D80-B59D-07C8228D7380@jerris.com>

sofia profile <profile_name> siptrace on

There is also a config param, it should be documented int he current default configs.

Mike


On Jan 29, 2010, at 11:20 PM, paul gore wrote:

> Hi there,
> I am running FS 1.0.trunk (14501) (I know it's old but we serve a small community and don't have time to upgrade/test the latest/greatest). I am having troubles understanding how to switch SIP trace in log files, I tried
> 
> fsctl loglevel debug
> sofia tracelevel debug
> 
> but it seem to have no effect, I only get sofia debug messages but no detailed SIP info.
> What also puzzling me is if I do 
> 
> console loglevel 0
> 
> I still get debug information on console.
> What am I doing wrong?




From mike at jerris.com  Sun Jan 31 22:05:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 01:05:37 -0500
Subject: [Freeswitch-users] Freeswitch core dump after upgrade to latest
	version
In-Reply-To: <d361ca7b1001300601r29530dcfm65c67e9121d3ddde@mail.gmail.com>
References: <d361ca7b1001230906j736d5b2dm2d9260b2628ffd3c@mail.gmail.com>
	<D56D65EB-B150-4A59-8B66-68BF4C58108B@jerris.com>
	<d361ca7b1001241004kcb73f11hc400bdb37c9d372b@mail.gmail.com>
	<3A27F063-E0C0-4178-A3AF-068956B55846@jerris.com>
	<d361ca7b1001241117q7f1379bewe1bc85279ef67c83@mail.gmail.com>
	<alpine.GSO.2.00.1001250716120.17161@h1.pp.uhwv.np.vy>
	<d361ca7b1001242150t29b36ebcy6c2e40ca3d360c00@mail.gmail.com>
	<d361ca7b1001251143p7f396dd8h11d372731a26f4d7@mail.gmail.com>
	<224C684A-B357-42E4-98AA-0EE238A27A49@jerris.com>
	<d361ca7b1001251206x21242589he3014081a7e110d1@mail.gmail.com>
	<d361ca7b1001300601r29530dcfm65c67e9121d3ddde@mail.gmail.com>
Message-ID: <08874DB8-D35F-47CA-8A5A-B6BF33C9D6B4@jerris.com>

Thanks again for you help reproducing this so we could chase this issue down.

Mike

On Jan 30, 2010, at 9:01 AM, Yehavi Bourvine wrote:

> It works ok now (fixed on r16534).
>  
>                 Thanks! __Yehavi:
> 




From mike at jerris.com  Sun Jan 31 22:08:01 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 01:08:01 -0500
Subject: [Freeswitch-users] Establishing a Call from .Net based
	application
In-Reply-To: <df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
References: <874941.17255.qm@web33502.mail.mud.yahoo.com>
	<df97b9c51001310545n54195cadr9faa3db46917ca9e@mail.gmail.com>
Message-ID: <35B856D0-81DB-4EF9-A376-D0B32780FD30@jerris.com>

api would not have a call associated with it at all.


On Jan 31, 2010, at 8:45 AM, Scott Fernandez wrote:

> Hi,
>  
> Thx for the information. Can I have some detailed steps to configure mod_managed class call control and how do we write the API commands in .Net applications?
>  
> In addition, how do we get the current STATE of the call when I use webapi?. Because it is required for me to route the call to the user upon it is answered or disconnect  it.
>  




From mike at jerris.com  Sun Jan 31 22:09:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 01:09:20 -0500
Subject: [Freeswitch-users] How to populate Voice recording metadata
	with custom CRM data?
In-Reply-To: <003c01caa23b$83ed0800$8bc71800$@co.za>
References: <003c01caa23b$83ed0800$8bc71800$@co.za>
Message-ID: <A86F32B9-6BF0-4344-B948-675AF39FF56F@jerris.com>

There is a later windows installer on the downloads site as well

http://files.freeswitch.org/windows_installer/

On Jan 31, 2010, at 1:05 AM, Roly Maz wrote:

> Hi All,  
>  
> I am using Freeswitch 1.0.4 (Latest Windows Installer version) on Windows 7 Ultimate for testing

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From mike at jerris.com  Sun Jan 31 22:11:58 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 01:11:58 -0500
Subject: [Freeswitch-users] Trunk Version Number
In-Reply-To: <2360B060EDF44D368707DD12064BD416@bp1.ad.bp.com>
References: <2360B060EDF44D368707DD12064BD416@bp1.ad.bp.com>
Message-ID: <8976E2C4-CF8C-433A-AC79-B58F65105D08@jerris.com>

it should.  This can happen if you build from an svn checkout and the svn client your using is newer than our static linked svnversion.exe.  If anyone can make me a newer stripped down version like that I would appreciate it I have not had the time.

On Jan 31, 2010, at 9:30 AM, Dave Stevenson wrote:

> Hi,
>  
> Running the latest SVN (16453) under Windows, the console "Version" command displays :-
>  
> "FreeSWITCH Version 1.0.trunk (UNKNOWN)"
>  
> Should the version number not include a meaningful build version in the brackets ?
>  
> regards
> Dave
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Sun Jan 31 22:14:14 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 01:14:14 -0500
Subject: [Freeswitch-users] determining the source of receyved call in
	public context
In-Reply-To: <4B65E649.2040007@gmail.com>
References: <4B65E649.2040007@gmail.com>
Message-ID: <46204926-1DCE-4C45-B31B-30CC698AE636@jerris.com>

What does the IP have to do with the numbers?  Can you rephrase the question?

On Jan 31, 2010, at 3:21 PM, Meftah Tayeb wrote:
> how do i determine the gateway or the Ip of a receyved call from ITSP's?
> i am calling my did from my mobile, but i see is processing the the 
> mobile number no the ITSP User or did




From mike at jerris.com  Sun Jan 31 22:26:36 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Feb 2010 01:26:36 -0500
Subject: [Freeswitch-users] vm-disk-quota
In-Reply-To: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>
References: <28f27f5d1001311246h17b426a4x39e0d48d3d305342@mail.gmail.com>
Message-ID: <7DFBD163-A72A-4186-BB89-6D468FD9ABA2@jerris.com>

If I read the code right (mod_voicemail.c:3051) it looks like it is measuring in seconds of vocicemail, but the wiki indicates number of voicemails, neither seems to match the name of the param.

Tamas-  Can you comment on how this was intended to work?

Mike

On Jan 31, 2010, at 3:46 PM, mbsip wrote:

> Hi ALL,
> 
> Maybe this question will be piece of cake for most of you, but it
> makes me think.
> 
> I would like to configure "vm-disk-quota" for all users i have.
> I followed the wiki page and provided:
> <param name="vm-disk-quota" value="10"/>
> to /conf/directory/default/1000.xml
> 
> After reloadxml, incoming call give me "mod_voicemail.c:3057 Voicemail
> disk quota is exceeded" feedback
> No surprise for me because i had more less 10 voice mails already
> recorded (before the vm-disk-quota was set up).
> Strange is that increasing value even to 100 does not change anything.
> The same thing with deleting recordings from user directory.
> The only wayout is to set it to default value=0 (even FS shutdown
> doesn't change anything)
> 
> I am wondering why vm-disk-quota produces "Voicemail disk quota is
> exceeded" all the time
> Where the module is looking for stored voicemail recordings.
> 
> Below is part of my configuration.
> 1) /conf/autoload_configs/voicemail.conf.xml
> <param name="storage-dir" value="/usr/local/freeswitch/vm"/>
> 2) /conf/directory/default/1000.xml
> <param name="vm-disk-quota" value="100"/>
> 3) /vm/FS_ip_address/1000 is empty